diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2009-12-08 07:47:46 -0800 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2009-12-08 07:47:46 -0800 |
| commit | a421018e8c10e5593a1fee076af72a66c3fe8ca3 (patch) | |
| tree | 2854511845d0e07d33726a13eda6de1059a5c9df /sound | |
| parent | 3ad1f3b35e8309ec93454dbf89beaafcdb5312da (diff) | |
| parent | 86e1d57e4f24ca27ce813bdc2afaac4adafcbaf4 (diff) | |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (294 commits)
S3C64XX: Staticise platform data for PCM devices
ASoC: Rename controls with a / in wm_hubs
snd-fm801: autodetect SF64-PCR (tuner-only) card
ALSA: tea575x-tuner: fix mute
ASoC: au1x: dbdma2: plug memleak in pcm device creation error path
ASoC: au1x: dbdma2: fix oops on soc device removal.
ALSA: hda - Fix memory leaks in the previous patch
ALSA: hda - Add ALC661/259, ALC892/888VD support
ALSA: opti9xx: remove snd_opti9xx fields
ALSA: aaci - Clean up duplicate code
ALSA: usb - Fix mixer map for Hercules Gamesurround Muse Pocket LT
ALSA: hda - Add position_fix quirk for HP dv3
ALSA: hda - Add a pin-fix for FSC Amilo Pi1505
ALSA: hda - Fix Cxt5047 test mode
ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API
ASoC: sh: fsi: Add runtime PM support
sh: ms7724se: Add runtime PM support for FSI
ALSA: hda - Add a position_fix quirk for MSI Wind U115
ALSA: opti-miro: add PnP detection
ALSA: opti-miro: separate comon probing code
...
Diffstat (limited to 'sound')
242 files changed, 19504 insertions, 7520 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 439e15c8faa..b3e53e616ec 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -58,7 +58,7 @@ config SOUND_OSS_CORE_PRECLAIM Please read Documentation/feature-removal-schedule.txt for details. - If unusre, say Y. + If unsure, say Y. source "sound/oss/dmasound/Kconfig" diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 5a549ed6c8a..8c0c851d464 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -3,7 +3,7 @@ # obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o -snd-aaci-objs := aaci.o devdma.o +snd-aaci-objs := aaci.o obj-$(CONFIG_SND_PXA2XX_PCM) += snd-pxa2xx-pcm.o snd-pxa2xx-pcm-objs := pxa2xx-pcm.o diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 6c160a038b2..1497dce1b04 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -18,10 +18,7 @@ #include <linux/interrupt.h> #include <linux/err.h> #include <linux/amba/bus.h> - -#include <asm/io.h> -#include <asm/irq.h> -#include <asm/sizes.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/initval.h> @@ -30,7 +27,6 @@ #include <sound/pcm_params.h> #include "aaci.h" -#include "devdma.h" #define DRIVER_NAME "aaci-pl041" @@ -492,7 +488,7 @@ static int aaci_pcm_hw_free(struct snd_pcm_substream *substream) /* * Clear out the DMA and any allocated buffers. */ - devdma_hw_free(NULL, substream); + snd_pcm_lib_free_pages(substream); return 0; } @@ -509,20 +505,14 @@ static int aaci_pcm_hw_params(struct snd_pcm_substream *substream, aacirun->pcm_open = 0; } - err = devdma_hw_alloc(NULL, substream, - params_buffer_bytes(params)); + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(params)); if (err < 0) goto out; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[0].slots); - else - err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), - params_channels(params), - aacirun->pcm->r[0].slots); - + err = snd_ac97_pcm_open(aacirun->pcm, params_rate(params), + params_channels(params), + aacirun->pcm->r[0].slots); if (err) goto out; @@ -538,7 +528,7 @@ static int aaci_pcm_prepare(struct snd_pcm_substream *substream) struct aaci_runtime *aacirun = runtime->private_data; aacirun->start = (void *)runtime->dma_area; - aacirun->end = aacirun->start + runtime->dma_bytes; + aacirun->end = aacirun->start + snd_pcm_lib_buffer_bytes(substream); aacirun->ptr = aacirun->start; aacirun->period = aacirun->bytes = frames_to_bytes(runtime, runtime->period_size); @@ -555,11 +545,6 @@ static snd_pcm_uframes_t aaci_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(runtime, bytes); } -static int aaci_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - return devdma_mmap(NULL, substream, vma); -} - /* * Playback specific ALSA stuff @@ -726,7 +711,6 @@ static struct snd_pcm_ops aaci_playback_ops = { .prepare = aaci_pcm_prepare, .trigger = aaci_pcm_playback_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; static int aaci_pcm_capture_hw_params(struct snd_pcm_substream *substream, @@ -854,7 +838,6 @@ static struct snd_pcm_ops aaci_capture_ops = { .prepare = aaci_pcm_capture_prepare, .trigger = aaci_pcm_capture_trigger, .pointer = aaci_pcm_pointer, - .mmap = aaci_pcm_mmap, }; /* @@ -1044,6 +1027,8 @@ static int __devinit aaci_init_pcm(struct aaci *aaci) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &aaci_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &aaci_capture_ops); + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + NULL, 0, 64 * 104); } return ret; diff --git a/sound/arm/devdma.c b/sound/arm/devdma.c deleted file mode 100644 index 9d1e6665b54..00000000000 --- a/sound/arm/devdma.c +++ /dev/null @@ -1,80 +0,0 @@ -/* - * linux/sound/arm/devdma.c - * - * Copyright (C) 2003-2004 Russell King, All rights reserved. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * ARM DMA shim for ALSA. - */ -#include <linux/device.h> -#include <linux/dma-mapping.h> - -#include <sound/core.h> -#include <sound/pcm.h> - -#include "devdma.h" - -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - - if (runtime->dma_area == NULL) - return; - - if (buf != &substream->dma_buffer) { - dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, buf->addr); - kfree(runtime->dma_buffer_p); - } - - snd_pcm_set_runtime_buffer(substream, NULL); -} - -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = runtime->dma_buffer_p; - int ret = 0; - - if (buf) { - if (buf->bytes >= size) - goto out; - devdma_hw_free(dev, substream); - } - - if (substream->dma_buffer.area != NULL && substream->dma_buffer.bytes >= size) { - buf = &substream->dma_buffer; - } else { - buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); - if (!buf) - goto nomem; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = dev; - buf->area = dma_alloc_coherent(dev, size, &buf->addr, GFP_KERNEL); - buf->bytes = size; - buf->private_data = NULL; - - if (!buf->area) - goto free; - } - snd_pcm_set_runtime_buffer(substream, buf); - ret = 1; - out: - runtime->dma_bytes = size; - return ret; - - free: - kfree(buf); - nomem: - return -ENOMEM; -} - -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - return dma_mmap_coherent(dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); -} diff --git a/sound/arm/devdma.h b/sound/arm/devdma.h deleted file mode 100644 index d025329c8a0..00000000000 --- a/sound/arm/devdma.h +++ /dev/null @@ -1,3 +0,0 @@ -void devdma_hw_free(struct device *dev, struct snd_pcm_substream *substream); -int devdma_hw_alloc(struct device *dev, struct snd_pcm_substream *substream, size_t size); -int devdma_mmap(struct device *dev, struct snd_pcm_substream *substream, struct vm_area_struct *vma); diff --git a/sound/core/control.c b/sound/core/control.c index a8b7fabe645..268ab747122 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -75,7 +75,7 @@ static int snd_ctl_open(struct inode *inode, struct file *file) ctl->card = card; ctl->prefer_pcm_subdevice = -1; ctl->prefer_rawmidi_subdevice = -1; - ctl->pid = current->pid; + ctl->pid = get_pid(task_pid(current)); file->private_data = ctl; write_lock_irqsave(&card->ctl_files_rwlock, flags); list_add_tail(&ctl->list, &card->ctl_files); @@ -125,6 +125,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); snd_ctl_empty_read_queue(ctl); + put_pid(ctl->pid); kfree(ctl); module_put(card->module); snd_card_file_remove(card, file); @@ -672,7 +673,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, info->access |= SNDRV_CTL_ELEM_ACCESS_LOCK; if (vd->owner == ctl) info->access |= SNDRV_CTL_ELEM_ACCESS_OWNER; - info->owner = vd->owner_pid; + info->owner = pid_vnr(vd->owner->pid); } else { info->owner = -1; } @@ -827,7 +828,6 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, result = -EBUSY; else { vd->owner = file; - vd->owner_pid = current->pid; result = 0; } } @@ -858,7 +858,6 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, result = -EPERM; else { vd->owner = NULL; - vd->owner_pid = 0; result = 0; } } @@ -1120,7 +1119,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, goto __kctl_end; } if (vd->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - if (file && vd->owner != NULL && vd->owner != file) { + if (vd->owner != NULL && vd->owner != file) { err = -EPERM; goto __kctl_end; } diff --git a/sound/core/isadma.c b/sound/core/isadma.c index 79f0f16af33..950e19ba91f 100644 --- a/sound/core/isadma.c +++ b/sound/core/isadma.c @@ -85,16 +85,24 @@ EXPORT_SYMBOL(snd_dma_disable); unsigned int snd_dma_pointer(unsigned long dma, unsigned int size) { unsigned long flags; - unsigned int result; + unsigned int result, result1; flags = claim_dma_lock(); clear_dma_ff(dma); if (!isa_dma_bridge_buggy) disable_dma(dma); result = get_dma_residue(dma); + /* + * HACK - read the counter again and choose higher value in order to + * avoid reading during counter lower byte roll over if the + * isa_dma_bridge_buggy is set. + */ + result1 = get_dma_residue(dma); if (!isa_dma_bridge_buggy) enable_dma(dma); release_dma_lock(flags); + if (unlikely(result < result1)) + result = result1; #ifdef CONFIG_SND_DEBUG if (result > size) snd_printk(KERN_ERR "pointer (0x%x) for DMA #%ld is greater than transfer size (0x%x)\n", result, dma, size); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 772423889eb..54e2eb56e4c 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1251,7 +1251,9 @@ static void snd_mixer_oss_build(struct snd_mixer_oss *mixer) { SOUND_MIXER_SYNTH, "FM", 0 }, /* fallback */ { SOUND_MIXER_SYNTH, "Music", 0 }, /* fallback */ { SOUND_MIXER_PCM, "PCM", 0 }, - { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, + { SOUND_MIXER_SPEAKER, "Beep", 0 }, + { SOUND_MIXER_SPEAKER, "PC Speaker", 0 }, /* fallback */ + { SOUND_MIXER_SPEAKER, "Speaker", 0 }, /* fallback */ { SOUND_MIXER_LINE, "Line", 0 }, { SOUND_MIXER_MIC, "Mic", 0 }, { SOUND_MIXER_CD, "CD", 0 }, diff --git a/sound/core/pcm.c b/sound/core/pcm.c index c69c60b2a48..6884ae031f6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -435,6 +435,7 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, return; } snd_iprintf(buffer, "state: %s\n", snd_pcm_state_name(status.state)); + snd_iprintf(buffer, "owner_pid : %d\n", pid_vnr(substream->pid)); snd_iprintf(buffer, "trigger_time: %ld.%09ld\n", status.trigger_tstamp.tv_sec, status.trigger_tstamp.tv_nsec); snd_iprintf(buffer, "tstamp : %ld.%09ld\n", @@ -809,7 +810,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, card = pcm->card; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { prefer_subdevice = kctl->prefer_pcm_subdevice; if (prefer_subdevice != -1) break; @@ -900,6 +901,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->private_data = pcm->private_data; substream->ref_count = 1; substream->f_flags = file->f_flags; + substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; return 0; @@ -921,6 +923,8 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) kfree(runtime->hw_constraints.rules); kfree(runtime); substream->runtime = NULL; + put_pid(substream->pid); + substream->pid = NULL; substream->pstr->substream_opened--; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ab73edf2c89..29ab46a12e1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -26,6 +26,7 @@ #include <linux/time.h> #include <linux/pm_qos_params.h> #include <linux/uio.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/control.h> #include <sound/info.h> @@ -3061,6 +3062,27 @@ static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file } #endif /* coherent mmap */ +static inline struct page * +snd_pcm_default_page_ops(struct snd_pcm_substream *substream, unsigned long ofs) +{ + void *vaddr = substream->runtime->dma_area + ofs; +#if defined(CONFIG_MIPS) && defined(CONFIG_DMA_NONCOHERENT) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return virt_to_page(CAC_ADDR(vaddr)); +#endif +#if defined(CONFIG_PPC32) && defined(CONFIG_NOT_COHERENT_CACHE) + if (substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) { + dma_addr_t addr = substream->runtime->dma_addr + ofs; + addr -= get_dma_offset(substream->dma_buffer.dev.dev); + /* assume dma_handle set via pfn_to_phys() in + * mm/dma-noncoherent.c + */ + return pfn_to_page(addr >> PAGE_SHIFT); + } +#endif + return virt_to_page(vaddr); +} + /* * fault callback for mmapping a RAM page */ @@ -3071,7 +3093,6 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, struct snd_pcm_runtime *runtime; unsigned long offset; struct page * page; - void *vaddr; size_t dma_bytes; if (substream == NULL) @@ -3081,36 +3102,53 @@ static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) return VM_FAULT_SIGBUS; - if (substream->ops->page) { + if (substream->ops->page) page = substream->ops->page(substream, offset); - if (!page) - return VM_FAULT_SIGBUS; - } else { - vaddr = runtime->dma_area + offset; - page = virt_to_page(vaddr); - } + else + page = snd_pcm_default_page_ops(substream, offset); + if (!page) + return VM_FAULT_SIGBUS; get_page(page); vmf->page = page; return 0; } -static const struct vm_operations_struct snd_pcm_vm_ops_data = -{ +static const struct vm_operations_struct snd_pcm_vm_ops_data = { + .open = snd_pcm_mmap_data_open, + .close = snd_pcm_mmap_data_close, +}; + +static const struct vm_operations_struct snd_pcm_vm_ops_data_fault = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, .fault = snd_pcm_mmap_data_fault, }; +#ifndef ARCH_HAS_DMA_MMAP_COHERENT +/* This should be defined / handled globally! */ +#ifdef CONFIG_ARM +#define ARCH_HAS_DMA_MMAP_COHERENT +#endif +#endif + /* * mmap the DMA buffer on RAM */ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *area) { - area->vm_ops = &snd_pcm_vm_ops_data; - area->vm_private_data = substream; area->vm_flags |= VM_RESERVED; - atomic_inc(&substream->mmap_count); +#ifdef ARCH_HAS_DMA_MMAP_COHERENT + if (!substream->ops->page && + substream->dma_buffer.dev.type == SNDRV_DMA_TYPE_DEV) + return dma_mmap_coherent(substream->dma_buffer.dev.dev, + area, + substream->runtime->dma_area, + substream->runtime->dma_addr, + area->vm_end - area->vm_start); +#endif /* ARCH_HAS_DMA_MMAP_COHERENT */ + /* mmap with fault handler */ + area->vm_ops = &snd_pcm_vm_ops_data_fault; return 0; } @@ -3118,12 +3156,6 @@ static int snd_pcm_default_mmap(struct snd_pcm_substream *substream, * mmap the DMA buffer on I/O memory area */ #if SNDRV_PCM_INFO_MMAP_IOMEM -static const struct vm_operations_struct snd_pcm_vm_ops_data_mmio = -{ - .open = snd_pcm_mmap_data_open, - .close = snd_pcm_mmap_data_close, -}; - int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_struct *area) { @@ -3133,8 +3165,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, #ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); #endif - area->vm_ops = &snd_pcm_vm_ops_data_mmio; - area->vm_private_data = substream; area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; @@ -3142,7 +3172,6 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, (substream->runtime->dma_addr + offset) >> PAGE_SHIFT, size, area->vm_page_prot)) return -EAGAIN; - atomic_inc(&substream->mmap_count); return 0; } @@ -3159,6 +3188,7 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, long size; unsigned long offset; size_t dma_bytes; + int err; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (!(area->vm_flags & (VM_WRITE|VM_READ))) @@ -3183,10 +3213,15 @@ int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, if (offset > dma_bytes - size) return -EINVAL; + area->vm_ops = &snd_pcm_vm_ops_data; + area->vm_private_data = substream; if (substream->ops->mmap) - return substream->ops->mmap(substream, area); + err = substream->ops->mmap(substream, area); else - return snd_pcm_default_mmap(substream, area); + err = snd_pcm_default_mmap(substream, area); + if (!err) + atomic_inc(&substream->mmap_count); + return err; } EXPORT_SYMBOL(snd_pcm_mmap_data); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 70d6f25ba52..2f766123b15 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -242,8 +242,6 @@ static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, return -ENXIO; if (subdevice >= 0 && subdevice >= s->substream_count) return -ENODEV; - if (s->substream_opened >= s->substream_count) - return -EAGAIN; list_for_each_entry(substream, &s->substreams, list) { if (substream->opened) { @@ -280,9 +278,10 @@ static int open_substream(struct snd_rawmidi *rmidi, substream->active_sensing = 0; if (mode & SNDRV_RAWMIDI_LFLG_APPEND) substream->append = 1; + substream->pid = get_pid(task_pid(current)); + rmidi->streams[substream->stream].substream_opened++; } substream->use_count++; - rmidi->streams[substream->stream].substream_opened++; return 0; } @@ -413,7 +412,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) subdevice = -1; read_lock(&card->ctl_files_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { - if (kctl->pid == current->pid) { + if (kctl->pid == task_pid(current)) { subdevice = kctl->prefer_rawmidi_subdevice; if (subdevice != -1) break; @@ -466,7 +465,6 @@ static void close_substream(struct snd_rawmidi *rmidi, struct snd_rawmidi_substream *substream, int cleanup) { - rmidi->streams[substream->stream].substream_opened--; if (--substream->use_count) return; @@ -491,6 +489,9 @@ static void close_substream(struct snd_rawmidi *rmidi, snd_rawmidi_runtime_free(substream); substream->opened = 0; substream->append = 0; + put_pid(substream->pid); + substream->pid = NULL; + rmidi->streams[substream->stream].substream_opened--; } static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) @@ -1338,6 +1339,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Mode : %s\n" @@ -1359,6 +1363,9 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, substream->number, (unsigned long) substream->bytes); if (substream->opened) { + snd_iprintf(buffer, + " Owner PID : %d\n", + pid_vnr(substream->pid)); runtime = substream->runtime; snd_iprintf(buffer, " Buffer size : %lu\n" diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index b60cef257b5..f165c77d627 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -26,6 +26,7 @@ MODULE_ALIAS("platform:pcspkr"); static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +static int nopcm; /* Disable PCM capability of the driver */ module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); @@ -33,6 +34,8 @@ module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable PC-Speaker sound."); +module_param(nopcm, bool, 0444); +MODULE_PARM_DESC(nopcm, "Disable PC-Speaker PCM sound. Only beeps remain."); struct snd_pcsp pcsp_chip; @@ -43,13 +46,16 @@ static int __devinit snd_pcsp_create(struct snd_card *card) int err; int div, min_div, order; - hrtimer_get_res(CLOCK_MONOTONIC, &tp); - if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { - printk(KERN_ERR "PCSP: Timer resolution is not sufficient " - "(%linS)\n", tp.tv_nsec); - printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " - "enabled.\n"); - return -EIO; + if (!nopcm) { + hrtimer_get_res(CLOCK_MONOTONIC, &tp); + if (tp.tv_sec || tp.tv_nsec > PCSP_MAX_PERIOD_NS) { + printk(KERN_ERR "PCSP: Timer resolution is not sufficient " + "(%linS)\n", tp.tv_nsec); + printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " + "enabled.\n"); + printk(KERN_ERR "PCSP: Turned into nopcm mode.\n"); + nopcm = 1; + } } if (loops_per_jiffy >= PCSP_MIN_LPJ && tp.tv_nsec <= PCSP_MIN_PERIOD_NS) @@ -107,12 +113,14 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev) snd_card_free(card); return err; } - err = snd_pcsp_new_pcm(&pcsp_chip); - if (err < 0) { - snd_card_free(card); - return err; + if (!nopcm) { + err = snd_pcsp_new_pcm(&pcsp_chip); + if (err < 0) { + snd_card_free(card); + return err; + } } - err = snd_pcsp_new_mixer(&pcsp_chip); + err = snd_pcsp_new_mixer(&pcsp_chip, nopcm); if (err < 0) { snd_card_free(card); return err; diff --git a/sound/drivers/pcsp/pcsp.h b/sound/drivers/pcsp/pcsp.h index 174dd2ff0f2..1e123077923 100644 --- a/sound/drivers/pcsp/pcsp.h +++ b/sound/drivers/pcsp/pcsp.h @@ -83,6 +83,6 @@ extern enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle); extern void pcsp_sync_stop(struct snd_pcsp *chip); extern int snd_pcsp_new_pcm(struct snd_pcsp *chip); -extern int snd_pcsp_new_mixer(struct snd_pcsp *chip); +extern int snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm); #endif diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index 903bc846763..6f633f4f3b9 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -119,24 +119,43 @@ static int pcsp_pcspkr_put(struct snd_kcontrol *kcontrol, .put = pcsp_##ctl_type##_put, \ } -static struct snd_kcontrol_new __devinitdata snd_pcsp_controls[] = { +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_pcm[] = { PCSP_MIXER_CONTROL(enable, "Master Playback Switch"), PCSP_MIXER_CONTROL(treble, "BaseFRQ Playback Volume"), - PCSP_MIXER_CONTROL(pcspkr, "PC Speaker Playback Switch"), }; -int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip) +static struct snd_kcontrol_new __devinitdata snd_pcsp_controls_spkr[] = { + PCSP_MIXER_CONTROL(pcspkr, "Beep Playback Switch"), +}; + +static int __devinit snd_pcsp_ctls_add(struct snd_pcsp *chip, + struct snd_kcontrol_new *ctls, int num) { - struct snd_card *card = chip->card; int i, err; + struct snd_card *card = chip->card; + for (i = 0; i < num; i++) { + err = snd_ctl_add(card, snd_ctl_new1(ctls + i, chip)); + if (err < 0) + return err; + } + return 0; +} + +int __devinit snd_pcsp_new_mixer(struct snd_pcsp *chip, int nopcm) +{ + int err; + struct snd_card *card = chip->card; - for (i = 0; i < ARRAY_SIZE(snd_pcsp_controls); i++) { - err = snd_ctl_add(card, - snd_ctl_new1(snd_pcsp_controls + i, - chip)); + if (!nopcm) { + err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_pcm, + ARRAY_SIZE(snd_pcsp_controls_pcm)); if (err < 0) return err; } + err = snd_pcsp_ctls_add(chip, snd_pcsp_controls_spkr, + ARRAY_SIZE(snd_pcsp_controls_spkr)); + if (err < 0) + return err; strcpy(card->mixername, "PC-Speaker"); diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 020a5d51247..04ae8704cdc 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -23,6 +23,7 @@ #include <linux/slab.h> #include <linux/delay.h> #include <linux/init.h> +#include <linux/bitrev.h> #include <asm/unaligned.h> #include <sound/core.h> #include <sound/control.h> @@ -55,18 +56,6 @@ struct cs8427 { struct cs8427_stream capture; }; -static unsigned char swapbits(unsigned char val) -{ - int bit; - unsigned char res = 0; - for (bit = 0; bit < 8; bit++) { - res <<= 1; - res |= val & 1; - val >>= 1; - } - return res; -} - int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg, unsigned char val) { @@ -149,7 +138,7 @@ static int snd_cs8427_send_corudata(struct snd_i2c_device *device, } data[0] = CS8427_REG_AUTOINC | CS8427_REG_CORU_DATABUF; for (idx = 0; idx < count; idx++) - data[idx + 1] = swapbits(ndata[idx]); + data[idx + 1] = bitrev8(ndata[idx]); if (snd_i2c_sendbytes(device, data, count + 1) != count + 1) return -EIO; return 1; diff --git a/sound/i2c/other/Makefile b/sound/i2c/other/Makefile index 703d954238f..2dad40f3f62 100644 --- a/sound/i2c/other/Makefile +++ b/sound/i2c/other/Makefile @@ -5,6 +5,7 @@ snd-ak4114-objs := ak4114.o snd-ak4117-objs := ak4117.o +snd-ak4113-objs := ak4113.o snd-ak4xxx-adda-objs := ak4xxx-adda.o snd-pt2258-objs := pt2258.o snd-tea575x-tuner-objs := tea575x-tuner.o @@ -12,5 +13,5 @@ snd-tea575x-tuner-objs := tea575x-tuner.o # Module Dependency obj-$(CONFIG_SND_PDAUDIOCF) += snd-ak4117.o obj-$(CONFIG_SND_ICE1712) += snd-ak4xxx-adda.o -obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4xxx-adda.o snd-pt2258.o +obj-$(CONFIG_SND_ICE1724) += snd-ak4114.o snd-ak4113.o snd-ak4xxx-adda.o snd-pt2258.o obj-$(CONFIG_SND_FM801_TEA575X) += snd-tea575x-tuner.o diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c new file mode 100644 index 00000000000..fff62cc8607 --- /dev/null +++ b/sound/i2c/other/ak4113.c @@ -0,0 +1,639 @@ +/* + * Routines for control of the AK4113 via I2C/4-wire serial interface + * IEC958 (S/PDIF) receiver by Asahi Kasei + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * Copyright (c) by Pavel Hofman <pavel.hofman@ivitera.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/slab.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/ak4113.h> +#include <sound/asoundef.h> +#include <sound/info.h> + +MODULE_AUTHOR("Pavel Hofman <pavel.hofman@ivitera.com>"); +MODULE_DESCRIPTION("AK4113 IEC958 (S/PDIF) receiver by Asahi Kasei"); +MODULE_LICENSE("GPL"); + +#define AK4113_ADDR 0x00 /* fixed address */ + +static void ak4113_stats(struct work_struct *work); +static void ak4113_init_regs(struct ak4113 *chip); + + +static void reg_write(struct ak4113 *ak4113, unsigned char reg, + unsigned char val) +{ + ak4113->write(ak4113->private_data, reg, val); + if (reg < sizeof(ak4113->regmap)) + ak4113->regmap[reg] = val; +} + +static inline unsigned char reg_read(struct ak4113 *ak4113, unsigned char reg) +{ + return ak4113->read(ak4113->private_data, reg); +} + +static void snd_ak4113_free(struct ak4113 *chip) +{ + chip->init = 1; /* don't schedule new work */ + mb(); + cancel_delayed_work(&chip->work); + flush_scheduled_work(); + kfree(chip); +} + +static int snd_ak4113_dev_free(struct snd_device *device) +{ + struct ak4113 *chip = device->device_data; + snd_ak4113_free(chip); + return 0; +} + +int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, + ak4113_write_t *write, const unsigned char pgm[5], + void *private_data, struct ak4113 **r_ak4113) +{ + struct ak4113 *chip; + int err = 0; + unsigned char reg; + static struct snd_device_ops ops = { + .dev_free = snd_ak4113_dev_free, + }; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + spin_lock_init(&chip->lock); + chip->card = card; + chip->read = read; + chip->write = write; + chip->private_data = private_data; + INIT_DELAYED_WORK(&chip->work, ak4113_stats); + + for (reg = 0; reg < AK4113_WRITABLE_REGS ; reg++) + chip->regmap[reg] = pgm[reg]; + ak4113_init_regs(chip); + + chip->rcs0 = reg_read(chip, AK4113_REG_RCS0) & ~(AK4113_QINT | + AK4113_CINT | AK4113_STC); + chip->rcs1 = reg_read(chip, AK4113_REG_RCS1); + chip->rcs2 = reg_read(chip, AK4113_REG_RCS2); + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto __fail; + + if (r_ak4113) + *r_ak4113 = chip; + return 0; + +__fail: + snd_ak4113_free(chip); + return err < 0 ? err : -EIO; +} +EXPORT_SYMBOL_GPL(snd_ak4113_create); + +void snd_ak4113_reg_write(struct ak4113 *chip, unsigned char reg, + unsigned char mask, unsigned char val) +{ + if (reg >= AK4113_WRITABLE_REGS) + return; + reg_write(chip, reg, (chip->regmap[reg] & ~mask) | val); +} +EXPORT_SYMBOL_GPL(snd_ak4113_reg_write); + +static void ak4113_init_regs(struct ak4113 *chip) +{ + unsigned char old = chip->regmap[AK4113_REG_PWRDN], reg; + + /* bring the chip to reset state and powerdown state */ + reg_write(chip, AK4113_REG_PWRDN, old & ~(AK4113_RST|AK4113_PWN)); + udelay(200); + /* release reset, but leave powerdown */ + reg_write(chip, AK4113_REG_PWRDN, (old | AK4113_RST) & ~AK4113_PWN); + udelay(200); + for (reg = 1; reg < AK4113_WRITABLE_REGS; reg++) + reg_write(chip, reg, chip->regmap[reg]); + /* release powerdown, everything is initialized now */ + reg_write(chip, AK4113_REG_PWRDN, old | AK4113_RST | AK4113_PWN); +} + +void snd_ak4113_reinit(struct ak4113 *chip) +{ + chip->init = 1; + mb(); + flush_scheduled_work(); + ak4113_init_regs(chip); + /* bring up statistics / event queing */ + chip->init = 0; + if (chip->kctls[0]) + schedule_delayed_work(&chip->work, HZ / 10); +} +EXPORT_SYMBOL_GPL(snd_ak4113_reinit); + +static unsigned int external_rate(unsigned char rcs1) +{ + switch (rcs1 & (AK4113_FS0|AK4113_FS1|AK4113_FS2|AK4113_FS3)) { + case AK4113_FS_8000HZ: + return 8000; + case AK4113_FS_11025HZ: + return 11025; + case AK4113_FS_16000HZ: + return 16000; + case AK4113_FS_22050HZ: + return 22050; + case AK4113_FS_24000HZ: + return 24000; + case AK4113_FS_32000HZ: + return 32000; + case AK4113_FS_44100HZ: + return 44100; + case AK4113_FS_48000HZ: + return 48000; + case AK4113_FS_64000HZ: + return 64000; + case AK4113_FS_88200HZ: + return 88200; + case AK4113_FS_96000HZ: + return 96000; + case AK4113_FS_176400HZ: + return 176400; + case AK4113_FS_192000HZ: + return 192000; + default: + return 0; + } +} + +static int snd_ak4113_in_error_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = LONG_MAX; + return 0; +} + +static int snd_ak4113_in_error_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + long *ptr; + + spin_lock_irq(&chip->lock); + ptr = (long *)(((char *)chip) + kcontrol->private_value); + ucontrol->value.integer.value[0] = *ptr; + *ptr = 0; + spin_unlock_irq(&chip->lock); + return 0; +} + +#define snd_ak4113_in_bit_info snd_ctl_boolean_mono_info + +static int snd_ak4113_in_bit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned char reg = kcontrol->private_value & 0xff; + unsigned char bit = (kcontrol->private_value >> 8) & 0xff; + unsigned char inv = (kcontrol->private_value >> 31) & 1; + + ucontrol->value.integer.value[0] = + ((reg_read(chip, reg) & (1 << bit)) ? 1 : 0) ^ inv; + return 0; +} + +static int snd_ak4113_rx_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 5; + return 0; +} + +static int snd_ak4113_rx_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + (AK4113_IPS(chip->regmap[AK4113_REG_IO1])); + return 0; +} + +static int snd_ak4113_rx_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + int change; + u8 old_val; + + spin_lock_irq(&chip->lock); + old_val = chip->regmap[AK4113_REG_IO1]; + change = ucontrol->value.integer.value[0] != AK4113_IPS(old_val); + if (change) + reg_write(chip, AK4113_REG_IO1, + (old_val & (~AK4113_IPS(0xff))) | + (AK4113_IPS(ucontrol->value.integer.value[0]))); + spin_unlock_irq(&chip->lock); + return change; +} + +static int snd_ak4113_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 192000; + return 0; +} + +static int snd_ak4113_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = external_rate(reg_read(chip, + AK4113_REG_RCS1)); + return 0; +} + +static int snd_ak4113_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_ak4113_spdif_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned i; + + for (i = 0; i < AK4113_REG_RXCSB_SIZE; i++) + ucontrol->value.iec958.status[i] = reg_read(chip, + AK4113_REG_RXCSB0 + i); + return 0; +} + +static int snd_ak4113_spdif_mask_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_ak4113_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + memset(ucontrol->value.iec958.status, 0xff, AK4113_REG_RXCSB_SIZE); + return 0; +} + +static int snd_ak4113_spdif_pinfo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0xffff; + uinfo->count = 4; + return 0; +} + +static int snd_ak4113_spdif_pget(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned short tmp; + + ucontrol->value.integer.value[0] = 0xf8f2; + ucontrol->value.integer.value[1] = 0x4e1f; + tmp = reg_read(chip, AK4113_REG_Pc0) | + (reg_read(chip, AK4113_REG_Pc1) << 8); + ucontrol->value.integer.value[2] = tmp; + tmp = reg_read(chip, AK4113_REG_Pd0) | + (reg_read(chip, AK4113_REG_Pd1) << 8); + ucontrol->value.integer.value[3] = tmp; + return 0; +} + +static int snd_ak4113_spdif_qinfo(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = AK4113_REG_QSUB_SIZE; + return 0; +} + +static int snd_ak4113_spdif_qget(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct ak4113 *chip = snd_kcontrol_chip(kcontrol); + unsigned i; + + for (i = 0; i < AK4113_REG_QSUB_SIZE; i++) + ucontrol->value.bytes.data[i] = reg_read(chip, + AK4113_REG_QSUB_ADDR + i); + return 0; +} + +/* Don't forget to change AK4113_CONTROLS define!!! */ +static struct snd_kcontrol_new snd_ak4113_iec958_controls[] = { +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Parity Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, parity_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 V-Bit Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, v_bit_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 C-CRC Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, ccrc_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Q-CRC Errors", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_error_info, + .get = snd_ak4113_in_error_get, + .private_value = offsetof(struct ak4113, qcrc_errors), +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 External Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_rate_info, + .get = snd_ak4113_rate_get, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, MASK), + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .info = snd_ak4113_spdif_mask_info, + .get = snd_ak4113_spdif_mask_get, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", CAPTURE, DEFAULT), + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_spdif_info, + .get = snd_ak4113_spdif_get, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Preample Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_spdif_pinfo, + .get = snd_ak4113_spdif_pget, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Q-subcode Capture Default", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_spdif_qinfo, + .get = snd_ak4113_spdif_qget, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Audio", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_bit_info, + .get = snd_ak4113_in_bit_get, + .private_value = (1<<31) | (1<<8) | AK4113_REG_RCS0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 Non-PCM Bitstream", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_bit_info, + .get = snd_ak4113_in_bit_get, + .private_value = (0<<8) | AK4113_REG_RCS1, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "IEC958 DTS Bitstream", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_ak4113_in_bit_info, + .get = snd_ak4113_in_bit_get, + .private_value = (1<<8) | AK4113_REG_RCS1, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "AK4113 Input Select", + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_WRITE, + .info = snd_ak4113_rx_info, + .get = snd_ak4113_rx_get, + .put = snd_ak4113_rx_put, +} +}; + +static void snd_ak4113_proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct ak4113 *ak4113 = entry->private_data; + int reg, val; + /* all ak4113 registers 0x00 - 0x1c */ + for (reg = 0; reg < 0x1d; reg++) { + val = reg_read(ak4113, reg); + snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); + } +} + +static void snd_ak4113_proc_init(struct ak4113 *ak4113) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ak4113->card, "ak4113", &entry)) + snd_info_set_text_ops(entry, ak4113, snd_ak4113_proc_regs_read); +} + +int snd_ak4113_build(struct ak4113 *ak4113, + struct snd_pcm_substream *cap_substream) +{ + struct snd_kcontrol *kctl; + unsigned int idx; + int err; + + if (snd_BUG_ON(!cap_substream)) + return -EINVAL; + ak4113->substream = cap_substream; + for (idx = 0; idx < AK4113_CONTROLS; idx++) { + kctl = snd_ctl_new1(&snd_ak4113_iec958_controls[idx], ak4113); + if (kctl == NULL) + return -ENOMEM; + kctl->id.device = cap_substream->pcm->device; + kctl->id.subdevice = cap_substream->number; + err = snd_ctl_add(ak4113->card, kctl); + if (err < 0) + return err; + ak4113->kctls[idx] = kctl; + } + snd_ak4113_proc_init(ak4113); + /* trigger workq */ + schedule_delayed_work(&ak4113->work, HZ / 10); + return 0; +} +EXPORT_SYMBOL_GPL(snd_ak4113_build); + +int snd_ak4113_external_rate(struct ak4113 *ak4113) +{ + unsigned char rcs1; + + rcs1 = reg_read(ak4113, AK4113_REG_RCS1); + return external_rate(rcs1); +} +EXPORT_SYMBOL_GPL(snd_ak4113_external_rate); + +int snd_ak4113_check_rate_and_errors(struct ak4113 *ak4113, unsigned int flags) +{ + struct snd_pcm_runtime *runtime = + ak4113->substream ? ak4113->substream->runtime : NULL; + unsigned long _flags; + int res = 0; + unsigned char rcs0, rcs1, rcs2; + unsigned char c0, c1; + + rcs1 = reg_read(ak4113, AK4113_REG_RCS1); + if (flags & AK4113_CHECK_NO_STAT) + goto __rate; + rcs0 = reg_read(ak4113, AK4113_REG_RCS0); + rcs2 = reg_read(ak4113, AK4113_REG_RCS2); + spin_lock_irqsave(&ak4113->lock, _flags); + if (rcs0 & AK4113_PAR) + ak4113->parity_errors++; + if (rcs0 & AK4113_V) + ak4113->v_bit_errors++; + if (rcs2 & AK4113_CCRC) + ak4113->ccrc_errors++; + if (rcs2 & AK4113_QCRC) + ak4113->qcrc_errors++; + c0 = (ak4113->rcs0 & (AK4113_QINT | AK4113_CINT | AK4113_STC | + AK4113_AUDION | AK4113_AUTO | AK4113_UNLCK)) ^ + (rcs0 & (AK4113_QINT | AK4113_CINT | AK4113_STC | + AK4113_AUDION | AK4113_AUTO | AK4113_UNLCK)); + c1 = (ak4113->rcs1 & (AK4113_DTSCD | AK4113_NPCM | AK4113_PEM | + AK4113_DAT | 0xf0)) ^ + (rcs1 & (AK4113_DTSCD | AK4113_NPCM | AK4113_PEM | + AK4113_DAT | 0xf0)); + ak4113->rcs0 = rcs0 & ~(AK4113_QINT | AK4113_CINT | AK4113_STC); + ak4113->rcs1 = rcs1; + ak4113->rcs2 = rcs2; + spin_unlock_irqrestore(&ak4113->lock, _flags); + + if (rcs0 & AK4113_PAR) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[0]->id); + if (rcs0 & AK4113_V) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[1]->id); + if (rcs2 & AK4113_CCRC) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[2]->id); + if (rcs2 & AK4113_QCRC) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[3]->id); + + /* rate change */ + if (c1 & 0xf0) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[4]->id); + + if ((c1 & AK4113_PEM) | (c0 & AK4113_CINT)) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[6]->id); + if (c0 & AK4113_QINT) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[8]->id); + + if (c0 & AK4113_AUDION) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[9]->id); + if (c1 & AK4113_NPCM) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[10]->id); + if (c1 & AK4113_DTSCD) + snd_ctl_notify(ak4113->card, SNDRV_CTL_EVENT_MASK_VALUE, + &ak4113->kctls[11]->id); + + if (ak4113->change_callback && (c0 | c1) != 0) + ak4113->change_callback(ak4113, c0, c1); + +__rate: + /* compare rate */ + res = external_rate(rcs1); + if (!(flags & AK4113_CHECK_NO_RATE) && runtime && + (runtime->rate != res)) { + snd_pcm_stream_lock_irqsave(ak4113->substream, _flags); + if (snd_pcm_running(ak4113->substream)) { + /*printk(KERN_DEBUG "rate changed (%i <- %i)\n", + * runtime->rate, res); */ + snd_pcm_stop(ak4113->substream, + SNDRV_PCM_STATE_DRAINING); + wake_up(&runtime->sleep); + res = 1; + } + snd_pcm_stream_unlock_irqrestore(ak4113->substream, _flags); + } + return res; +} +EXPORT_SYMBOL_GPL(snd_ak4113_check_rate_and_errors); + +static void ak4113_stats(struct work_struct *work) +{ + struct ak4113 *chip = container_of(work, struct ak4113, work.work); + + if (!chip->init) + snd_ak4113_check_rate_and_errors(chip, chip->check_flags); + + schedule_delayed_work(&chip->work, HZ / 10); +} diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index ee47abab764..1adb8a3c2b6 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -19,7 +19,7 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * - */ + */ #include <asm/io.h> #include <linux/delay.h> @@ -29,6 +29,7 @@ #include <sound/control.h> #include <sound/tlv.h> #include <sound/ak4xxx-adda.h> +#include <sound/info.h> MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, Takashi Iwai <tiwai@suse.de>"); MODULE_DESCRIPTION("Routines for control of AK452x / AK43xx AD/DA converters"); @@ -52,26 +53,21 @@ EXPORT_SYMBOL(snd_akm4xxx_write); static void ak4524_reset(struct snd_akm4xxx *ak, int state) { unsigned int chip; - unsigned char reg, maxreg; + unsigned char reg; - if (ak->type == SND_AK4528) - maxreg = 0x06; - else - maxreg = 0x08; for (chip = 0; chip < ak->num_dacs/2; chip++) { snd_akm4xxx_write(ak, chip, 0x01, state ? 0x00 : 0x03); if (state) continue; /* DAC volumes */ - for (reg = 0x04; reg < maxreg; reg++) + for (reg = 0x04; reg < ak->total_regs; reg++) snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg)); } } /* reset procedure for AK4355 and AK4358 */ -static void ak435X_reset(struct snd_akm4xxx *ak, int state, - unsigned char total_regs) +static void ak435X_reset(struct snd_akm4xxx *ak, int state) { unsigned char reg; @@ -79,7 +75,7 @@ static void ak435X_reset(struct snd_akm4xxx *ak, int state, snd_akm4xxx_write(ak, 0, 0x01, 0x02); /* reset and soft-mute */ return; } - for (reg = 0x00; reg < total_regs; reg++) + for (reg = 0x00; reg < ak->total_regs; reg++) if (reg != 0x01) snd_akm4xxx_write(ak, 0, reg, snd_akm4xxx_get(ak, 0, reg)); @@ -91,12 +87,11 @@ static void ak4381_reset(struct snd_akm4xxx *ak, int state) { unsigned int chip; unsigned char reg; - for (chip = 0; chip < ak->num_dacs/2; chip++) { snd_akm4xxx_write(ak, chip, 0x00, state ? 0x0c : 0x0f); if (state) continue; - for (reg = 0x01; reg < 0x05; reg++) + for (reg = 0x01; reg < ak->total_regs; reg++) snd_akm4xxx_write(ak, chip, reg, snd_akm4xxx_get(ak, chip, reg)); } @@ -113,16 +108,17 @@ void snd_akm4xxx_reset(struct snd_akm4xxx *ak, int state) switch (ak->type) { case SND_AK4524: case SND_AK4528: + case SND_AK4620: ak4524_reset(ak, state); break; case SND_AK4529: /* FIXME: needed for ak4529? */ break; case SND_AK4355: - ak435X_reset(ak, state, 0x0b); + ak435X_reset(ak, state); break; case SND_AK4358: - ak435X_reset(ak, state, 0x10); + ak435X_reset(ak, state); break; case SND_AK4381: ak4381_reset(ak, state); @@ -139,7 +135,7 @@ EXPORT_SYMBOL(snd_akm4xxx_reset); * Volume conversion table for non-linear volumes * from -63.5dB (mute) to 0dB step 0.5dB * - * Used for AK4524 input/ouput attenuation, AK4528, and + * Used for AK4524/AK4620 input/ouput attenuation, AK4528, and * AK5365 input attenuation */ static const unsigned char vol_cvt_datt[128] = { @@ -259,8 +255,22 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) 0x00, 0x0f, /* 0: power-up, un-reset */ 0xff, 0xff }; + static const unsigned char inits_ak4620[] = { + 0x00, 0x07, /* 0: normal */ + 0x01, 0x00, /* 0: reset */ + 0x01, 0x02, /* 1: RSTAD */ + 0x01, 0x03, /* 1: RSTDA */ + 0x01, 0x0f, /* 1: normal */ + 0x02, 0x60, /* 2: 24bit I2S */ + 0x03, 0x01, /* 3: deemphasis off */ + 0x04, 0x00, /* 4: LIN muted */ + 0x05, 0x00, /* 5: RIN muted */ + 0x06, 0x00, /* 6: LOUT muted */ + 0x07, 0x00, /* 7: ROUT muted */ + 0xff, 0xff + }; - int chip, num_chips; + int chip; const unsigned char *ptr, *inits; unsigned char reg, data; @@ -270,42 +280,64 @@ void snd_akm4xxx_init(struct snd_akm4xxx *ak) switch (ak->type) { case SND_AK4524: inits = inits_ak4524; - num_chips = ak->num_dacs / 2; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4524"; + ak->total_regs = 0x08; break; case SND_AK4528: inits = inits_ak4528; - num_chips = ak->num_dacs / 2; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4528"; + ak->total_regs = 0x06; break; case SND_AK4529: inits = inits_ak4529; - num_chips = 1; + ak->num_chips = 1; + ak->name = "ak4529"; + ak->total_regs = 0x0d; break; case SND_AK4355: inits = inits_ak4355; - num_chips = 1; + ak->num_chips = 1; + ak->name = "ak4355"; + ak->total_regs = 0x0b; break; case SND_AK4358: inits = inits_ak4358; - num_chips = 1; + ak->num_chips = 1; + ak->name = "ak4358"; + ak->total_regs = 0x10; break; case SND_AK4381: inits = inits_ak4381; - num_chips = ak->num_dacs / 2; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4381"; + ak->total_regs = 0x05; break; case SND_AK5365: /* FIXME: any init sequence? */ + ak->num_chips = 1; + ak->name = "ak5365"; + ak->total_regs = 0x08; return; + case SND_AK4620: + inits = inits_ak4620; + ak->num_chips = ak->num_dacs / 2; + ak->name = "ak4620"; + ak->total_regs = 0x08; + break; default: snd_BUG(); return; } - for (chip = 0; chip < num_chips; chip++) { + for (chip = 0; chip < ak->num_chips; chip++) { ptr = inits; while (*ptr != 0xff) { reg = *ptr++; data = *ptr++; snd_akm4xxx_write(ak, chip, reg, data); + udelay(10); } } } @@ -688,6 +720,12 @@ static int build_dac_controls(struct snd_akm4xxx *ak) AK_COMPOSE(idx/2, (idx%2) + 3, 0, 255); knew.tlv.p = db_scale_linear; break; + case SND_AK4620: + /* register 6 & 7 */ + knew.private_value = + AK_COMPOSE(idx/2, (idx%2) + 6, 0, 255); + knew.tlv.p = db_scale_linear; + break; default: return -EINVAL; } @@ -704,10 +742,12 @@ static int build_dac_controls(struct snd_akm4xxx *ak) static int build_adc_controls(struct snd_akm4xxx *ak) { - int idx, err, mixer_ch, num_stereo; + int idx, err, mixer_ch, num_stereo, max_steps; struct snd_kcontrol_new knew; mixer_ch = 0; + if (ak->type == SND_AK4528) + return 0; /* no controls */ for (idx = 0; idx < ak->num_adcs;) { memset(&knew, 0, sizeof(knew)); if (! ak->adc_info || ! ak->adc_info[mixer_ch].name) { @@ -733,13 +773,12 @@ static int build_adc_controls(struct snd_akm4xxx *ak) } /* register 4 & 5 */ if (ak->type == SND_AK5365) - knew.private_value = - AK_COMPOSE(idx/2, (idx%2) + 4, 0, 151) | - AK_VOL_CVT | AK_IPGA; + max_steps = 152; else - knew.private_value = - AK_COMPOSE(idx/2, (idx%2) + 4, 0, 163) | - AK_VOL_CVT | AK_IPGA; + max_steps = 164; + knew.private_value = + AK_COMPOSE(idx/2, (idx%2) + 4, 0, max_steps) | + AK_VOL_CVT | AK_IPGA; knew.tlv.p = db_scale_vol_datt; err = snd_ctl_add(ak->card, snd_ctl_new1(&knew, ak)); if (err < 0) @@ -808,6 +847,7 @@ static int build_deemphasis(struct snd_akm4xxx *ak, int num_emphs) switch (ak->type) { case SND_AK4524: case SND_AK4528: + case SND_AK4620: /* register 3 */ knew.private_value = AK_COMPOSE(idx, 3, 0, 0); break; @@ -834,6 +874,35 @@ static int build_deemphasis(struct snd_akm4xxx *ak, int num_emphs) return 0; } +#ifdef CONFIG_PROC_FS +static void proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_akm4xxx *ak = (struct snd_akm4xxx *)entry->private_data; + int reg, val, chip; + for (chip = 0; chip < ak->num_chips; chip++) { + for (reg = 0; reg < ak->total_regs; reg++) { + val = snd_akm4xxx_get(ak, chip, reg); + snd_iprintf(buffer, "chip %d: 0x%02x = 0x%02x\n", chip, + reg, val); + } + } +} + +static int proc_init(struct snd_akm4xxx *ak) +{ + struct snd_info_entry *entry; + int err; + err = snd_card_proc_new(ak->card, ak->name, &entry); + if (err < 0) + return err; + snd_info_set_text_ops(entry, ak, proc_regs_read); + return 0; +} +#else /* !CONFIG_PROC_FS */ +static int proc_init(struct snd_akm4xxx *ak) {} +#endif + int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) { int err, num_emphs; @@ -845,18 +914,21 @@ int snd_akm4xxx_build_controls(struct snd_akm4xxx *ak) err = build_adc_controls(ak); if (err < 0) return err; - if (ak->type == SND_AK4355 || ak->type == SND_AK4358) num_emphs = 1; + else if (ak->type == SND_AK4620) + num_emphs = 0; else num_emphs = ak->num_dacs / 2; err = build_deemphasis(ak, num_emphs); if (err < 0) return err; + err = proc_init(ak); + if (err < 0) + return err; return 0; } - EXPORT_SYMBOL(snd_akm4xxx_build_controls); static int __init alsa_akm4xxx_module_init(void) diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index d31c373e076..c4c6ef73f9b 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -225,7 +225,7 @@ static int vidioc_s_ctrl(struct file *file, void *priv, case V4L2_CID_AUDIO_MUTE: if (tea->ops->mute) { tea->ops->mute(tea, ctrl->value); - tea->mute = 1; + tea->mute = ctrl->value; return 0; } } diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 51a7e3777e1..02fe81ca88f 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -372,15 +372,21 @@ config SND_SGALAXY config SND_SSCAPE tristate "Ensoniq SoundScape driver" - select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB + select FW_LOADER help Say Y here to include support for Ensoniq SoundScape - soundcards. + and Ensoniq OEM soundcards. The PCM audio is supported on SoundScape Classic, Elite, PnP - and VIVO cards. The MIDI support is very experimental. + and VIVO cards. The supported OEM cards are SPEA Media FX and + Reveal SC-600. + The MIDI support is very experimental and requires binary + firmware files called "scope.cod" and "sndscape.co?" where the + ? is digit 0, 1, 2, 3 or 4. The firmware files can be found + in DOS or Windows driver packages. One has to put the firmware + files into the /lib/firmware directory. To compile this driver as a module, choose M here: the module will be called snd-sscape. diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 02f79d25271..8246aae32ab 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -237,7 +237,7 @@ WSS_DOUBLE("Wavetable Capture Volume", 0, CMI8330_WAVGAIN, CMI8330_WAVGAIN, 4, 0, 15, 0), WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), -WSS_SINGLE("PC Speaker Playback Volume", 0, +WSS_SINGLE("Beep Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), WSS_DOUBLE("FM Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), @@ -262,7 +262,7 @@ SB_DOUBLE("SB Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, SB_DOUBLE("SB Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31), SB_SINGLE("SB Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1), SB_SINGLE("SB Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), -SB_SINGLE("SB PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3), +SB_SINGLE("SB Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3), SB_DOUBLE("SB Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3), SB_DOUBLE("SB Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3), SB_SINGLE("SB Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1), diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index a076a6ce807..93fa6720d19 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -394,21 +394,15 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) return -EBUSY; } - err = snd_wss_create(card, port[dev], cport[dev], + err = snd_cs4236_create(card, port[dev], cport[dev], irq[dev], dma1[dev], dma2[dev], WSS_HW_DETECT3, 0, &chip); if (err < 0) return err; + + acard->chip = chip; if (chip->hardware & WSS_HW_CS4236B_MASK) { - snd_wss_free(chip); - err = snd_cs4236_create(card, - port[dev], cport[dev], - irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; err = snd_cs4236_pcm(chip, 0, &pcm); if (err < 0) @@ -418,7 +412,6 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) if (err < 0) return err; } else { - acard->chip = chip; err = snd_wss_pcm(chip, 0, &pcm); if (err < 0) return err; diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 38835f31298..c5adca30063 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -87,6 +87,8 @@ #include <sound/core.h> #include <sound/wss.h> #include <sound/asoundef.h> +#include <sound/initval.h> +#include <sound/tlv.h> /* * @@ -264,7 +266,10 @@ static void snd_cs4236_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ - +/* + * This function does no fail if the chip is not CS4236B or compatible. + * It just an equivalent to the snd_wss_create() then. + */ int snd_cs4236_create(struct snd_card *card, unsigned long port, unsigned long cport, @@ -281,21 +286,17 @@ int snd_cs4236_create(struct snd_card *card, *rchip = NULL; if (hardware == WSS_HW_DETECT) hardware = WSS_HW_DETECT3; - if (cport < 0x100) { - snd_printk(KERN_ERR "please, specify control port " - "for CS4236+ chips\n"); - return -ENODEV; - } + err = snd_wss_create(card, port, cport, irq, dma1, dma2, hardware, hwshare, &chip); if (err < 0) return err; - if (!(chip->hardware & WSS_HW_CS4236B_MASK)) { - snd_printk(KERN_ERR "CS4236+: MODE3 and extended registers " - "not available, hardware=0x%x\n", chip->hardware); - snd_device_free(card, chip); - return -ENODEV; + if ((chip->hardware & WSS_HW_CS4236B_MASK) == 0) { + snd_printd("chip is not CS4236+, hardware=0x%x\n", + chip->hardware); + *rchip = chip; + return 0; } #if 0 { @@ -308,9 +309,16 @@ int snd_cs4236_create(struct snd_card *card, idx, snd_cs4236_ctrl_in(chip, idx)); } #endif + if (cport < 0x100 || cport == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please, specify control port " + "for CS4236+ chips\n"); + snd_device_free(card, chip); + return -ENODEV; + } ver1 = snd_cs4236_ctrl_in(chip, 1); ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION); - snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2); + snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", + cport, ver1, ver2); if (ver1 != ver2) { snd_printk(KERN_ERR "CS4236+ chip detected, but " "control port 0x%lx is not valid\n", cport); @@ -321,13 +329,17 @@ int snd_cs4236_create(struct snd_card *card, snd_cs4236_ctrl_out(chip, 2, 0xff); snd_cs4236_ctrl_out(chip, 3, 0x00); snd_cs4236_ctrl_out(chip, 4, 0x80); - snd_cs4236_ctrl_out(chip, 5, ((IEC958_AES1_CON_PCM_CODER & 3) << 6) | IEC958_AES0_CON_EMPHASIS_NONE); + reg = ((IEC958_AES1_CON_PCM_CODER & 3) << 6) | + IEC958_AES0_CON_EMPHASIS_NONE; + snd_cs4236_ctrl_out(chip, 5, reg); snd_cs4236_ctrl_out(chip, 6, IEC958_AES1_CON_PCM_CODER >> 2); snd_cs4236_ctrl_out(chip, 7, 0x00); - /* 0x8c for C8 is valid for Turtle Beach Malibu - the IEC-958 output */ - /* is working with this setup, other hardware should have */ - /* different signal paths and this value should be selectable */ - /* in the future */ + /* + * 0x8c for C8 is valid for Turtle Beach Malibu - the IEC-958 + * output is working with this setup, other hardware should + * have different signal paths and this value should be + * selectable in the future + */ snd_cs4236_ctrl_out(chip, 8, 0x8c); chip->rate_constraint = snd_cs4236_xrate; chip->set_playback_format = snd_cs4236_playback_format; @@ -339,9 +351,10 @@ int snd_cs4236_create(struct snd_card *card, /* initialize extended registers */ for (reg = 0; reg < sizeof(snd_cs4236_ext_map); reg++) - snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), snd_cs4236_ext_map[reg]); + snd_cs4236_ext_out(chip, CS4236_I23VAL(reg), + snd_cs4236_ext_map[reg]); - /* initialize compatible but more featured registers */ + /* initialize compatible but more featured registers */ snd_wss_out(chip, CS4231_LEFT_INPUT, 0x40); snd_wss_out(chip, CS4231_RIGHT_INPUT, 0x40); snd_wss_out(chip, CS4231_AUX1_LEFT_INPUT, 0xff); @@ -387,6 +400,14 @@ int snd_cs4236_pcm(struct snd_wss *chip, int device, struct snd_pcm **rpcm) .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } +#define CS4236_SINGLE_TLV(xname, xindex, reg, shift, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_single, \ + .get = snd_cs4236_get_single, .put = snd_cs4236_put_single, \ + .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 16) & 0xff; @@ -490,6 +511,16 @@ static int snd_cs4236_put_singlec(struct snd_kcontrol *kcontrol, struct snd_ctl_ .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double, .put = snd_cs4236_put_double, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { int mask = (kcontrol->private_value >> 24) & 0xff; @@ -560,12 +591,23 @@ static int snd_cs4236_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } -#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +#define CS4236_DOUBLE1(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +#define CS4236_DOUBLE1_TLV(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert, xtlv) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_cs4236_info_double, \ + .get = snd_cs4236_get_double1, .put = snd_cs4236_put_double1, \ + .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | \ + (shift_right << 19) | (mask << 24) | (invert << 22), \ + .tlv = { .p = (xtlv) } } + static int snd_cs4236_get_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_wss *chip = snd_kcontrol_chip(kcontrol); @@ -619,16 +661,18 @@ static int snd_cs4236_put_double1(struct snd_kcontrol *kcontrol, struct snd_ctl_ return change; } -#define CS4236_MASTER_DIGITAL(xname, xindex) \ +#define CS4236_MASTER_DIGITAL(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4236_get_master_digital, .put = snd_cs4236_put_master_digital, \ - .private_value = 71 << 24 } + .private_value = 71 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4236_mixer_master_digital_invert_volume(int vol) { return (vol < 64) ? 63 - vol : 64 + (71 - vol); -} +} static int snd_cs4236_get_master_digital(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -661,11 +705,13 @@ static int snd_cs4236_put_master_digital(struct snd_kcontrol *kcontrol, struct s return change; } -#define CS4235_OUTPUT_ACCU(xname, xindex) \ +#define CS4235_OUTPUT_ACCU(xname, xindex, xtlv) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ .info = snd_cs4236_info_double, \ .get = snd_cs4235_get_output_accu, .put = snd_cs4235_put_output_accu, \ - .private_value = 3 << 24 } + .private_value = 3 << 24, \ + .tlv = { .p = (xtlv) } } static inline int snd_cs4235_mixer_output_accu_get_volume(int vol) { @@ -720,41 +766,56 @@ static int snd_cs4235_put_output_accu(struct snd_kcontrol *kcontrol, struct snd_ return change; } +static const DECLARE_TLV_DB_SCALE(db_scale_7bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_6bit_12db_max, -8250, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_22db_max, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); + static struct snd_kcontrol_new snd_cs4236_controls[] = { CS4236_DOUBLE("Master Digital Playback Switch", 0, CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), CS4236_DOUBLE("Master Digital Capture Switch", 0, CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), +CS4236_MASTER_DIGITAL("Master Digital Volume", 0, db_scale_7bit), -CS4236_DOUBLE("Capture Boost Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Boost Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE_TLV("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Playback Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), -CS4236_DOUBLE("DSP Playback Volume", 0, - CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("DSP Playback Volume", 0, + CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("FM Playback Switch", 0, CS4236_LEFT_FM, CS4236_RIGHT_FM, 7, 7, 1, 1), -CS4236_DOUBLE("FM Playback Volume", 0, - CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("FM Playback Volume", 0, + CS4236_LEFT_FM, CS4236_RIGHT_FM, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("Wavetable Playback Switch", 0, CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 7, 7, 1, 1), -CS4236_DOUBLE("Wavetable Playback Volume", 0, - CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1), +CS4236_DOUBLE_TLV("Wavetable Playback Volume", 0, + CS4236_LEFT_WAVE, CS4236_RIGHT_WAVE, 0, 0, 63, 1, + db_scale_6bit_12db_max), WSS_DOUBLE("Synth Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Synth Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Synth Capture Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), WSS_DOUBLE("Synth Capture Bypass", 0, @@ -764,14 +825,16 @@ CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), -CS4236_DOUBLE("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 0, 0, 31, 1), -CS4236_DOUBLE("Mic Playback Boost", 0, +CS4236_DOUBLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, + 0, 0, 31, 1, db_scale_5bit_22db_max), +CS4236_DOUBLE("Mic Playback Boost (+20dB)", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 5, 5, 1, 0), WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Line Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), WSS_DOUBLE("Line Capture Bypass", 0, @@ -779,57 +842,63 @@ WSS_DOUBLE("Line Capture Bypass", 0, WSS_DOUBLE("CD Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("CD Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("CD Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("CD Capture Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), CS4236_DOUBLE1("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), -CS4236_DOUBLE1("Mono Playback Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), -WSS_SINGLE("Mono Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), +WSS_SINGLE_TLV("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, + 0, 0, 15, 0, db_scale_rec_gain), WSS_DOUBLE("Analog Loopback Capture Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), -WSS_SINGLE("Digital Loopback Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -CS4236_DOUBLE1("Digital Loopback Playback Volume", 0, - CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1) +WSS_SINGLE("Loopback Digital Playback Switch", 0, CS4231_LOOPBACK, 0, 1, 0), +CS4236_DOUBLE1_TLV("Loopback Digital Playback Volume", 0, + CS4231_LOOPBACK, CS4236_RIGHT_LOOPBACK, 2, 0, 63, 1, + db_scale_6bit), }; +static const DECLARE_TLV_DB_SCALE(db_scale_5bit_6db_max, -5600, 200, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_2bit_16db_max, -2400, 800, 0); + static struct snd_kcontrol_new snd_cs4235_controls[] = { -WSS_DOUBLE("Master Switch", 0, +WSS_DOUBLE("Master Playback Switch", 0, CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 7, 7, 1, 1), -WSS_DOUBLE("Master Volume", 0, - CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1), - -CS4235_OUTPUT_ACCU("Playback Volume", 0), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + CS4235_LEFT_MASTER, CS4235_RIGHT_MASTER, 0, 0, 31, 1, + db_scale_5bit_6db_max), -CS4236_DOUBLE("Master Digital Playback Switch", 0, - CS4236_LEFT_MASTER, CS4236_RIGHT_MASTER, 7, 7, 1, 1), -CS4236_DOUBLE("Master Digital Capture Switch", 0, - CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), -CS4236_MASTER_DIGITAL("Master Digital Volume", 0), +CS4235_OUTPUT_ACCU("Playback Volume", 0, db_scale_2bit_16db_max), -WSS_DOUBLE("Master Digital Playback Switch", 1, +WSS_DOUBLE("Synth Playback Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Master Digital Capture Switch", 1, +WSS_DOUBLE("Synth Capture Switch", 1, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 6, 6, 1, 1), -WSS_DOUBLE("Master Digital Volume", 1, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Synth Volume", 1, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), -CS4236_DOUBLE("Capture Volume", 0, - CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1), +CS4236_DOUBLE_TLV("Capture Volume", 0, + CS4236_LEFT_MIX_CTRL, CS4236_RIGHT_MIX_CTRL, 5, 5, 3, 1, + db_scale_2bit), -WSS_DOUBLE("PCM Switch", 0, +WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE("PCM Capture Switch", 0, + CS4236_DAC_MUTE, CS4236_DAC_MUTE, 7, 6, 1, 1), +WSS_DOUBLE_TLV("PCM Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), CS4236_DOUBLE("DSP Switch", 0, CS4236_LEFT_DSP, CS4236_RIGHT_DSP, 7, 7, 1, 1), @@ -842,29 +911,29 @@ CS4236_DOUBLE("Mic Capture Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 7, 7, 1, 1), CS4236_DOUBLE("Mic Playback Switch", 0, CS4236_LEFT_MIC, CS4236_RIGHT_MIC, 6, 6, 1, 1), -CS4236_SINGLE("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1), -CS4236_SINGLE("Mic Playback Boost", 0, CS4236_LEFT_MIC, 5, 1, 0), +CS4236_SINGLE_TLV("Mic Volume", 0, CS4236_LEFT_MIC, 0, 31, 1, + db_scale_5bit_22db_max), +CS4236_SINGLE("Mic Boost (+20dB)", 0, CS4236_LEFT_MIC, 5, 1, 0), -WSS_DOUBLE("Aux Playback Switch", 0, +WSS_DOUBLE("Line Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 0, +WSS_DOUBLE("Line Capture Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +WSS_DOUBLE_TLV("Line Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), -WSS_DOUBLE("Aux Playback Switch", 1, +WSS_DOUBLE("CD Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Capture Switch", 1, +WSS_DOUBLE("CD Capture Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 6, 6, 1, 1), -WSS_DOUBLE("Aux Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), - -CS4236_DOUBLE1("Master Mono Switch", 0, - CS4231_MONO_CTRL, CS4236_RIGHT_MIX_CTRL, 6, 7, 1, 1), +WSS_DOUBLE_TLV("CD Volume", 1, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), -CS4236_DOUBLE1("Mono Switch", 0, +CS4236_DOUBLE1("Beep Playback Switch", 0, CS4231_MONO_CTRL, CS4236_LEFT_MIX_CTRL, 7, 7, 1, 1), -WSS_SINGLE("Mono Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE("Beep Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), WSS_DOUBLE("Analog Loopback Switch", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 7, 7, 1, 0), diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 4c6e14f87f2..c76bb00c9d1 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -982,7 +982,7 @@ ES1688_DOUBLE("CD Playback Volume", 0, ES1688_CD_DEV, ES1688_CD_DEV, 4, 0, 15, 0 ES1688_DOUBLE("FM Playback Volume", 0, ES1688_FM_DEV, ES1688_FM_DEV, 4, 0, 15, 0), ES1688_DOUBLE("Mic Playback Volume", 0, ES1688_MIC_DEV, ES1688_MIC_DEV, 4, 0, 15, 0), ES1688_DOUBLE("Aux Playback Volume", 0, ES1688_AUX_DEV, ES1688_AUX_DEV, 4, 0, 15, 0), -ES1688_SINGLE("PC Speaker Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0), +ES1688_SINGLE("Beep Playback Volume", 0, ES1688_SPEAKER_DEV, 0, 7, 0), ES1688_DOUBLE("Capture Volume", 0, ES1688_RECLEV_DEV, ES1688_RECLEV_DEV, 4, 0, 15, 0), ES1688_SINGLE("Capture Switch", 0, ES1688_REC_DEV, 4, 1, 1), { diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 8cfbff73a83..9a43baae725 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -102,8 +102,6 @@ struct snd_es18xx { unsigned long port; /* port of ESS chip */ - unsigned long mpu_port; /* MPU-401 port of ESS chip */ - unsigned long fm_port; /* FM port */ unsigned long ctrl_port; /* Control port of ESS chip */ struct resource *res_port; struct resource *res_mpu_port; @@ -116,12 +114,9 @@ struct snd_es18xx { unsigned short audio2_vol; /* volume level of audio2 */ unsigned short active; /* active channel mask */ - unsigned int dma1_size; - unsigned int dma2_size; unsigned int dma1_shift; unsigned int dma2_shift; - struct snd_card *card; struct snd_pcm *pcm; struct snd_pcm_substream *playback_a_substream; struct snd_pcm_substream *capture_a_substream; @@ -136,14 +131,9 @@ struct snd_es18xx { spinlock_t reg_lock; spinlock_t mixer_lock; - spinlock_t ctrl_lock; #ifdef CONFIG_PM unsigned char pm_reg; #endif -}; - -struct snd_audiodrive { - struct snd_es18xx *chip; #ifdef CONFIG_PNP struct pnp_dev *dev; struct pnp_dev *devc; @@ -359,7 +349,7 @@ static inline int snd_es18xx_mixer_writable(struct snd_es18xx *chip, unsigned ch } -static int snd_es18xx_reset(struct snd_es18xx *chip) +static int __devinit snd_es18xx_reset(struct snd_es18xx *chip) { int i; outb(0x03, chip->port + 0x06); @@ -495,8 +485,6 @@ static int snd_es18xx_playback1_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma2_size = size; - snd_es18xx_rate_set(chip, substream, DAC2); /* Transfer Count Reload */ @@ -596,8 +584,6 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -664,8 +650,6 @@ static int snd_es18xx_playback2_prepare(struct snd_es18xx *chip, unsigned int size = snd_pcm_lib_buffer_bytes(substream); unsigned int count = snd_pcm_lib_period_bytes(substream); - chip->dma1_size = size; - snd_es18xx_reset_fifo(chip); /* Set stereo/mono */ @@ -755,7 +739,8 @@ static int snd_es18xx_playback_trigger(struct snd_pcm_substream *substream, static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) { - struct snd_es18xx *chip = dev_id; + struct snd_card *card = dev_id; + struct snd_es18xx *chip = card->private_data; unsigned char status; if (chip->caps & ES18XX_CONTROL) { @@ -805,12 +790,16 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) int split = 0; if (chip->caps & ES18XX_HWV) { split = snd_es18xx_mixer_read(chip, 0x64) & 0x80; - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->hw_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->hw_volume->id); } if (!split) { - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_switch->id); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &chip->master_volume->id); } /* ack interrupt */ snd_es18xx_mixer_write(chip, 0x66, 0x00); @@ -821,17 +810,18 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id) static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (substream->number == 0 && (chip->caps & ES18XX_PCM2)) { if (!(chip->active & DAC2)) return 0; - pos = snd_dma_pointer(chip->dma2, chip->dma2_size); + pos = snd_dma_pointer(chip->dma2, size); return pos >> chip->dma2_shift; } else { if (!(chip->active & DAC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } } @@ -839,11 +829,12 @@ static snd_pcm_uframes_t snd_es18xx_playback_pointer(struct snd_pcm_substream *s static snd_pcm_uframes_t snd_es18xx_capture_pointer(struct snd_pcm_substream *substream) { struct snd_es18xx *chip = snd_pcm_substream_chip(substream); + unsigned int size = snd_pcm_lib_buffer_bytes(substream); int pos; if (!(chip->active & ADC1)) return 0; - pos = snd_dma_pointer(chip->dma1, chip->dma1_size); + pos = snd_dma_pointer(chip->dma1, size); return pos >> chip->dma1_shift; } @@ -974,9 +965,6 @@ static int snd_es18xx_capture_close(struct snd_pcm_substream *substream) static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts4Source[4] = { - "Mic", "CD", "Line", "Master" - }; static char *texts5Source[5] = { "Mic", "CD", "Line", "Master", "Mix" }; @@ -994,7 +982,8 @@ static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_ele uinfo->value.enumerated.items = 4; if (uinfo->value.enumerated.item > 3) uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts4Source[uinfo->value.enumerated.item]); + strcpy(uinfo->value.enumerated.name, + texts5Source[uinfo->value.enumerated.item]); break; case 0x1887: case 0x1888: @@ -1313,7 +1302,7 @@ ES18XX_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0) * The chipset specific mixer controls */ static struct snd_kcontrol_new snd_es18xx_opt_speaker = - ES18XX_SINGLE("PC Speaker Playback Volume", 0, 0x3c, 0, 7, 0); + ES18XX_SINGLE("Beep Playback Volume", 0, 0x3c, 0, 7, 0); static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), @@ -1378,11 +1367,9 @@ ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), static int __devinit snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; - unsigned long flags; - spin_lock_irqsave(&chip->ctrl_lock, flags); + outb(reg, chip->ctrl_port); data = inb(chip->ctrl_port + 1); - spin_unlock_irqrestore(&chip->ctrl_lock, flags); return data; } @@ -1398,7 +1385,9 @@ static void __devinit snd_es18xx_config_write(struct snd_es18xx *chip, #endif } -static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) +static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { int mask = 0; @@ -1412,15 +1401,15 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) if (chip->caps & ES18XX_CONTROL) { /* Hardware volume IRQ */ snd_es18xx_config_write(chip, 0x27, chip->irq); - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { /* FM I/O */ - snd_es18xx_config_write(chip, 0x62, chip->fm_port >> 8); - snd_es18xx_config_write(chip, 0x63, chip->fm_port & 0xff); + snd_es18xx_config_write(chip, 0x62, fm_port >> 8); + snd_es18xx_config_write(chip, 0x63, fm_port & 0xff); } - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU-401 I/O */ - snd_es18xx_config_write(chip, 0x64, chip->mpu_port >> 8); - snd_es18xx_config_write(chip, 0x65, chip->mpu_port & 0xff); + snd_es18xx_config_write(chip, 0x64, mpu_port >> 8); + snd_es18xx_config_write(chip, 0x65, mpu_port & 0xff); /* MPU-401 IRQ */ snd_es18xx_config_write(chip, 0x28, chip->irq); } @@ -1507,11 +1496,12 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) snd_es18xx_mixer_write(chip, 0x7A, 0x68); /* Enable and set hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x06); - if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { + if (mpu_port > 0 && mpu_port != SNDRV_AUTO_PORT) { /* MPU401 share irq with audio Joystick enabled FM enabled */ - snd_es18xx_mixer_write(chip, 0x40, 0x43 | (chip->mpu_port & 0xf0) >> 1); + snd_es18xx_mixer_write(chip, 0x40, + 0x43 | (mpu_port & 0xf0) >> 1); } snd_es18xx_mixer_write(chip, 0x7f, ((irqmask + 1) << 1) | 0x01); } @@ -1629,7 +1619,9 @@ static int __devinit snd_es18xx_identify(struct snd_es18xx *chip) return 0; } -static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) +static int __devinit snd_es18xx_probe(struct snd_es18xx *chip, + unsigned long mpu_port, + unsigned long fm_port) { if (snd_es18xx_identify(chip) < 0) { snd_printk(KERN_ERR PFX "[0x%lx] ESS chip not found\n", chip->port); @@ -1650,8 +1642,6 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_I2S | ES18XX_CONTROL | ES18XX_HWV; break; case 0x1887: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; - break; case 0x1888: chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; break; @@ -1666,7 +1656,7 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) if (chip->dma1 == chip->dma2) chip->caps &= ~(ES18XX_PCM2 | ES18XX_DUPLEX_SAME); - return snd_es18xx_initialize(chip); + return snd_es18xx_initialize(chip, mpu_port, fm_port); } static struct snd_pcm_ops snd_es18xx_playback_ops = { @@ -1691,8 +1681,10 @@ static struct snd_pcm_ops snd_es18xx_capture_ops = { .pointer = snd_es18xx_capture_pointer, }; -static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct snd_pcm ** rpcm) +static int __devinit snd_es18xx_pcm(struct snd_card *card, int device, + struct snd_pcm **rpcm) { + struct snd_es18xx *chip = card->private_data; struct snd_pcm *pcm; char str[16]; int err; @@ -1701,9 +1693,9 @@ static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct *rpcm = NULL; sprintf(str, "ES%x", chip->version); if (chip->caps & ES18XX_PCM2) - err = snd_pcm_new(chip->card, str, device, 2, 1, &pcm); + err = snd_pcm_new(card, str, device, 2, 1, &pcm); else - err = snd_pcm_new(chip->card, str, device, 1, 1, &pcm); + err = snd_pcm_new(card, str, device, 1, 1, &pcm); if (err < 0) return err; @@ -1734,10 +1726,9 @@ static int __devinit snd_es18xx_pcm(struct snd_es18xx *chip, int device, struct #ifdef CONFIG_PM static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D3hot); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); snd_pcm_suspend_all(chip->pcm); @@ -1752,24 +1743,25 @@ static int snd_es18xx_suspend(struct snd_card *card, pm_message_t state) static int snd_es18xx_resume(struct snd_card *card) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip = acard->chip; + struct snd_es18xx *chip = card->private_data; /* restore PM register, we won't wake till (not 0x07) i/o activity though */ snd_es18xx_write(chip, ES18XX_PM, chip->pm_reg ^= ES18XX_PM_FM); - snd_power_change_state(chip->card, SNDRV_CTL_POWER_D0); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } #endif /* CONFIG_PM */ -static int snd_es18xx_free(struct snd_es18xx *chip) +static int snd_es18xx_free(struct snd_card *card) { + struct snd_es18xx *chip = card->private_data; + release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_ctrl_port); release_and_free_resource(chip->res_mpu_port); if (chip->irq >= 0) - free_irq(chip->irq, (void *) chip); + free_irq(chip->irq, (void *) card); if (chip->dma1 >= 0) { disable_dma(chip->dma1); free_dma(chip->dma1); @@ -1778,93 +1770,82 @@ static int snd_es18xx_free(struct snd_es18xx *chip) disable_dma(chip->dma2); free_dma(chip->dma2); } - kfree(chip); return 0; } static int snd_es18xx_dev_free(struct snd_device *device) { - struct snd_es18xx *chip = device->device_data; - return snd_es18xx_free(chip); + return snd_es18xx_free(device->card); } static int __devinit snd_es18xx_new_device(struct snd_card *card, unsigned long port, unsigned long mpu_port, unsigned long fm_port, - int irq, int dma1, int dma2, - struct snd_es18xx ** rchip) + int irq, int dma1, int dma2) { - struct snd_es18xx *chip; + struct snd_es18xx *chip = card->private_data; static struct snd_device_ops ops = { .dev_free = snd_es18xx_dev_free, }; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); - spin_lock_init(&chip->ctrl_lock); - chip->card = card; chip->port = port; - chip->mpu_port = mpu_port; - chip->fm_port = fm_port; chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; chip->audio2_vol = 0x00; chip->active = 0; - if ((chip->res_port = request_region(port, 16, "ES18xx")) == NULL) { - snd_es18xx_free(chip); + chip->res_port = request_region(port, 16, "ES18xx"); + if (chip->res_port == NULL) { + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap ports 0x%lx-0x%lx\n", port, port + 16 - 1); return -EBUSY; } - if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", (void *) chip)) { - snd_es18xx_free(chip); + if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", + (void *) card)) { + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq); return -EBUSY; } chip->irq = irq; if (request_dma(dma1, "ES18xx DMA 1")) { - snd_es18xx_free(chip); + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap DMA1 %d\n", dma1); return -EBUSY; } chip->dma1 = dma1; if (dma2 != dma1 && request_dma(dma2, "ES18xx DMA 2")) { - snd_es18xx_free(chip); + snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap DMA2 %d\n", dma2); return -EBUSY; } chip->dma2 = dma2; - if (snd_es18xx_probe(chip) < 0) { - snd_es18xx_free(chip); - return -ENODEV; - } - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { - snd_es18xx_free(chip); + if (snd_es18xx_probe(chip, mpu_port, fm_port) < 0) { + snd_es18xx_free(card); + return -ENODEV; + } + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_es18xx_free(card); return err; } - *rchip = chip; return 0; } -static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) +static int __devinit snd_es18xx_mixer(struct snd_card *card) { - struct snd_card *card; + struct snd_es18xx *chip = card->private_data; int err; unsigned int idx; - card = chip->card; - strcpy(card->mixername, chip->pcm->name); for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_base_controls); idx++) { @@ -1986,7 +1967,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */ #ifdef CONFIG_PNP -static int isapnp[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; #endif static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260,0x280 */ #ifndef CONFIG_PNP @@ -2063,11 +2044,11 @@ static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev) return 0; } -static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard, +static int __devinit snd_audiodrive_pnp(int dev, struct snd_es18xx *chip, struct pnp_dev *pdev) { - acard->dev = pdev; - if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) + chip->dev = pdev; + if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0) return -EBUSY; return 0; } @@ -2093,26 +2074,26 @@ static struct pnp_card_device_id snd_audiodrive_pnpids[] = { MODULE_DEVICE_TABLE(pnp_card, snd_audiodrive_pnpids); -static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, +static int __devinit snd_audiodrive_pnpc(int dev, struct snd_es18xx *chip, struct pnp_card_link *card, const struct pnp_card_device_id *id) { - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) + chip->dev = pnp_request_card_device(card, id->devs[0].id, NULL); + if (chip->dev == NULL) return -EBUSY; - acard->devc = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devc == NULL) + chip->devc = pnp_request_card_device(card, id->devs[1].id, NULL); + if (chip->devc == NULL) return -EBUSY; /* Control port initialization */ - if (pnp_activate_dev(acard->devc) < 0) { + if (pnp_activate_dev(chip->devc) < 0) { snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n"); return -EAGAIN; } snd_printdd("pnp: port=0x%llx\n", - (unsigned long long)pnp_port_start(acard->devc, 0)); - if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) + (unsigned long long)pnp_port_start(chip->devc, 0)); + if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0) return -EBUSY; return 0; @@ -2128,24 +2109,20 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard, static int snd_es18xx_card_new(int dev, struct snd_card **cardp) { return snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_audiodrive), cardp); + sizeof(struct snd_es18xx), cardp); } static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) { - struct snd_audiodrive *acard = card->private_data; - struct snd_es18xx *chip; + struct snd_es18xx *chip = card->private_data; struct snd_opl3 *opl3; int err; - if ((err = snd_es18xx_new_device(card, - port[dev], - mpu_port[dev], - fm_port[dev], - irq[dev], dma1[dev], dma2[dev], - &chip)) < 0) + err = snd_es18xx_new_device(card, + port[dev], mpu_port[dev], fm_port[dev], + irq[dev], dma1[dev], dma2[dev]); + if (err < 0) return err; - acard->chip = chip; sprintf(card->driver, "ES%x", chip->version); @@ -2161,26 +2138,32 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) chip->port, irq[dev], dma1[dev]); - if ((err = snd_es18xx_pcm(chip, 0, NULL)) < 0) + err = snd_es18xx_pcm(card, 0, NULL); + if (err < 0) return err; - if ((err = snd_es18xx_mixer(chip)) < 0) + err = snd_es18xx_mixer(card); + if (err < 0) return err; if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { - if (snd_opl3_create(card, chip->fm_port, chip->fm_port + 2, OPL3_HW_OPL3, 0, &opl3) < 0) { - snd_printk(KERN_WARNING PFX "opl3 not detected at 0x%lx\n", chip->fm_port); + if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) { + snd_printk(KERN_WARNING PFX + "opl3 not detected at 0x%lx\n", + fm_port[dev]); } else { - if ((err = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) return err; } } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { - if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, - chip->mpu_port, 0, - irq[dev], 0, - &chip->rmidi)) < 0) + err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, + mpu_port[dev], 0, + irq[dev], 0, &chip->rmidi); + if (err < 0) return err; } diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 02e30d7c6a9..6123c753111 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -25,6 +25,7 @@ #include <linux/init.h> #include <linux/err.h> #include <linux/isa.h> +#include <linux/pnp.h> #include <linux/delay.h> #include <linux/slab.h> #include <linux/ioport.h> @@ -40,7 +41,7 @@ #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> -#include "miro.h" +#include <sound/aci.h> MODULE_AUTHOR("Martin Langer <martin-langer@gmx.de>"); MODULE_LICENSE("GPL"); @@ -60,6 +61,9 @@ static int dma1 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int dma2 = SNDRV_DEFAULT_DMA1; /* 0,1,3 */ static int wss; static int ide; +#ifdef CONFIG_PNP +static int isapnp = 1; /* Enable ISA PnP detection */ +#endif module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for miro soundcard."); @@ -83,6 +87,10 @@ module_param(wss, int, 0444); MODULE_PARM_DESC(wss, "wss mode"); module_param(ide, int, 0444); MODULE_PARM_DESC(ide, "enable ide port"); +#ifdef CONFIG_PNP +module_param(isapnp, bool, 0444); +MODULE_PARM_DESC(isapnp, "Enable ISA PnP detection for specified soundcard."); +#endif #define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 @@ -96,7 +104,6 @@ MODULE_PARM_DESC(ide, "enable ide port"); #define OPTi9XX_MC_REG(n) n - struct snd_miro { unsigned short hardware; unsigned char password; @@ -110,7 +117,6 @@ struct snd_miro { unsigned long pwd_reg; spinlock_t lock; - struct snd_card *card; struct snd_pcm *pcm; long wss_base; @@ -118,23 +124,13 @@ struct snd_miro { int dma1; int dma2; - long fm_port; - long mpu_port; int mpu_irq; - unsigned long aci_port; - int aci_vendor; - int aci_product; - int aci_version; - int aci_amp; - int aci_preamp; - int aci_solomode; - - struct mutex aci_mutex; + struct snd_miro_aci *aci; }; -static void snd_miro_proc_init(struct snd_miro * miro); +static struct snd_miro_aci aci_device; static char * snd_opti9xx_names[] = { "unkown", @@ -143,17 +139,33 @@ static char * snd_opti9xx_names[] = { "82C930", "82C931", "82C933" }; +static int snd_miro_pnp_is_probed; + +#ifdef CONFIG_PNP + +static struct pnp_card_device_id snd_miro_pnpids[] = { + /* PCM20 and PCM12 in PnP mode */ + { .id = "MIR0924", + .devs = { { "MIR0000" }, { "MIR0002" }, { "MIR0005" } }, }, + { .id = "" } +}; + +MODULE_DEVICE_TABLE(pnp_card, snd_miro_pnpids); + +#endif /* CONFIG_PNP */ + /* * ACI control */ -static int aci_busy_wait(struct snd_miro * miro) +static int aci_busy_wait(struct snd_miro_aci *aci) { long timeout; unsigned char byte; - for (timeout = 1; timeout <= ACI_MINTIME+30; timeout++) { - if (((byte=inb(miro->aci_port + ACI_REG_BUSY)) & 1) == 0) { + for (timeout = 1; timeout <= ACI_MINTIME + 30; timeout++) { + byte = inb(aci->aci_port + ACI_REG_BUSY); + if ((byte & 1) == 0) { if (timeout >= ACI_MINTIME) snd_printd("aci ready in round %ld.\n", timeout-ACI_MINTIME); @@ -179,10 +191,10 @@ static int aci_busy_wait(struct snd_miro * miro) return -EBUSY; } -static inline int aci_write(struct snd_miro * miro, unsigned char byte) +static inline int aci_write(struct snd_miro_aci *aci, unsigned char byte) { - if (aci_busy_wait(miro) >= 0) { - outb(byte, miro->aci_port + ACI_REG_COMMAND); + if (aci_busy_wait(aci) >= 0) { + outb(byte, aci->aci_port + ACI_REG_COMMAND); return 0; } else { snd_printk(KERN_ERR "aci busy, aci_write(0x%x) stopped.\n", byte); @@ -190,12 +202,12 @@ static inline int aci_write(struct snd_miro * miro, unsigned char byte) } } -static inline int aci_read(struct snd_miro * miro) +static inline int aci_read(struct snd_miro_aci *aci) { unsigned char byte; - if (aci_busy_wait(miro) >= 0) { - byte=inb(miro->aci_port + ACI_REG_STATUS); + if (aci_busy_wait(aci) >= 0) { + byte = inb(aci->aci_port + ACI_REG_STATUS); return byte; } else { snd_printk(KERN_ERR "aci busy, aci_read() stopped.\n"); @@ -203,39 +215,49 @@ static inline int aci_read(struct snd_miro * miro) } } -static int aci_cmd(struct snd_miro * miro, int write1, int write2, int write3) +int snd_aci_cmd(struct snd_miro_aci *aci, int write1, int write2, int write3) { int write[] = {write1, write2, write3}; int value, i; - if (mutex_lock_interruptible(&miro->aci_mutex)) + if (mutex_lock_interruptible(&aci->aci_mutex)) return -EINTR; for (i=0; i<3; i++) { if (write[i]< 0 || write[i] > 255) break; else { - value = aci_write(miro, write[i]); + value = aci_write(aci, write[i]); if (value < 0) goto out; } } - value = aci_read(miro); + value = aci_read(aci); -out: mutex_unlock(&miro->aci_mutex); +out: mutex_unlock(&aci->aci_mutex); return value; } +EXPORT_SYMBOL(snd_aci_cmd); + +static int aci_getvalue(struct snd_miro_aci *aci, unsigned char index) +{ + return snd_aci_cmd(aci, ACI_STATUS, index, -1); +} -static int aci_getvalue(struct snd_miro * miro, unsigned char index) +static int aci_setvalue(struct snd_miro_aci *aci, unsigned char index, + int value) { - return aci_cmd(miro, ACI_STATUS, index, -1); + return snd_aci_cmd(aci, index, value, -1); } -static int aci_setvalue(struct snd_miro * miro, unsigned char index, int value) +struct snd_miro_aci *snd_aci_get_aci(void) { - return aci_cmd(miro, index, value, -1); + if (aci_device.aci_port == 0) + return NULL; + return &aci_device; } +EXPORT_SYMBOL(snd_aci_get_aci); /* * MIXER part @@ -249,8 +271,10 @@ static int snd_miro_get_capture(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if ((value = aci_getvalue(miro, ACI_S_GENERAL)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_S_GENERAL); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", + value); return value; } @@ -267,13 +291,15 @@ static int snd_miro_put_capture(struct snd_kcontrol *kcontrol, value = !(ucontrol->value.integer.value[0]); - if ((error = aci_setvalue(miro, ACI_SET_SOLOMODE, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_SOLOMODE, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", + error); return error; } - change = (value != miro->aci_solomode); - miro->aci_solomode = value; + change = (value != miro->aci->aci_solomode); + miro->aci->aci_solomode = value; return change; } @@ -295,7 +321,7 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, struct snd_miro *miro = snd_kcontrol_chip(kcontrol); int value; - if (miro->aci_version <= 176) { + if (miro->aci->aci_version <= 176) { /* OSS says it's not readable with versions < 176. @@ -303,12 +329,14 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, which is a PCM12 with aci_version = 176. */ - ucontrol->value.integer.value[0] = miro->aci_preamp; + ucontrol->value.integer.value[0] = miro->aci->aci_preamp; return 0; } - if ((value = aci_getvalue(miro, ACI_GET_PREAMP)) < 0) { - snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", value); + value = aci_getvalue(miro->aci, ACI_GET_PREAMP); + if (value < 0) { + snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", + value); return value; } @@ -325,13 +353,15 @@ static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_PREAMP, value)) < 0) { - snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", error); + error = aci_setvalue(miro->aci, ACI_SET_PREAMP, value); + if (error < 0) { + snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", + error); return error; } - change = (value != miro->aci_preamp); - miro->aci_preamp = value; + change = (value != miro->aci->aci_preamp); + miro->aci->aci_preamp = value; return change; } @@ -342,7 +372,7 @@ static int snd_miro_get_amp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = miro->aci_amp; + ucontrol->value.integer.value[0] = miro->aci->aci_amp; return 0; } @@ -355,13 +385,14 @@ static int snd_miro_put_amp(struct snd_kcontrol *kcontrol, value = ucontrol->value.integer.value[0]; - if ((error = aci_setvalue(miro, ACI_SET_POWERAMP, value)) < 0) { + error = aci_setvalue(miro->aci, ACI_SET_POWERAMP, value); + if (error < 0) { snd_printk(KERN_ERR "snd_miro_put_amp() to %d failed: %d\n", value, error); return error; } - change = (value != miro->aci_amp); - miro->aci_amp = value; + change = (value != miro->aci->aci_amp); + miro->aci->aci_amp = value; return change; } @@ -410,12 +441,14 @@ static int snd_miro_get_double(struct snd_kcontrol *kcontrol, int right_reg = kcontrol->private_value & 0xff; int left_reg = right_reg + 1; - if ((right_val = aci_getvalue(miro, right_reg)) < 0) { + right_val = aci_getvalue(miro->aci, right_reg); + if (right_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", right_reg, right_val); return right_val; } - if ((left_val = aci_getvalue(miro, left_reg)) < 0) { + left_val = aci_getvalue(miro->aci, left_reg); + if (left_val < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", left_reg, left_val); return left_val; } @@ -451,6 +484,7 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_miro *miro = snd_kcontrol_chip(kcontrol); + struct snd_miro_aci *aci = miro->aci; int left, right, left_old, right_old; int setreg_left, setreg_right, getreg_left, getreg_right; int change, error; @@ -459,21 +493,21 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right = ucontrol->value.integer.value[1]; setreg_right = (kcontrol->private_value >> 8) & 0xff; - if (setreg_right == ACI_SET_MASTER) { - setreg_left = setreg_right + 1; - } else { - setreg_left = setreg_right + 8; - } + setreg_left = setreg_right + 8; + if (setreg_right == ACI_SET_MASTER) + setreg_left -= 7; getreg_right = kcontrol->private_value & 0xff; getreg_left = getreg_right + 1; - if ((left_old = aci_getvalue(miro, getreg_left)) < 0) { + left_old = aci_getvalue(aci, getreg_left); + if (left_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_left, left_old); return left_old; } - if ((right_old = aci_getvalue(miro, getreg_right)) < 0) { + right_old = aci_getvalue(aci, getreg_right); + if (right_old < 0) { snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_right, right_old); return right_old; } @@ -492,13 +526,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, right_old = 0x80 - right_old; if (left >= 0) { - if ((error = aci_setvalue(miro, setreg_left, left)) < 0) { + error = aci_setvalue(aci, setreg_left, left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", left, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_left, 0x80 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x80 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - left, error); return error; @@ -506,13 +542,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, } if (right >= 0) { - if ((error = aci_setvalue(miro, setreg_right, right)) < 0) { + error = aci_setvalue(aci, setreg_right, right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", right, error); return error; } } else { - if ((error = aci_setvalue(miro, setreg_right, 0x80 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x80 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x80 - right, error); return error; @@ -530,12 +568,14 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, left_old = 0x20 - left_old; right_old = 0x20 - right_old; - if ((error = aci_setvalue(miro, setreg_left, 0x20 - left)) < 0) { + error = aci_setvalue(aci, setreg_left, 0x20 - left); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - left, error); return error; } - if ((error = aci_setvalue(miro, setreg_right, 0x20 - right)) < 0) { + error = aci_setvalue(aci, setreg_right, 0x20 - right); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", 0x20 - right, error); return error; @@ -633,11 +673,13 @@ static unsigned char aci_init_values[][2] __devinitdata = { static int __devinit snd_set_aci_init_values(struct snd_miro *miro) { int idx, error; + struct snd_miro_aci *aci = miro->aci; /* enable WSS on PCM1 */ - if ((miro->aci_product == 'A') && wss) { - if ((error = aci_setvalue(miro, ACI_SET_WSS, wss)) < 0) { + if ((aci->aci_product == 'A') && wss) { + error = aci_setvalue(aci, ACI_SET_WSS, wss); + if (error < 0) { snd_printk(KERN_ERR "enabling WSS mode failed\n"); return error; } @@ -646,7 +688,8 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* enable IDE port */ if (ide) { - if ((error = aci_setvalue(miro, ACI_SET_IDE, ide)) < 0) { + error = aci_setvalue(aci, ACI_SET_IDE, ide); + if (error < 0) { snd_printk(KERN_ERR "enabling IDE port failed\n"); return error; } @@ -654,32 +697,31 @@ static int __devinit snd_set_aci_init_values(struct snd_miro *miro) /* set common aci values */ - for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) - if ((error = aci_setvalue(miro, aci_init_values[idx][0], - aci_init_values[idx][1])) < 0) { + for (idx = 0; idx < ARRAY_SIZE(aci_init_values); idx++) { + error = aci_setvalue(aci, aci_init_values[idx][0], + aci_init_values[idx][1]); + if (error < 0) { snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", aci_init_values[idx][0], error); return error; } - - miro->aci_amp = 0; - miro->aci_preamp = 0; - miro->aci_solomode = 1; + } + aci->aci_amp = 0; + aci->aci_preamp = 0; + aci->aci_solomode = 1; return 0; } -static int __devinit snd_miro_mixer(struct snd_miro *miro) +static int __devinit snd_miro_mixer(struct snd_card *card, + struct snd_miro *miro) { - struct snd_card *card; unsigned int idx; int err; - if (snd_BUG_ON(!miro || !miro->card)) + if (snd_BUG_ON(!miro || !card)) return -EINVAL; - card = miro->card; - switch (miro->hardware) { case OPTi9XX_HW_82C924: strcpy(card->mixername, "ACI & OPTi924"); @@ -697,7 +739,8 @@ static int __devinit snd_miro_mixer(struct snd_miro *miro) return err; } - if ((miro->aci_product == 'A') || (miro->aci_product == 'B')) { + if ((miro->aci->aci_product == 'A') || + (miro->aci->aci_product == 'B')) { /* PCM1/PCM12 with power-amp and Line 2 */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_line_control[0], miro))) < 0) return err; @@ -705,16 +748,17 @@ static int __devinit snd_miro_mixer(struct snd_miro *miro) return err; } - if ((miro->aci_product == 'B') || (miro->aci_product == 'C')) { + if ((miro->aci->aci_product == 'B') || + (miro->aci->aci_product == 'C')) { /* PCM12/PCM20 with mic-preamp */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_preamp_control[0], miro))) < 0) return err; - if (miro->aci_version >= 176) + if (miro->aci->aci_version >= 176) if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_capture_control[0], miro))) < 0) return err; } - if (miro->aci_product == 'C') { + if (miro->aci->aci_product == 'C') { /* PCM20 with radio and 7 band equalizer */ if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_miro_radio_control[0], miro))) < 0) return err; @@ -757,21 +801,26 @@ static int __devinit snd_miro_init(struct snd_miro *chip, chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; - chip->fm_port = -1; chip->mpu_port = -1; chip->mpu_irq = -1; + chip->pwd_reg = 3; + +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; + else +#endif + chip->mc_base = 0xf8c; + switch (hardware) { case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = 0xe3; - chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: - chip->mc_base = 0xf8c; chip->password = 0xe5; - chip->pwd_reg = 3; break; default: @@ -853,14 +902,15 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, struct snd_info_buffer *buffer) { struct snd_miro *miro = (struct snd_miro *) entry->private_data; + struct snd_miro_aci *aci = miro->aci; char* model = "unknown"; /* miroSOUND PCM1 pro, early PCM12 */ if ((miro->hardware == OPTi9XX_HW_82C929) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'A')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'A')) { + switch (aci->aci_version) { case 3: model = "miroSOUND PCM1 pro"; break; @@ -873,9 +923,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM12, PCM12 (Rev. E), PCM12 pnp */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'B')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'B')) { + switch (aci->aci_version) { case 4: model = "miroSOUND PCM12"; break; @@ -891,9 +941,9 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, /* miroSOUND PCM20 radio */ if ((miro->hardware == OPTi9XX_HW_82C924) && - (miro->aci_vendor == 'm') && - (miro->aci_product == 'C')) { - switch(miro->aci_version) { + (aci->aci_vendor == 'm') && + (aci->aci_product == 'C')) { + switch (aci->aci_version) { case 7: model = "miroSOUND PCM20 radio (Rev. E)"; break; @@ -917,17 +967,17 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "ACI information:\n"); snd_iprintf(buffer, " vendor : "); - switch(miro->aci_vendor) { + switch (aci->aci_vendor) { case 'm': snd_iprintf(buffer, "Miro\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_vendor); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_vendor); break; } snd_iprintf(buffer, " product : "); - switch(miro->aci_product) { + switch (aci->aci_product) { case 'A': snd_iprintf(buffer, "miroSOUND PCM1 pro / (early) PCM12\n"); break; @@ -938,26 +988,27 @@ static void snd_miro_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "miroSOUND PCM20 radio\n"); break; default: - snd_iprintf(buffer, "unknown (0x%x)\n", miro->aci_product); + snd_iprintf(buffer, "unknown (0x%x)\n", aci->aci_product); break; } snd_iprintf(buffer, " firmware: %d (0x%x)\n", - miro->aci_version, miro->aci_version); + aci->aci_version, aci->aci_version); snd_iprintf(buffer, " port : 0x%lx-0x%lx\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); snd_iprintf(buffer, " wss : 0x%x\n", wss); snd_iprintf(buffer, " ide : 0x%x\n", ide); - snd_iprintf(buffer, " solomode: 0x%x\n", miro->aci_solomode); - snd_iprintf(buffer, " amp : 0x%x\n", miro->aci_amp); - snd_iprintf(buffer, " preamp : 0x%x\n", miro->aci_preamp); + snd_iprintf(buffer, " solomode: 0x%x\n", aci->aci_solomode); + snd_iprintf(buffer, " amp : 0x%x\n", aci->aci_amp); + snd_iprintf(buffer, " preamp : 0x%x\n", aci->aci_preamp); } -static void __devinit snd_miro_proc_init(struct snd_miro * miro) +static void __devinit snd_miro_proc_init(struct snd_card *card, + struct snd_miro *miro) { struct snd_info_entry *entry; - if (! snd_card_proc_new(miro->card, "miro", &entry)) + if (!snd_card_proc_new(card, "miro", &entry)) snd_info_set_text_ops(entry, miro, snd_miro_proc_read); } @@ -974,37 +1025,40 @@ static int __devinit snd_miro_configure(struct snd_miro *chip) unsigned char mpu_irq_bits; unsigned long flags; + snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ + snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); + switch (chip->hardware) { case OPTi9XX_HW_82C924: snd_miro_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; case OPTi9XX_HW_82C929: /* untested init commands for OPTi929 */ - snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); /* OPL4 */ snd_miro_write_mask(chip, OPTi9XX_MC_REG(4), 0x00, 0x0c); - snd_miro_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); break; default: snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); return -EINVAL; } - switch (chip->wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (chip->wss_base & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: @@ -1122,75 +1176,92 @@ __skip_mpu: return 0; } +static int __devinit snd_miro_opti_check(struct snd_miro *chip) +{ + unsigned char value; + + chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, + "OPTi9xx MC"); + if (chip->res_mc_base == NULL) + return -ENOMEM; + + value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); + if (value != 0xff && value != inb(chip->mc_base + OPTi9XX_MC_REG(1))) + if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) + return 0; + + release_and_free_resource(chip->res_mc_base); + chip->res_mc_base = NULL; + + return -ENODEV; +} + static int __devinit snd_card_miro_detect(struct snd_card *card, struct snd_miro *chip) { int i, err; - unsigned char value; for (i = OPTi9XX_HW_82C929; i <= OPTi9XX_HW_82C924; i++) { if ((err = snd_miro_init(chip, i)) < 0) return err; - if ((chip->res_mc_base = request_region(chip->mc_base, chip->mc_base_size, "OPTi9xx MC")) == NULL) - continue; - - value = snd_miro_read(chip, OPTi9XX_MC_REG(1)); - if ((value != 0xff) && (value != inb(chip->mc_base + 1))) - if (value == snd_miro_read(chip, OPTi9XX_MC_REG(1))) - return 1; - - release_and_free_resource(chip->res_mc_base); - chip->res_mc_base = NULL; - + err = snd_miro_opti_check(chip); + if (err == 0) + return 1; } return -ENODEV; } static int __devinit snd_card_miro_aci_detect(struct snd_card *card, - struct snd_miro * miro) + struct snd_miro *miro) { unsigned char regval; int i; + struct snd_miro_aci *aci = &aci_device; + + miro->aci = aci; - mutex_init(&miro->aci_mutex); + mutex_init(&aci->aci_mutex); /* get ACI port from OPTi9xx MC 4 */ - miro->mc_base = 0xf8c; regval=inb(miro->mc_base + 4); - miro->aci_port = (regval & 0x10) ? 0x344: 0x354; + aci->aci_port = (regval & 0x10) ? 0x344 : 0x354; - if ((miro->res_aci_port = request_region(miro->aci_port, 3, "miro aci")) == NULL) { + miro->res_aci_port = request_region(aci->aci_port, 3, "miro aci"); + if (miro->res_aci_port == NULL) { snd_printk(KERN_ERR "aci i/o area 0x%lx-0x%lx already used.\n", - miro->aci_port, miro->aci_port+2); + aci->aci_port, aci->aci_port+2); return -ENOMEM; } /* force ACI into a known state */ for (i = 0; i < 3; i++) - if (aci_cmd(miro, ACI_ERROR_OP, -1, -1) < 0) { + if (snd_aci_cmd(aci, ACI_ERROR_OP, -1, -1) < 0) { snd_printk(KERN_ERR "can't force aci into known state.\n"); return -ENXIO; } - if ((miro->aci_vendor=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0 || - (miro->aci_product=aci_cmd(miro, ACI_READ_IDCODE, -1, -1)) < 0) { - snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", miro->aci_port); + aci->aci_vendor = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + aci->aci_product = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); + if (aci->aci_vendor < 0 || aci->aci_product < 0) { + snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", + aci->aci_port); return -ENXIO; } - if ((miro->aci_version=aci_cmd(miro, ACI_READ_VERSION, -1, -1)) < 0) { + aci->aci_version = snd_aci_cmd(aci, ACI_READ_VERSION, -1, -1); + if (aci->aci_version < 0) { snd_printk(KERN_ERR "can't read aci version on 0x%lx.\n", - miro->aci_port); + aci->aci_port); return -ENXIO; } - if (aci_cmd(miro, ACI_INIT, -1, -1) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || - aci_cmd(miro, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { + if (snd_aci_cmd(aci, ACI_INIT, -1, -1) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || + snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { snd_printk(KERN_ERR "can't initialize aci.\n"); return -ENXIO; } @@ -1201,157 +1272,80 @@ static int __devinit snd_card_miro_aci_detect(struct snd_card *card, static void snd_card_miro_free(struct snd_card *card) { struct snd_miro *miro = card->private_data; - + release_and_free_resource(miro->res_aci_port); + if (miro->aci) + miro->aci->aci_port = 0; release_and_free_resource(miro->res_mc_base); } -static int __devinit snd_miro_match(struct device *devptr, unsigned int n) -{ - return 1; -} - -static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) +static int __devinit snd_miro_probe(struct snd_card *card) { - static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; - static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; - static int possible_irqs[] = {11, 9, 10, 7, -1}; - static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; - static int possible_dma1s[] = {3, 1, 0, -1}; - static int possible_dma2s[][2] = {{1,-1}, {0,-1}, {-1,-1}, {0,-1}}; - int error; - struct snd_miro *miro; + struct snd_miro *miro = card->private_data; struct snd_wss *codec; struct snd_timer *timer; - struct snd_card *card; struct snd_pcm *pcm; struct snd_rawmidi *rmidi; - error = snd_card_create(index, id, THIS_MODULE, - sizeof(struct snd_miro), &card); - if (error < 0) - return error; - - card->private_free = snd_card_miro_free; - miro = card->private_data; - miro->card = card; - - if ((error = snd_card_miro_aci_detect(card, miro)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to detect aci chip\n"); - return -ENODEV; + if (!miro->res_mc_base) { + miro->res_mc_base = request_region(miro->mc_base, + miro->mc_base_size, + "miro (OPTi9xx MC)"); + if (miro->res_mc_base == NULL) { + snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); + return -ENOMEM; + } } - /* init proc interface */ - snd_miro_proc_init(miro); - - if ((error = snd_card_miro_detect(card, miro)) < 0) { + error = snd_card_miro_aci_detect(card, miro); + if (error < 0) { snd_card_free(card); - snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + snd_printk(KERN_ERR "unable to detect aci chip\n"); return -ENODEV; } - if (! miro->res_mc_base && - (miro->res_mc_base = request_region(miro->mc_base, miro->mc_base_size, - "miro (OPTi9xx MC)")) == NULL) { - snd_card_free(card); - snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); - return -ENOMEM; - } - miro->wss_base = port; - miro->fm_port = fm_port; miro->mpu_port = mpu_port; miro->irq = irq; miro->mpu_irq = mpu_irq; miro->dma1 = dma1; miro->dma2 = dma2; - if (miro->wss_base == SNDRV_AUTO_PORT) { - if ((miro->wss_base = snd_legacy_find_free_ioport(possible_ports, 4)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free WSS port\n"); - return -EBUSY; - } - } - - if (miro->mpu_port == SNDRV_AUTO_PORT) { - if ((miro->mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); - return -EBUSY; - } - } - if (miro->irq == SNDRV_AUTO_IRQ) { - if ((miro->irq = snd_legacy_find_free_irq(possible_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free IRQ\n"); - return -EBUSY; - } - } - if (miro->mpu_irq == SNDRV_AUTO_IRQ) { - if ((miro->mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); - return -EBUSY; - } - } - if (miro->dma1 == SNDRV_AUTO_DMA) { - if ((miro->dma1 = snd_legacy_find_free_dma(possible_dma1s)) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA1\n"); - return -EBUSY; - } - } - if (miro->dma2 == SNDRV_AUTO_DMA) { - if ((miro->dma2 = snd_legacy_find_free_dma(possible_dma2s[miro->dma1 % 4])) < 0) { - snd_card_free(card); - snd_printk(KERN_ERR "unable to find a free DMA2\n"); - return -EBUSY; - } - } + /* init proc interface */ + snd_miro_proc_init(card, miro); error = snd_miro_configure(miro); - if (error) { - snd_card_free(card); + if (error) return error; - } error = snd_wss_create(card, miro->wss_base + 4, -1, - miro->irq, miro->dma1, miro->dma2, - WSS_HW_AD1845, 0, &codec); - if (error < 0) { - snd_card_free(card); + miro->irq, miro->dma1, miro->dma2, + WSS_HW_DETECT, 0, &codec); + if (error < 0) return error; - } error = snd_wss_pcm(codec, 0, &pcm); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_mixer(codec); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } + error = snd_wss_timer(codec, 0, &timer); - if (error < 0) { - snd_card_free(card); + if (error < 0) return error; - } miro->pcm = pcm; - if ((error = snd_miro_mixer(miro)) < 0) { - snd_card_free(card); + error = snd_miro_mixer(card, miro); + if (error < 0) return error; - } - if (miro->aci_vendor == 'm') { + if (miro->aci->aci_vendor == 'm') { /* It looks like a miro sound card. */ - switch (miro->aci_product) { + switch (miro->aci->aci_product) { case 'A': sprintf(card->shortname, "miroSOUND PCM1 pro / PCM12"); @@ -1380,30 +1374,131 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) card->shortname, miro->name, pcm->name, miro->wss_base + 4, miro->irq, miro->dma1, miro->dma2); - if (miro->mpu_port <= 0 || miro->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - miro->mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, - &rmidi))) - snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", miro->mpu_port); + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, miro->mpu_irq, IRQF_DISABLED, + &rmidi); + if (error < 0) + snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", + mpu_port); + } - if (miro->fm_port > 0 && miro->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; struct snd_opl4 *opl4; - if (snd_opl4_create(card, miro->fm_port, miro->fm_port - 8, + + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) - snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", miro->fm_port); + snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", + fm_port); } - if ((error = snd_set_aci_init_values(miro)) < 0) { - snd_card_free(card); + error = snd_set_aci_init_values(miro); + if (error < 0) return error; + + return snd_card_register(card); +} + +static int __devinit snd_miro_isa_match(struct device *devptr, unsigned int n) +{ +#ifdef CONFIG_PNP + if (snd_miro_pnp_is_probed) + return 0; + if (isapnp) + return 0; +#endif + return 1; +} + +static int __devinit snd_miro_isa_probe(struct device *devptr, unsigned int n) +{ + static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; + static long possible_mpu_ports[] = {0x330, 0x300, 0x310, 0x320, -1}; + static int possible_irqs[] = {11, 9, 10, 7, -1}; + static int possible_mpu_irqs[] = {10, 5, 9, 7, -1}; + static int possible_dma1s[] = {3, 1, 0, -1}; + static int possible_dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, + {0, -1} }; + + int error; + struct snd_miro *miro; + struct snd_card *card; + + error = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (error < 0) + return error; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + error = snd_card_miro_detect(card, miro); + if (error < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + return -ENODEV; + } + + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free WSS port\n"); + return -EBUSY; + } + } + + if (mpu_port == SNDRV_AUTO_PORT) { + mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); + if (mpu_port < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 port\n"); + return -EBUSY; + } + } + + if (irq == SNDRV_AUTO_IRQ) { + irq = snd_legacy_find_free_irq(possible_irqs); + if (irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (mpu_irq == SNDRV_AUTO_IRQ) { + mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs); + if (mpu_irq < 0) { + snd_card_free(card); + snd_printk(KERN_ERR + "unable to find a free MPU401 IRQ\n"); + return -EBUSY; + } + } + if (dma1 == SNDRV_AUTO_DMA) { + dma1 = snd_legacy_find_free_dma(possible_dma1s); + if (dma1 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2 == SNDRV_AUTO_DMA) { + dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4]); + if (dma2 < 0) { + snd_card_free(card); + snd_printk(KERN_ERR "unable to find a free DMA2\n"); + return -EBUSY; + } } snd_card_set_dev(card, devptr); - if ((error = snd_card_register(card))) { + error = snd_miro_probe(card); + if (error < 0) { snd_card_free(card); return error; } @@ -1412,7 +1507,8 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n) return 0; } -static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) +static int __devexit snd_miro_isa_remove(struct device *devptr, + unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); dev_set_drvdata(devptr, NULL); @@ -1422,23 +1518,164 @@ static int __devexit snd_miro_remove(struct device *devptr, unsigned int dev) #define DEV_NAME "miro" static struct isa_driver snd_miro_driver = { - .match = snd_miro_match, - .probe = snd_miro_probe, - .remove = __devexit_p(snd_miro_remove), + .match = snd_miro_isa_match, + .probe = snd_miro_isa_probe, + .remove = __devexit_p(snd_miro_isa_remove), /* FIXME: suspend/resume */ .driver = { .name = DEV_NAME }, }; +#ifdef CONFIG_PNP + +static int __devinit snd_card_miro_pnp(struct snd_miro *chip, + struct pnp_card_link *card, + const struct pnp_card_device_id *pid) +{ + struct pnp_dev *pdev; + int err; + struct pnp_dev *devmpu; + struct pnp_dev *devmc; + + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) + return -EBUSY; + + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + if (devmpu == NULL) + return -EBUSY; + + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); + return err; + } + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + err); + return err; + } + + port = pnp_port_start(pdev, 1); + fm_port = pnp_port_start(pdev, 2) + 8; + + /* + * The MC(0) is never accessed and the miroSOUND PCM20 card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; + + irq = pnp_irq(pdev, 0); + dma1 = pnp_dma(pdev, 0); + dma2 = pnp_dma(pdev, 1); + + if (mpu_port > 0) { + err = pnp_activate_dev(devmpu); + if (err < 0) { + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); + mpu_port = -1; + return err; + } + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); + } + return 0; +} + +static int __devinit snd_miro_pnp_probe(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + struct snd_card *card; + int err; + struct snd_miro *miro; + + if (snd_miro_pnp_is_probed) + return -EBUSY; + if (!isapnp) + return -ENODEV; + err = snd_card_create(index, id, THIS_MODULE, + sizeof(struct snd_miro), &card); + if (err < 0) + return err; + + card->private_free = snd_card_miro_free; + miro = card->private_data; + + err = snd_card_miro_pnp(miro, pcard, pid); + if (err) { + snd_card_free(card); + return err; + } + + /* only miroSOUND PCM20 and PCM12 == OPTi924 */ + err = snd_miro_init(miro, OPTi9XX_HW_82C924); + if (err) { + snd_card_free(card); + return err; + } + + err = snd_miro_opti_check(miro); + if (err) { + snd_printk(KERN_ERR "OPTI chip not found\n"); + snd_card_free(card); + return err; + } + + snd_card_set_dev(card, &pcard->card->dev); + err = snd_miro_probe(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pnp_set_card_drvdata(pcard, card); + snd_miro_pnp_is_probed = 1; + return 0; +} + +static void __devexit snd_miro_pnp_remove(struct pnp_card_link * pcard) +{ + snd_card_free(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); + snd_miro_pnp_is_probed = 0; +} + +static struct pnp_card_driver miro_pnpc_driver = { + .flags = PNP_DRIVER_RES_DISABLE, + .name = "miro", + .id_table = snd_miro_pnpids, + .probe = snd_miro_pnp_probe, + .remove = __devexit_p(snd_miro_pnp_remove), +}; +#endif + static int __init alsa_card_miro_init(void) { +#ifdef CONFIG_PNP + pnp_register_card_driver(&miro_pnpc_driver); + if (snd_miro_pnp_is_probed) + return 0; + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif return isa_register_driver(&snd_miro_driver, 1); } static void __exit alsa_card_miro_exit(void) { - isa_unregister_driver(&snd_miro_driver); + if (!snd_miro_pnp_is_probed) { + isa_unregister_driver(&snd_miro_driver); + return; + } +#ifdef CONFIG_PNP + pnp_unregister_card_driver(&miro_pnpc_driver); +#endif } module_init(alsa_card_miro_init) diff --git a/sound/isa/opti9xx/miro.h b/sound/isa/opti9xx/miro.h deleted file mode 100644 index 6e1385b8e07..00000000000 --- a/sound/isa/opti9xx/miro.h +++ /dev/null @@ -1,73 +0,0 @@ -#ifndef _MIRO_H_ -#define _MIRO_H_ - -#define ACI_REG_COMMAND 0 /* write register offset */ -#define ACI_REG_STATUS 1 /* read register offset */ -#define ACI_REG_BUSY 2 /* busy register offset */ -#define ACI_REG_RDS 2 /* PCM20: RDS register offset */ -#define ACI_MINTIME 500 /* ACI time out limit */ - -#define ACI_SET_MUTE 0x0d -#define ACI_SET_POWERAMP 0x0f -#define ACI_SET_TUNERMUTE 0xa3 -#define ACI_SET_TUNERMONO 0xa4 -#define ACI_SET_IDE 0xd0 -#define ACI_SET_WSS 0xd1 -#define ACI_SET_SOLOMODE 0xd2 -#define ACI_SET_PREAMP 0x03 -#define ACI_GET_PREAMP 0x21 -#define ACI_WRITE_TUNE 0xa7 -#define ACI_READ_TUNERSTEREO 0xa8 -#define ACI_READ_TUNERSTATION 0xa9 -#define ACI_READ_VERSION 0xf1 -#define ACI_READ_IDCODE 0xf2 -#define ACI_INIT 0xff -#define ACI_STATUS 0xf0 -#define ACI_S_GENERAL 0x00 -#define ACI_ERROR_OP 0xdf - -/* ACI Mixer */ - -/* These are the values for the right channel GET registers. - Add an offset of 0x01 for the left channel register. - (left=right+0x01) */ - -#define ACI_GET_MASTER 0x03 -#define ACI_GET_MIC 0x05 -#define ACI_GET_LINE 0x07 -#define ACI_GET_CD 0x09 -#define ACI_GET_SYNTH 0x0b -#define ACI_GET_PCM 0x0d -#define ACI_GET_LINE1 0x10 /* Radio on PCM20 */ -#define ACI_GET_LINE2 0x12 - -#define ACI_GET_EQ1 0x22 /* from Bass ... */ -#define ACI_GET_EQ2 0x24 -#define ACI_GET_EQ3 0x26 -#define ACI_GET_EQ4 0x28 -#define ACI_GET_EQ5 0x2a -#define ACI_GET_EQ6 0x2c -#define ACI_GET_EQ7 0x2e /* ... to Treble */ - -/* And these are the values for the right channel SET registers. - For left channel access you have to add an offset of 0x08. - MASTER is an exception, which needs an offset of 0x01 */ - -#define ACI_SET_MASTER 0x00 -#define ACI_SET_MIC 0x30 -#define ACI_SET_LINE 0x31 -#define ACI_SET_CD 0x34 -#define ACI_SET_SYNTH 0x33 -#define ACI_SET_PCM 0x32 -#define ACI_SET_LINE1 0x35 /* Radio on PCM20 */ -#define ACI_SET_LINE2 0x36 - -#define ACI_SET_EQ1 0x40 /* from Bass ... */ -#define ACI_SET_EQ2 0x41 -#define ACI_SET_EQ3 0x42 -#define ACI_SET_EQ4 0x43 -#define ACI_SET_EQ5 0x44 -#define ACI_SET_EQ6 0x45 -#define ACI_SET_EQ7 0x46 /* ... to Treble */ - -#endif /* _MIRO_H_ */ diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 5cd555325b9..d08c3890644 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -141,15 +141,7 @@ struct snd_opti9xx { spinlock_t lock; - long wss_base; int irq; - int dma1; - int dma2; - - long fm_port; - - long mpu_port; - int mpu_irq; #ifdef CONFIG_PNP struct pnp_dev *dev; @@ -216,13 +208,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, spin_lock_init(&chip->lock); - chip->wss_base = -1; chip->irq = -1; - chip->dma1 = -1; - chip->dma2 = -1; - chip->fm_port = -1; - chip->mpu_port = -1; - chip->mpu_irq = -1; switch (hardware) { #ifndef OPTi93X @@ -348,7 +334,10 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, (snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask))) -static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) +static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, + long wss_base, + int irq, int dma1, int dma2, + long mpu_port, int mpu_irq) { unsigned char wss_base_bits; unsigned char irq_bits; @@ -416,7 +405,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) return -EINVAL; } - switch (chip->wss_base) { + switch (wss_base) { case 0x530: wss_base_bits = 0x00; break; @@ -430,14 +419,13 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", - chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); __skip_base: - switch (chip->irq) { + switch (irq) { //#ifdef OPTi93X case 5: irq_bits = 0x05; @@ -456,11 +444,11 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", irq); goto __skip_resources; } - switch (chip->dma1) { + switch (dma1) { case 0: dma_bits = 0x01; break; @@ -471,38 +459,36 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", - chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) - if (chip->dma1 == chip->dma2) { + if (dma1 == dma2) { snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } - switch (chip->dma2) { + switch (dma2) { case 0: case 1: break; default: - snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", - chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", dma2); goto __skip_resources; } dma_bits |= 0x04; #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, chip->wss_base); + outb(irq_bits << 3 | dma_bits, wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ __skip_resources: if (chip->hardware > OPTi9XX_HW_82C928) { - switch (chip->mpu_port) { + switch (mpu_port) { case 0: case -1: break; @@ -520,12 +506,11 @@ __skip_resources: break; default: snd_printk(KERN_WARNING - "MPU-401 port 0x%lx not valid\n", - chip->mpu_port); + "MPU-401 port 0x%lx not valid\n", mpu_port); goto __skip_mpu; } - switch (chip->mpu_irq) { + switch (mpu_irq) { case 5: mpu_irq_bits = 0x02; break; @@ -540,12 +525,12 @@ __skip_resources: break; default: snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", - chip->mpu_irq); + mpu_irq); goto __skip_mpu; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), - (chip->mpu_port <= 0) ? 0x00 : + (mpu_port <= 0) ? 0x00 : 0x80 | mpu_port_bits << 5 | mpu_irq_bits << 3, 0xf8); } @@ -701,6 +686,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) { static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; + int xdma2; struct snd_opti9xx *chip = card->private_data; struct snd_wss *codec; #ifdef CS4231 @@ -715,31 +701,25 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) "OPTi9xx MC")) == NULL) return -ENOMEM; - chip->wss_base = port; - chip->fm_port = fm_port; - chip->mpu_port = mpu_port; - chip->irq = irq; - chip->mpu_irq = mpu_irq; - chip->dma1 = dma1; #if defined(CS4231) || defined(OPTi93X) - chip->dma2 = dma2; + xdma2 = dma2; #else - chip->dma2 = -1; + xdma2 = -1; #endif - if (chip->wss_base == SNDRV_AUTO_PORT) { - chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); - if (chip->wss_base < 0) { + if (port == SNDRV_AUTO_PORT) { + port = snd_legacy_find_free_ioport(possible_ports, 4); + if (port < 0) { snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } - error = snd_opti9xx_configure(chip); + error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2, + mpu_port, mpu_irq); if (error) return error; - error = snd_wss_create(card, chip->wss_base + 4, -1, - chip->irq, chip->dma1, chip->dma2, + error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -763,35 +743,35 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) return error; #endif #ifdef OPTi93X - error = request_irq(chip->irq, snd_opti93x_interrupt, + error = request_irq(irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec); if (error < 0) { snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); return error; } #endif + chip->irq = irq; strcpy(card->driver, chip->name); sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1, chip->dma2); + card->shortname, pcm->name, port + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, chip->wss_base + 4, - chip->irq, chip->dma1); + card->shortname, pcm->name, port + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ - if (chip->mpu_port <= 0 || chip->mpu_port == SNDRV_AUTO_PORT) + if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) rmidi = NULL; - else - if ((error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - chip->mpu_port, 0, chip->mpu_irq, IRQF_DISABLED, - &rmidi))) + else { + error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port, 0, mpu_irq, IRQF_DISABLED, &rmidi); + if (error) snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - chip->mpu_port); + mpu_port); + } - if (chip->fm_port > 0 && chip->fm_port != SNDRV_AUTO_PORT) { + if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { struct snd_opl3 *opl3 = NULL; #ifndef OPTi93X if (chip->hardware == OPTi9XX_HW_82C928 || @@ -801,9 +781,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) /* assume we have an OPL4 */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x20, 0x20); - if (snd_opl4_create(card, - chip->fm_port, - chip->fm_port - 8, + if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) { /* no luck, use OPL3 instead */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), @@ -811,12 +789,10 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) } } #endif /* !OPTi93X */ - if (!opl3 && snd_opl3_create(card, - chip->fm_port, - chip->fm_port + 2, + if (!opl3 && snd_opl3_create(card, fm_port, fm_port + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - chip->fm_port, chip->fm_port + 4 - 1); + fm_port, fm_port + 4 - 1); } if (opl3) { error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 475220bbcc9..318ff0c823e 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -631,7 +631,7 @@ static struct sbmix_elem snd_sb16_ctl_mic_play_switch = static struct sbmix_elem snd_sb16_ctl_mic_play_vol = SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31); static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol = - SB_SINGLE("PC Speaker Volume", SB_DSP4_SPEAKER_DEV, 6, 3); + SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3); static struct sbmix_elem snd_sb16_ctl_capture_vol = SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3); static struct sbmix_elem snd_sb16_ctl_play_vol = @@ -689,7 +689,7 @@ static struct sbmix_elem snd_dt019x_ctl_cd_play_vol = static struct sbmix_elem snd_dt019x_ctl_mic_play_vol = SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7); static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol = - SB_SINGLE("PC Speaker Volume", SB_DT019X_SPKR_DEV, 0, 7); + SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7); static struct sbmix_elem snd_dt019x_ctl_line_play_vol = SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15); static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch = diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 66187122377..e2d5d2d3ed9 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1,5 +1,5 @@ /* - * Low-level ALSA driver for the ENSONIQ SoundScape PnP + * Low-level ALSA driver for the ENSONIQ SoundScape * Copyright (c) by Chris Rankin * * This driver was written in part using information obtained from @@ -25,31 +25,36 @@ #include <linux/err.h> #include <linux/isa.h> #include <linux/delay.h> +#include <linux/firmware.h> #include <linux/pnp.h> #include <linux/spinlock.h> #include <linux/moduleparam.h> #include <asm/dma.h> #include <sound/core.h> -#include <sound/hwdep.h> #include <sound/wss.h> #include <sound/mpu401.h> #include <sound/initval.h> -#include <sound/sscape_ioctl.h> - MODULE_AUTHOR("Chris Rankin"); -MODULE_DESCRIPTION("ENSONIQ SoundScape PnP driver"); +MODULE_DESCRIPTION("ENSONIQ SoundScape driver"); MODULE_LICENSE("GPL"); - -static int index[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IDX; -static char* id[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_STR; -static long port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; -static long wss_port[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_PORT; -static int irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; -static int mpu_irq[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_IRQ; -static int dma[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; -static int dma2[SNDRV_CARDS] __devinitdata = SNDRV_DEFAULT_DMA; +MODULE_FIRMWARE("sndscape.co0"); +MODULE_FIRMWARE("sndscape.co1"); +MODULE_FIRMWARE("sndscape.co2"); +MODULE_FIRMWARE("sndscape.co3"); +MODULE_FIRMWARE("sndscape.co4"); +MODULE_FIRMWARE("scope.cod"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static bool joystick[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index number for SoundScape soundcard"); @@ -75,6 +80,9 @@ MODULE_PARM_DESC(dma, "DMA # for SoundScape driver."); module_param_array(dma2, int, NULL, 0444); MODULE_PARM_DESC(dma2, "DMA2 # for SoundScape driver."); +module_param_array(joystick, bool, NULL, 0444); +MODULE_PARM_DESC(joystick, "Enable gameport."); + #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -101,14 +109,14 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #define RX_READY 0x01 #define TX_READY 0x02 -#define CMD_ACK 0x80 -#define CMD_SET_MIDI_VOL 0x84 -#define CMD_GET_MIDI_VOL 0x85 -#define CMD_XXX_MIDI_VOL 0x86 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d +#define CMD_ACK 0x80 +#define CMD_SET_MIDI_VOL 0x84 +#define CMD_GET_MIDI_VOL 0x85 +#define CMD_XXX_MIDI_VOL 0x86 +#define CMD_SET_EXTMIDI 0x8a +#define CMD_GET_EXTMIDI 0x8b +#define CMD_SET_MT32 0x8c +#define CMD_GET_MT32 0x8d enum GA_REG { GA_INTSTAT_REG = 0, @@ -127,7 +135,8 @@ enum GA_REG { enum card_type { - SSCAPE, + MEDIA_FX, /* Sequoia S-1000 */ + SSCAPE, /* Sequoia S-2000 */ SSCAPE_PNP, SSCAPE_VIVO, }; @@ -140,16 +149,7 @@ struct soundscape { struct resource *io_res; struct resource *wss_res; struct snd_wss *chip; - struct snd_mpu401 *mpu; - struct snd_hwdep *hw; - /* - * The MIDI device won't work until we've loaded - * its firmware via a hardware-dependent device IOCTL - */ - spinlock_t fwlock; - int hw_in_use; - unsigned long midi_usage; unsigned char midi_vol; }; @@ -161,28 +161,21 @@ static inline struct soundscape *get_card_soundscape(struct snd_card *c) return (struct soundscape *) (c->private_data); } -static inline struct soundscape *get_mpu401_soundscape(struct snd_mpu401 * mpu) -{ - return (struct soundscape *) (mpu->private_data); -} - -static inline struct soundscape *get_hwdep_soundscape(struct snd_hwdep * hw) -{ - return (struct soundscape *) (hw->private_data); -} - - /* * Allocates some kernel memory that we can use for DMA. * I think this means that the memory has to map to * contiguous pages of physical memory. */ -static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, unsigned long size) +static struct snd_dma_buffer *get_dmabuf(struct snd_dma_buffer *buf, + unsigned long size) { if (buf) { - if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), + if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, + snd_dma_isa_data(), size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate %lu bytes for DMA\n", size); + snd_printk(KERN_ERR "sscape: Failed to allocate " + "%lu bytes for DMA\n", + size); return NULL; } } @@ -199,13 +192,13 @@ static void free_dmabuf(struct snd_dma_buffer *buf) snd_dma_free_pages(buf); } - /* * This function writes to the SoundScape's control registers, * but doesn't do any locking. It's up to the caller to do that. * This is why this function is "unsafe" ... */ -static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsigned char val) +static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, + unsigned char val) { outb(reg, ODIE_ADDR_IO(io_base)); outb(val, ODIE_DATA_IO(io_base)); @@ -215,7 +208,8 @@ static inline void sscape_write_unsafe(unsigned io_base, enum GA_REG reg, unsign * Write to the SoundScape's control registers, and do the * necessary locking ... */ -static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char val) +static void sscape_write(struct soundscape *s, enum GA_REG reg, + unsigned char val) { unsigned long flags; @@ -228,7 +222,8 @@ static void sscape_write(struct soundscape *s, enum GA_REG reg, unsigned char va * Read from the SoundScape's control registers, but leave any * locking to the caller. This is why the function is "unsafe" ... */ -static inline unsigned char sscape_read_unsafe(unsigned io_base, enum GA_REG reg) +static inline unsigned char sscape_read_unsafe(unsigned io_base, + enum GA_REG reg) { outb(reg, ODIE_ADDR_IO(io_base)); return inb(ODIE_DATA_IO(io_base)); @@ -257,9 +252,8 @@ static inline void set_midi_mode_unsafe(unsigned io_base) static inline int host_read_unsafe(unsigned io_base) { int data = -1; - if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) { + if ((inb(HOST_CTRL_IO(io_base)) & RX_READY) != 0) data = inb(HOST_DATA_IO(io_base)); - } return data; } @@ -301,7 +295,7 @@ static inline int host_write_unsafe(unsigned io_base, unsigned char data) * Also leaves all locking-issues to the caller ... */ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, - unsigned timeout) + unsigned timeout) { int err; @@ -320,7 +314,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, * * NOTE: This check is based upon observation, not documentation. */ -static inline int verify_mpu401(const struct snd_mpu401 * mpu) +static inline int verify_mpu401(const struct snd_mpu401 *mpu) { return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } @@ -328,7 +322,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) /* * This is apparently the standard way to initailise an MPU-401 */ -static inline void initialise_mpu401(const struct snd_mpu401 * mpu) +static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { outb(0, MPU401D(mpu)); } @@ -338,9 +332,10 @@ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) * The AD1845 detection fails if we *don't* do this, so I * think that this is a good idea ... */ -static inline void activate_ad1845_unsafe(unsigned io_base) +static void activate_ad1845_unsafe(unsigned io_base) { - sscape_write_unsafe(io_base, GA_HMCTL_REG, (sscape_read_unsafe(io_base, GA_HMCTL_REG) & 0xcf) | 0x10); + unsigned char val = sscape_read_unsafe(io_base, GA_HMCTL_REG); + sscape_write_unsafe(io_base, GA_HMCTL_REG, (val & 0xcf) | 0x10); sscape_write_unsafe(io_base, GA_CDCFG_REG, 0x80); } @@ -359,24 +354,27 @@ static void soundscape_free(struct snd_card *c) * Tell the SoundScape to begin a DMA tranfer using the given channel. * All locking issues are left to the caller. */ -static inline void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) +static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) { - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) | 0x01); - sscape_write_unsafe(io_base, reg, sscape_read_unsafe(io_base, reg) & 0xfe); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) | 0x01); + sscape_write_unsafe(io_base, reg, + sscape_read_unsafe(io_base, reg) & 0xfe); } /* * Wait for a DMA transfer to complete. This is a "limited busy-wait", * and all locking issues are left to the caller. */ -static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned timeout) +static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, + unsigned timeout) { while (!(sscape_read_unsafe(io_base, reg) & 0x01) && (timeout != 0)) { udelay(100); --timeout; } /* while */ - return (sscape_read_unsafe(io_base, reg) & 0x01); + return sscape_read_unsafe(io_base, reg) & 0x01; } /* @@ -392,12 +390,12 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); - if ((x & 0xfe) == 0xfe) + if (x == 0xfe || x == 0xff) return 1; msleep(10); @@ -419,10 +417,10 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) do { unsigned long flags; - unsigned char x; + int x; spin_lock_irqsave(&s->lock, flags); - x = inb(HOST_DATA_IO(s->io_base)); + x = host_read_unsafe(s->io_base); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; @@ -436,15 +434,15 @@ static int host_startup_ack(struct soundscape *s, unsigned timeout) /* * Upload a byte-stream into the SoundScape using DMA channel A. */ -static int upload_dma_data(struct soundscape *s, - const unsigned char __user *data, - size_t size) +static int upload_dma_data(struct soundscape *s, const unsigned char *data, + size_t size) { unsigned long flags; struct snd_dma_buffer dma; int ret; + unsigned char val; - if (!get_dmabuf(&dma, PAGE_ALIGN(size))) + if (!get_dmabuf(&dma, PAGE_ALIGN(32 * 1024))) return -ENOMEM; spin_lock_irqsave(&s->lock, flags); @@ -452,70 +450,57 @@ static int upload_dma_data(struct soundscape *s, /* * Reset the board ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val & 0x3f); /* * Enable the DMA channels and configure them ... */ - sscape_write_unsafe(s->io_base, GA_DMACFG_REG, 0x50); - sscape_write_unsafe(s->io_base, GA_DMAA_REG, (s->chip->dma1 << 4) | DMA_8BIT); + val = (s->chip->dma1 << 4) | DMA_8BIT; + sscape_write_unsafe(s->io_base, GA_DMAA_REG, val); sscape_write_unsafe(s->io_base, GA_DMAB_REG, 0x20); /* * Take the board out of reset ... */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x80); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x80); /* - * Upload the user's data (firmware?) to the SoundScape + * Upload the firmware to the SoundScape * board through the DMA channel ... */ while (size != 0) { unsigned long len; - /* - * Apparently, copying to/from userspace can sleep. - * We are therefore forbidden from holding any - * spinlocks while we copy ... - */ - spin_unlock_irqrestore(&s->lock, flags); - - /* - * Remember that the data that we want to DMA - * comes from USERSPACE. We have already verified - * the userspace pointer ... - */ len = min(size, dma.bytes); - len -= __copy_from_user(dma.area, data, len); + memcpy(dma.area, data, len); data += len; size -= len; - /* - * Grab that spinlock again, now that we've - * finished copying! - */ - spin_lock_irqsave(&s->lock, flags); - snd_dma_program(s->chip->dma1, dma.addr, len, DMA_MODE_WRITE); sscape_start_dma_unsafe(s->io_base, GA_DMAA_REG); if (!sscape_wait_dma_unsafe(s->io_base, GA_DMAA_REG, 5000)) { /* - * Don't forget to release this spinlock we're holding ... + * Don't forget to release this spinlock we're holding */ spin_unlock_irqrestore(&s->lock, flags); - snd_printk(KERN_ERR "sscape: DMA upload has timed out\n"); + snd_printk(KERN_ERR + "sscape: DMA upload has timed out\n"); ret = -EAGAIN; goto _release_dma; } } /* while */ set_host_mode_unsafe(s->io_base); + outb(0x0, s->io_base); /* * Boot the board ... (I think) */ - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, sscape_read_unsafe(s->io_base, GA_HMCTL_REG) | 0x40); + val = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, val | 0x40); spin_unlock_irqrestore(&s->lock, flags); /* @@ -525,10 +510,12 @@ static int upload_dma_data(struct soundscape *s, */ ret = 0; if (!obp_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); + snd_printk(KERN_ERR "sscape: No response " + "from on-board processor after upload\n"); ret = -EAGAIN; } else if (!host_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); + snd_printk(KERN_ERR + "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -536,7 +523,7 @@ _release_dma: /* * NOTE!!! We are NOT holding any spinlocks at this point !!! */ - sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_ODIE ? 0x70 : 0x40)); + sscape_write(s, GA_DMAA_REG, (s->ic_type == IC_OPUS ? 0x40 : 0x70)); free_dmabuf(&dma); return ret; @@ -546,167 +533,76 @@ _release_dma: * Upload the bootblock(?) into the SoundScape. The only * purpose of this block of code seems to be to tell * us which version of the microcode we should be using. - * - * NOTE: The boot-block data resides in USER-SPACE!!! - * However, we have already verified its memory - * addresses by the time we get here. */ -static int sscape_upload_bootblock(struct soundscape *sscape, struct sscape_bootblock __user *bb) +static int sscape_upload_bootblock(struct snd_card *card) { + struct soundscape *sscape = get_card_soundscape(card); unsigned long flags; + const struct firmware *init_fw = NULL; int data = 0; int ret; - ret = upload_dma_data(sscape, bb->code, sizeof(bb->code)); - - spin_lock_irqsave(&sscape->lock, flags); - if (ret == 0) { - data = host_read_ctrl_unsafe(sscape->io_base, 100); - } - set_midi_mode_unsafe(sscape->io_base); - spin_unlock_irqrestore(&sscape->lock, flags); - - if (ret == 0) { - if (data < 0) { - snd_printk(KERN_ERR "sscape: timeout reading firmware version\n"); - ret = -EAGAIN; - } - else if (__copy_to_user(&bb->version, &data, sizeof(bb->version))) { - ret = -EFAULT; - } + ret = request_firmware(&init_fw, "scope.cod", card->dev); + if (ret < 0) { + snd_printk(KERN_ERR "sscape: Error loading scope.cod"); + return ret; } + ret = upload_dma_data(sscape, init_fw->data, init_fw->size); - return ret; -} - -/* - * Upload the microcode into the SoundScape. The - * microcode is 64K of data, and if we try to copy - * it into a local variable then we will SMASH THE - * KERNEL'S STACK! We therefore leave it in USER - * SPACE, and save ourselves from copying it at all. - */ -static int sscape_upload_microcode(struct soundscape *sscape, - const struct sscape_microcode __user *mc) -{ - unsigned long flags; - char __user *code; - int err; + release_firmware(init_fw); - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now. - * - * NOTE: This buffer is 64K long! That's WAY too big to - * copy into a stack-temporary anyway. - */ - if ( get_user(code, &mc->code) || - !access_ok(VERIFY_READ, code, SSCAPE_MICROCODE_SIZE) ) - return -EFAULT; + spin_lock_irqsave(&sscape->lock, flags); + if (ret == 0) + data = host_read_ctrl_unsafe(sscape->io_base, 100); - if ((err = upload_dma_data(sscape, code, SSCAPE_MICROCODE_SIZE)) == 0) { - snd_printk(KERN_INFO "sscape: MIDI firmware loaded\n"); - } + if (data & 0x10) + sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2f); - spin_lock_irqsave(&sscape->lock, flags); - set_midi_mode_unsafe(sscape->io_base); spin_unlock_irqrestore(&sscape->lock, flags); - initialise_mpu401(sscape->mpu); + data &= 0xf; + if (ret == 0 && data > 7) { + snd_printk(KERN_ERR + "sscape: timeout reading firmware version\n"); + ret = -EAGAIN; + } - return err; + return (ret == 0) ? data : ret; } /* - * Hardware-specific device functions, to implement special - * IOCTLs for the SoundScape card. This is how we upload - * the microcode into the card, for example, and so we - * must ensure that no two processes can open this device - * simultaneously, and that we can't open it at all if - * someone is using the MIDI device. + * Upload the microcode into the SoundScape. */ -static int sscape_hw_open(struct snd_hwdep * hw, struct file *file) +static int sscape_upload_microcode(struct snd_card *card, int version) { - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; + struct soundscape *sscape = get_card_soundscape(card); + const struct firmware *init_fw = NULL; + char name[14]; int err; - spin_lock_irqsave(&sscape->fwlock, flags); + snprintf(name, sizeof(name), "sndscape.co%d", version); - if ((sscape->midi_usage != 0) || sscape->hw_in_use) { - err = -EBUSY; - } else { - sscape->hw_in_use = 1; - err = 0; + err = request_firmware(&init_fw, name, card->dev); + if (err < 0) { + snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d", + version); + return err; } + err = upload_dma_data(sscape, init_fw->data, init_fw->size); + if (err == 0) + snd_printk(KERN_INFO "sscape: MIDI firmware loaded %d KBs\n", + init_fw->size >> 10); - spin_unlock_irqrestore(&sscape->fwlock, flags); - return err; -} - -static int sscape_hw_release(struct snd_hwdep * hw, struct file *file) -{ - register struct soundscape *sscape = get_hwdep_soundscape(hw); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - sscape->hw_in_use = 0; - spin_unlock_irqrestore(&sscape->fwlock, flags); - return 0; -} - -static int sscape_hw_ioctl(struct snd_hwdep * hw, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct soundscape *sscape = get_hwdep_soundscape(hw); - int err = -EBUSY; - - switch (cmd) { - case SND_SSCAPE_LOAD_BOOTB: - { - register struct sscape_bootblock __user *bb = (struct sscape_bootblock __user *) arg; - - /* - * We are going to have to copy this data into a special - * DMA-able buffer before we can upload it. We shall therefore - * just check that the data pointer is valid for now ... - */ - if ( !access_ok(VERIFY_READ, bb->code, sizeof(bb->code)) ) - return -EFAULT; - - /* - * Now check that we can write the firmware version number too... - */ - if ( !access_ok(VERIFY_WRITE, &bb->version, sizeof(bb->version)) ) - return -EFAULT; - - err = sscape_upload_bootblock(sscape, bb); - } - break; - - case SND_SSCAPE_LOAD_MCODE: - { - register const struct sscape_microcode __user *mc = (const struct sscape_microcode __user *) arg; - - err = sscape_upload_microcode(sscape, mc); - } - break; - - default: - err = -EINVAL; - break; - } /* switch */ + release_firmware(init_fw); return err; } - /* * Mixer control for the SoundScape's MIDI device. */ static int sscape_midi_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *uinfo) + struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; @@ -716,7 +612,7 @@ static int sscape_midi_info(struct snd_kcontrol *ctl, } static int sscape_midi_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; @@ -730,16 +626,18 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, } static int sscape_midi_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *uctl) + struct snd_ctl_elem_value *uctl) { struct snd_wss *chip = snd_kcontrol_chip(kctl); struct snd_card *card = chip->card; - register struct soundscape *s = get_card_soundscape(card); + struct soundscape *s = get_card_soundscape(card); unsigned long flags; int change; + unsigned char new_val; spin_lock_irqsave(&s->lock, flags); + new_val = uctl->value.integer.value[0] & 127; /* * We need to put the board into HOST mode before we * can send any volume-changing HOST commands ... @@ -752,15 +650,16 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, * and then perform another volume-related command. Perhaps the * first command is an "open" and the second command is a "close"? */ - if (s->midi_vol == ((unsigned char) uctl->value.integer. value[0] & 127)) { + if (s->midi_vol == new_val) { change = 0; goto __skip_change; } - change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) - && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) - && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); - s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; - __skip_change: + change = host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100) + && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100) + && host_write_ctrl_unsafe(s->io_base, new_val, 100); + s->midi_vol = new_val; +__skip_change: /* * Take the board out of HOST mode and back into MIDI mode ... @@ -784,20 +683,25 @@ static struct snd_kcontrol_new midi_mixer_ctl = { * These IRQs are encoded as bit patterns so that they can be * written to the control registers. */ -static unsigned __devinit get_irq_config(int irq) +static unsigned __devinit get_irq_config(int sscape_type, int irq) { static const int valid_irq[] = { 9, 5, 7, 10 }; + static const int old_irq[] = { 9, 7, 5, 15 }; unsigned cfg; - for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) { - if (irq == valid_irq[cfg]) - return cfg; - } /* for */ + if (sscape_type == MEDIA_FX) { + for (cfg = 0; cfg < ARRAY_SIZE(old_irq); ++cfg) + if (irq == old_irq[cfg]) + return cfg; + } else { + for (cfg = 0; cfg < ARRAY_SIZE(valid_irq); ++cfg) + if (irq == valid_irq[cfg]) + return cfg; + } return INVALID_IRQ; } - /* * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. @@ -842,11 +746,38 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) if (s->type != SSCAPE_VIVO && (d & 0x9f) != 0x0e) goto _done; - d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; - sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + if (s->ic_type == IC_OPUS) + activate_ad1845_unsafe(s->io_base); if (s->type == SSCAPE_VIVO) wss_io += 4; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); + + /* wait for WSS codec */ + for (d = 0; d < 500; d++) { + if ((inb(wss_io) & 0x80) == 0) + break; + spin_unlock_irqrestore(&s->lock, flags); + msleep(1); + spin_lock_irqsave(&s->lock, flags); + } + + if ((inb(wss_io) & 0x80) != 0) + goto _done; + + if (inb(wss_io + 2) == 0xff) + goto _done; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG) & 0x3f; + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d); + + if ((inb(wss_io) & 0x80) != 0) + s->type = MEDIA_FX; + + d = sscape_read_unsafe(s->io_base, GA_HMCTL_REG); + sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); /* wait for WSS codec */ for (d = 0; d < 500; d++) { if ((inb(wss_io) & 0x80) == 0) @@ -855,14 +786,13 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) msleep(1); spin_lock_irqsave(&s->lock, flags); } - snd_printd(KERN_INFO "init delay = %d ms\n", d); /* * SoundScape successfully detected! */ retval = 1; - _done: +_done: spin_unlock_irqrestore(&s->lock, flags); return retval; } @@ -873,63 +803,35 @@ static int __devinit detect_sscape(struct soundscape *s, long wss_io) * to crash the machine. Also check that someone isn't using the hardware * IOCTL device. */ -static int mpu401_open(struct snd_mpu401 * mpu) +static int mpu401_open(struct snd_mpu401 *mpu) { - int err; - if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, please load firmware\n"); - err = -ENODEV; - } else { - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - - if (sscape->hw_in_use || (sscape->midi_usage == ULONG_MAX)) { - err = -EBUSY; - } else { - ++(sscape->midi_usage); - err = 0; - } - - spin_unlock_irqrestore(&sscape->fwlock, flags); + snd_printk(KERN_ERR "sscape: MIDI disabled, " + "please load firmware\n"); + return -ENODEV; } - return err; -} - -static void mpu401_close(struct snd_mpu401 * mpu) -{ - register struct soundscape *sscape = get_mpu401_soundscape(mpu); - unsigned long flags; - - spin_lock_irqsave(&sscape->fwlock, flags); - --(sscape->midi_usage); - spin_unlock_irqrestore(&sscape->fwlock, flags); + return 0; } /* * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. */ -static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) +static int __devinit create_mpu401(struct snd_card *card, int devnum, + unsigned long port, int irq) { struct soundscape *sscape = get_card_soundscape(card); struct snd_rawmidi *rawmidi; int err; - if ((err = snd_mpu401_uart_new(card, devnum, - MPU401_HW_MPU401, - port, MPU401_INFO_INTEGRATED, - irq, IRQF_DISABLED, - &rawmidi)) == 0) { - struct snd_mpu401 *mpu = (struct snd_mpu401 *) rawmidi->private_data; + err = snd_mpu401_uart_new(card, devnum, MPU401_HW_MPU401, port, + MPU401_INFO_INTEGRATED, irq, IRQF_DISABLED, + &rawmidi); + if (err == 0) { + struct snd_mpu401 *mpu = rawmidi->private_data; mpu->open_input = mpu401_open; mpu->open_output = mpu401_open; - mpu->close_input = mpu401_close; - mpu->close_output = mpu401_close; mpu->private_data = sscape; - sscape->mpu = mpu; initialise_mpu401(mpu); } @@ -950,32 +852,34 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, register struct soundscape *sscape = get_card_soundscape(card); struct snd_wss *chip; int err; + int codec_type = WSS_HW_DETECT; - if (sscape->type == SSCAPE_VIVO) - port += 4; + switch (sscape->type) { + case MEDIA_FX: + case SSCAPE: + /* + * There are some freak examples of early Soundscape cards + * with CS4231 instead of AD1848/CS4248. Unfortunately, the + * CS4231 works only in CS4248 compatibility mode on + * these cards so force it. + */ + if (sscape->ic_type != IC_OPUS) + codec_type = WSS_HW_AD1848; + break; - if (dma1 == dma2) - dma2 = -1; + case SSCAPE_VIVO: + port += 4; + break; + default: + break; + } err = snd_wss_create(card, port, -1, irq, dma1, dma2, - WSS_HW_DETECT, WSS_HWSHARE_DMA1, &chip); + codec_type, WSS_HWSHARE_DMA1, &chip); if (!err) { unsigned long flags; struct snd_pcm *pcm; -/* - * It turns out that the PLAYBACK_ENABLE bit is set - * by the lowlevel driver ... - * -#define AD1845_IFACE_CONFIG \ - (CS4231_AUTOCALIB | CS4231_RECORD_ENABLE | CS4231_PLAYBACK_ENABLE) - snd_wss_mce_up(chip); - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_IFACE_CTRL, AD1845_IFACE_CONFIG); - spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_mce_down(chip); - */ - if (sscape->type != SSCAPE_VIVO) { /* * The input clock frequency on the SoundScape must @@ -1022,17 +926,10 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, } } - strcpy(card->driver, "SoundScape"); - strcpy(card->shortname, pcm->name); - snprintf(card->longname, sizeof(card->longname), - "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", - pcm->name, chip->port, chip->irq, - chip->dma1, chip->dma2); - sscape->chip = chip; } - _error: +_error: return err; } @@ -1051,21 +948,8 @@ static int __devinit create_sscape(int dev, struct snd_card *card) struct resource *wss_res; unsigned long flags; int err; - - /* - * Check that the user didn't pass us garbage data ... - */ - irq_cfg = get_irq_config(irq[dev]); - if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; - } - - mpu_irq_cfg = get_irq_config(mpu_irq[dev]); - if (mpu_irq_cfg == INVALID_IRQ) { - printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; - } + int val; + const char *name; /* * Grab IO ports that we will need to probe so that we @@ -1098,41 +982,51 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } spin_lock_init(&sscape->lock); - spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); + printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + sscape->io_base); err = -ENODEV; goto _release_dma; } - printk(KERN_INFO "sscape: hardware detected at 0x%x, using IRQ %d, DMA %d\n", - sscape->io_base, irq[dev], dma[dev]); + switch (sscape->type) { + case MEDIA_FX: + name = "MediaFX/SoundFX"; + break; + case SSCAPE: + name = "Soundscape"; + break; + case SSCAPE_PNP: + name = "Soundscape PnP"; + break; + case SSCAPE_VIVO: + name = "Soundscape VIVO"; + break; + default: + name = "unknown Soundscape"; + break; + } - if (sscape->type != SSCAPE_VIVO) { - /* - * Now create the hardware-specific device so that we can - * load the microcode into the on-board processor. - * We cannot use the MPU-401 MIDI system until this firmware - * has been loaded into the card. - */ - err = snd_hwdep_new(card, "MC68EC000", 0, &(sscape->hw)); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "firmware device\n"); - goto _release_dma; - } - strlcpy(sscape->hw->name, "SoundScape M68K", - sizeof(sscape->hw->name)); - sscape->hw->name[sizeof(sscape->hw->name) - 1] = '\0'; - sscape->hw->iface = SNDRV_HWDEP_IFACE_SSCAPE; - sscape->hw->ops.open = sscape_hw_open; - sscape->hw->ops.release = sscape_hw_release; - sscape->hw->ops.ioctl = sscape_hw_ioctl; - sscape->hw->private_data = sscape; + printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", + name, sscape->io_base, irq[dev], dma[dev]); + + /* + * Check that the user didn't pass us garbage data ... + */ + irq_cfg = get_irq_config(sscape->type, irq[dev]); + if (irq_cfg == INVALID_IRQ) { + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + return -ENXIO; + } + + mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); + if (mpu_irq_cfg == INVALID_IRQ) { + snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + return -ENXIO; } /* @@ -1141,9 +1035,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) */ spin_lock_irqsave(&sscape->lock, flags); - activate_ad1845_unsafe(sscape->io_base); - - sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x00); /* disable */ sscape_write_unsafe(sscape->io_base, GA_SMCFGA_REG, 0x2e); sscape_write_unsafe(sscape->io_base, GA_SMCFGB_REG, 0x00); @@ -1151,15 +1042,23 @@ static int __devinit create_sscape(int dev, struct snd_card *card) * Enable and configure the DMA channels ... */ sscape_write_unsafe(sscape->io_base, GA_DMACFG_REG, 0x50); - dma_cfg = (sscape->ic_type == IC_ODIE ? 0x70 : 0x40); + dma_cfg = (sscape->ic_type == IC_OPUS ? 0x40 : 0x70); sscape_write_unsafe(sscape->io_base, GA_DMAA_REG, dma_cfg); sscape_write_unsafe(sscape->io_base, GA_DMAB_REG, 0x20); - sscape_write_unsafe(sscape->io_base, - GA_INTCFG_REG, 0xf0 | (mpu_irq_cfg << 2) | mpu_irq_cfg); + mpu_irq_cfg |= mpu_irq_cfg << 2; + val = sscape_read_unsafe(sscape->io_base, GA_HMCTL_REG) & 0xF7; + if (joystick[dev]) + val |= 8; + sscape_write_unsafe(sscape->io_base, GA_HMCTL_REG, val | 0x10); + sscape_write_unsafe(sscape->io_base, GA_INTCFG_REG, 0xf0 | mpu_irq_cfg); sscape_write_unsafe(sscape->io_base, GA_CDCFG_REG, 0x09 | DMA_8BIT | (dma[dev] << 4) | (irq_cfg << 1)); + /* + * Enable the master IRQ ... + */ + sscape_write_unsafe(sscape->io_base, GA_INTENA_REG, 0x80); spin_unlock_irqrestore(&sscape->lock, flags); @@ -1170,32 +1069,56 @@ static int __devinit create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - printk(KERN_ERR "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + snd_printk(KERN_ERR + "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); goto _release_dma; } + strcpy(card->driver, "SoundScape"); + strcpy(card->shortname, name); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx, IRQ %d, DMA1 %d, DMA2 %d\n", + name, sscape->chip->port, sscape->chip->irq, + sscape->chip->dma1, sscape->chip->dma2); + #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); - if (err < 0) { - printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); - goto _release_dma; - } + err = sscape_upload_bootblock(card); + if (err >= 0) + err = sscape_upload_microcode(card, err); - /* - * Enable the master IRQ ... - */ - sscape_write(sscape, GA_INTENA_REG, 0x80); + if (err == 0) { + err = create_mpu401(card, MIDI_DEVNUM, port[dev], + mpu_irq[dev]); + if (err < 0) { + snd_printk(KERN_ERR "sscape: Failed to create " + "MPU-401 device at 0x%lx\n", + port[dev]); + goto _release_dma; + } - /* - * Initialize mixer - */ - sscape->midi_vol = 0; - host_write_ctrl_unsafe(sscape->io_base, CMD_SET_MIDI_VOL, 100); - host_write_ctrl_unsafe(sscape->io_base, 0, 100); - host_write_ctrl_unsafe(sscape->io_base, CMD_XXX_MIDI_VOL, 100); + /* + * Initialize mixer + */ + spin_lock_irqsave(&sscape->lock, flags); + sscape->midi_vol = 0; + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_XXX_MIDI_VOL, 100); + host_write_ctrl_unsafe(sscape->io_base, + sscape->midi_vol, 100); + host_write_ctrl_unsafe(sscape->io_base, + CMD_SET_EXTMIDI, 100); + host_write_ctrl_unsafe(sscape->io_base, + 0, 100); + host_write_ctrl_unsafe(sscape->io_base, CMD_ACK, 100); + + set_midi_mode_unsafe(sscape->io_base); + spin_unlock_irqrestore(&sscape->lock, flags); + } } /* @@ -1231,7 +1154,8 @@ static int __devinit snd_sscape_match(struct device *pdev, unsigned int i) mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { printk(KERN_INFO - "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); + "sscape: insufficient parameters, " + "need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1253,13 +1177,15 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev) sscape->type = SSCAPE; dma[dev] &= 0x03; + snd_card_set_dev(card, pdev); + ret = create_sscape(dev, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, pdev); - if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + ret = snd_card_register(card); + if (ret < 0) { + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } dev_set_drvdata(pdev, card); @@ -1311,36 +1237,20 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, * Allow this function to fail *quietly* if all the ISA PnP * devices were configured using module parameters instead. */ - if ((idx = get_next_autoindex(idx)) >= SNDRV_CARDS) + idx = get_next_autoindex(idx); + if (idx >= SNDRV_CARDS) return -ENOSPC; /* - * We have found a candidate ISA PnP card. Now we - * have to check that it has the devices that we - * expect it to have. - * - * We will NOT try and autoconfigure all of the resources - * needed and then activate the card as we are assuming that - * has already been done at boot-time using /proc/isapnp. - * We shall simply try to give each active card the resources - * that it wants. This is a sensible strategy for a modular - * system where unused modules are unloaded regularly. - * - * This strategy is utterly useless if we compile the driver - * into the kernel, of course. - */ - // printk(KERN_INFO "sscape: %s\n", card->name); - - /* * Check that we still have room for another sound card ... */ dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); - if (! dev) + if (!dev) return -ENODEV; if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - printk(KERN_INFO "sscape: device is inactive\n"); + snd_printk(KERN_INFO "sscape: device is inactive\n"); return -EBUSY; } } @@ -1378,14 +1288,15 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard, wss_port[idx] = pnp_port_start(dev, 1); dma2[idx] = pnp_dma(dev, 1); } + snd_card_set_dev(card, &pcard->card->dev); ret = create_sscape(idx, card); if (ret < 0) goto _release_card; - snd_card_set_dev(card, &pcard->card->dev); - if ((ret = snd_card_register(card)) < 0) { - printk(KERN_ERR "sscape: Failed to register sound card\n"); + ret = snd_card_register(card); + if (ret < 0) { + snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); goto _release_card; } diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 5d2ba1b749a..5b9d6c18bc4 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1682,7 +1682,7 @@ static void snd_wss_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ -int snd_wss_free(struct snd_wss *chip) +static int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); @@ -1705,7 +1705,6 @@ int snd_wss_free(struct snd_wss *chip) kfree(chip); return 0; } -EXPORT_SYMBOL(snd_wss_free); static int snd_wss_dev_free(struct snd_device *device) { @@ -2198,84 +2197,61 @@ EXPORT_SYMBOL(snd_wss_put_double); static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_5bit_12db_max, -3450, 150, 0); static const DECLARE_TLV_DB_SCALE(db_scale_rec_gain, 0, 150, 0); +static const DECLARE_TLV_DB_SCALE(db_scale_4bit, -4500, 300, 0); -static struct snd_kcontrol_new snd_ad1848_controls[] = { -WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, - 7, 7, 1, 1), +static struct snd_kcontrol_new snd_wss_controls[] = { +WSS_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, - db_scale_6bit), + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1, + db_scale_6bit), WSS_DOUBLE("Aux Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE("Aux Playback Switch", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), WSS_DOUBLE_TLV("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, - db_scale_5bit_12db_max), + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1, + db_scale_5bit_12db_max), WSS_DOUBLE_TLV("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0, db_scale_rec_gain), { - .name = "Capture Source", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", .info = snd_wss_info_mux, .get = snd_wss_get_mux, .put = snd_wss_put_mux, }, -WSS_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 1, 63, 0, - db_scale_6bit), -}; - -static struct snd_kcontrol_new snd_wss_controls[] = { -WSS_DOUBLE("PCM Playback Switch", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 63, 1), +WSS_DOUBLE("Mic Boost (+20dB)", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +WSS_SINGLE("Loopback Capture Switch", 0, + CS4231_LOOPBACK, 0, 1, 0), +WSS_SINGLE_TLV("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1, + db_scale_6bit), WSS_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), -WSS_DOUBLE("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1), -WSS_DOUBLE("Aux Playback Switch", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -WSS_DOUBLE("Aux Playback Switch", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -WSS_DOUBLE("Aux Playback Volume", 1, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), -WSS_SINGLE("Mono Playback Switch", 0, +WSS_DOUBLE_TLV("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 31, 1, + db_scale_5bit_12db_max), +WSS_SINGLE("Beep Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), -WSS_SINGLE("Mono Playback Volume", 0, - CS4231_MONO_CTRL, 0, 15, 1), +WSS_SINGLE_TLV("Beep Playback Volume", 0, + CS4231_MONO_CTRL, 0, 15, 1, + db_scale_4bit), WSS_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), -WSS_SINGLE("Mono Output Playback Bypass", 0, +WSS_SINGLE("Beep Bypass Playback Switch", 0, CS4231_MONO_CTRL, 5, 1, 0), -WSS_DOUBLE("Capture Volume", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_wss_info_mux, - .get = snd_wss_get_mux, - .put = snd_wss_put_mux, -}, -WSS_DOUBLE("Mic Boost", 0, - CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), -WSS_SINGLE("Loopback Capture Switch", 0, - CS4231_LOOPBACK, 0, 1, 0), -WSS_SINGLE("Loopback Capture Volume", 0, - CS4231_LOOPBACK, 2, 63, 1) }; static struct snd_kcontrol_new snd_opti93x_controls[] = { WSS_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -WSS_DOUBLE("Master Playback Volume", 0, - OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +WSS_DOUBLE_TLV("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1, + db_scale_6bit), WSS_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), WSS_DOUBLE("PCM Playback Volume", 0, @@ -2334,22 +2310,21 @@ int snd_wss_mixer(struct snd_wss *chip) if (err < 0) return err; } - else if (chip->hardware & WSS_HW_AD1848_MASK) - for (idx = 0; idx < ARRAY_SIZE(snd_ad1848_controls); idx++) { - err = snd_ctl_add(card, - snd_ctl_new1(&snd_ad1848_controls[idx], - chip)); - if (err < 0) - return err; - } - else - for (idx = 0; idx < ARRAY_SIZE(snd_wss_controls); idx++) { + else { + int count = ARRAY_SIZE(snd_wss_controls); + + /* Use only the first 11 entries on AD1848 */ + if (chip->hardware & WSS_HW_AD1848_MASK) + count = 11; + + for (idx = 0; idx < count; idx++) { err = snd_ctl_add(card, snd_ctl_new1(&snd_wss_controls[idx], chip)); if (err < 0) return err; } + } return 0; } EXPORT_SYMBOL(snd_wss_mixer); diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index bcf2a0698d5..135a2b77cc4 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -287,18 +287,6 @@ config SOUND_DMAP Say Y unless you have 16MB or more RAM or a PCI sound card. -config SOUND_SSCAPE - tristate "Ensoniq SoundScape support" - help - Answer Y if you have a sound card based on the Ensoniq SoundScape - chipset. Such cards are being manufactured at least by Ensoniq, Spea - and Reveal (Reveal makes also other cards). - - If you compile the driver into the kernel, you have to add - "sscape=<io>,<irq>,<dma>,<mpuio>,<mpuirq>" to the kernel command - line. - - config SOUND_VMIDI tristate "Loopback MIDI device support" help diff --git a/sound/oss/Makefile b/sound/oss/Makefile index e0ae4d4d6a5..567b8a74178 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -13,7 +13,6 @@ obj-$(CONFIG_SOUND_SH_DAC_AUDIO) += sh_dac_audio.o obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o -obj-$(CONFIG_SOUND_SSCAPE) += sscape.o ad1848.o mpu401.o obj-$(CONFIG_SOUND_MSS) += ad1848.o obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o diff --git a/sound/oss/audio.c b/sound/oss/audio.c index b69c05b7ea7..7df48a25c4e 100644 --- a/sound/oss/audio.c +++ b/sound/oss/audio.c @@ -838,7 +838,7 @@ static int dma_ioctl(int dev, unsigned int cmd, void __user *arg) if ((err = audio_devs[dev]->d->prepare_for_input(dev, dmap_in->fragment_size, dmap_in->nbufs)) < 0) { spin_unlock_irqrestore(&dmap_in->lock,flags); - return -err; + return err; } dmap_in->dma_mode = DMODE_INPUT; audio_devs[dev]->enable_bits |= PCM_ENABLE_INPUT; diff --git a/sound/oss/midi_synth.c b/sound/oss/midi_synth.c index 9e450988ed3..3bc7104c537 100644 --- a/sound/oss/midi_synth.c +++ b/sound/oss/midi_synth.c @@ -426,7 +426,7 @@ midi_synth_open(int dev, int mode) int err; struct midi_input_info *inc; - if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL) + if (orig_dev < 0 || orig_dev >= num_midis || midi_devs[orig_dev] == NULL) return -ENXIO; midi2synth[orig_dev] = dev; diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 734b8f9e2f7..0af9d24feb8 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -770,7 +770,7 @@ static int mpu_synth_ioctl(int dev, unsigned int cmd, void __user *arg) midi_dev = synth_devs[dev]->midi_dev; - if (midi_dev < 0 || midi_dev > num_midis || midi_devs[midi_dev] == NULL) + if (midi_dev < 0 || midi_dev >= num_midis || midi_devs[midi_dev] == NULL) return -ENXIO; devc = &dev_conf[midi_dev]; diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index b2ed8757542..4153752507e 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -164,9 +164,6 @@ static ssize_t dac_audio_write(struct file *file, const char *buf, size_t count, int free; int nbytes; - if (count < 0) - return -EINVAL; - if (!count) { dac_audio_sync(); return 0; diff --git a/sound/oss/sscape.c b/sound/oss/sscape.c deleted file mode 100644 index 30c36d1f35d..00000000000 --- a/sound/oss/sscape.c +++ /dev/null @@ -1,1480 +0,0 @@ -/* - * sound/oss/sscape.c - * - * Low level driver for Ensoniq SoundScape - * - * - * Copyright (C) by Hannu Savolainen 1993-1997 - * - * OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL) - * Version 2 (June 1991). See the "COPYING" file distributed with this software - * for more info. - * - * - * Thomas Sailer : ioctl code reworked (vmalloc/vfree removed) - * Sergey Smitienko : ensoniq p'n'p support - * Christoph Hellwig : adapted to module_init/module_exit - * Bartlomiej Zolnierkiewicz : added __init to attach_sscape() - * Chris Rankin : Specify that this module owns the coprocessor - * Arnaldo C. de Melo : added missing restore_flags in sscape_pnp_upload_file - */ - -#include <linux/init.h> -#include <linux/module.h> - -#include "sound_config.h" -#include "sound_firmware.h" - -#include <linux/types.h> -#include <linux/errno.h> -#include <linux/signal.h> -#include <linux/fcntl.h> -#include <linux/ctype.h> -#include <linux/stddef.h> -#include <linux/kmod.h> -#include <asm/dma.h> -#include <asm/io.h> -#include <linux/wait.h> -#include <linux/slab.h> -#include <linux/ioport.h> -#include <linux/delay.h> -#include <linux/proc_fs.h> -#include <linux/mm.h> -#include <linux/spinlock.h> - -#include "coproc.h" - -#include "ad1848.h" -#include "mpu401.h" - -/* - * I/O ports - */ -#define MIDI_DATA 0 -#define MIDI_CTRL 1 -#define HOST_CTRL 2 -#define TX_READY 0x02 -#define RX_READY 0x01 -#define HOST_DATA 3 -#define ODIE_ADDR 4 -#define ODIE_DATA 5 - -/* - * Indirect registers - */ - -#define GA_INTSTAT_REG 0 -#define GA_INTENA_REG 1 -#define GA_DMAA_REG 2 -#define GA_DMAB_REG 3 -#define GA_INTCFG_REG 4 -#define GA_DMACFG_REG 5 -#define GA_CDCFG_REG 6 -#define GA_SMCFGA_REG 7 -#define GA_SMCFGB_REG 8 -#define GA_HMCTL_REG 9 - -/* - * DMA channel identifiers (A and B) - */ - -#define SSCAPE_DMA_A 0 -#define SSCAPE_DMA_B 1 - -#define PORT(name) (devc->base+name) - -/* - * Host commands recognized by the OBP microcode - */ - -#define CMD_GEN_HOST_ACK 0x80 -#define CMD_GEN_MPU_ACK 0x81 -#define CMD_GET_BOARD_TYPE 0x82 -#define CMD_SET_CONTROL 0x88 /* Old firmware only */ -#define CMD_GET_CONTROL 0x89 /* Old firmware only */ -#define CTL_MASTER_VOL 0 -#define CTL_MIC_MODE 2 -#define CTL_SYNTH_VOL 4 -#define CTL_WAVE_VOL 7 -#define CMD_SET_EXTMIDI 0x8a -#define CMD_GET_EXTMIDI 0x8b -#define CMD_SET_MT32 0x8c -#define CMD_GET_MT32 0x8d - -#define CMD_ACK 0x80 - -#define IC_ODIE 1 -#define IC_OPUS 2 - -typedef struct sscape_info -{ - int base, irq, dma; - - int codec, codec_irq; /* required to setup pnp cards*/ - int codec_type; - int ic_type; - char* raw_buf; - unsigned long raw_buf_phys; - int buffsize; /* -------------------------- */ - spinlock_t lock; - int ok; /* Properly detected */ - int failed; - int dma_allocated; - int codec_audiodev; - int opened; - int *osp; - int my_audiodev; -} sscape_info; - -static struct sscape_info adev_info = { - 0 -}; - -static struct sscape_info *devc = &adev_info; -static int sscape_mididev = -1; - -/* Some older cards have assigned interrupt bits differently than new ones */ -static char valid_interrupts_old[] = { - 9, 7, 5, 15 -}; - -static char valid_interrupts_new[] = { - 9, 5, 7, 10 -}; - -static char *valid_interrupts = valid_interrupts_new; - -/* - * See the bottom of the driver. This can be set by spea =0/1. - */ - -#ifdef REVEAL_SPEA -static char old_hardware = 1; -#else -static char old_hardware; -#endif - -static void sleep(unsigned howlong) -{ - current->state = TASK_INTERRUPTIBLE; - schedule_timeout(howlong); -} - -static unsigned char sscape_read(struct sscape_info *devc, int reg) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&devc->lock,flags); - outb(reg, PORT(ODIE_ADDR)); - val = inb(PORT(ODIE_DATA)); - spin_unlock_irqrestore(&devc->lock,flags); - return val; -} - -static void __sscape_write(int reg, int data) -{ - outb(reg, PORT(ODIE_ADDR)); - outb(data, PORT(ODIE_DATA)); -} - -static void sscape_write(struct sscape_info *devc, int reg, int data) -{ - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - __sscape_write(reg, data); - spin_unlock_irqrestore(&devc->lock,flags); -} - -static unsigned char sscape_pnp_read_codec(sscape_info* devc, unsigned char reg) -{ - unsigned char res; - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - outb( reg, devc -> codec); - res = inb (devc -> codec + 1); - spin_unlock_irqrestore(&devc->lock,flags); - return res; - -} - -static void sscape_pnp_write_codec(sscape_info* devc, unsigned char reg, unsigned char data) -{ - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - outb( reg, devc -> codec); - outb( data, devc -> codec + 1); - spin_unlock_irqrestore(&devc->lock,flags); -} - -static void host_open(struct sscape_info *devc) -{ - outb((0x00), PORT(HOST_CTRL)); /* Put the board to the host mode */ -} - -static void host_close(struct sscape_info *devc) -{ - outb((0x03), PORT(HOST_CTRL)); /* Put the board to the MIDI mode */ -} - -static int host_write(struct sscape_info *devc, unsigned char *data, int count) -{ - unsigned long flags; - int i, timeout_val; - - spin_lock_irqsave(&devc->lock,flags); - /* - * Send the command and data bytes - */ - - for (i = 0; i < count; i++) - { - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - if (inb(PORT(HOST_CTRL)) & TX_READY) - break; - - if (timeout_val <= 0) - { - spin_unlock_irqrestore(&devc->lock,flags); - return 0; - } - outb(data[i], PORT(HOST_DATA)); - } - spin_unlock_irqrestore(&devc->lock,flags); - return 1; -} - -static int host_read(struct sscape_info *devc) -{ - unsigned long flags; - int timeout_val; - unsigned char data; - - spin_lock_irqsave(&devc->lock,flags); - /* - * Read a byte - */ - - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - if (inb(PORT(HOST_CTRL)) & RX_READY) - break; - - if (timeout_val <= 0) - { - spin_unlock_irqrestore(&devc->lock,flags); - return -1; - } - data = inb(PORT(HOST_DATA)); - spin_unlock_irqrestore(&devc->lock,flags); - return data; -} - -#if 0 /* unused */ -static int host_command1(struct sscape_info *devc, int cmd) -{ - unsigned char buf[10]; - buf[0] = (unsigned char) (cmd & 0xff); - return host_write(devc, buf, 1); -} -#endif /* unused */ - - -static int host_command2(struct sscape_info *devc, int cmd, int parm1) -{ - unsigned char buf[10]; - - buf[0] = (unsigned char) (cmd & 0xff); - buf[1] = (unsigned char) (parm1 & 0xff); - - return host_write(devc, buf, 2); -} - -static int host_command3(struct sscape_info *devc, int cmd, int parm1, int parm2) -{ - unsigned char buf[10]; - - buf[0] = (unsigned char) (cmd & 0xff); - buf[1] = (unsigned char) (parm1 & 0xff); - buf[2] = (unsigned char) (parm2 & 0xff); - return host_write(devc, buf, 3); -} - -static void set_mt32(struct sscape_info *devc, int value) -{ - host_open(devc); - host_command2(devc, CMD_SET_MT32, value ? 1 : 0); - if (host_read(devc) != CMD_ACK) - { - /* printk( "SNDSCAPE: Setting MT32 mode failed\n"); */ - } - host_close(devc); -} - -static void set_control(struct sscape_info *devc, int ctrl, int value) -{ - host_open(devc); - host_command3(devc, CMD_SET_CONTROL, ctrl, value); - if (host_read(devc) != CMD_ACK) - { - /* printk( "SNDSCAPE: Setting control (%d) failed\n", ctrl); */ - } - host_close(devc); -} - -static void do_dma(struct sscape_info *devc, int dma_chan, unsigned long buf, int blk_size, int mode) -{ - unsigned char temp; - - if (dma_chan != SSCAPE_DMA_A) - { - printk(KERN_WARNING "soundscape: Tried to use DMA channel != A. Why?\n"); - return; - } - audio_devs[devc->codec_audiodev]->flags &= ~DMA_AUTOMODE; - DMAbuf_start_dma(devc->codec_audiodev, buf, blk_size, mode); - audio_devs[devc->codec_audiodev]->flags |= DMA_AUTOMODE; - - temp = devc->dma << 4; /* Setup DMA channel select bits */ - if (devc->dma <= 3) - temp |= 0x80; /* 8 bit DMA channel */ - - temp |= 1; /* Trigger DMA */ - sscape_write(devc, GA_DMAA_REG, temp); - temp &= 0xfe; /* Clear DMA trigger */ - sscape_write(devc, GA_DMAA_REG, temp); -} - -static int verify_mpu(struct sscape_info *devc) -{ - /* - * The SoundScape board could be in three modes (MPU, 8250 and host). - * If the card is not in the MPU mode, enabling the MPU driver will - * cause infinite loop (the driver believes that there is always some - * received data in the buffer. - * - * Detect this by looking if there are more than 10 received MIDI bytes - * (0x00) in the buffer. - */ - - int i; - - for (i = 0; i < 10; i++) - { - if (inb(devc->base + HOST_CTRL) & 0x80) - return 1; - - if (inb(devc->base) != 0x00) - return 1; - } - printk(KERN_WARNING "SoundScape: The device is not in the MPU-401 mode\n"); - return 0; -} - -static int sscape_coproc_open(void *dev_info, int sub_device) -{ - if (sub_device == COPR_MIDI) - { - set_mt32(devc, 0); - if (!verify_mpu(devc)) - return -EIO; - } - return 0; -} - -static void sscape_coproc_close(void *dev_info, int sub_device) -{ - struct sscape_info *devc = dev_info; - unsigned long flags; - - spin_lock_irqsave(&devc->lock,flags); - if (devc->dma_allocated) - { - __sscape_write(GA_DMAA_REG, 0x20); /* DMA channel disabled */ - devc->dma_allocated = 0; - } - spin_unlock_irqrestore(&devc->lock,flags); - return; -} - -static void sscape_coproc_reset(void *dev_info) -{ -} - -static int sscape_download_boot(struct sscape_info *devc, unsigned char *block, int size, int flag) -{ - unsigned long flags; - unsigned char temp; - volatile int done, timeout_val; - static unsigned char codec_dma_bits; - - if (flag & CPF_FIRST) - { - /* - * First block. Have to allocate DMA and to reset the board - * before continuing. - */ - - spin_lock_irqsave(&devc->lock,flags); - codec_dma_bits = sscape_read(devc, GA_CDCFG_REG); - - if (devc->dma_allocated == 0) - devc->dma_allocated = 1; - - spin_unlock_irqrestore(&devc->lock,flags); - - sscape_write(devc, GA_HMCTL_REG, - (temp = sscape_read(devc, GA_HMCTL_REG)) & 0x3f); /*Reset */ - - for (timeout_val = 10000; timeout_val > 0; timeout_val--) - sscape_read(devc, GA_HMCTL_REG); /* Delay */ - - /* Take board out of reset */ - sscape_write(devc, GA_HMCTL_REG, - (temp = sscape_read(devc, GA_HMCTL_REG)) | 0x80); - } - /* - * Transfer one code block using DMA - */ - if (audio_devs[devc->codec_audiodev]->dmap_out->raw_buf == NULL) - { - printk(KERN_WARNING "soundscape: DMA buffer not available\n"); - return 0; - } - memcpy(audio_devs[devc->codec_audiodev]->dmap_out->raw_buf, block, size); - - spin_lock_irqsave(&devc->lock,flags); - - /******** INTERRUPTS DISABLED NOW ********/ - - do_dma(devc, SSCAPE_DMA_A, - audio_devs[devc->codec_audiodev]->dmap_out->raw_buf_phys, - size, DMA_MODE_WRITE); - - /* - * Wait until transfer completes. - */ - - done = 0; - timeout_val = 30; - while (!done && timeout_val-- > 0) - { - int resid; - - if (HZ / 50) - sleep(HZ / 50); - clear_dma_ff(devc->dma); - if ((resid = get_dma_residue(devc->dma)) == 0) - done = 1; - } - - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - return 0; - - if (flag & CPF_LAST) - { - /* - * Take the board out of reset - */ - outb((0x00), PORT(HOST_CTRL)); - outb((0x00), PORT(MIDI_CTRL)); - - temp = sscape_read(devc, GA_HMCTL_REG); - temp |= 0x40; - sscape_write(devc, GA_HMCTL_REG, temp); /* Kickstart the board */ - - /* - * Wait until the ODB wakes up - */ - spin_lock_irqsave(&devc->lock,flags); - done = 0; - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - unsigned char x; - - sleep(1); - x = inb(PORT(HOST_DATA)); - if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ - { - DDB(printk("Soundscape: Acknowledge = %x\n", x)); - done = 1; - } - } - sscape_write(devc, GA_CDCFG_REG, codec_dma_bits); - - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - { - printk(KERN_ERR "soundscape: The OBP didn't respond after code download\n"); - return 0; - } - spin_lock_irqsave(&devc->lock,flags); - done = 0; - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - sleep(1); - if (inb(PORT(HOST_DATA)) == 0xfe) /* Host startup acknowledge */ - done = 1; - } - spin_unlock_irqrestore(&devc->lock,flags); - if (!done) - { - printk(KERN_ERR "soundscape: OBP Initialization failed.\n"); - return 0; - } - printk(KERN_INFO "SoundScape board initialized OK\n"); - set_control(devc, CTL_MASTER_VOL, 100); - set_control(devc, CTL_SYNTH_VOL, 100); - -#ifdef SSCAPE_DEBUG3 - /* - * Temporary debugging aid. Print contents of the registers after - * downloading the code. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (new value)\n", i, sscape_read(devc, i)); - } -#endif - - } - return 1; -} - -static int download_boot_block(void *dev_info, copr_buffer * buf) -{ - if (buf->len <= 0 || buf->len > sizeof(buf->data)) - return -EINVAL; - - if (!sscape_download_boot(devc, buf->data, buf->len, buf->flags)) - { - printk(KERN_ERR "soundscape: Unable to load microcode block to the OBP.\n"); - return -EIO; - } - return 0; -} - -static int sscape_coproc_ioctl(void *dev_info, unsigned int cmd, void __user *arg, int local) -{ - copr_buffer *buf; - int err; - - switch (cmd) - { - case SNDCTL_COPR_RESET: - sscape_coproc_reset(dev_info); - return 0; - - case SNDCTL_COPR_LOAD: - buf = (copr_buffer *) vmalloc(sizeof(copr_buffer)); - if (buf == NULL) - return -ENOSPC; - if (copy_from_user(buf, arg, sizeof(copr_buffer))) - { - vfree(buf); - return -EFAULT; - } - err = download_boot_block(dev_info, buf); - vfree(buf); - return err; - - default: - return -EINVAL; - } -} - -static coproc_operations sscape_coproc_operations = -{ - "SoundScape M68K", - THIS_MODULE, - sscape_coproc_open, - sscape_coproc_close, - sscape_coproc_ioctl, - sscape_coproc_reset, - &adev_info -}; - -static struct resource *sscape_ports; -static int sscape_is_pnp; - -static void __init attach_sscape(struct address_info *hw_config) -{ -#ifndef SSCAPE_REGS - /* - * Config register values for Spea/V7 Media FX and Ensoniq S-2000. - * These values are card - * dependent. If you have another SoundScape based card, you have to - * find the correct values. Do the following: - * - Compile this driver with SSCAPE_DEBUG1 defined. - * - Shut down and power off your machine. - * - Boot with DOS so that the SSINIT.EXE program is run. - * - Warm boot to {Linux|SYSV|BSD} and write down the lines displayed - * when detecting the SoundScape. - * - Modify the following list to use the values printed during boot. - * Undefine the SSCAPE_DEBUG1 - */ -#define SSCAPE_REGS { \ -/* I0 */ 0x00, \ -/* I1 */ 0xf0, /* Note! Ignored. Set always to 0xf0 */ \ -/* I2 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \ -/* I3 */ 0x20, /* Note! Ignored. Set always to 0x20 */ \ -/* I4 */ 0xf5, /* Ignored */ \ -/* I5 */ 0x10, \ -/* I6 */ 0x00, \ -/* I7 */ 0x2e, /* I7 MEM config A. Likely to vary between models */ \ -/* I8 */ 0x00, /* I8 MEM config B. Likely to vary between models */ \ -/* I9 */ 0x40 /* Ignored */ \ - } -#endif - - unsigned long flags; - static unsigned char regs[10] = SSCAPE_REGS; - - int i, irq_bits = 0xff; - - if (old_hardware) - { - valid_interrupts = valid_interrupts_old; - conf_printf("Ensoniq SoundScape (old)", hw_config); - } - else - conf_printf("Ensoniq SoundScape", hw_config); - - for (i = 0; i < 4; i++) - { - if (hw_config->irq == valid_interrupts[i]) - { - irq_bits = i; - break; - } - } - if (hw_config->irq > 15 || (regs[4] = irq_bits == 0xff)) - { - printk(KERN_ERR "Invalid IRQ%d\n", hw_config->irq); - release_region(devc->base, 2); - release_region(devc->base + 2, 6); - if (sscape_is_pnp) - release_region(devc->codec, 2); - return; - } - - if (!sscape_is_pnp) { - - spin_lock_irqsave(&devc->lock,flags); - /* Host interrupt enable */ - sscape_write(devc, 1, 0xf0); /* All interrupts enabled */ - /* DMA A status/trigger register */ - sscape_write(devc, 2, 0x20); /* DMA channel disabled */ - /* DMA B status/trigger register */ - sscape_write(devc, 3, 0x20); /* DMA channel disabled */ - /* Host interrupt config reg */ - sscape_write(devc, 4, 0xf0 | (irq_bits << 2) | irq_bits); - /* Don't destroy CD-ROM DMA config bits (0xc0) */ - sscape_write(devc, 5, (regs[5] & 0x3f) | (sscape_read(devc, 5) & 0xc0)); - /* CD-ROM config (WSS codec actually) */ - sscape_write(devc, 6, regs[6]); - sscape_write(devc, 7, regs[7]); - sscape_write(devc, 8, regs[8]); - /* Master control reg. Don't modify CR-ROM bits. Disable SB emul */ - sscape_write(devc, 9, (sscape_read(devc, 9) & 0xf0) | 0x08); - spin_unlock_irqrestore(&devc->lock,flags); - } -#ifdef SSCAPE_DEBUG2 - /* - * Temporary debugging aid. Print contents of the registers after - * changing them. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (new value)\n", i, sscape_read(devc, i)); - } -#endif - - if (probe_mpu401(hw_config, sscape_ports)) - hw_config->always_detect = 1; - hw_config->name = "SoundScape"; - - hw_config->irq *= -1; /* Negative value signals IRQ sharing */ - attach_mpu401(hw_config, THIS_MODULE); - hw_config->irq *= -1; /* Restore it */ - - if (hw_config->slots[1] != -1) /* The MPU driver installed itself */ - { - sscape_mididev = hw_config->slots[1]; - midi_devs[hw_config->slots[1]]->coproc = &sscape_coproc_operations; - } - sscape_write(devc, GA_INTENA_REG, 0x80); /* Master IRQ enable */ - devc->ok = 1; - devc->failed = 0; -} - -static int detect_ga(sscape_info * devc) -{ - unsigned char save; - - DDB(printk("Entered Soundscape detect_ga(%x)\n", devc->base)); - - /* - * First check that the address register of "ODIE" is - * there and that it has exactly 4 writable bits. - * First 4 bits - */ - - if ((save = inb(PORT(ODIE_ADDR))) & 0xf0) - { - DDB(printk("soundscape: Detect error A\n")); - return 0; - } - outb((0x00), PORT(ODIE_ADDR)); - if (inb(PORT(ODIE_ADDR)) != 0x00) - { - DDB(printk("soundscape: Detect error B\n")); - return 0; - } - outb((0xff), PORT(ODIE_ADDR)); - if (inb(PORT(ODIE_ADDR)) != 0x0f) - { - DDB(printk("soundscape: Detect error C\n")); - return 0; - } - outb((save), PORT(ODIE_ADDR)); - - /* - * Now verify that some indirect registers return zero on some bits. - * This may break the driver with some future revisions of "ODIE" but... - */ - - if (sscape_read(devc, 0) & 0x0c) - { - DDB(printk("soundscape: Detect error D (%x)\n", sscape_read(devc, 0))); - return 0; - } - if (sscape_read(devc, 1) & 0x0f) - { - DDB(printk("soundscape: Detect error E\n")); - return 0; - } - if (sscape_read(devc, 5) & 0x0f) - { - DDB(printk("soundscape: Detect error F\n")); - return 0; - } - return 1; -} - -static int sscape_read_host_ctrl(sscape_info* devc) -{ - return host_read(devc); -} - -static void sscape_write_host_ctrl2(sscape_info *devc, int a, int b) -{ - host_command2(devc, a, b); -} - -static int sscape_alloc_dma(sscape_info *devc) -{ - char *start_addr, *end_addr; - int dma_pagesize; - int sz, size; - struct page *page; - - if (devc->raw_buf != NULL) return 0; /* Already done */ - dma_pagesize = (devc->dma < 4) ? (64 * 1024) : (128 * 1024); - devc->raw_buf = NULL; - devc->buffsize = 8192*4; - if (devc->buffsize > dma_pagesize) devc->buffsize = dma_pagesize; - start_addr = NULL; - /* - * Now loop until we get a free buffer. Try to get smaller buffer if - * it fails. Don't accept smaller than 8k buffer for performance - * reasons. - */ - while (start_addr == NULL && devc->buffsize > PAGE_SIZE) { - for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1); - devc->buffsize = PAGE_SIZE * (1 << sz); - start_addr = (char *) __get_free_pages(GFP_ATOMIC|GFP_DMA, sz); - if (start_addr == NULL) devc->buffsize /= 2; - } - - if (start_addr == NULL) { - printk(KERN_ERR "sscape pnp init error: Couldn't allocate DMA buffer\n"); - return 0; - } else { - /* make some checks */ - end_addr = start_addr + devc->buffsize - 1; - /* now check if it fits into the same dma-pagesize */ - - if (((long) start_addr & ~(dma_pagesize - 1)) != ((long) end_addr & ~(dma_pagesize - 1)) - || end_addr >= (char *) (MAX_DMA_ADDRESS)) { - printk(KERN_ERR "sscape pnp: Got invalid address 0x%lx for %db DMA-buffer\n", (long) start_addr, devc->buffsize); - return 0; - } - } - devc->raw_buf = start_addr; - devc->raw_buf_phys = virt_to_bus(start_addr); - - for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) - SetPageReserved(page); - return 1; -} - -static void sscape_free_dma(sscape_info *devc) -{ - int sz, size; - unsigned long start_addr, end_addr; - struct page *page; - - if (devc->raw_buf == NULL) return; - for (sz = 0, size = PAGE_SIZE; size < devc->buffsize; sz++, size <<= 1); - start_addr = (unsigned long) devc->raw_buf; - end_addr = start_addr + devc->buffsize; - - for (page = virt_to_page(start_addr); page <= virt_to_page(end_addr); page++) - ClearPageReserved(page); - - free_pages((unsigned long) devc->raw_buf, sz); - devc->raw_buf = NULL; -} - -/* Intel version !!!!!!!!! */ - -static int sscape_start_dma(int chan, unsigned long physaddr, int count, int dma_mode) -{ - unsigned long flags; - - flags = claim_dma_lock(); - disable_dma(chan); - clear_dma_ff(chan); - set_dma_mode(chan, dma_mode); - set_dma_addr(chan, physaddr); - set_dma_count(chan, count); - enable_dma(chan); - release_dma_lock(flags); - return 0; -} - -static void sscape_pnp_start_dma(sscape_info* devc, int arg ) -{ - int reg; - if (arg == 0) reg = 2; - else reg = 3; - - sscape_write(devc, reg, sscape_read( devc, reg) | 0x01); - sscape_write(devc, reg, sscape_read( devc, reg) & 0xFE); -} - -static int sscape_pnp_wait_dma (sscape_info* devc, int arg ) -{ - int reg; - unsigned long i; - unsigned char d; - - if (arg == 0) reg = 2; - else reg = 3; - - sleep ( 1 ); - i = 0; - do { - d = sscape_read(devc, reg) & 1; - if ( d == 1) break; - i++; - } while (i < 500000); - d = sscape_read(devc, reg) & 1; - return d; -} - -static int sscape_pnp_alloc_dma(sscape_info* devc) -{ - /* printk(KERN_INFO "sscape: requesting dma\n"); */ - if (request_dma(devc -> dma, "sscape")) return 0; - /* printk(KERN_INFO "sscape: dma channel allocated\n"); */ - if (!sscape_alloc_dma(devc)) { - free_dma(devc -> dma); - return 0; - }; - return 1; -} - -static void sscape_pnp_free_dma(sscape_info* devc) -{ - sscape_free_dma( devc); - free_dma(devc -> dma ); - /* printk(KERN_INFO "sscape: dma released\n"); */ -} - -static int sscape_pnp_upload_file(sscape_info* devc, char* fn) -{ - int done = 0; - int timeout_val; - char* data,*dt; - int len,l; - unsigned long flags; - - sscape_write( devc, 9, sscape_read(devc, 9 ) & 0x3F ); - sscape_write( devc, 2, (devc -> dma << 4) | 0x80 ); - sscape_write( devc, 3, 0x20 ); - sscape_write( devc, 9, sscape_read( devc, 9 ) | 0x80 ); - - len = mod_firmware_load(fn, &data); - if (len == 0) { - printk(KERN_ERR "sscape: file not found: %s\n", fn); - return 0; - } - dt = data; - spin_lock_irqsave(&devc->lock,flags); - while ( len > 0 ) { - if (len > devc -> buffsize) l = devc->buffsize; - else l = len; - len -= l; - memcpy(devc->raw_buf, dt, l); dt += l; - sscape_start_dma(devc->dma, devc->raw_buf_phys, l, 0x48); - sscape_pnp_start_dma ( devc, 0 ); - if (sscape_pnp_wait_dma ( devc, 0 ) == 0) { - spin_unlock_irqrestore(&devc->lock,flags); - return 0; - } - } - - spin_unlock_irqrestore(&devc->lock,flags); - vfree(data); - - outb(0, devc -> base + 2); - outb(0, devc -> base); - - sscape_write ( devc, 9, sscape_read( devc, 9 ) | 0x40); - - timeout_val = 5 * HZ; - while (!done && timeout_val-- > 0) - { - unsigned char x; - sleep(1); - x = inb( devc -> base + 3); - if (x == 0xff || x == 0xfe) /* OBP startup acknowledge */ - { - //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x); - done = 1; - } - } - timeout_val = 5 * HZ; - done = 0; - while (!done && timeout_val-- > 0) - { - unsigned char x; - sleep(1); - x = inb( devc -> base + 3); - if (x == 0xfe) /* OBP startup acknowledge */ - { - //printk(KERN_ERR "Soundscape: Acknowledge = %x\n", x); - done = 1; - } - } - - if ( !done ) printk(KERN_ERR "soundscape: OBP Initialization failed.\n"); - - sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40); - sscape_write( devc, 3, (devc -> dma << 4) + 0x80); - return 1; -} - -static void __init sscape_pnp_init_hw(sscape_info* devc) -{ - unsigned char midi_irq = 0, sb_irq = 0; - unsigned i; - static char code_file_name[23] = "/sndscape/sndscape.cox"; - - int sscape_joystic_enable = 0x7f; - int sscape_mic_enable = 0; - int sscape_ext_midi = 0; - - if ( !sscape_pnp_alloc_dma(devc) ) { - printk(KERN_ERR "sscape: faild to allocate dma\n"); - return; - } - - for (i = 0; i < 4; i++) { - if ( devc -> irq == valid_interrupts[i] ) - midi_irq = i; - if ( devc -> codec_irq == valid_interrupts[i] ) - sb_irq = i; - } - - sscape_write( devc, 5, 0x50); - sscape_write( devc, 7, 0x2e); - sscape_write( devc, 8, 0x00); - - sscape_write( devc, 2, devc->ic_type == IC_ODIE ? 0x70 : 0x40); - sscape_write( devc, 3, ( devc -> dma << 4) | 0x80); - - sscape_write (devc, 4, 0xF0 | (midi_irq<<2) | midi_irq); - - i = 0x10; //sscape_read(devc, 9) & (devc->ic_type == IC_ODIE ? 0xf0 : 0xc0); - if (sscape_joystic_enable) i |= 8; - - sscape_write (devc, 9, i); - sscape_write (devc, 6, 0x80); - sscape_write (devc, 1, 0x80); - - if (devc -> codec_type == 2) { - sscape_pnp_write_codec( devc, 0x0C, 0x50); - sscape_pnp_write_codec( devc, 0x10, sscape_pnp_read_codec( devc, 0x10) & 0x3F); - sscape_pnp_write_codec( devc, 0x11, sscape_pnp_read_codec( devc, 0x11) | 0xC0); - sscape_pnp_write_codec( devc, 29, 0x20); - } - - if (sscape_pnp_upload_file(devc, "/sndscape/scope.cod") == 0 ) { - printk(KERN_ERR "sscape: faild to upload file /sndscape/scope.cod\n"); - sscape_pnp_free_dma(devc); - return; - } - - i = sscape_read_host_ctrl( devc ); - - if ( (i & 0x0F) > 7 ) { - printk(KERN_ERR "sscape: scope.cod faild\n"); - sscape_pnp_free_dma(devc); - return; - } - if ( i & 0x10 ) sscape_write( devc, 7, 0x2F); - code_file_name[21] = (char) ( i & 0x0F) + 0x30; - if (sscape_pnp_upload_file( devc, code_file_name) == 0) { - printk(KERN_ERR "sscape: faild to upload file %s\n", code_file_name); - sscape_pnp_free_dma(devc); - return; - } - - if (devc->ic_type != IC_ODIE) { - sscape_pnp_write_codec( devc, 10, (sscape_pnp_read_codec(devc, 10) & 0x7f) | - ( sscape_mic_enable == 0 ? 0x00 : 0x80) ); - } - sscape_write_host_ctrl2( devc, 0x84, 0x64 ); /* MIDI volume */ - sscape_write_host_ctrl2( devc, 0x86, 0x64 ); /* MIDI volume?? */ - sscape_write_host_ctrl2( devc, 0x8A, sscape_ext_midi); - - sscape_pnp_write_codec ( devc, 6, 0x3f ); //WAV_VOL - sscape_pnp_write_codec ( devc, 7, 0x3f ); //WAV_VOL - sscape_pnp_write_codec ( devc, 2, 0x1F ); //WD_CDXVOLL - sscape_pnp_write_codec ( devc, 3, 0x1F ); //WD_CDXVOLR - - if (devc -> codec_type == 1) { - sscape_pnp_write_codec ( devc, 4, 0x1F ); - sscape_pnp_write_codec ( devc, 5, 0x1F ); - sscape_write_host_ctrl2( devc, 0x88, sscape_mic_enable); - } else { - int t; - sscape_pnp_write_codec ( devc, 0x10, 0x1F << 1); - sscape_pnp_write_codec ( devc, 0x11, 0xC0 | (0x1F << 1)); - - t = sscape_pnp_read_codec( devc, 0x00) & 0xDF; - if ( (sscape_mic_enable == 0)) t |= 0; - else t |= 0x20; - sscape_pnp_write_codec ( devc, 0x00, t); - t = sscape_pnp_read_codec( devc, 0x01) & 0xDF; - if ( (sscape_mic_enable == 0) ) t |= 0; - else t |= 0x20; - sscape_pnp_write_codec ( devc, 0x01, t); - sscape_pnp_write_codec ( devc, 0x40 | 29 , 0x20); - outb(0, devc -> codec); - } - if (devc -> ic_type == IC_OPUS ) { - int i = sscape_read( devc, 9 ); - sscape_write( devc, 9, i | 3 ); - sscape_write( devc, 3, 0x40); - - if (request_region(0x228, 1, "sscape setup junk")) { - outb(0, 0x228); - release_region(0x228,1); - } - sscape_write( devc, 3, (devc -> dma << 4) | 0x80); - sscape_write( devc, 9, i ); - } - - host_close ( devc ); - sscape_pnp_free_dma(devc); -} - -static int __init detect_sscape_pnp(sscape_info* devc) -{ - long i, irq_bits = 0xff; - unsigned int d; - - DDB(printk("Entered detect_sscape_pnp(%x)\n", devc->base)); - - if (!request_region(devc->codec, 2, "sscape codec")) { - printk(KERN_ERR "detect_sscape_pnp: port %x is not free\n", devc->codec); - return 0; - } - - if ((inb(devc->base + 2) & 0x78) != 0) - goto fail; - - d = inb ( devc -> base + 4) & 0xF0; - if (d & 0x80) - goto fail; - - if (d == 0) { - devc->codec_type = 1; - devc->ic_type = IC_ODIE; - } else if ( (d & 0x60) != 0) { - devc->codec_type = 2; - devc->ic_type = IC_OPUS; - } else if ( (d & 0x40) != 0) { /* WTF? */ - devc->codec_type = 2; - devc->ic_type = IC_ODIE; - } else - goto fail; - - sscape_is_pnp = 1; - - outb(0xFA, devc -> base+4); - if ((inb( devc -> base+4) & 0x9F) != 0x0A) - goto fail; - outb(0xFE, devc -> base+4); - if ( (inb(devc -> base+4) & 0x9F) != 0x0E) - goto fail; - if ( (inb(devc -> base+5) & 0x9F) != 0x0E) - goto fail; - - if (devc->codec_type == 2) { - if (devc->codec != devc->base + 8) { - printk("soundscape warning: incorrect codec port specified\n"); - goto fail; - } - d = 0x10 | (sscape_read(devc, 9) & 0xCF); - sscape_write(devc, 9, d); - sscape_write(devc, 6, 0x80); - } else { - //todo: check codec is not base + 8 - } - - d = (sscape_read(devc, 9) & 0x3F) | 0xC0; - sscape_write(devc, 9, d); - - for (i = 0; i < 550000; i++) - if ( !(inb(devc -> codec) & 0x80) ) break; - - d = inb(devc -> codec); - if (d & 0x80) - goto fail; - if ( inb(devc -> codec + 2) == 0xFF) - goto fail; - - sscape_write(devc, 9, sscape_read(devc, 9) & 0x3F ); - - d = inb(devc -> codec) & 0x80; - if ( d == 0) { - printk(KERN_INFO "soundscape: hardware detected\n"); - valid_interrupts = valid_interrupts_new; - } else { - printk(KERN_INFO "soundscape: board looks like media fx\n"); - valid_interrupts = valid_interrupts_old; - old_hardware = 1; - } - - sscape_write( devc, 9, 0xC0 | (sscape_read(devc, 9) & 0x3F) ); - - for (i = 0; i < 550000; i++) - if ( !(inb(devc -> codec) & 0x80)) - break; - - sscape_pnp_init_hw(devc); - - for (i = 0; i < 4; i++) - { - if (devc->codec_irq == valid_interrupts[i]) { - irq_bits = i; - break; - } - } - sscape_write(devc, GA_INTENA_REG, 0x00); - sscape_write(devc, GA_DMACFG_REG, 0x50); - sscape_write(devc, GA_DMAA_REG, 0x70); - sscape_write(devc, GA_DMAB_REG, 0x20); - sscape_write(devc, GA_INTCFG_REG, 0xf0); - sscape_write(devc, GA_CDCFG_REG, 0x89 | (devc->dma << 4) | (irq_bits << 1)); - - sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 0) | 0x20); - sscape_pnp_write_codec( devc, 0, sscape_pnp_read_codec( devc, 1) | 0x20); - - return 1; -fail: - release_region(devc->codec, 2); - return 0; -} - -static int __init probe_sscape(struct address_info *hw_config) -{ - devc->base = hw_config->io_base; - devc->irq = hw_config->irq; - devc->dma = hw_config->dma; - devc->osp = hw_config->osp; - -#ifdef SSCAPE_DEBUG1 - /* - * Temporary debugging aid. Print contents of the registers before - * changing them. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x (old value)\n", i, sscape_read(devc, i)); - } -#endif - devc->failed = 1; - - sscape_ports = request_region(devc->base, 2, "mpu401"); - if (!sscape_ports) - return 0; - - if (!request_region(devc->base + 2, 6, "SoundScape")) { - release_region(devc->base, 2); - return 0; - } - - if (!detect_ga(devc)) { - if (detect_sscape_pnp(devc)) - return 1; - release_region(devc->base, 2); - release_region(devc->base + 2, 6); - return 0; - } - - if (old_hardware) /* Check that it's really an old Spea/Reveal card. */ - { - unsigned char tmp; - int cc; - - if (!((tmp = sscape_read(devc, GA_HMCTL_REG)) & 0xc0)) - { - sscape_write(devc, GA_HMCTL_REG, tmp | 0x80); - for (cc = 0; cc < 200000; ++cc) - inb(devc->base + ODIE_ADDR); - } - } - return 1; -} - -static int __init init_ss_ms_sound(struct address_info *hw_config) -{ - int i, irq_bits = 0xff; - int ad_flags = 0; - struct resource *ports; - - if (devc->failed) - { - printk(KERN_ERR "soundscape: Card not detected\n"); - return 0; - } - if (devc->ok == 0) - { - printk(KERN_ERR "soundscape: Invalid initialization order.\n"); - return 0; - } - for (i = 0; i < 4; i++) - { - if (hw_config->irq == valid_interrupts[i]) - { - irq_bits = i; - break; - } - } - if (irq_bits == 0xff) { - printk(KERN_ERR "soundscape: Invalid MSS IRQ%d\n", hw_config->irq); - return 0; - } - - if (old_hardware) - ad_flags = 0x12345677; /* Tell that we may have a CS4248 chip (Spea-V7 Media FX) */ - else if (sscape_is_pnp) - ad_flags = 0x87654321; /* Tell that we have a soundscape pnp with 1845 chip */ - - ports = request_region(hw_config->io_base, 4, "ad1848"); - if (!ports) { - printk(KERN_ERR "soundscape: ports busy\n"); - return 0; - } - - if (!ad1848_detect(ports, &ad_flags, hw_config->osp)) { - release_region(hw_config->io_base, 4); - return 0; - } - - if (!sscape_is_pnp) /*pnp is already setup*/ - { - /* - * Setup the DMA polarity. - */ - sscape_write(devc, GA_DMACFG_REG, 0x50); - - /* - * Take the gate-array off of the DMA channel. - */ - sscape_write(devc, GA_DMAB_REG, 0x20); - - /* - * Init the AD1848 (CD-ROM) config reg. - */ - sscape_write(devc, GA_CDCFG_REG, 0x89 | (hw_config->dma << 4) | (irq_bits << 1)); - } - - if (hw_config->irq == devc->irq) - printk(KERN_WARNING "soundscape: Warning! The WSS mode can't share IRQ with MIDI\n"); - - hw_config->slots[0] = ad1848_init( - sscape_is_pnp ? "SoundScape" : "SoundScape PNP", - ports, - hw_config->irq, - hw_config->dma, - hw_config->dma, - 0, - devc->osp, - THIS_MODULE); - - - if (hw_config->slots[0] != -1) /* The AD1848 driver installed itself */ - { - audio_devs[hw_config->slots[0]]->coproc = &sscape_coproc_operations; - devc->codec_audiodev = hw_config->slots[0]; - devc->my_audiodev = hw_config->slots[0]; - - /* Set proper routings here (what are they) */ - AD1848_REROUTE(SOUND_MIXER_LINE1, SOUND_MIXER_LINE); - } - -#ifdef SSCAPE_DEBUG5 - /* - * Temporary debugging aid. Print contents of the registers - * after the AD1848 device has been initialized. - */ - { - int i; - - for (i = 0; i < 13; i++) - printk("I%d = %02x\n", i, sscape_read(devc, i)); - } -#endif - return 1; -} - -static void __exit unload_sscape(struct address_info *hw_config) -{ - release_region(devc->base + 2, 6); - unload_mpu401(hw_config); - if (sscape_is_pnp) - release_region(devc->codec, 2); -} - -static void __exit unload_ss_ms_sound(struct address_info *hw_config) -{ - ad1848_unload(hw_config->io_base, - hw_config->irq, - devc->dma, - devc->dma, - 0); - sound_unload_audiodev(hw_config->slots[0]); -} - -static struct address_info cfg; -static struct address_info cfg_mpu; - -static int __initdata spea = -1; -static int mss = 0; -static int __initdata dma = -1; -static int __initdata irq = -1; -static int __initdata io = -1; -static int __initdata mpu_irq = -1; -static int __initdata mpu_io = -1; - -module_param(dma, int, 0); -module_param(irq, int, 0); -module_param(io, int, 0); -module_param(spea, int, 0); /* spea=0/1 set the old_hardware */ -module_param(mpu_irq, int, 0); -module_param(mpu_io, int, 0); -module_param(mss, int, 0); - -static int __init init_sscape(void) -{ - printk(KERN_INFO "Soundscape driver Copyright (C) by Hannu Savolainen 1993-1996\n"); - - cfg.irq = irq; - cfg.dma = dma; - cfg.io_base = io; - - cfg_mpu.irq = mpu_irq; - cfg_mpu.io_base = mpu_io; - /* WEH - Try to get right dma channel */ - cfg_mpu.dma = dma; - - devc->codec = cfg.io_base; - devc->codec_irq = cfg.irq; - devc->codec_type = 0; - devc->ic_type = 0; - devc->raw_buf = NULL; - spin_lock_init(&devc->lock); - - if (cfg.dma == -1 || cfg.irq == -1 || cfg.io_base == -1) { - printk(KERN_ERR "DMA, IRQ, and IO port must be specified.\n"); - return -EINVAL; - } - - if (cfg_mpu.irq == -1 && cfg_mpu.io_base != -1) { - printk(KERN_ERR "MPU_IRQ must be specified if MPU_IO is set.\n"); - return -EINVAL; - } - - if(spea != -1) { - old_hardware = spea; - printk(KERN_INFO "Forcing %s hardware support.\n", - spea?"new":"old"); - } - if (probe_sscape(&cfg_mpu) == 0) - return -ENODEV; - - attach_sscape(&cfg_mpu); - - mss = init_ss_ms_sound(&cfg); - - return 0; -} - -static void __exit cleanup_sscape(void) -{ - if (mss) - unload_ss_ms_sound(&cfg); - unload_sscape(&cfg_mpu); -} - -module_init(init_sscape); -module_exit(cleanup_sscape); - -#ifndef MODULE -static int __init setup_sscape(char *str) -{ - /* io, irq, dma, mpu_io, mpu_irq */ - int ints[6]; - - str = get_options(str, ARRAY_SIZE(ints), ints); - - io = ints[1]; - irq = ints[2]; - dma = ints[3]; - mpu_io = ints[4]; - mpu_irq = ints[5]; - - return 1; -} - -__setup("sscape=", setup_sscape); -#endif -MODULE_LICENSE("GPL"); diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 75c602b5b13..351654cf7b0 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -570,6 +570,7 @@ config SND_ICE1712 tristate "ICEnsemble ICE1712 (Envy24)" select SND_MPU401_UART select SND_AC97_CODEC + select BITREVERSE help Say Y here to include support for soundcards based on the ICE1712 (Envy24) chip. diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 78288dbfc17..20cb60afb20 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -603,8 +603,8 @@ AC97_SINGLE("Tone Control - Treble", AC97_MASTER_TONE, 0, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_pc_beep[2] = { -AC97_SINGLE("PC Speaker Playback Switch", AC97_PC_BEEP, 15, 1, 1), -AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) +AC97_SINGLE("Beep Playback Switch", AC97_PC_BEEP, 15, 1, 1), +AC97_SINGLE("Beep Playback Volume", AC97_PC_BEEP, 1, 15, 1) }; static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = @@ -1393,7 +1393,7 @@ static int snd_ac97_mixer_build(struct snd_ac97 * ac97) } } - /* build PC Speaker controls */ + /* build Beep controls */ if (!(ac97->flags & AC97_HAS_NO_PC_BEEP) && ((ac97->flags & AC97_HAS_PC_BEEP) || snd_ac97_try_volume_mix(ac97, AC97_PC_BEEP))) { diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 7337abdbe4e..139cf3b2b9d 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -800,12 +800,12 @@ AC97_SINGLE("Mono Switch", AC97_MASTER_TONE, 7, 1, 1), AC97_SINGLE("Mono ZC Switch", AC97_MASTER_TONE, 6, 1, 0), AC97_SINGLE("Mono Volume", AC97_MASTER_TONE, 0, 31, 1), -AC97_SINGLE("PC Beep to Headphone Switch", AC97_AUX, 15, 1, 1), -AC97_SINGLE("PC Beep to Headphone Volume", AC97_AUX, 12, 7, 1), -AC97_SINGLE("PC Beep to Master Switch", AC97_AUX, 11, 1, 1), -AC97_SINGLE("PC Beep to Master Volume", AC97_AUX, 8, 7, 1), -AC97_SINGLE("PC Beep to Mono Switch", AC97_AUX, 7, 1, 1), -AC97_SINGLE("PC Beep to Mono Volume", AC97_AUX, 4, 7, 1), +AC97_SINGLE("Beep to Headphone Switch", AC97_AUX, 15, 1, 1), +AC97_SINGLE("Beep to Headphone Volume", AC97_AUX, 12, 7, 1), +AC97_SINGLE("Beep to Master Switch", AC97_AUX, 11, 1, 1), +AC97_SINGLE("Beep to Master Volume", AC97_AUX, 8, 7, 1), +AC97_SINGLE("Beep to Mono Switch", AC97_AUX, 7, 1, 1), +AC97_SINGLE("Beep to Mono Volume", AC97_AUX, 4, 7, 1), AC97_SINGLE("Voice to Headphone Switch", AC97_PCM, 15, 1, 1), AC97_SINGLE("Voice to Headphone Volume", AC97_PCM, 12, 7, 1), diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 8451a0169f3..69867ace786 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -830,8 +830,8 @@ static struct snd_kcontrol_new snd_azf3328_mixer_controls[] __devinitdata = { AZF3328_MIXER_SWITCH("Mic Boost (+20dB)", IDX_MIXER_MIC, 6, 0), AZF3328_MIXER_SWITCH("Line Playback Switch", IDX_MIXER_LINEIN, 15, 1), AZF3328_MIXER_VOL_STEREO("Line Playback Volume", IDX_MIXER_LINEIN, 0x1f, 1), - AZF3328_MIXER_SWITCH("PC Speaker Playback Switch", IDX_MIXER_PCBEEP, 15, 1), - AZF3328_MIXER_VOL_SPECIAL("PC Speaker Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1), + AZF3328_MIXER_SWITCH("Beep Playback Switch", IDX_MIXER_PCBEEP, 15, 1), + AZF3328_MIXER_VOL_SPECIAL("Beep Playback Volume", IDX_MIXER_PCBEEP, 0x0f, 1, 1), AZF3328_MIXER_SWITCH("Video Playback Switch", IDX_MIXER_VIDEO, 15, 1), AZF3328_MIXER_VOL_STEREO("Video Playback Volume", IDX_MIXER_VIDEO, 0x1f, 1), AZF3328_MIXER_SWITCH("Aux Playback Switch", IDX_MIXER_AUX, 15, 1), diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index c8c6f437f5b..8f443a9d61e 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -792,8 +792,8 @@ int __devinit snd_ca0106_mixer(struct snd_ca0106 *emu) "Phone Playback Volume", "Video Playback Switch", "Video Playback Volume", - "PC Speaker Playback Switch", - "PC Speaker Playback Volume", + "Beep Playback Switch", + "Beep Playback Volume", "Mono Output Select", "Capture Source", "Capture Switch", diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index c62b7d10ec6..15523e60351 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -304,7 +304,7 @@ static void snd_ca0106_proc_reg_write32(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; - if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + if (reg < 0x40 && val <= 0xffffffff) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -405,7 +405,7 @@ static void snd_ca0106_proc_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) ) + if (reg < 0x80 && val <= 0xffffffff && channel_id <= 3) snd_ca0106_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index ddcd4a9fd7e..a312bae08f5 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2302,7 +2302,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_SB_VOL_MONO("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31), CMIPCI_SB_SW_MONO("Mic Playback Switch", 0), CMIPCI_DOUBLE("Mic Capture Switch", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0, 1, 0, 0), - CMIPCI_SB_VOL_MONO("PC Speaker Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3), + CMIPCI_SB_VOL_MONO("Beep Playback Volume", SB_DSP4_SPEAKER_DEV, 6, 3), CMIPCI_MIXER_VOL_STEREO("Aux Playback Volume", CM_REG_AUX_VOL, 4, 0, 15), CMIPCI_MIXER_SW_STEREO("Aux Playback Switch", CM_REG_MIXER2, CM_VAUXLM_SHIFT, CM_VAUXRM_SHIFT, 0), CMIPCI_MIXER_SW_STEREO("Aux Capture Switch", CM_REG_MIXER2, CM_RAUXLEN_SHIFT, CM_RAUXREN_SHIFT, 0), @@ -2310,7 +2310,7 @@ static struct snd_kcontrol_new snd_cmipci_mixers[] __devinitdata = { CMIPCI_MIXER_VOL_MONO("Mic Capture Volume", CM_REG_MIXER2, CM_VADMIC_SHIFT, 7), CMIPCI_SB_VOL_MONO("Phone Playback Volume", CM_REG_EXTENT_IND, 5, 7), CMIPCI_DOUBLE("Phone Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 4, 4, 1, 0, 0), - CMIPCI_DOUBLE("PC Speaker Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), + CMIPCI_DOUBLE("Beep Playback Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 3, 3, 1, 0, 0), CMIPCI_DOUBLE("Mic Boost Capture Switch", CM_REG_EXTENT_IND, CM_REG_EXTENT_IND, 0, 0, 1, 0, 0), }; diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 75454648d50..cb65bd0dd35 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -240,7 +240,7 @@ static int select_rom(unsigned int pitch) } else if (pitch == 0x02000000) { /* pitch == 2 */ return 3; - } else if (pitch >= 0x0 && pitch <= 0x08000000) { + } else if (pitch <= 0x08000000) { /* 0 <= pitch <= 8 */ return 0; } else { diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 36e08bd2b3c..6b8ae7b5cd5 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1040,8 +1040,7 @@ static void snd_emu10k1x_proc_reg_write(struct snd_info_entry *entry, if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0x49) && (reg >= 0) && (val <= 0xffffffff) - && (channel_id >= 0) && (channel_id <= 2) ) + if (reg < 0x49 && val <= 0xffffffff && channel_id <= 2) snd_emu10k1x_ptr_write(emu, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index b0fb6c917c3..05afe06e353 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -1818,8 +1818,8 @@ int __devinit snd_emu10k1_mixer(struct snd_emu10k1 *emu, "Master Playback Switch", "Master Capture Switch", "Master Playback Volume", "Master Capture Volume", "Wave Master Playback Volume", "Master Playback Volume", - "PC Speaker Playback Switch", "PC Speaker Capture Switch", - "PC Speaker Playback Volume", "PC Speaker Capture Volume", + "Beep Playback Switch", "Beep Capture Switch", + "Beep Playback Volume", "Beep Capture Volume", "Phone Playback Switch", "Phone Capture Switch", "Phone Playback Volume", "Phone Capture Volume", "Mic Playback Switch", "Mic Capture Switch", diff --git a/sound/pci/emu10k1/emuproc.c b/sound/pci/emu10k1/emuproc.c index 216f9748aff..baa7cd508cd 100644 --- a/sound/pci/emu10k1/emuproc.c +++ b/sound/pci/emu10k1/emuproc.c @@ -451,7 +451,7 @@ static void snd_emu_proc_io_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; - if ((reg < 0x40) && (reg >= 0) && (val <= 0xffffffff) ) { + if (reg < 0x40 && val <= 0xffffffff) { spin_lock_irqsave(&emu->emu_lock, flags); outl(val, emu->port + (reg & 0xfffffffc)); spin_unlock_irqrestore(&emu->emu_lock, flags); @@ -527,7 +527,7 @@ static void snd_emu_proc_ptr_reg_write(struct snd_info_entry *entry, while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) continue; - if ((reg < 0xa0) && (reg >= 0) && (val <= 0xffffffff) && (channel_id >= 0) && (channel_id <= 3) ) + if (reg < 0xa0 && val <= 0xffffffff && channel_id <= 3) snd_ptr_write(emu, iobase, reg, channel_id, val); } } diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index c1a5aa15af8..5ef7080e14d 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -256,7 +256,7 @@ int snd_emu1010_fpga_write(struct snd_emu10k1 * emu, u32 reg, u32 value) if (reg > 0x3f) return 1; reg += 0x40; /* 0x40 upwards are registers. */ - if (value < 0 || value > 0x3f) /* 0 to 0x3f are values */ + if (value > 0x3f) /* 0 to 0x3f are values */ return 1; spin_lock_irqsave(&emu->emu_lock, flags); outl(reg, emu->port + A_IOCFG); diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 820318ee62c..fb83e1ffa5c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1387,7 +1387,7 @@ ES1938_DOUBLE_TLV("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0, db_scale_line), ES1938_DOUBLE_TLV("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0, db_scale_capture), -ES1938_SINGLE("PC Speaker Volume", 0, 0x3c, 0, 7, 0), +ES1938_SINGLE("Beep Volume", 0, 0x3c, 0, 7, 0), ES1938_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0), ES1938_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), { diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 60cdb9e0b68..83508b3964f 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -55,7 +55,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * * 1 = MediaForte 256-PCS * 2 = MediaForte 256-PCPR * 3 = MediaForte 64-PCR - * 16 = setup tuner only (this is additional bit), i.e. SF-64-PCR FM card + * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 */ static int tea575x_tuner[SNDRV_CARDS]; @@ -67,7 +67,10 @@ MODULE_PARM_DESC(id, "ID string for the FM801 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "Enable TEA575x tuner."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); + +#define TUNER_ONLY (1<<4) +#define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) /* * Direct registers @@ -160,7 +163,7 @@ struct fm801 { unsigned int multichannel: 1, /* multichannel support */ secondary: 1; /* secondary codec */ unsigned char secondary_addr; /* address of the secondary codec */ - unsigned int tea575x_tuner; /* tuner flags */ + unsigned int tea575x_tuner; /* tuner access method & flags */ unsigned short ply_ctrl; /* playback control */ unsigned short cap_ctrl; /* capture control */ @@ -1287,7 +1290,7 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) { unsigned short cmdw; - if (chip->tea575x_tuner & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __ac97_ok; /* codec cold reset + AC'97 warm reset */ @@ -1296,11 +1299,13 @@ static int snd_fm801_chip_init(struct fm801 *chip, int resume) udelay(100); outw(0, FM801_REG(chip, CODEC_CTRL)); - if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) { - snd_printk(KERN_ERR "Primary AC'97 codec not found\n"); - if (! resume) - return -EIO; - } + if (wait_for_codec(chip, 0, AC97_RESET, msecs_to_jiffies(750)) < 0) + if (!resume) { + snd_printk(KERN_INFO "Primary AC'97 codec not found, " + "assume SF64-PCR (tuner-only)\n"); + chip->tea575x_tuner = 3 | TUNER_ONLY; + goto __ac97_ok; + } if (chip->multichannel) { if (chip->secondary_addr) { @@ -1414,7 +1419,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, return err; } chip->port = pci_resource_start(pci, 0); - if ((tea575x_tuner & 0x0010) == 0) { + if ((tea575x_tuner & TUNER_ONLY) == 0) { if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, "FM801", chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->irq); @@ -1429,6 +1434,14 @@ static int __devinit snd_fm801_create(struct snd_card *card, chip->multichannel = 1; snd_fm801_chip_init(chip, 0); + /* init might set tuner access method */ + tea575x_tuner = chip->tea575x_tuner; + + if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) { + pci_clear_master(pci); + free_irq(chip->irq, chip); + chip->irq = -1; + } if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); @@ -1438,12 +1451,13 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO - if (tea575x_tuner > 0 && (tea575x_tuner & 0x000f) < 4) { + if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && + (tea575x_tuner & TUNER_TYPE_MASK) < 4) { chip->tea.dev_nr = tea575x_tuner >> 16; chip->tea.card = card; chip->tea.freq_fixup = 10700; chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & 0x000f) - 1]; + chip->tea.ops = &snd_fm801_tea_ops[(tea575x_tuner & TUNER_TYPE_MASK) - 1]; snd_tea575x_init(&chip->tea); } #endif @@ -1483,7 +1497,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, chip->port, chip->irq); - if (tea575x_tuner[dev] & 0x0010) + if (chip->tea575x_tuner & TUNER_ONLY) goto __fm801_tuner_only; if ((err = snd_fm801_pcm(chip, 0, NULL)) < 0) { diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 55545e0818b..556cff937be 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -38,9 +38,20 @@ config SND_HDA_INPUT_BEEP Say Y here to build a digital beep interface for HD-audio driver. This interface is used to generate digital beeps. +config SND_HDA_INPUT_BEEP_MODE + int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)" + depends on SND_HDA_INPUT_BEEP=y + default "1" + range 0 2 + help + Set 0 to disable the digital beep interface for HD-audio by default. + Set 1 to always enable the digital beep interface for HD-audio by + default. Set 2 to control the beep device registration to input + layer using a "Beep Switch" in mixer applications. + config SND_HDA_INPUT_JACK bool "Support jack plugging notification via input layer" - depends on INPUT=y || INPUT=SND_HDA_INTEL + depends on INPUT=y || INPUT=SND select SND_JACK help Say Y here to enable the jack plugging notification via diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 3f51a981e60..5fe34a8d8c8 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -113,23 +113,25 @@ static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, return 0; } -int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +static void snd_hda_do_detach(struct hda_beep *beep) +{ + input_unregister_device(beep->dev); + beep->dev = NULL; + cancel_work_sync(&beep->beep_work); + /* turn off beep for sure */ + snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + AC_VERB_SET_BEEP_CONTROL, 0); +} + +static int snd_hda_do_attach(struct hda_beep *beep) { struct input_dev *input_dev; - struct hda_beep *beep; + struct hda_codec *codec = beep->codec; int err; - if (!snd_hda_get_bool_hint(codec, "beep")) - return 0; /* disabled explicitly */ - - beep = kzalloc(sizeof(*beep), GFP_KERNEL); - if (beep == NULL) - return -ENOMEM; - snprintf(beep->phys, sizeof(beep->phys), - "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); input_dev = input_allocate_device(); if (!input_dev) { - kfree(beep); + printk(KERN_INFO "hda_beep: unable to allocate input device\n"); return -ENOMEM; } @@ -151,21 +153,96 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) err = input_register_device(input_dev); if (err < 0) { input_free_device(input_dev); - kfree(beep); + printk(KERN_INFO "hda_beep: unable to register input device\n"); return err; } + beep->dev = input_dev; + return 0; +} + +static void snd_hda_do_register(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, register_work); + + mutex_lock(&beep->mutex); + if (beep->enabled && !beep->dev) + snd_hda_do_attach(beep); + mutex_unlock(&beep->mutex); +} + +static void snd_hda_do_unregister(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, unregister_work.work); + + mutex_lock(&beep->mutex); + if (!beep->enabled && beep->dev) + snd_hda_do_detach(beep); + mutex_unlock(&beep->mutex); +} +int snd_hda_enable_beep_device(struct hda_codec *codec, int enable) +{ + struct hda_beep *beep = codec->beep; + enable = !!enable; + if (beep == NULL) + return 0; + if (beep->enabled != enable) { + beep->enabled = enable; + if (!enable) { + /* turn off beep */ + snd_hda_codec_write_cache(beep->codec, beep->nid, 0, + AC_VERB_SET_BEEP_CONTROL, 0); + } + if (beep->mode == HDA_BEEP_MODE_SWREG) { + if (enable) { + cancel_delayed_work(&beep->unregister_work); + schedule_work(&beep->register_work); + } else { + schedule_delayed_work(&beep->unregister_work, + HZ); + } + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_enable_beep_device); + +int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +{ + struct hda_beep *beep; + + if (!snd_hda_get_bool_hint(codec, "beep")) + return 0; /* disabled explicitly by hints */ + if (codec->beep_mode == HDA_BEEP_MODE_OFF) + return 0; /* disabled by module option */ + + beep = kzalloc(sizeof(*beep), GFP_KERNEL); + if (beep == NULL) + return -ENOMEM; + snprintf(beep->phys, sizeof(beep->phys), + "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); /* enable linear scale */ snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, 0x01); beep->nid = nid; - beep->dev = input_dev; beep->codec = codec; - beep->enabled = 1; + beep->mode = codec->beep_mode; codec->beep = beep; + INIT_WORK(&beep->register_work, &snd_hda_do_register); + INIT_DELAYED_WORK(&beep->unregister_work, &snd_hda_do_unregister); INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); + mutex_init(&beep->mutex); + + if (beep->mode == HDA_BEEP_MODE_ON) { + beep->enabled = 1; + snd_hda_do_register(&beep->register_work); + } + return 0; } EXPORT_SYMBOL_HDA(snd_hda_attach_beep_device); @@ -174,11 +251,12 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) { struct hda_beep *beep = codec->beep; if (beep) { - cancel_work_sync(&beep->beep_work); - - input_unregister_device(beep->dev); - kfree(beep); + cancel_work_sync(&beep->register_work); + cancel_delayed_work(&beep->unregister_work); + if (beep->enabled) + snd_hda_do_detach(beep); codec->beep = NULL; + kfree(beep); } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 0c3de787c71..f1de1bac042 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -24,19 +24,29 @@ #include "hda_codec.h" +#define HDA_BEEP_MODE_OFF 0 +#define HDA_BEEP_MODE_ON 1 +#define HDA_BEEP_MODE_SWREG 2 + /* beep information */ struct hda_beep { struct input_dev *dev; struct hda_codec *codec; + unsigned int mode; char phys[32]; int tone; hda_nid_t nid; unsigned int enabled:1; + unsigned int request_enable:1; unsigned int linear_tone:1; /* linear tone for IDT/STAC codec */ + struct work_struct register_work; /* registration work */ + struct delayed_work unregister_work; /* unregistration work */ struct work_struct beep_work; /* scheduled task for beep event */ + struct mutex mutex; }; #ifdef CONFIG_SND_HDA_INPUT_BEEP +int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index af989f660cc..9cfdb771928 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -30,6 +30,7 @@ #include <sound/tlv.h> #include <sound/initval.h> #include "hda_local.h" +#include "hda_beep.h" #include <sound/hda_hwdep.h> /* @@ -93,6 +94,13 @@ static void hda_keep_power_on(struct hda_codec *codec); static inline void hda_keep_power_on(struct hda_codec *codec) {} #endif +/** + * snd_hda_get_jack_location - Give a location string of the jack + * @cfg: pin default config value + * + * Parse the pin default config value and returns the string of the + * jack location, e.g. "Rear", "Front", etc. + */ const char *snd_hda_get_jack_location(u32 cfg) { static char *bases[7] = { @@ -120,6 +128,13 @@ const char *snd_hda_get_jack_location(u32 cfg) } EXPORT_SYMBOL_HDA(snd_hda_get_jack_location); +/** + * snd_hda_get_jack_connectivity - Give a connectivity string of the jack + * @cfg: pin default config value + * + * Parse the pin default config value and returns the string of the + * jack connectivity, i.e. external or internal connection. + */ const char *snd_hda_get_jack_connectivity(u32 cfg) { static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" }; @@ -128,6 +143,13 @@ const char *snd_hda_get_jack_connectivity(u32 cfg) } EXPORT_SYMBOL_HDA(snd_hda_get_jack_connectivity); +/** + * snd_hda_get_jack_type - Give a type string of the jack + * @cfg: pin default config value + * + * Parse the pin default config value and returns the string of the + * jack type, i.e. the purpose of the jack, such as Line-Out or CD. + */ const char *snd_hda_get_jack_type(u32 cfg) { static char *jack_types[16] = { @@ -515,6 +537,7 @@ static int snd_hda_bus_dev_register(struct snd_device *device) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { snd_hda_hwdep_add_sysfs(codec); + snd_hda_hwdep_add_power_sysfs(codec); } return 0; } @@ -820,6 +843,16 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, return 0; } +/** + * snd_hda_codec_set_pincfg - Override a pin default configuration + * @codec: the HDA codec + * @nid: NID to set the pin config + * @cfg: the pin default config value + * + * Override a pin default configuration value in the cache. + * This value can be read by snd_hda_codec_get_pincfg() in a higher + * priority than the real hardware value. + */ int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, unsigned int cfg) { @@ -827,7 +860,15 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); -/* get the current pin config value of the given pin NID */ +/** + * snd_hda_codec_get_pincfg - Obtain a pin-default configuration + * @codec: the HDA codec + * @nid: NID to get the pin config + * + * Get the current pin config value of the given pin NID. + * If the pincfg value is cached or overridden via sysfs or driver, + * returns the cached value. + */ unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) { struct hda_pincfg *pin; @@ -944,7 +985,7 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 60); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { @@ -1026,6 +1067,15 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr } EXPORT_SYMBOL_HDA(snd_hda_codec_new); +/** + * snd_hda_codec_configure - (Re-)configure the HD-audio codec + * @codec: the HDA codec + * + * Start parsing of the given codec tree and (re-)initialize the whole + * patch instance. + * + * Returns 0 if successful or a negative error code. + */ int snd_hda_codec_configure(struct hda_codec *codec) { int err; @@ -1088,6 +1138,11 @@ void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_setup_stream); +/** + * snd_hda_codec_cleanup_stream - clean up the codec for closing + * @codec: the CODEC to clean up + * @nid: the NID to clean up + */ void snd_hda_codec_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) { if (!nid) @@ -1163,8 +1218,17 @@ get_alloc_amp_hash(struct hda_codec *codec, u32 key) return (struct hda_amp_info *)get_alloc_hash(&codec->amp_cache, key); } -/* - * query AMP capabilities for the given widget and direction +/** + * query_amp_caps - query AMP capabilities + * @codec: the HD-auio codec + * @nid: the NID to query + * @direction: either #HDA_INPUT or #HDA_OUTPUT + * + * Query AMP capabilities for the given widget and direction. + * Returns the obtained capability bits. + * + * When cap bits have been already read, this doesn't read again but + * returns the cached value. */ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { @@ -1187,6 +1251,19 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) } EXPORT_SYMBOL_HDA(query_amp_caps); +/** + * snd_hda_override_amp_caps - Override the AMP capabilities + * @codec: the CODEC to clean up + * @nid: the NID to clean up + * @direction: either #HDA_INPUT or #HDA_OUTPUT + * @caps: the capability bits to set + * + * Override the cached AMP caps bits value by the given one. + * This function is useful if the driver needs to adjust the AMP ranges, + * e.g. limit to 0dB, etc. + * + * Returns zero if successful or a negative error code. + */ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps) { @@ -1222,6 +1299,17 @@ static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); } +/** + * snd_hda_query_pin_caps - Query PIN capabilities + * @codec: the HD-auio codec + * @nid: the NID to query + * + * Query PIN capabilities for the given widget. + * Returns the obtained capability bits. + * + * When cap bits have been already read, this doesn't read again but + * returns the cached value. + */ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) { return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), @@ -1229,6 +1317,40 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) } EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); +/** + * snd_hda_pin_sense - execute pin sense measurement + * @codec: the CODEC to sense + * @nid: the pin NID to sense + * + * Execute necessary pin sense measurement and return its Presence Detect, + * Impedance, ELD Valid etc. status bits. + */ +u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) +{ + u32 pincap = snd_hda_query_pin_caps(codec, nid); + + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + + return snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_SENSE, 0); +} +EXPORT_SYMBOL_HDA(snd_hda_pin_sense); + +/** + * snd_hda_jack_detect - query pin Presence Detect status + * @codec: the CODEC to sense + * @nid: the pin NID to sense + * + * Query and return the pin's Presence Detect status. + */ +int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + u32 sense = snd_hda_pin_sense(codec, nid); + return !!(sense & AC_PINSENSE_PRESENCE); +} +EXPORT_SYMBOL_HDA(snd_hda_jack_detect); + /* * read the current volume to info * if the cache exists, read the cache value. @@ -1269,8 +1391,15 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, info->vol[ch] = val; } -/* - * read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. +/** + * snd_hda_codec_amp_read - Read AMP value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @index: the index value (only for input direction) + * + * Read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. */ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index) @@ -1283,8 +1412,18 @@ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_read); -/* - * update the AMP value, mask = bit mask to set, val = the value +/** + * snd_hda_codec_amp_update - update the AMP value + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @ch: channel (left=0 or right=1) + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the AMP value with a bit mask. + * Returns 0 if the value is unchanged, 1 if changed. */ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) @@ -1303,8 +1442,17 @@ int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, } EXPORT_SYMBOL_HDA(snd_hda_codec_amp_update); -/* - * update the AMP stereo with the same mask and value +/** + * snd_hda_codec_amp_stereo - update the AMP stereo values + * @codec: HD-audio codec + * @nid: NID to read the AMP value + * @direction: #HDA_INPUT or #HDA_OUTPUT + * @idx: the index value (only for input direction) + * @mask: bit mask to set + * @val: the bits value to set + * + * Update the AMP values like snd_hda_codec_amp_update(), but for a + * stereo widget with the same mask and value. */ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, int direction, int idx, int mask, int val) @@ -1318,7 +1466,12 @@ int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_codec_amp_stereo); #ifdef SND_HDA_NEEDS_RESUME -/* resume the all amp commands from the cache */ +/** + * snd_hda_codec_resume_amp - Resume all AMP commands from the cache + * @codec: HD-audio codec + * + * Resume the all amp commands from the cache. + */ void snd_hda_codec_resume_amp(struct hda_codec *codec) { struct hda_amp_info *buffer = codec->amp_cache.buf.list; @@ -1344,7 +1497,12 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) EXPORT_SYMBOL_HDA(snd_hda_codec_resume_amp); #endif /* SND_HDA_NEEDS_RESUME */ -/* volume */ +/** + * snd_hda_mixer_amp_volume_info - Info callback for a standard AMP mixer + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1400,6 +1558,12 @@ update_amp_value(struct hda_codec *codec, hda_nid_t nid, HDA_AMP_VOLMASK, val); } +/** + * snd_hda_mixer_amp_volume_get - Get callback for a standard AMP mixer volume + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1419,6 +1583,12 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get); +/** + * snd_hda_mixer_amp_volume_put - Put callback for a standard AMP mixer volume + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1443,6 +1613,12 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_put); +/** + * snd_hda_mixer_amp_volume_put - TLV callback for a standard AMP mixer volume + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *_tlv) { @@ -1472,8 +1648,16 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_tlv); -/* - * set (static) TLV for virtual master volume; recalculated as max 0dB +/** + * snd_hda_set_vmaster_tlv - Set TLV for a virtual master control + * @codec: HD-audio codec + * @nid: NID of a reference widget + * @dir: #HDA_INPUT or #HDA_OUTPUT + * @tlv: TLV data to be stored, at least 4 elements + * + * Set (static) TLV data for a virtual master volume using the AMP caps + * obtained from the reference NID. + * The volume range is recalculated as if the max volume is 0dB. */ void snd_hda_set_vmaster_tlv(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int *tlv) @@ -1507,6 +1691,13 @@ _snd_hda_find_mixer_ctl(struct hda_codec *codec, return snd_ctl_find_id(codec->bus->card, &id); } +/** + * snd_hda_find_mixer_ctl - Find a mixer control element with the given name + * @codec: HD-audio codec + * @name: ctl id name string + * + * Get the control element with the given id string and IFACE_MIXER. + */ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name) { @@ -1514,30 +1705,57 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); -/* Add a control element and assign to the codec */ -int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) +/** + * snd_hda_ctl-add - Add a control element and assign to the codec + * @codec: HD-audio codec + * @nid: corresponding NID (optional) + * @kctl: the control element to assign + * + * Add the given control element to an array inside the codec instance. + * All control elements belonging to a codec are supposed to be added + * by this function so that a proper clean-up works at the free or + * reconfiguration time. + * + * If non-zero @nid is passed, the NID is assigned to the control element. + * The assignment is shown in the codec proc file. + * + * snd_hda_ctl_add() checks the control subdev id field whether + * #HDA_SUBDEV_NID_FLAG bit is set. If set (and @nid is zero), the lower + * bits value is taken as the NID to assign. + */ +int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, + struct snd_kcontrol *kctl) { int err; - struct snd_kcontrol **knewp; + struct hda_nid_item *item; + if (kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) { + if (nid == 0) + nid = kctl->id.subdevice & 0xffff; + kctl->id.subdevice = 0; + } err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) return err; - knewp = snd_array_new(&codec->mixers); - if (!knewp) + item = snd_array_new(&codec->mixers); + if (!item) return -ENOMEM; - *knewp = kctl; + item->kctl = kctl; + item->nid = nid; return 0; } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); -/* Clear all controls assigned to the given codec */ +/** + * snd_hda_ctls_clear - Clear all controls assigned to the given codec + * @codec: HD-audio codec + */ void snd_hda_ctls_clear(struct hda_codec *codec) { int i; - struct snd_kcontrol **kctls = codec->mixers.list; + struct hda_nid_item *items = codec->mixers.list; for (i = 0; i < codec->mixers.used; i++) - snd_ctl_remove(codec->bus->card, kctls[i]); + snd_ctl_remove(codec->bus->card, items[i].kctl); snd_array_free(&codec->mixers); } @@ -1563,6 +1781,16 @@ static void hda_unlock_devices(struct snd_card *card) spin_unlock(&card->files_lock); } +/** + * snd_hda_codec_reset - Clear all objects assigned to the codec + * @codec: HD-audio codec + * + * This frees the all PCM and control elements assigned to the codec, and + * clears the caches and restores the pin default configurations. + * + * When a device is being used, it returns -EBSY. If successfully freed, + * returns zero. + */ int snd_hda_codec_reset(struct hda_codec *codec) { struct snd_card *card = codec->bus->card; @@ -1626,7 +1854,22 @@ int snd_hda_codec_reset(struct hda_codec *codec) return 0; } -/* create a virtual master control and add slaves */ +/** + * snd_hda_add_vmaster - create a virtual master control and add slaves + * @codec: HD-audio codec + * @name: vmaster control name + * @tlv: TLV data (optional) + * @slaves: slave control names (optional) + * + * Create a virtual master control with the given name. The TLV data + * must be either NULL or a valid data. + * + * @slaves is a NULL-terminated array of strings, each of which is a + * slave control name. All controls with these names are assigned to + * the new virtual master control. + * + * This function returns zero if successful or a negative error code. + */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves) { @@ -1643,7 +1886,7 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, kctl = snd_ctl_make_virtual_master(name, tlv); if (!kctl) return -ENOMEM; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; @@ -1668,7 +1911,12 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, } EXPORT_SYMBOL_HDA(snd_hda_add_vmaster); -/* switch */ +/** + * snd_hda_mixer_amp_switch_info - Info callback for a standard AMP mixer switch + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1682,6 +1930,12 @@ int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_info); +/** + * snd_hda_mixer_amp_switch_get - Get callback for a standard AMP mixer switch + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1702,6 +1956,12 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get); +/** + * snd_hda_mixer_amp_switch_put - Put callback for a standard AMP mixer switch + * + * The control element is supposed to have the private_value field + * set up via HDA_COMPOSE_AMP_VAL*() or related macros. + */ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1733,6 +1993,25 @@ int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); +#ifdef CONFIG_SND_HDA_INPUT_BEEP +/** + * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch + * + * This function calls snd_hda_enable_beep_device(), which behaves differently + * depending on beep_mode option. + */ +int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + + snd_hda_enable_beep_device(codec, *valp); + return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +} +EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); +#endif /* CONFIG_SND_HDA_INPUT_BEEP */ + /* * bound volume controls * @@ -1742,6 +2021,12 @@ EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put); #define AMP_VAL_IDX_SHIFT 19 #define AMP_VAL_IDX_MASK (0x0f<<19) +/** + * snd_hda_mixer_bind_switch_get - Get callback for a bound volume control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_MUTE*() macros. + */ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1759,6 +2044,12 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_get); +/** + * snd_hda_mixer_bind_switch_put - Put callback for a bound volume control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_MUTE*() macros. + */ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1783,8 +2074,11 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_switch_put); -/* - * generic bound volume/swtich controls +/** + * snd_hda_mixer_bind_ctls_info - Info callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. */ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1803,6 +2097,12 @@ int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_info); +/** + * snd_hda_mixer_bind_ctls_get - Get callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. + */ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1820,6 +2120,12 @@ int snd_hda_mixer_bind_ctls_get(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_get); +/** + * snd_hda_mixer_bind_ctls_put - Put callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() or HDA_BIND_SW() macros. + */ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1843,6 +2149,12 @@ int snd_hda_mixer_bind_ctls_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_bind_ctls_put); +/** + * snd_hda_mixer_bind_tlv - TLV callback for a generic bound control + * + * The control element is supposed to have the private_value field + * set up via HDA_BIND_VOL() macro. + */ int snd_hda_mixer_bind_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { @@ -2126,7 +2438,7 @@ int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid) return -ENOMEM; kctl->id.index = idx; kctl->private_value = nid; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; } @@ -2165,14 +2477,19 @@ static struct snd_kcontrol_new spdif_share_sw = { .put = spdif_share_sw_put, }; +/** + * snd_hda_create_spdif_share_sw - create Default PCM switch + * @codec: the HDA codec + * @mout: multi-out instance + */ int snd_hda_create_spdif_share_sw(struct hda_codec *codec, struct hda_multi_out *mout) { if (!mout->dig_out_nid) return 0; /* ATTENTION: here mout is passed as private_data, instead of codec */ - return snd_hda_ctl_add(codec, - snd_ctl_new1(&spdif_share_sw, mout)); + return snd_hda_ctl_add(codec, mout->dig_out_nid, + snd_ctl_new1(&spdif_share_sw, mout)); } EXPORT_SYMBOL_HDA(snd_hda_create_spdif_share_sw); @@ -2276,7 +2593,7 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) if (!kctl) return -ENOMEM; kctl->private_value = nid; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, nid, kctl); if (err < 0) return err; } @@ -2332,7 +2649,12 @@ int snd_hda_codec_write_cache(struct hda_codec *codec, hda_nid_t nid, } EXPORT_SYMBOL_HDA(snd_hda_codec_write_cache); -/* resume the all commands from the cache */ +/** + * snd_hda_codec_resume_cache - Resume the all commands from the cache + * @codec: HD-audio codec + * + * Execute all verbs recorded in the command caches to resume. + */ void snd_hda_codec_resume_cache(struct hda_codec *codec) { struct hda_cache_head *buffer = codec->cmd_cache.buf.list; @@ -2452,9 +2774,11 @@ static void hda_call_codec_suspend(struct hda_codec *codec) codec->afg ? codec->afg : codec->mfg, AC_PWRST_D3); #ifdef CONFIG_SND_HDA_POWER_SAVE + snd_hda_update_power_acct(codec); cancel_delayed_work(&codec->power_work); codec->power_on = 0; codec->power_transition = 0; + codec->power_jiffies = jiffies; #endif } @@ -2756,8 +3080,12 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, } /** - * snd_hda_is_supported_format - check whether the given node supports - * the format val + * snd_hda_is_supported_format - Check the validity of the format + * @codec: HD-audio codec + * @nid: NID to check + * @format: the HD-audio format value to check + * + * Check whether the given node supports the format value. * * Returns 1 if supported, 0 if not. */ @@ -2877,51 +3205,36 @@ static int set_pcm_default_values(struct hda_codec *codec, return 0; } +/* global */ +const char *snd_hda_pcm_type_name[HDA_PCM_NTYPES] = { + "Audio", "SPDIF", "HDMI", "Modem" +}; + /* * get the empty PCM device number to assign */ static int get_empty_pcm_device(struct hda_bus *bus, int type) { - static const char *dev_name[HDA_PCM_NTYPES] = { - "Audio", "SPDIF", "HDMI", "Modem" - }; - /* starting device index for each PCM type */ - static int dev_idx[HDA_PCM_NTYPES] = { - [HDA_PCM_TYPE_AUDIO] = 0, - [HDA_PCM_TYPE_SPDIF] = 1, - [HDA_PCM_TYPE_HDMI] = 3, - [HDA_PCM_TYPE_MODEM] = 6 + /* audio device indices; not linear to keep compatibility */ + static int audio_idx[HDA_PCM_NTYPES][5] = { + [HDA_PCM_TYPE_AUDIO] = { 0, 2, 4, 5, -1 }, + [HDA_PCM_TYPE_SPDIF] = { 1, -1 }, + [HDA_PCM_TYPE_HDMI] = { 3, 7, 8, 9, -1 }, + [HDA_PCM_TYPE_MODEM] = { 6, -1 }, }; - /* normal audio device indices; not linear to keep compatibility */ - static int audio_idx[4] = { 0, 2, 4, 5 }; - int i, dev; - - switch (type) { - case HDA_PCM_TYPE_AUDIO: - for (i = 0; i < ARRAY_SIZE(audio_idx); i++) { - dev = audio_idx[i]; - if (!test_bit(dev, bus->pcm_dev_bits)) - goto ok; - } - snd_printk(KERN_WARNING "Too many audio devices\n"); - return -EAGAIN; - case HDA_PCM_TYPE_SPDIF: - case HDA_PCM_TYPE_HDMI: - case HDA_PCM_TYPE_MODEM: - dev = dev_idx[type]; - if (test_bit(dev, bus->pcm_dev_bits)) { - snd_printk(KERN_WARNING "%s already defined\n", - dev_name[type]); - return -EAGAIN; - } - break; - default: + int i; + + if (type >= HDA_PCM_NTYPES) { snd_printk(KERN_WARNING "Invalid PCM type %d\n", type); return -EINVAL; } - ok: - set_bit(dev, bus->pcm_dev_bits); - return dev; + + for (i = 0; audio_idx[type][i] >= 0 ; i++) + if (!test_and_set_bit(audio_idx[type][i], bus->pcm_dev_bits)) + return audio_idx[type][i]; + + snd_printk(KERN_WARNING "Too many %s devices\n", snd_hda_pcm_type_name[type]); + return -EAGAIN; } /* @@ -3159,14 +3472,14 @@ EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config); */ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) { - int err; + int err; for (; knew->name; knew++) { struct snd_kcontrol *kctl; kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) { if (!codec->addr) return err; @@ -3174,7 +3487,7 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) if (!kctl) return -ENOMEM; kctl->id.device = codec->addr; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } @@ -3207,8 +3520,27 @@ static void hda_keep_power_on(struct hda_codec *codec) { codec->power_count++; codec->power_on = 1; + codec->power_jiffies = jiffies; } +/* update the power on/off account with the current jiffies */ +void snd_hda_update_power_acct(struct hda_codec *codec) +{ + unsigned long delta = jiffies - codec->power_jiffies; + if (codec->power_on) + codec->power_on_acct += delta; + else + codec->power_off_acct += delta; + codec->power_jiffies += delta; +} + +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ void snd_hda_power_up(struct hda_codec *codec) { struct hda_bus *bus = codec->bus; @@ -3217,7 +3549,9 @@ void snd_hda_power_up(struct hda_codec *codec) if (codec->power_on || codec->power_transition) return; + snd_hda_update_power_acct(codec); codec->power_on = 1; + codec->power_jiffies = jiffies; if (bus->ops.pm_notify) bus->ops.pm_notify(bus); hda_call_codec_resume(codec); @@ -3229,9 +3563,13 @@ EXPORT_SYMBOL_HDA(snd_hda_power_up); #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) -#define power_save(codec) \ - ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) - +/** + * snd_hda_power_down - Power-down the codec + * @codec: HD-audio codec + * + * Decrement the power-up counter and schedules the power-off work if + * the counter rearches to zero. + */ void snd_hda_power_down(struct hda_codec *codec) { --codec->power_count; @@ -3245,6 +3583,19 @@ void snd_hda_power_down(struct hda_codec *codec) } EXPORT_SYMBOL_HDA(snd_hda_power_down); +/** + * snd_hda_check_amp_list_power - Check the amp list and update the power + * @codec: HD-audio codec + * @check: the object containing an AMP list and the status + * @nid: NID to check / update + * + * Check whether the given NID is in the amp list. If it's in the list, + * check the current AMP status, and update the the power-status according + * to the mute status. + * + * This function is supposed to be set or called from the check_power_status + * patch ops. + */ int snd_hda_check_amp_list_power(struct hda_codec *codec, struct hda_loopback_check *check, hda_nid_t nid) @@ -3286,6 +3637,10 @@ EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); /* * Channel mode helper */ + +/** + * snd_hda_ch_mode_info - Info callback helper for the channel mode enum + */ int snd_hda_ch_mode_info(struct hda_codec *codec, struct snd_ctl_elem_info *uinfo, const struct hda_channel_mode *chmode, @@ -3302,6 +3657,9 @@ int snd_hda_ch_mode_info(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_ch_mode_info); +/** + * snd_hda_ch_mode_get - Get callback helper for the channel mode enum + */ int snd_hda_ch_mode_get(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, @@ -3320,6 +3678,9 @@ int snd_hda_ch_mode_get(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_ch_mode_get); +/** + * snd_hda_ch_mode_put - Put callback helper for the channel mode enum + */ int snd_hda_ch_mode_put(struct hda_codec *codec, struct snd_ctl_elem_value *ucontrol, const struct hda_channel_mode *chmode, @@ -3344,6 +3705,10 @@ EXPORT_SYMBOL_HDA(snd_hda_ch_mode_put); /* * input MUX helper */ + +/** + * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + */ int snd_hda_input_mux_info(const struct hda_input_mux *imux, struct snd_ctl_elem_info *uinfo) { @@ -3362,6 +3727,9 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, } EXPORT_SYMBOL_HDA(snd_hda_input_mux_info); +/** + * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + */ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux, struct snd_ctl_elem_value *ucontrol, @@ -3421,8 +3789,29 @@ static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) } } -/* - * open the digital out in the exclusive mode +/** + * snd_hda_bus_reboot_notify - call the reboot notifier of each codec + * @bus: HD-audio bus + */ +void snd_hda_bus_reboot_notify(struct hda_bus *bus) +{ + struct hda_codec *codec; + + if (!bus) + return; + list_for_each_entry(codec, &bus->codec_list, list) { +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!codec->power_on) + continue; +#endif + if (codec->patch_ops.reboot_notify) + codec->patch_ops.reboot_notify(codec); + } +} +EXPORT_SYMBOL_HDA(snd_hda_bus_reboot_notify); + +/** + * snd_hda_multi_out_dig_open - open the digital out in the exclusive mode */ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) @@ -3437,6 +3826,9 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_open); +/** + * snd_hda_multi_out_dig_prepare - prepare the digital out stream + */ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, struct hda_multi_out *mout, unsigned int stream_tag, @@ -3450,6 +3842,9 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare); +/** + * snd_hda_multi_out_dig_cleanup - clean-up the digital out stream + */ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) { @@ -3460,8 +3855,8 @@ int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup); -/* - * release the digital out +/** + * snd_hda_multi_out_dig_close - release the digital out stream */ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) @@ -3473,8 +3868,12 @@ int snd_hda_multi_out_dig_close(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_close); -/* - * set up more restrictions for analog out +/** + * snd_hda_multi_out_analog_open - open analog outputs + * + * Open analog outputs and set up the hw-constraints. + * If the digital outputs can be opened as slave, open the digital + * outputs, too. */ int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, @@ -3519,9 +3918,11 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_open); -/* - * set up the i/o for analog out - * when the digital out is available, copy the front out to digital out, too. +/** + * snd_hda_multi_out_analog_prepare - Preapre the analog outputs. + * + * Set up the i/o for analog out. + * When the digital out is available, copy the front out to digital out, too. */ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, @@ -3578,8 +3979,8 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_analog_prepare); -/* - * clean up the setting for analog out +/** + * snd_hda_multi_out_analog_cleanup - clean up the setting for analog out */ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout) @@ -3965,8 +4366,14 @@ EXPORT_SYMBOL_HDA(snd_hda_resume); * generic arrays */ -/* get a new element from the given array - * if it exceeds the pre-allocated array size, re-allocate the array +/** + * snd_array_new - get a new element from the given array + * @array: the array object + * + * Get a new element from the given array. If it exceeds the + * pre-allocated array size, re-allocate the array. + * + * Returns NULL if allocation failed. */ void *snd_array_new(struct snd_array *array) { @@ -3990,7 +4397,10 @@ void *snd_array_new(struct snd_array *array) } EXPORT_SYMBOL_HDA(snd_array_new); -/* free the given array elements */ +/** + * snd_array_free - free the given array elements + * @array: the array object + */ void snd_array_free(struct snd_array *array) { kfree(array->list); @@ -4000,7 +4410,12 @@ void snd_array_free(struct snd_array *array) } EXPORT_SYMBOL_HDA(snd_array_free); -/* +/** + * snd_print_pcm_rates - Print the supported PCM rates to the string buffer + * @pcm: PCM caps bits + * @buf: the string buffer to write + * @buflen: the max buffer length + * * used by hda_proc.c and hda_eld.c */ void snd_print_pcm_rates(int pcm, char *buf, int buflen) @@ -4019,6 +4434,14 @@ void snd_print_pcm_rates(int pcm, char *buf, int buflen) } EXPORT_SYMBOL_HDA(snd_print_pcm_rates); +/** + * snd_print_pcm_bits - Print the supported PCM fmt bits to the string buffer + * @pcm: PCM caps bits + * @buf: the string buffer to write + * @buflen: the max buffer length + * + * used by hda_proc.c and hda_eld.c + */ void snd_print_pcm_bits(int pcm, char *buf, int buflen) { static unsigned int bits[] = { 8, 16, 20, 24, 32 }; diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 99552fb5f75..2d627613aea 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -286,6 +286,10 @@ enum { #define AC_PWRST_D1SUP (1<<1) #define AC_PWRST_D2SUP (1<<2) #define AC_PWRST_D3SUP (1<<3) +#define AC_PWRST_D3COLDSUP (1<<4) +#define AC_PWRST_S3D3COLDSUP (1<<29) +#define AC_PWRST_CLKSTOP (1<<30) +#define AC_PWRST_EPSS (1U<<31) /* Power state values */ #define AC_PWRST_SETTING (0xf<<0) @@ -674,6 +678,7 @@ struct hda_codec_ops { #ifdef CONFIG_SND_HDA_POWER_SAVE int (*check_power_status)(struct hda_codec *codec, hda_nid_t nid); #endif + void (*reboot_notify)(struct hda_codec *codec); }; /* record for amp information cache */ @@ -771,6 +776,7 @@ struct hda_codec { /* beep device */ struct hda_beep *beep; + unsigned int beep_mode; /* widget capabilities cache */ unsigned int num_nodes; @@ -811,6 +817,9 @@ struct hda_codec { unsigned int power_transition :1; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ + unsigned long power_on_acct; + unsigned long power_off_acct; + unsigned long power_jiffies; #endif /* codec-specific additional proc output */ @@ -910,6 +919,7 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, * Misc */ void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen); +void snd_hda_bus_reboot_notify(struct hda_bus *bus); /* * power management @@ -933,6 +943,7 @@ const char *snd_hda_get_jack_location(u32 cfg); void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); #define snd_hda_codec_needs_resume(codec) codec->power_count +void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 9446a5abea1..4228f2fe595 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -309,17 +309,12 @@ out_fail: return -EINVAL; } -static int hdmi_present_sense(struct hda_codec *codec, hda_nid_t nid) -{ - return snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); -} - static int hdmi_eld_valid(struct hda_codec *codec, hda_nid_t nid) { int eldv; int present; - present = hdmi_present_sense(codec, nid); + present = snd_hda_pin_sense(codec, nid); eldv = (present & AC_PINSENSE_ELDV); present = (present & AC_PINSENSE_PRESENCE); @@ -477,6 +472,8 @@ static void hdmi_print_eld_info(struct snd_info_entry *entry, [4 ... 7] = "reserved" }; + snd_iprintf(buffer, "monitor_present\t\t%d\n", e->monitor_present); + snd_iprintf(buffer, "eld_valid\t\t%d\n", e->eld_valid); snd_iprintf(buffer, "monitor_name\t\t%s\n", e->monitor_name); snd_iprintf(buffer, "connection_type\t\t%s\n", eld_connection_type_names[e->conn_type]); @@ -518,7 +515,11 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, * monitor_name manufacture_id product_id * eld_version edid_version */ - if (!strcmp(name, "connection_type")) + if (!strcmp(name, "monitor_present")) + e->monitor_present = val; + else if (!strcmp(name, "eld_valid")) + e->eld_valid = val; + else if (!strcmp(name, "connection_type")) e->conn_type = val; else if (!strcmp(name, "port_id")) e->port_id = val; @@ -560,13 +561,14 @@ static void hdmi_write_eld_info(struct snd_info_entry *entry, } -int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld) +int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, + int index) { char name[32]; struct snd_info_entry *entry; int err; - snprintf(name, sizeof(name), "eld#%d", codec->addr); + snprintf(name, sizeof(name), "eld#%d.%d", codec->addr, index); err = snd_card_proc_new(codec->bus->card, name, &entry); if (err < 0) return err; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b36f6c5a92d..092c6a7c2ff 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -727,7 +727,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, if (is_loopback) add_input_loopback(codec, node->nid, HDA_INPUT, index); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -737,7 +738,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, if (is_loopback) add_input_loopback(codec, node->nid, HDA_OUTPUT, 0); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -751,7 +753,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) { knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT); snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -759,7 +762,8 @@ static int create_mixer(struct hda_codec *codec, struct hda_gnode *node, (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) { knew = (struct snd_kcontrol_new)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT); snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; created = 1; @@ -857,7 +861,7 @@ static int build_input_controls(struct hda_codec *codec) } /* create input MUX if multiple sources are available */ - err = snd_hda_ctl_add(codec, snd_ctl_new1(&cap_sel, codec)); + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cap_sel, codec)); if (err < 0) return err; @@ -875,7 +879,8 @@ static int build_input_controls(struct hda_codec *codec) HDA_CODEC_VOLUME(name, adc_node->nid, spec->input_mux.items[i].index, HDA_INPUT); - err = snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + err = snd_hda_ctl_add(codec, adc_node->nid, + snd_ctl_new1(&knew, codec)); if (err < 0) return err; } diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index cc24e6721d7..d24328661c6 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -154,6 +154,44 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) return 0; } +#ifdef CONFIG_SND_HDA_POWER_SAVE +static ssize_t power_on_acct_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + snd_hda_update_power_acct(codec); + return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_on_acct)); +} + +static ssize_t power_off_acct_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + snd_hda_update_power_acct(codec); + return sprintf(buf, "%u\n", jiffies_to_msecs(codec->power_off_acct)); +} + +static struct device_attribute power_attrs[] = { + __ATTR_RO(power_on_acct), + __ATTR_RO(power_off_acct), +}; + +int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec) +{ + struct snd_hwdep *hwdep = codec->hwdep; + int i; + + for (i = 0; i < ARRAY_SIZE(power_attrs); i++) + snd_add_device_sysfs_file(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card, + hwdep->device, &power_attrs[i]); + return 0; +} +#endif /* CONFIG_SND_HDA_POWER_SAVE */ + #ifdef CONFIG_SND_HDA_RECONFIG /* diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 6517f589d01..d822bfc6cad 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -60,10 +60,14 @@ static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; static int probe_only[SNDRV_CARDS]; static int single_cmd; -static int enable_msi; +static int enable_msi = -1; #ifdef CONFIG_SND_HDA_PATCH_LOADER static char *patch[SNDRV_CARDS]; #endif +#ifdef CONFIG_SND_HDA_INPUT_BEEP +static int beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = + CONFIG_SND_HDA_INPUT_BEEP_MODE}; +#endif module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -91,6 +95,11 @@ MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)"); module_param_array(patch, charp, NULL, 0444); MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface."); #endif +#ifdef CONFIG_SND_HDA_INPUT_BEEP +module_param_array(beep_mode, int, NULL, 0444); +MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " + "(0=off, 1=on, 2=mute switch on/off) (default=1)."); +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; @@ -404,6 +413,7 @@ struct azx { unsigned short codec_mask; int codec_probe_mask; /* copied from probe_mask option */ struct hda_bus *bus; + unsigned int beep_mode; /* CORB/RIRB */ struct azx_rb corb; @@ -677,6 +687,14 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, } } + if (!chip->polling_mode) { + snd_printk(KERN_WARNING SFX "azx_get_response timeout, " + "switching to polling mode: last cmd=0x%08x\n", + chip->last_cmd[addr]); + chip->polling_mode = 1; + goto again; + } + if (chip->msi) { snd_printk(KERN_WARNING SFX "No response from codec, " "disabling MSI: last cmd=0x%08x\n", @@ -692,14 +710,6 @@ static unsigned int azx_rirb_get_response(struct hda_bus *bus, goto again; } - if (!chip->polling_mode) { - snd_printk(KERN_WARNING SFX "azx_get_response timeout, " - "switching to polling mode: last cmd=0x%08x\n", - chip->last_cmd[addr]); - chip->polling_mode = 1; - goto again; - } - if (chip->probing) { /* If this critical timeout happens during the codec probing * phase, this is likely an access to a non-existing codec @@ -1404,6 +1414,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) err = snd_hda_codec_new(chip->bus, c, &codec); if (err < 0) continue; + codec->beep_mode = chip->beep_mode; codecs++; } } @@ -2154,6 +2165,7 @@ static int azx_resume(struct pci_dev *pci) static int azx_halt(struct notifier_block *nb, unsigned long event, void *buf) { struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); azx_stop_chip(chip); return NOTIFY_OK; } @@ -2221,7 +2233,9 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), + SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), {} }; @@ -2304,11 +2318,9 @@ static void __devinit check_probe_mask(struct azx *chip, int dev) } /* - * white-list for enable_msi + * white/black-list for enable_msi */ -static struct snd_pci_quirk msi_white_list[] __devinitdata = { - SND_PCI_QUIRK(0x103c, 0x30f7, "HP Pavilion dv4t-1300", 1), - SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1), +static struct snd_pci_quirk msi_black_list[] __devinitdata = { {} }; @@ -2316,10 +2328,12 @@ static void __devinit check_msi(struct azx *chip) { const struct snd_pci_quirk *q; - chip->msi = enable_msi; - if (chip->msi) + if (enable_msi >= 0) { + chip->msi = !!enable_msi; return; - q = snd_pci_quirk_lookup(chip->pci, msi_white_list); + } + chip->msi = 1; /* enable MSI as default */ + q = snd_pci_quirk_lookup(chip->pci, msi_black_list); if (q) { printk(KERN_INFO "hda_intel: msi for device %04x:%04x set to %d\n", @@ -2578,6 +2592,10 @@ static int __devinit azx_probe(struct pci_dev *pci, goto out_free; card->private_data = chip; +#ifdef CONFIG_SND_HDA_INPUT_BEEP + chip->beep_mode = beep_mode[dev]; +#endif + /* create codec instances */ err = azx_codec_create(chip, model[dev]); if (err < 0) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 5f1dcc59002..5778ae882b8 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -23,6 +23,15 @@ #ifndef __SOUND_HDA_LOCAL_H #define __SOUND_HDA_LOCAL_H +/* We abuse kcontrol_new.subdev field to pass the NID corresponding to + * the given new control. If id.subdev has a bit flag HDA_SUBDEV_NID_FLAG, + * snd_hda_ctl_add() takes the lower-bit subdev value as a valid NID. + * + * Note that the subdevice field is cleared again before the real registration + * in snd_hda_ctl_add(), so that this value won't appear in the outside. + */ +#define HDA_SUBDEV_NID_FLAG (1U << 31) + /* * for mixer controls */ @@ -33,6 +42,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -53,6 +63,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -66,6 +77,28 @@ /* stereo mute switch */ #define HDA_CODEC_MUTE(xname, nid, xindex, direction) \ HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction) +#ifdef CONFIG_SND_HDA_INPUT_BEEP +/* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ +#define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ + .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = snd_hda_mixer_amp_switch_put_beep, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) } +#else +/* no digital beep - just the standard one */ +#define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, ch, xidx, dir) \ + HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, ch, xidx, dir) +#endif /* CONFIG_SND_HDA_INPUT_BEEP */ +/* special beep mono mute switch */ +#define HDA_CODEC_MUTE_BEEP_MONO(xname, nid, channel, xindex, direction) \ + HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, 0, nid, channel, xindex, direction) +/* special beep stereo mute switch */ +#define HDA_CODEC_MUTE_BEEP(xname, nid, xindex, direction) \ + HDA_CODEC_MUTE_BEEP_MONO(xname, nid, 3, xindex, direction) + +extern const char *snd_hda_pcm_type_name[]; int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); @@ -81,6 +114,10 @@ int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +#ifdef CONFIG_SND_HDA_INPUT_BEEP +int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +#endif /* lowlevel accessor with caching; use carefully */ int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index); @@ -424,8 +461,16 @@ u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); +u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl); +struct hda_nid_item { + struct snd_kcontrol *kctl; + hda_nid_t nid; +}; + +int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, + struct snd_kcontrol *kctl); void snd_hda_ctls_clear(struct hda_codec *codec); /* @@ -437,6 +482,15 @@ int snd_hda_create_hwdep(struct hda_codec *codec); static inline int snd_hda_create_hwdep(struct hda_codec *codec) { return 0; } #endif +#if defined(CONFIG_SND_HDA_POWER_SAVE) && defined(CONFIG_SND_HDA_HWDEP) +int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec); +#else +static inline int snd_hda_hwdep_add_power_sysfs(struct hda_codec *codec) +{ + return 0; +} +#endif + #ifdef CONFIG_SND_HDA_RECONFIG int snd_hda_hwdep_add_sysfs(struct hda_codec *codec); #else @@ -490,7 +544,8 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, * AMP control callbacks */ /* retrieve parameters from private_value */ -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) +#define get_amp_nid_(pv) ((pv) & 0xffff) +#define get_amp_nid(kc) get_amp_nid_((kc)->private_value) #define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) @@ -516,9 +571,11 @@ struct cea_sad { * ELD: EDID Like Data */ struct hdmi_eld { + bool monitor_present; + bool eld_valid; int eld_size; int baseline_len; - int eld_ver; /* (eld_ver == 0) indicates invalid ELD */ + int eld_ver; int cea_edid_ver; char monitor_name[ELD_MAX_MNL + 1]; int manufacture_id; @@ -541,11 +598,13 @@ int snd_hdmi_get_eld(struct hdmi_eld *, struct hda_codec *, hda_nid_t); void snd_hdmi_show_eld(struct hdmi_eld *eld); #ifdef CONFIG_PROC_FS -int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld); +int snd_hda_eld_proc_new(struct hda_codec *codec, struct hdmi_eld *eld, + int index); void snd_hda_eld_proc_free(struct hda_codec *codec, struct hdmi_eld *eld); #else static inline int snd_hda_eld_proc_new(struct hda_codec *codec, - struct hdmi_eld *eld) + struct hdmi_eld *eld, + int index) { return 0; } diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 95f24e4729f..09476fc1ab6 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -26,6 +26,21 @@ #include "hda_codec.h" #include "hda_local.h" +static char *bits_names(unsigned int bits, char *names[], int size) +{ + int i, n; + static char buf[128]; + + for (i = 0, n = 0; i < size; i++) { + if (bits & (1U<<i) && names[i]) + n += snprintf(buf + n, sizeof(buf) - n, " %s", + names[i]); + } + buf[n] = '\0'; + + return buf; +} + static const char *get_wid_type_name(unsigned int wid_value) { static char *names[16] = { @@ -46,6 +61,41 @@ static const char *get_wid_type_name(unsigned int wid_value) return "UNKNOWN Widget"; } +static void print_nid_mixers(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i; + struct hda_nid_item *items = codec->mixers.list; + struct snd_kcontrol *kctl; + for (i = 0; i < codec->mixers.used; i++) { + if (items[i].nid == nid) { + kctl = items[i].kctl; + snd_iprintf(buffer, + " Control: name=\"%s\", index=%i, device=%i\n", + kctl->id.name, kctl->id.index, kctl->id.device); + } + } +} + +static void print_nid_pcms(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int pcm, type; + struct hda_pcm *cpcm; + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + cpcm = &codec->pcm_info[pcm]; + for (type = 0; type < 2; type++) { + if (cpcm->stream[type].nid != nid || cpcm->pcm == NULL) + continue; + snd_iprintf(buffer, " Device: name=\"%s\", " + "type=\"%s\", device=%i\n", + cpcm->name, + snd_hda_pcm_type_name[cpcm->pcm_type], + cpcm->pcm->device); + } + } +} + static void print_amp_caps(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid, int dir) { @@ -363,8 +413,24 @@ static const char *get_pwr_state(u32 state) static void print_power_state(struct snd_info_buffer *buffer, struct hda_codec *codec, hda_nid_t nid) { + static char *names[] = { + [ilog2(AC_PWRST_D0SUP)] = "D0", + [ilog2(AC_PWRST_D1SUP)] = "D1", + [ilog2(AC_PWRST_D2SUP)] = "D2", + [ilog2(AC_PWRST_D3SUP)] = "D3", + [ilog2(AC_PWRST_D3COLDSUP)] = "D3cold", + [ilog2(AC_PWRST_S3D3COLDSUP)] = "S3D3cold", + [ilog2(AC_PWRST_CLKSTOP)] = "CLKSTOP", + [ilog2(AC_PWRST_EPSS)] = "EPSS", + }; + + int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE); int pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); + if (sup) + snd_iprintf(buffer, " Power states: %s\n", + bits_names(sup, names, ARRAY_SIZE(names))); + snd_iprintf(buffer, " Power: setting=%s, actual=%s\n", get_pwr_state(pwr & AC_PWRST_SETTING), get_pwr_state((pwr & AC_PWRST_ACTUAL) >> @@ -457,6 +523,7 @@ static void print_gpio(struct snd_info_buffer *buffer, (data & (1<<i)) ? 1 : 0, (unsol & (1<<i)) ? 1 : 0); /* FIXME: add GPO and GPI pin information */ + print_nid_mixers(buffer, codec, nid); } static void print_codec_info(struct snd_info_entry *entry, @@ -536,6 +603,9 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " CP"); snd_iprintf(buffer, "\n"); + print_nid_mixers(buffer, codec, nid); + print_nid_pcms(buffer, codec, nid); + /* volume knob is a special widget that always have connection * list */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2d603f6aba6..455a0494f90 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -156,15 +156,19 @@ static const char *ad_slave_sws[] = { static void ad198x_free_kctls(struct hda_codec *codec); +#ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ static struct snd_kcontrol_new ad_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT), + HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_OUTPUT), { } /* end */ }; #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ +#else +#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#endif static int ad198x_build_controls(struct hda_codec *codec) { @@ -194,6 +198,7 @@ static int ad198x_build_controls(struct hda_codec *codec) } /* create beep controls if needed */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP if (spec->beep_amp) { struct snd_kcontrol_new *knew; for (knew = ad_beep_mixer; knew->name; knew++) { @@ -202,11 +207,14 @@ static int ad198x_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, + get_amp_nid_(spec->beep_amp), + kctl); if (err < 0) return err; } } +#endif /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { @@ -712,10 +720,10 @@ static struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { static void ad1986a_automic(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_PIN_SENSE, 0); + present = snd_hda_jack_detect(codec, 0x1f); /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */ snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL, - (present & AC_PINSENSE_PRESENCE) ? 0 : 2); + present ? 0 : 2); } #define AD1986A_MIC_EVENT 0x36 @@ -754,10 +762,8 @@ static void ad1986a_update_hp(struct hda_codec *codec) static void ad1986a_hp_automute(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = !!(present & 0x80000000); + spec->jack_present = snd_hda_jack_detect(codec, 0x1a); if (spec->inv_jack_detect) spec->jack_present = !spec->jack_present; ad1986a_update_hp(codec); @@ -1547,8 +1553,7 @@ static void ad1981_hp_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x06, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x06); snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } @@ -1568,8 +1573,7 @@ static void ad1981_hp_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x08, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x08); if (present) snd_hda_sequence_write(codec, mic_jack_on); else @@ -2524,7 +2528,7 @@ static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) { if ((res >> 26) != AD1988_HP_EVENT) return; - if (snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) & (1 << 31)) + if (snd_hda_jack_detect(codec, 0x11)) snd_hda_sequence_write(codec, ad1988_laptop_hp_on); else snd_hda_sequence_write(codec, ad1988_laptop_hp_off); @@ -2569,6 +2573,8 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; + if (get_amp_nid_(val)) + knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); knew->private_value = val; return 0; } @@ -3768,8 +3774,7 @@ static void ad1884a_hp_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x11, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x11); snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, @@ -3781,8 +3786,7 @@ static void ad1884a_hp_automic(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x14); snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, present ? 0 : 1); } @@ -3817,13 +3821,9 @@ static void ad1884a_laptop_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0); - present &= AC_PINSENSE_PRESENCE; - if (!present) { - present = snd_hda_codec_read(codec, 0x12, 0, - AC_VERB_GET_PIN_SENSE, 0); - present &= AC_PINSENSE_PRESENCE; - } + present = snd_hda_jack_detect(codec, 0x11); + if (!present) + present = snd_hda_jack_detect(codec, 0x12); snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, @@ -3835,11 +3835,9 @@ static void ad1884a_laptop_automic(struct hda_codec *codec) { unsigned int idx; - if (snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE) + if (snd_hda_jack_detect(codec, 0x14)) idx = 0; - else if (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE) + else if (snd_hda_jack_detect(codec, 0x1c)) idx = 4; else idx = 1; @@ -4008,8 +4006,7 @@ static void ad1984a_thinkpad_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x11); snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } @@ -4117,14 +4114,12 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* switch to external mic if plugged */ static void ad1984a_touchsmart_automic(struct hda_codec *codec) { - if (snd_hda_codec_read(codec, 0x1c, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000) { + if (snd_hda_jack_detect(codec, 0x1c)) snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, 0x4); - } else { + else snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, 0x5); - } } diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index d08353d3bb7..af478019088 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -144,7 +144,7 @@ static int _add_switch(struct hda_codec *codec, hda_nid_t nid, const char *pfx, struct snd_kcontrol_new knew = HDA_CODEC_MUTE_MONO(namestr, nid, chan, 0, type); sprintf(namestr, "%s %s Switch", pfx, dirstr[dir]); - return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, @@ -155,7 +155,7 @@ static int _add_volume(struct hda_codec *codec, hda_nid_t nid, const char *pfx, struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_MONO(namestr, nid, chan, 0, type); sprintf(namestr, "%s %s Volume", pfx, dirstr[dir]); - return snd_hda_ctl_add(codec, snd_ctl_new1(&knew, codec)); + return snd_hda_ctl_add(codec, nid, snd_ctl_new1(&knew, codec)); } #define add_out_switch(codec, nid, pfx) _add_switch(codec, nid, pfx, 3, 0) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 8ba306856d3..2439e84dcb2 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -500,7 +500,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); - return snd_hda_ctl_add(codec, *kctlp); + return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } static int add_volume(struct hda_codec *codec, const char *name, @@ -513,7 +513,7 @@ static int add_volume(struct hda_codec *codec, const char *name, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); - return snd_hda_ctl_add(codec, *kctlp); + return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) @@ -536,14 +536,14 @@ static int add_vmaster(struct hda_codec *codec, hda_nid_t dac) spec->vmaster_sw = snd_ctl_make_virtual_master("Master Playback Switch", NULL); - err = snd_hda_ctl_add(codec, spec->vmaster_sw); + err = snd_hda_ctl_add(codec, dac, spec->vmaster_sw); if (err < 0) return err; snd_hda_set_vmaster_tlv(codec, dac, HDA_OUTPUT, tlv); spec->vmaster_vol = snd_ctl_make_virtual_master("Master Playback Volume", tlv); - err = snd_hda_ctl_add(codec, spec->vmaster_vol); + err = snd_hda_ctl_add(codec, dac, spec->vmaster_vol); if (err < 0) return err; return 0; @@ -756,13 +756,13 @@ static int build_input(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = (long)spec->capture_bind[i]; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } if (spec->num_inputs > 1 && !spec->mic_detect) { - err = snd_hda_ctl_add(codec, + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cs_capture_source, codec)); if (err < 0) return err; @@ -807,7 +807,7 @@ static void cs_automute(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int caps, present, hp_present; + unsigned int caps, hp_present; hda_nid_t nid; int i; @@ -817,12 +817,7 @@ static void cs_automute(struct hda_codec *codec) caps = snd_hda_query_pin_caps(codec, nid); if (!(caps & AC_PINCAP_PRES_DETECT)) continue; - if (caps & AC_PINCAP_TRIG_REQ) - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - hp_present |= (present & AC_PINSENSE_PRESENCE) != 0; + hp_present = snd_hda_jack_detect(codec, nid); if (hp_present) break; } @@ -844,15 +839,11 @@ static void cs_automic(struct hda_codec *codec) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; - unsigned int caps, present; + unsigned int present; nid = cfg->input_pins[spec->automic_idx]; - caps = snd_hda_query_pin_caps(codec, nid); - if (caps & AC_PINCAP_TRIG_REQ) - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - if (present & AC_PINSENSE_PRESENCE) + present = snd_hda_jack_detect(codec, nid); + if (present) change_cur_input(codec, spec->automic_idx, 0); else { unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ? diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 780e1a72114..85c81feb10c 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -197,8 +197,8 @@ static struct snd_kcontrol_new cmi9880_basic_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x23, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x23, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x23, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x23, 0, HDA_OUTPUT), { } /* end */ }; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 905859d4f4d..a09c03c3f62 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -397,9 +397,7 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) for (i = 0; i < spec->jacks.used; i++) { if (jacks->nid == nid) { unsigned int present; - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, nid); present = (present) ? jacks->type : 0 ; @@ -750,8 +748,7 @@ static void cxt5045_hp_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x12, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x12); if (present) snd_hda_sequence_write(codec, mic_jack_on); else @@ -765,8 +762,7 @@ static void cxt5045_hp_automute(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int bits; - spec->hp_present = snd_hda_codec_read(codec, 0x11, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + spec->hp_present = snd_hda_jack_detect(codec, 0x11); bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x10, HDA_OUTPUT, 0, @@ -1175,9 +1171,10 @@ static int patch_cxt5045(struct hda_codec *codec) switch (codec->subsystem_id >> 16) { case 0x103c: - /* HP laptop has a really bad sound over 0dB on NID 0x17. - * Fix max PCM level to 0 dB - * (originall it has 0x2b steps with 0dB offset 0x14) + case 0x1734: + /* HP & Fujitsu-Siemens laptops have really bad sound over 0dB + * on NID 0x17. Fix max PCM level to 0 dB + * (originally it has 0x2b steps with 0dB offset 0x14) */ snd_hda_override_amp_caps(codec, 0x17, HDA_INPUT, (0x14 << AC_AMPCAP_OFFSET_SHIFT) | @@ -1243,8 +1240,7 @@ static void cxt5047_hp_automute(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int bits; - spec->hp_present = snd_hda_codec_read(codec, 0x13, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + spec->hp_present = snd_hda_jack_detect(codec, 0x13); bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; /* See the note in cxt5047_hp_master_sw_put */ @@ -1267,8 +1263,7 @@ static void cxt5047_hp_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); if (present) snd_hda_sequence_write(codec, mic_jack_on); else @@ -1415,16 +1410,7 @@ static struct snd_kcontrol_new cxt5047_test_mixer[] = { .get = conexant_mux_enum_get, .put = conexant_mux_enum_put, }, - HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Volume", 0x1a, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -1621,9 +1607,7 @@ static void cxt5051_portb_automic(struct hda_codec *codec) if (spec->no_auto_mic) return; - present = snd_hda_codec_read(codec, 0x17, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x17); snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_CONNECT_SEL, present ? 0x01 : 0x00); @@ -1638,9 +1622,7 @@ static void cxt5051_portc_automic(struct hda_codec *codec) if (spec->no_auto_mic) return; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x18); if (present) spec->cur_adc_idx = 1; else @@ -1661,9 +1643,7 @@ static void cxt5051_hp_automute(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - spec->hp_present = snd_hda_codec_read(codec, 0x16, 0, - AC_VERB_GET_PIN_SENSE, 0) & - AC_PINSENSE_PRESENCE; + spec->hp_present = snd_hda_jack_detect(codec, 0x16); cxt5051_update_speaker(codec); } @@ -2011,8 +1991,47 @@ static void cxt5066_automic(struct hda_codec *codec) }; unsigned int present; - present = snd_hda_codec_read(codec, 0x1a, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x1a); + if (present) { + snd_printdd("CXT5066: external microphone detected\n"); + snd_hda_sequence_write(codec, ext_mic_present); + } else { + snd_printdd("CXT5066: external microphone absent\n"); + snd_hda_sequence_write(codec, ext_mic_absent); + } +} + +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_vostro_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + unsigned int present; + + struct hda_verb ext_mic_present[] = { + /* enable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, + + /* switch to external mic input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0}, + + /* disable internal digital mic */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + static struct hda_verb ext_mic_absent[] = { + /* enable internal mic, port C */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + /* switch to internal mic input */ + {0x14, AC_VERB_SET_CONNECT_SEL, 2}, + + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {} + }; + + present = snd_hda_jack_detect(codec, 0x1a); if (present) { snd_printdd("CXT5066: external microphone detected\n"); snd_hda_sequence_write(codec, ext_mic_present); @@ -2029,12 +2048,10 @@ static void cxt5066_hp_automute(struct hda_codec *codec) unsigned int portA, portD; /* Port A */ - portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + portA = snd_hda_jack_detect(codec, 0x19); /* Port D */ - portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE) << 1; + portD = snd_hda_jack_detect(codec, 0x1c); spec->hp_present = !!(portA | portD); snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n", @@ -2056,6 +2073,20 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_vostro_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_vostro: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_vostro_automic(codec); + break; + } +} + static const struct hda_input_mux cxt5066_analog_mic_boost = { .num_items = 5, .items = { @@ -2297,6 +2328,67 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_vostro[] = { + /* Port A: headphones */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* Port B: external microphone */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port C: unused */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port D: unused */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port E: unused, but has primary EAPD */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + + /* Port F: unused */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Port G: internal speakers */ + {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ + + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + /* Audio input selectors */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable unsolicited events for Port A and B */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + static struct hda_verb cxt5066_init_verbs_portd_lo[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { } /* end */ @@ -2318,6 +2410,7 @@ enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ CXT5066_OLPC_XO_1_5, /* OLPC XO 1.5 */ + CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_MODELS }; @@ -2325,6 +2418,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_LAPTOP] = "laptop", [CXT5066_DELL_LAPTOP] = "dell-laptop", [CXT5066_OLPC_XO_1_5] = "olpc-xo-1_5", + [CXT5066_DELL_VOSTO] = "dell-vostro" }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -2333,6 +2427,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02f5, "Dell", CXT5066_DELL_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), {} }; @@ -2400,6 +2495,19 @@ static int patch_cxt5066(struct hda_codec *codec) /* input source automatically selected */ spec->input_mux = NULL; break; + case CXT5066_DELL_VOSTO: + codec->patch_ops.unsol_event = cxt5066_vostro_event; + spec->init_verbs[0] = cxt5066_init_verbs_vostro; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + spec->port_d_mode = 0; + + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + + /* input source automatically selected */ + spec->input_mux = NULL; + break; } return 0; @@ -2417,6 +2525,8 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5051 }, { .id = 0x14f15066, .name = "CX20582 (Pebble)", .patch = patch_cxt5066 }, + { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)", + .patch = patch_cxt5066 }, {} /* terminator */ }; @@ -2424,6 +2534,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15045"); MODULE_ALIAS("snd-hda-codec-id:14f15047"); MODULE_ALIAS("snd-hda-codec-id:14f15051"); MODULE_ALIAS("snd-hda-codec-id:14f15066"); +MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Conexant HD-audio codec"); diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 01a18ed475a..928df59be5d 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -33,15 +33,41 @@ #include "hda_codec.h" #include "hda_local.h" -static hda_nid_t cvt_nid; /* audio converter */ -static hda_nid_t pin_nid; /* HDMI output pin */ +/* + * The HDMI/DisplayPort configuration can be highly dynamic. A graphics device + * could support two independent pipes, each of them can be connected to one or + * more ports (DVI, HDMI or DisplayPort). + * + * The HDA correspondence of pipes/ports are converter/pin nodes. + */ +#define INTEL_HDMI_CVTS 2 +#define INTEL_HDMI_PINS 3 -#define INTEL_HDMI_EVENT_TAG 0x08 +static char *intel_hdmi_pcm_names[INTEL_HDMI_CVTS] = { + "INTEL HDMI 0", + "INTEL HDMI 1", +}; struct intel_hdmi_spec { - struct hda_multi_out multiout; - struct hda_pcm pcm_rec; - struct hdmi_eld sink_eld; + int num_cvts; + int num_pins; + hda_nid_t cvt[INTEL_HDMI_CVTS+1]; /* audio sources */ + hda_nid_t pin[INTEL_HDMI_PINS+1]; /* audio sinks */ + + /* + * source connection for each pin + */ + hda_nid_t pin_cvt[INTEL_HDMI_PINS+1]; + + /* + * HDMI sink attached to each pin + */ + struct hdmi_eld sink_eld[INTEL_HDMI_PINS]; + + /* + * export one pcm per pipe + */ + struct hda_pcm pcm_rec[INTEL_HDMI_CVTS]; }; struct hdmi_audio_infoframe { @@ -184,40 +210,186 @@ static struct cea_channel_speaker_allocation channel_allocations[] = { { .ca_index = 0x31, .speakers = { FRW, FLW, RR, RL, FC, LFE, FR, FL } }, }; + +/* + * HDA/HDMI auto parsing + */ + +static int hda_node_index(hda_nid_t *nids, hda_nid_t nid) +{ + int i; + + for (i = 0; nids[i]; i++) + if (nids[i] == nid) + return i; + + snd_printk(KERN_WARNING "HDMI: nid %d not registered\n", nid); + return -EINVAL; +} + +static int intel_hdmi_read_pin_conn(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct intel_hdmi_spec *spec = codec->spec; + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int conn_len, curr; + int index; + + if (!(get_wcaps(codec, pin_nid) & AC_WCAP_CONN_LIST)) { + snd_printk(KERN_WARNING + "HDMI: pin %d wcaps %#x " + "does not support connection list\n", + pin_nid, get_wcaps(codec, pin_nid)); + return -EINVAL; + } + + conn_len = snd_hda_get_connections(codec, pin_nid, conn_list, + HDA_MAX_CONNECTIONS); + if (conn_len > 1) + curr = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + else + curr = 0; + + index = hda_node_index(spec->pin, pin_nid); + if (index < 0) + return -EINVAL; + + spec->pin_cvt[index] = conn_list[curr]; + + return 0; +} + +static void hdmi_get_show_eld(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + if (!snd_hdmi_get_eld(eld, codec, pin_nid)) + snd_hdmi_show_eld(eld); +} + +static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_eld *eld) +{ + int present = snd_hda_pin_sense(codec, pin_nid); + + eld->monitor_present = !!(present & AC_PINSENSE_PRESENCE); + eld->eld_valid = !!(present & AC_PINSENSE_ELDV); + + if (present & AC_PINSENSE_ELDV) + hdmi_get_show_eld(codec, pin_nid, eld); +} + +static int intel_hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) +{ + struct intel_hdmi_spec *spec = codec->spec; + + if (spec->num_pins >= INTEL_HDMI_PINS) { + snd_printk(KERN_WARNING + "HDMI: no space for pin %d \n", pin_nid); + return -EINVAL; + } + + hdmi_present_sense(codec, pin_nid, &spec->sink_eld[spec->num_pins]); + + spec->pin[spec->num_pins] = pin_nid; + spec->num_pins++; + + /* + * It is assumed that converter nodes come first in the node list and + * hence have been registered and usable now. + */ + return intel_hdmi_read_pin_conn(codec, pin_nid); +} + +static int intel_hdmi_add_cvt(struct hda_codec *codec, hda_nid_t nid) +{ + struct intel_hdmi_spec *spec = codec->spec; + + if (spec->num_cvts >= INTEL_HDMI_CVTS) { + snd_printk(KERN_WARNING + "HDMI: no space for converter %d \n", nid); + return -EINVAL; + } + + spec->cvt[spec->num_cvts] = nid; + spec->num_cvts++; + + return 0; +} + +static int intel_hdmi_parse_codec(struct hda_codec *codec) +{ + hda_nid_t nid; + int i, nodes; + + nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid); + if (!nid || nodes < 0) { + snd_printk(KERN_WARNING "HDMI: failed to get afg sub nodes\n"); + return -EINVAL; + } + + for (i = 0; i < nodes; i++, nid++) { + unsigned int caps; + unsigned int type; + + caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP); + type = get_wcaps_type(caps); + + if (!(caps & AC_WCAP_DIGITAL)) + continue; + + switch (type) { + case AC_WID_AUD_OUT: + if (intel_hdmi_add_cvt(codec, nid) < 0) + return -EINVAL; + break; + case AC_WID_PIN: + caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + if (!(caps & AC_PINCAP_HDMI)) + continue; + if (intel_hdmi_add_pin(codec, nid) < 0) + return -EINVAL; + break; + } + } + + return 0; +} + /* * HDMI routines */ #ifdef BE_PARANOID -static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t nid, +static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) { int val; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_DIP_INDEX, 0); + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_INDEX, 0); *packet_index = val >> 5; *byte_index = val & 0x1f; } #endif -static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t nid, +static void hdmi_set_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int packet_index, int byte_index) { int val; val = (packet_index << 5) | (byte_index & 0x1f); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_INDEX, val); } -static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, +static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t pin_nid, unsigned char val) { - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); + snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_DATA, val); } -static void hdmi_enable_output(struct hda_codec *codec) +static void hdmi_enable_output(struct hda_codec *codec, hda_nid_t pin_nid) { /* Unmute */ if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) @@ -231,7 +403,8 @@ static void hdmi_enable_output(struct hda_codec *codec) /* * Enable Audio InfoFrame Transmission */ -static void hdmi_start_infoframe_trans(struct hda_codec *codec) +static void hdmi_start_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) { hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, @@ -241,59 +414,49 @@ static void hdmi_start_infoframe_trans(struct hda_codec *codec) /* * Disable Audio InfoFrame Transmission */ -static void hdmi_stop_infoframe_trans(struct hda_codec *codec) +static void hdmi_stop_infoframe_trans(struct hda_codec *codec, + hda_nid_t pin_nid) { hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT, AC_DIPXMIT_DISABLE); } -static int hdmi_get_channel_count(struct hda_codec *codec) +static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t nid) { - return 1 + snd_hda_codec_read(codec, cvt_nid, 0, + return 1 + snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CVT_CHAN_COUNT, 0); } -static void hdmi_set_channel_count(struct hda_codec *codec, int chs) +static void hdmi_set_channel_count(struct hda_codec *codec, + hda_nid_t nid, int chs) { - snd_hda_codec_write(codec, cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); - - if (chs != hdmi_get_channel_count(codec)) - snd_printd(KERN_INFO "HDMI channel count: expect %d, get %d\n", - chs, hdmi_get_channel_count(codec)); + if (chs != hdmi_get_channel_count(codec, nid)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CVT_CHAN_COUNT, chs - 1); } -static void hdmi_debug_channel_mapping(struct hda_codec *codec) +static void hdmi_debug_channel_mapping(struct hda_codec *codec, hda_nid_t nid) { #ifdef CONFIG_SND_DEBUG_VERBOSE int i; int slot; for (i = 0; i < 8; i++) { - slot = snd_hda_codec_read(codec, cvt_nid, 0, + slot = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_HDMI_CHAN_SLOT, i); printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n", - slot >> 4, slot & 0x7); + slot >> 4, slot & 0xf); } #endif } -static void hdmi_parse_eld(struct hda_codec *codec) -{ - struct intel_hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld = &spec->sink_eld; - - if (!snd_hdmi_get_eld(eld, codec, pin_nid)) - snd_hdmi_show_eld(eld); -} - /* * Audio InfoFrame routines */ -static void hdmi_debug_dip_size(struct hda_codec *codec) +static void hdmi_debug_dip_size(struct hda_codec *codec, hda_nid_t pin_nid) { #ifdef CONFIG_SND_DEBUG_VERBOSE int i; @@ -310,7 +473,7 @@ static void hdmi_debug_dip_size(struct hda_codec *codec) #endif } -static void hdmi_clear_dip_buffers(struct hda_codec *codec) +static void hdmi_clear_dip_buffers(struct hda_codec *codec, hda_nid_t pin_nid) { #ifdef BE_PARANOID int i, j; @@ -339,23 +502,35 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec) #endif } -static void hdmi_fill_audio_infoframe(struct hda_codec *codec, - struct hdmi_audio_infoframe *ai) +static void hdmi_checksum_audio_infoframe(struct hdmi_audio_infoframe *ai) { - u8 *params = (u8 *)ai; + u8 *bytes = (u8 *)ai; u8 sum = 0; int i; - hdmi_debug_dip_size(codec); - hdmi_clear_dip_buffers(codec); /* be paranoid */ + ai->checksum = 0; + + for (i = 0; i < sizeof(*ai); i++) + sum += bytes[i]; - for (i = 0; i < sizeof(ai); i++) - sum += params[i]; ai->checksum = - sum; +} + +static void hdmi_fill_audio_infoframe(struct hda_codec *codec, + hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + int i; + + hdmi_debug_dip_size(codec, pin_nid); + hdmi_clear_dip_buffers(codec, pin_nid); /* be paranoid */ + + hdmi_checksum_audio_infoframe(ai); hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); - for (i = 0; i < sizeof(ai); i++) - hdmi_write_dip_byte(codec, pin_nid, params[i]); + for (i = 0; i < sizeof(*ai); i++) + hdmi_write_dip_byte(codec, pin_nid, bytes[i]); } /* @@ -386,11 +561,11 @@ static void init_channel_allocations(void) * * TODO: it could select the wrong CA from multiple candidates. */ -static int hdmi_setup_channel_allocation(struct hda_codec *codec, +static int hdmi_setup_channel_allocation(struct hda_codec *codec, hda_nid_t nid, struct hdmi_audio_infoframe *ai) { struct intel_hdmi_spec *spec = codec->spec; - struct hdmi_eld *eld = &spec->sink_eld; + struct hdmi_eld *eld; int i; int spk_mask = 0; int channels = 1 + (ai->CC02_CT47 & 0x7); @@ -402,6 +577,11 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, if (channels <= 2) return 0; + i = hda_node_index(spec->pin_cvt, nid); + if (i < 0) + return 0; + eld = &spec->sink_eld[i]; + /* * HDMI sink's ELD info cannot always be retrieved for now, e.g. * in console or for audio devices. Assume the highest speakers @@ -439,8 +619,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, return ai->CA; } -static void hdmi_setup_channel_mapping(struct hda_codec *codec, - struct hdmi_audio_infoframe *ai) +static void hdmi_setup_channel_mapping(struct hda_codec *codec, hda_nid_t nid, + struct hdmi_audio_infoframe *ai) { int i; @@ -453,17 +633,41 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec, */ for (i = 0; i < 8; i++) - snd_hda_codec_write(codec, cvt_nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_HDMI_CHAN_SLOT, (i << 4) | i); - hdmi_debug_channel_mapping(codec); + hdmi_debug_channel_mapping(codec, nid); } +static bool hdmi_infoframe_uptodate(struct hda_codec *codec, hda_nid_t pin_nid, + struct hdmi_audio_infoframe *ai) +{ + u8 *bytes = (u8 *)ai; + u8 val; + int i; + + if (snd_hda_codec_read(codec, pin_nid, 0, AC_VERB_GET_HDMI_DIP_XMIT, 0) + != AC_DIPXMIT_BEST) + return false; + + hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0); + for (i = 0; i < sizeof(*ai); i++) { + val = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_HDMI_DIP_DATA, 0); + if (val != bytes[i]) + return false; + } -static void hdmi_setup_audio_infoframe(struct hda_codec *codec, + return true; +} + +static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid, struct snd_pcm_substream *substream) { + struct intel_hdmi_spec *spec = codec->spec; + hda_nid_t pin_nid; + int i; struct hdmi_audio_infoframe ai = { .type = 0x84, .ver = 0x01, @@ -471,11 +675,22 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, .CC02_CT47 = substream->runtime->channels - 1, }; - hdmi_setup_channel_allocation(codec, &ai); - hdmi_setup_channel_mapping(codec, &ai); + hdmi_setup_channel_allocation(codec, nid, &ai); + hdmi_setup_channel_mapping(codec, nid, &ai); - hdmi_fill_audio_infoframe(codec, &ai); - hdmi_start_infoframe_trans(codec); + for (i = 0; i < spec->num_pins; i++) { + if (spec->pin_cvt[i] != nid) + continue; + if (!spec->sink_eld[i].monitor_present) + continue; + + pin_nid = spec->pin[i]; + if (!hdmi_infoframe_uptodate(codec, pin_nid, &ai)) { + hdmi_stop_infoframe_trans(codec, pin_nid); + hdmi_fill_audio_infoframe(codec, pin_nid, &ai); + hdmi_start_infoframe_trans(codec, pin_nid); + } + } } @@ -485,27 +700,39 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) { + struct intel_hdmi_spec *spec = codec->spec; + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int pind = !!(res & AC_UNSOL_RES_PD); int eldv = !!(res & AC_UNSOL_RES_ELDV); + int index; printk(KERN_INFO - "HDMI hot plug event: Presence_Detect=%d ELD_Valid=%d\n", - pind, eldv); + "HDMI hot plug event: Pin=%d Presence_Detect=%d ELD_Valid=%d\n", + tag, pind, eldv); + + index = hda_node_index(spec->pin, tag); + if (index < 0) + return; + + spec->sink_eld[index].monitor_present = pind; + spec->sink_eld[index].eld_valid = eldv; if (pind && eldv) { - hdmi_parse_eld(codec); + hdmi_get_show_eld(codec, spec->pin[index], &spec->sink_eld[index]); /* TODO: do real things about ELD */ } } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) { + int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; int cp_state = !!(res & AC_UNSOL_RES_CP_STATE); int cp_ready = !!(res & AC_UNSOL_RES_CP_READY); printk(KERN_INFO - "HDMI content protection event: SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + "HDMI CP event: PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n", + tag, subtag, cp_state, cp_ready); @@ -520,10 +747,11 @@ static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) { + struct intel_hdmi_spec *spec = codec->spec; int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int subtag = (res & AC_UNSOL_RES_SUBTAG) >> AC_UNSOL_RES_SUBTAG_SHIFT; - if (tag != INTEL_HDMI_EVENT_TAG) { + if (hda_node_index(spec->pin, tag) < 0) { snd_printd(KERN_INFO "Unexpected HDMI event tag 0x%x\n", tag); return; } @@ -538,24 +766,29 @@ static void intel_hdmi_unsol_event(struct hda_codec *codec, unsigned int res) * Callbacks */ -static int intel_hdmi_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static void hdmi_setup_stream(struct hda_codec *codec, hda_nid_t nid, + u32 stream_tag, int format) { - struct intel_hdmi_spec *spec = codec->spec; - - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} + int tag; + int fmt; -static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct intel_hdmi_spec *spec = codec->spec; + tag = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0) >> 4; + fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); - hdmi_stop_infoframe_trans(codec); + snd_printdd("hdmi_setup_stream: " + "NID=0x%x, %sstream=0x%x, %sformat=0x%x\n", + nid, + tag == stream_tag ? "" : "new-", + stream_tag, + fmt == format ? "" : "new-", + format); - return snd_hda_multi_out_dig_close(codec, &spec->multiout); + if (tag != stream_tag) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CHANNEL_STREAMID, stream_tag << 4); + if (fmt != format) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_STREAM_FORMAT, format); } static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -564,43 +797,53 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int format, struct snd_pcm_substream *substream) { - struct intel_hdmi_spec *spec = codec->spec; - - snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); + hdmi_set_channel_count(codec, hinfo->nid, + substream->runtime->channels); - hdmi_set_channel_count(codec, substream->runtime->channels); + hdmi_setup_audio_infoframe(codec, hinfo->nid, substream); - hdmi_setup_audio_infoframe(codec, substream); + hdmi_setup_stream(codec, hinfo->nid, stream_tag, format); + return 0; +} +static int intel_hdmi_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ return 0; } static struct hda_pcm_stream intel_hdmi_pcm_playback = { .substreams = 1, .channels_min = 2, - .channels_max = 8, .ops = { - .open = intel_hdmi_playback_pcm_open, - .close = intel_hdmi_playback_pcm_close, - .prepare = intel_hdmi_playback_pcm_prepare + .prepare = intel_hdmi_playback_pcm_prepare, + .cleanup = intel_hdmi_playback_pcm_cleanup, }, }; static int intel_hdmi_build_pcms(struct hda_codec *codec) { struct intel_hdmi_spec *spec = codec->spec; - struct hda_pcm *info = &spec->pcm_rec; + struct hda_pcm *info = spec->pcm_rec; + int i; - codec->num_pcms = 1; + codec->num_pcms = spec->num_cvts; codec->pcm_info = info; - /* NID to query formats and rates and setup streams */ - intel_hdmi_pcm_playback.nid = cvt_nid; + for (i = 0; i < codec->num_pcms; i++, info++) { + unsigned int chans; - info->name = "INTEL HDMI"; - info->pcm_type = HDA_PCM_TYPE_HDMI; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback; + chans = get_wcaps(codec, spec->cvt[i]); + chans = get_wcaps_channels(chans); + + info->name = intel_hdmi_pcm_names[i]; + info->pcm_type = HDA_PCM_TYPE_HDMI; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + intel_hdmi_pcm_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->cvt[i]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; + } return 0; } @@ -609,29 +852,39 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) { struct intel_hdmi_spec *spec = codec->spec; int err; + int i; - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); - if (err < 0) - return err; + for (i = 0; i < codec->num_pcms; i++) { + err = snd_hda_create_spdif_out_ctls(codec, spec->cvt[i]); + if (err < 0) + return err; + } return 0; } static int intel_hdmi_init(struct hda_codec *codec) { - hdmi_enable_output(codec); + struct intel_hdmi_spec *spec = codec->spec; + int i; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | INTEL_HDMI_EVENT_TAG); + for (i = 0; spec->pin[i]; i++) { + hdmi_enable_output(codec, spec->pin[i]); + snd_hda_codec_write(codec, spec->pin[i], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | spec->pin[i]); + } return 0; } static void intel_hdmi_free(struct hda_codec *codec) { struct intel_hdmi_spec *spec = codec->spec; + int i; + + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_free(codec, &spec->sink_eld[i]); - snd_hda_eld_proc_free(codec, &spec->sink_eld); kfree(spec); } @@ -643,49 +896,38 @@ static struct hda_codec_ops intel_hdmi_patch_ops = { .unsol_event = intel_hdmi_unsol_event, }; -static int do_patch_intel_hdmi(struct hda_codec *codec) +static int patch_intel_hdmi(struct hda_codec *codec) { struct intel_hdmi_spec *spec; + int i; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; - spec->multiout.num_dacs = 0; /* no analog */ - spec->multiout.max_channels = 8; - spec->multiout.dig_out_nid = cvt_nid; - codec->spec = spec; + if (intel_hdmi_parse_codec(codec) < 0) { + codec->spec = NULL; + kfree(spec); + return -EINVAL; + } codec->patch_ops = intel_hdmi_patch_ops; - snd_hda_eld_proc_new(codec, &spec->sink_eld); + for (i = 0; i < spec->num_pins; i++) + snd_hda_eld_proc_new(codec, &spec->sink_eld[i], i); init_channel_allocations(); return 0; } -static int patch_intel_hdmi(struct hda_codec *codec) -{ - cvt_nid = 0x02; - pin_nid = 0x03; - return do_patch_intel_hdmi(codec); -} - -static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec) -{ - cvt_nid = 0x02; - pin_nid = 0x04; - return do_patch_intel_hdmi(codec); -} - static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x808629fb, .name = "G45 DEVCL", .patch = patch_intel_hdmi }, { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, - { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi_ibexpeak }, + { .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70583719282..d967836f36b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -961,18 +961,12 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, static void alc_automute_pin(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present, pincap; unsigned int nid = spec->autocfg.hp_pins[0]; int i; if (!nid) return; - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, nid); for (i = 0; i < ARRAY_SIZE(spec->autocfg.speaker_pins); i++) { nid = spec->autocfg.speaker_pins[i]; if (!nid) @@ -1012,9 +1006,7 @@ static void alc_mic_automute(struct hda_codec *codec) cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0]; - present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0, - AC_VERB_GET_PIN_SENSE, 0); - present &= AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, spec->ext_mic.pin); if (present) { alive = &spec->ext_mic; dead = &spec->int_mic; @@ -1402,6 +1394,17 @@ static void alc_pick_fixup(struct hda_codec *codec, add_verb(codec->spec, fix->verbs); } +static int alc_read_coef_idx(struct hda_codec *codec, + unsigned int coef_idx) +{ + unsigned int val; + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, + coef_idx); + val = snd_hda_codec_read(codec, 0x20, 0, + AC_VERB_GET_PROC_COEF, 0); + return val; +} + /* * ALC888 */ @@ -1513,7 +1516,7 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { static void alc_automute_amp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int val, mute, pincap; + unsigned int mute; hda_nid_t nid; int i; @@ -1522,13 +1525,7 @@ static void alc_automute_amp(struct hda_codec *codec) nid = spec->autocfg.hp_pins[i]; if (!nid) break; - pincap = snd_hda_query_pin_caps(codec, nid); - if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_SENSE, 0); - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - if (val & AC_PINSENSE_PRESENCE) { + if (snd_hda_jack_detect(codec, nid)) { spec->jack_present = 1; break; } @@ -1786,6 +1783,8 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; } static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) @@ -2410,12 +2409,14 @@ static const char *alc_slave_sws[] = { static void alc_free_kctls(struct hda_codec *codec); +#ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ static struct snd_kcontrol_new alc_beep_mixer[] = { HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE_BEEP("Beep Playback Switch", 0, 0, HDA_INPUT), { } /* end */ }; +#endif static int alc_build_controls(struct hda_codec *codec) { @@ -2452,6 +2453,7 @@ static int alc_build_controls(struct hda_codec *codec) return err; } +#ifdef CONFIG_SND_HDA_INPUT_BEEP /* create beep controls if needed */ if (spec->beep_amp) { struct snd_kcontrol_new *knew; @@ -2461,11 +2463,13 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, kctl); + err = snd_hda_ctl_add(codec, + get_amp_nid_(spec->beep_amp), kctl); if (err < 0) return err; } } +#endif /* if we have no master control, let's create it */ if (!spec->no_analog && @@ -2779,8 +2783,7 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x18); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } @@ -3480,7 +3483,7 @@ static int alc_build_pcms(struct hda_codec *codec) snprintf(spec->stream_name_analog, sizeof(spec->stream_name_analog), "%s Analog", codec->chip_name); info->name = spec->stream_name_analog; - + if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) return -EINVAL; @@ -4322,10 +4325,26 @@ static int add_control(struct alc_spec *spec, int type, const char *name, knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; + if (get_amp_nid_(val)) + knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); knew->private_value = val; return 0; } +static int add_control_with_pfx(struct alc_spec *spec, int type, + const char *pfx, const char *dir, + const char *sfx, unsigned long val) +{ + char name[32]; + snprintf(name, sizeof(name), "%s %s %s", pfx, dir, sfx); + return add_control(spec, type, name, val); +} + +#define add_pb_vol_ctrl(spec, type, pfx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Volume", val) +#define add_pb_sw_ctrl(spec, type, pfx, val) \ + add_control_with_pfx(spec, type, pfx, "Playback", "Switch", val) + #define alc880_is_fixed_pin(nid) ((nid) >= 0x14 && (nid) <= 0x17) #define alc880_fixed_pin_idx(nid) ((nid) - 0x14) #define alc880_is_multi_pin(nid) ((nid) >= 0x18) @@ -4379,7 +4398,6 @@ static int alc880_auto_fill_dac_nids(struct alc_spec *spec, static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - char name[32]; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; @@ -4392,26 +4410,26 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, nid = alc880_idx_to_mixer(alc880_dac_to_idx(spec->multiout.dac_nids[i])); if (i == 2) { /* Center/LFE */ - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Center Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "Center", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "LFE Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "LFE", HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "Center Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "Center", HDA_COMPOSE_AMP_VAL(nid, 1, 2, HDA_INPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "LFE Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "LFE", HDA_COMPOSE_AMP_VAL(nid, 2, 2, HDA_INPUT)); if (err < 0) @@ -4423,14 +4441,12 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec, pfx = "Speaker"; else pfx = chname[i]; - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) @@ -4446,7 +4462,6 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, { hda_nid_t nid; int err; - char name[32]; if (!pin) return 0; @@ -4460,21 +4475,18 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, spec->multiout.extra_out_nid[0] = nid; /* control HP volume/switch on the output mixer amp */ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin)); - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, 3, 2, HDA_INPUT)); if (err < 0) return err; } else if (alc880_is_multi_pin(pin)) { /* set manual connection */ /* we have only a switch on HP-out PIN */ - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -4487,16 +4499,13 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, const char *ctlname, int idx, hda_nid_t mix_nid) { - char name[32]; int err; - sprintf(name, "%s Playback Volume", ctlname); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", ctlname); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -4773,8 +4782,12 @@ static void set_capture_mixer(struct hda_codec *codec) } } +#ifdef CONFIG_SND_HDA_INPUT_BEEP #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) +#else +#define set_beep_amp(spec, nid, idx, dir) /* NOP */ +#endif /* * OK, here we have finally the patch for ALC880 @@ -5087,11 +5100,8 @@ static struct hda_verb alc260_hp_unsol_verbs[] = { static void alc260_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x10, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x10); alc260_hp_master_update(codec, 0x0f, 0x10, 0x11); } @@ -5156,11 +5166,8 @@ static struct hda_verb alc260_hp_3013_unsol_verbs[] = { static void alc260_hp_3013_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x15); alc260_hp_master_update(codec, 0x15, 0x10, 0x11); } @@ -5173,12 +5180,8 @@ static void alc260_hp_3013_unsol_event(struct hda_codec *codec, static void alc260_hp_3012_automute(struct hda_codec *codec) { - unsigned int present, bits; - - present = snd_hda_codec_read(codec, 0x10, 0, - AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; + unsigned int bits = snd_hda_jack_detect(codec, 0x10) ? 0 : PIN_OUT; - bits = present ? 0 : PIN_OUT; snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, bits); snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, @@ -5748,8 +5751,7 @@ static void alc260_replacer_672v_automute(struct hda_codec *codec) unsigned int present; /* speaker --> GPIO Data 0, hp or spdif --> GPIO data 1 */ - present = snd_hda_codec_read(codec, 0x0f, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x0f); if (present) { snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 1); @@ -5989,7 +5991,6 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, { hda_nid_t nid_vol; unsigned long vol_val, sw_val; - char name[32]; int err; if (nid >= 0x0f && nid < 0x11) { @@ -6009,14 +6010,12 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, if (!(*vol_bits & (1 << nid_vol))) { /* first control for the volume widget */ - snprintf(name, sizeof(name), "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, vol_val); if (err < 0) return err; *vol_bits |= (1 << nid_vol); } - snprintf(name, sizeof(name), "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, sw_val); + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, sw_val); if (err < 0) return err; return 1; @@ -7336,8 +7335,8 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), /* FIXME: this looks suspicious... - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x02, HDA_INPUT), */ { } /* end */ }; @@ -8184,12 +8183,8 @@ static void alc883_mitac_setup(struct hda_codec *codec) /* static void alc883_mitac_mic_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + unsigned char bits = snd_hda_jack_detect(codec, 0x18) ? HDA_AMP_MUTE : 0; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } */ @@ -8411,10 +8406,8 @@ static struct hda_channel_mode alc888_3st_hp_modes[3] = { /* toggle front-jack and RCA according to the hp-jack state */ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = snd_hda_jack_detect(codec, 0x1b); - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -8424,10 +8417,8 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) /* toggle RCA according to the front-jack state */ static void alc888_lenovo_ms7195_rca_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = snd_hda_jack_detect(codec, 0x14); - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } @@ -8468,8 +8459,7 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x18); snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } @@ -8520,24 +8510,16 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + int bits = snd_hda_jack_detect(codec, 0x14) ? HDA_AMP_MUTE : 0; - present = snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); } static void alc883_lenovo_101e_all_automute(struct hda_codec *codec) { - unsigned int present; - unsigned char bits; + int bits = snd_hda_jack_detect(codec, 0x1b) ? HDA_AMP_MUTE : 0; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -8688,8 +8670,7 @@ static void alc889A_mb31_automute(struct hda_codec *codec) /* Mute only in 2ch or 4ch mode */ if (snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_CONNECT_SEL, 0) == 0x00) { - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x15); snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, @@ -10032,10 +10013,8 @@ static void alc262_hp_master_update(struct hda_codec *codec) static void alc262_hp_bpc_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int presence; - presence = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE); + + spec->jack_present = snd_hda_jack_detect(codec, 0x1b); alc262_hp_master_update(codec); } @@ -10049,10 +10028,8 @@ static void alc262_hp_bpc_unsol_event(struct hda_codec *codec, unsigned int res) static void alc262_hp_wildwest_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int presence; - presence = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = !!(presence & AC_PINSENSE_PRESENCE); + + spec->jack_present = snd_hda_jack_detect(codec, 0x15); alc262_hp_master_update(codec); } @@ -10286,13 +10263,8 @@ static void alc262_hippo_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, hp_nid, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, hp_nid, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; + spec->jack_present = snd_hda_jack_detect(codec, hp_nid); alc262_hippo_master_update(codec); } @@ -10618,21 +10590,8 @@ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); - /* check laptop HP jack */ - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - /* check docking HP jack */ - present |= snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - if (present & AC_PINSENSE_PRESENCE) - spec->jack_present = 1; - else - spec->jack_present = 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x14) || + snd_hda_jack_detect(codec, 0x1b); spec->sense_updated = 1; } /* unmute internal speaker only if both HPs are unplugged and @@ -10677,12 +10636,7 @@ static void alc262_lenovo_3000_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present_int_hp; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); - present_int_hp = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present_int_hp & 0x80000000) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x1b); spec->sense_updated = 1; } if (spec->jack_present) { @@ -10874,12 +10828,7 @@ static void alc262_ultra_automute(struct hda_codec *codec) mute = 0; /* auto-mute only when HP is used as HP */ if (!spec->cur_mux[0]) { - unsigned int present; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, 0x15, 0, AC_VERB_SET_PIN_SENSE, 0); - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x15); if (spec->jack_present) mute = HDA_AMP_MUTE; } @@ -10956,7 +10905,6 @@ static int alc262_check_volbit(hda_nid_t nid) static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid, const char *pfx, int *vbits) { - char name[32]; unsigned long val; int vbit; @@ -10966,28 +10914,25 @@ static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid, if (*vbits & vbit) /* a volume control for this mixer already there */ return 0; *vbits |= vbit; - snprintf(name, sizeof(name), "%s Playback Volume", pfx); if (vbit == 2) val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT); else val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT); - return add_control(spec, ALC_CTL_WIDGET_VOL, name, val); + return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, val); } static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid, const char *pfx) { - char name[32]; unsigned long val; if (!nid) return 0; - snprintf(name, sizeof(name), "%s Playback Switch", pfx); if (nid == 0x16) val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT); else val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); - return add_control(spec, ALC_CTL_WIDGET_MUTE, name, val); + return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, val); } /* add playback controls from the parsed DAC table */ @@ -11463,8 +11408,10 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x9025, "Sony VAIO Z21MN", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x104d, 0x9035, "Sony VAIO VGN-FW170J", ALC262_AUTO), SND_PCI_QUIRK(0x104d, 0x9047, "Sony VAIO Type G", ALC262_AUTO), +#if 0 /* disable the quirk since model=auto works better in recent versions */ SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", ALC262_SONY_ASSAMD), +#endif SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), @@ -11923,10 +11870,7 @@ static void alc268_acer_automute(struct hda_codec *codec, int force) unsigned int mute; if (force || !spec->sense_updated) { - unsigned int present; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0); - spec->jack_present = (present & 0x80000000) != 0; + spec->jack_present = snd_hda_jack_detect(codec, 0x14); spec->sense_updated = 1; } if (spec->jack_present) @@ -12045,8 +11989,7 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -12327,11 +12270,9 @@ static struct snd_kcontrol_new alc268_test_mixer[] = { static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, const char *ctlname, int idx) { - char name[32]; hda_nid_t dac; int err; - sprintf(name, "%s Playback Volume", ctlname); switch (nid) { case 0x14: case 0x16: @@ -12345,7 +12286,7 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, } if (spec->multiout.dac_nids[0] != dac && spec->multiout.dac_nids[1] != dac) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, ctlname, HDA_COMPOSE_AMP_VAL(dac, 3, idx, HDA_OUTPUT)); if (err < 0) @@ -12353,12 +12294,11 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac; } - sprintf(name, "%s Playback Switch", ctlname); if (nid != 0x16) - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); else /* mono */ - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, ctlname, HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT)); if (err < 0) return err; @@ -12388,8 +12328,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->speaker_pins[0]; if (nid == 0x1d) { - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Speaker Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, "Speaker", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); if (err < 0) return err; @@ -12407,8 +12346,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->line_out_pins[1] | cfg->line_out_pins[2]; if (nid == 0x16) { - err = add_control(spec, ALC_CTL_WIDGET_MUTE, - "Mono Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, "Mono", HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -13034,8 +12972,7 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -13060,12 +12997,10 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec) unsigned char bits; /* Check laptop headphone socket */ - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); /* Check port replicator headphone socket */ - present |= snd_hda_codec_read(codec, 0x1a, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present |= snd_hda_jack_detect(codec, 0x1a); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, @@ -13089,11 +13024,8 @@ static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) unsigned int present_laptop; unsigned int present_dock; - present_laptop = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - - present_dock = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present_laptop = snd_hda_jack_detect(codec, 0x18); + present_dock = snd_hda_jack_detect(codec, 0x1b); /* Laptop mic port overrides dock mic port, design decision */ if (present_dock) @@ -13178,8 +13110,7 @@ static void alc269_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -13525,6 +13456,15 @@ static int patch_alc269(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x04, 15); + if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010){ + kfree(codec->chip_name); + codec->chip_name = kstrdup("ALC259", GFP_KERNEL); + if (!codec->chip_name) { + alc_free(codec); + return -ENOMEM; + } + } + board_config = snd_hda_check_board_config(codec, ALC269_MODEL_LAST, alc269_models, alc269_cfg_tbl); @@ -14157,10 +14097,8 @@ static struct hda_verb alc861_toshiba_init_verbs[] = { /* toggle speaker-output according to the hp-jack state */ static void alc861_toshiba_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = snd_hda_jack_detect(codec, 0x0f); - present = snd_hda_codec_read(codec, 0x0f, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x16, HDA_INPUT, 0, HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 3, @@ -14260,9 +14198,7 @@ static int alc861_auto_fill_dac_nids(struct hda_codec *codec, static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx, hda_nid_t nid, unsigned int chs) { - char name[32]; - snprintf(name, sizeof(name), "%s Playback Switch", pfx); - return add_control(codec->spec, ALC_CTL_WIDGET_MUTE, name, + return add_pb_sw_ctrl(codec->spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); } @@ -14627,6 +14563,27 @@ static struct alc_config_preset alc861_presets[] = { }, }; +/* Pin config fixes */ +enum { + PINFIX_FSC_AMILO_PI1505, +}; + +static struct alc_pincfg alc861_fsc_amilo_pi1505_pinfix[] = { + { 0x0b, 0x0221101f }, /* HP */ + { 0x0f, 0x90170310 }, /* speaker */ + { } +}; + +static const struct alc_fixup alc861_fixups[] = { + [PINFIX_FSC_AMILO_PI1505] = { + .pins = alc861_fsc_amilo_pi1505_pinfix + }, +}; + +static struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK(0x1734, 0x10c7, "FSC Amilo Pi1505", PINFIX_FSC_AMILO_PI1505), + {} +}; static int patch_alc861(struct hda_codec *codec) { @@ -14650,6 +14607,8 @@ static int patch_alc861(struct hda_codec *codec) board_config = ALC861_AUTO; } + alc_pick_fixup(codec, alc861_fixup_tbl, alc861_fixups); + if (board_config == ALC861_AUTO) { /* automatic parse from the BIOS config */ err = alc861_parse_auto_config(codec); @@ -15067,9 +15026,9 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x18); bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); } @@ -15386,7 +15345,6 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - char name[32]; static const char *chname[4] = {"Front", "Surround", "CLFE", "Side"}; hda_nid_t nid_v, nid_s; int i, err; @@ -15403,26 +15361,26 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, if (i == 2) { /* Center/LFE */ - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "Center Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "Center", HDA_COMPOSE_AMP_VAL(nid_v, 1, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_WIDGET_VOL, - "LFE Playback Volume", + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, + "LFE", HDA_COMPOSE_AMP_VAL(nid_v, 2, 0, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "Center Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "Center", HDA_COMPOSE_AMP_VAL(nid_s, 1, 2, HDA_INPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, - "LFE Playback Switch", + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, + "LFE", HDA_COMPOSE_AMP_VAL(nid_s, 2, 2, HDA_INPUT)); if (err < 0) @@ -15437,8 +15395,7 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, pfx = "PCM"; } else pfx = chname[i]; - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) @@ -15446,8 +15403,7 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec, if (cfg->line_outs == 1 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) pfx = "Speaker"; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) @@ -15465,7 +15421,6 @@ static int alc861vd_auto_create_extra_out(struct alc_spec *spec, { hda_nid_t nid_v, nid_s; int err; - char name[32]; if (!pin) return 0; @@ -15483,21 +15438,18 @@ static int alc861vd_auto_create_extra_out(struct alc_spec *spec, nid_s = alc861vd_idx_to_mixer_switch( alc880_fixed_pin_idx(pin)); - sprintf(name, "%s Playback Volume", pfx); - err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + err = add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid_v, 3, 0, HDA_OUTPUT)); if (err < 0) return err; - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_BIND_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid_s, 3, 2, HDA_INPUT)); if (err < 0) return err; } else if (alc880_is_multi_pin(pin)) { /* set manual connection */ /* we have only a switch on HP-out PIN */ - sprintf(name, "%s Playback Switch", pfx); - err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + err = add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -16387,9 +16339,9 @@ static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x14, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x14); bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); } @@ -16399,9 +16351,9 @@ static void alc662_lenovo_101e_all_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + present = snd_hda_jack_detect(codec, 0x1b); bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -16460,9 +16412,7 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -16475,9 +16425,7 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -16494,9 +16442,7 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, AMP_IN_MUTE(0), bits); @@ -16513,9 +16459,7 @@ static void alc662_f5z_speaker_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x1b); bits = present ? 0 : PIN_OUT; snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, bits); @@ -16525,12 +16469,8 @@ static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec) { unsigned int present1, present2; - present1 = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - present2 = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present1 = snd_hda_jack_detect(codec, 0x21); + present2 = snd_hda_jack_detect(codec, 0x15); if (present1 || present2) { snd_hda_codec_write_cache(codec, 0x14, 0, @@ -16545,12 +16485,8 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) { unsigned int present1, present2; - present1 = snd_hda_codec_read(codec, 0x1b, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - present2 = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present1 = snd_hda_jack_detect(codec, 0x1b); + present2 = snd_hda_jack_detect(codec, 0x15); if (present1 || present2) { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, @@ -16710,9 +16646,7 @@ static void alc663_g71v_hp_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); @@ -16725,9 +16659,7 @@ static void alc663_g71v_front_automute(struct hda_codec *codec) unsigned int present; unsigned char bits; - present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); @@ -17264,21 +17196,17 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec, return 0; } -static int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx, +static inline int alc662_add_vol_ctl(struct alc_spec *spec, const char *pfx, hda_nid_t nid, unsigned int chs) { - char name[32]; - sprintf(name, "%s Playback Volume", pfx); - return add_control(spec, ALC_CTL_WIDGET_VOL, name, + return add_pb_vol_ctrl(spec, ALC_CTL_WIDGET_VOL, pfx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); } -static int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx, +static inline int alc662_add_sw_ctl(struct alc_spec *spec, const char *pfx, hda_nid_t nid, unsigned int chs) { - char name[32]; - sprintf(name, "%s Playback Switch", pfx); - return add_control(spec, ALC_CTL_WIDGET_MUTE, name, + return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_INPUT)); } @@ -17356,13 +17284,11 @@ static int alc662_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, return 0; nid = alc662_look_for_dac(codec, pin); if (!nid) { - char name[32]; /* the corresponding DAC is already occupied */ if (!(get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)) return 0; /* no way */ /* create a switch only */ - sprintf(name, "%s Playback Switch", pfx); - return add_control(spec, ALC_CTL_WIDGET_MUTE, name, + return add_pb_sw_ctrl(spec, ALC_CTL_WIDGET_MUTE, pfx, HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); } @@ -17538,6 +17464,15 @@ static int patch_alc662(struct hda_codec *codec) alc_fix_pll_init(codec, 0x20, 0x04, 15); + if (alc_read_coef_idx(codec, 0)==0x8020){ + kfree(codec->chip_name); + codec->chip_name = kstrdup("ALC661", GFP_KERNEL); + if (!codec->chip_name) { + alc_free(codec); + return -ENOMEM; + } + } + board_config = snd_hda_check_board_config(codec, ALC662_MODEL_LAST, alc662_models, alc662_cfg_tbl); @@ -17604,6 +17539,20 @@ static int patch_alc662(struct hda_codec *codec) return 0; } +static int patch_alc888(struct hda_codec *codec) +{ + if ((alc_read_coef_idx(codec, 0) & 0x00f0)==0x0030){ + kfree(codec->chip_name); + codec->chip_name = kstrdup("ALC888-VD", GFP_KERNEL); + if (!codec->chip_name) { + alc_free(codec); + return -ENOMEM; + } + return patch_alc662(codec); + } + return patch_alc882(codec); +} + /* * patch entries */ @@ -17635,8 +17584,9 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 }, { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", .patch = patch_alc882 }, - { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 }, + { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc888 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 }, + { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 86de305fc9f..6b0bc040c3b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -93,6 +93,7 @@ enum { STAC_92HD83XXX_REF, STAC_92HD83XXX_PWR_REF, STAC_DELL_S14, + STAC_92HD83XXX_HP, STAC_92HD83XXX_MODELS }; @@ -1085,7 +1086,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (!spec->auto_mic && spec->num_dmuxes > 0 && snd_hda_get_bool_hint(codec, "separate_dmux") == 1) { stac_dmux_mixer.count = spec->num_dmuxes; - err = snd_hda_ctl_add(codec, + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&stac_dmux_mixer, codec)); if (err < 0) return err; @@ -1101,7 +1102,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) spec->spdif_mute = 1; } stac_smux_mixer.count = spec->num_smuxes; - err = snd_hda_ctl_add(codec, + err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&stac_smux_mixer, codec)); if (err < 0) return err; @@ -1624,6 +1625,7 @@ static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = "ref", [STAC_92HD83XXX_PWR_REF] = "mic-ref", [STAC_DELL_S14] = "dell-s14", + [STAC_92HD83XXX_HP] = "hp", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { @@ -1634,6 +1636,8 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { "DFI LanParty", STAC_92HD83XXX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, "unknown Dell", STAC_DELL_S14), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xff00, 0x3600, + "HP", STAC_92HD83XXX_HP), {} /* terminator */ }; @@ -2648,6 +2652,7 @@ static int stac92xx_clfe_switch_put(struct snd_kcontrol *kcontrol, enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, + STAC_CTL_WIDGET_MUTE_BEEP, STAC_CTL_WIDGET_MONO_MUX, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, @@ -2658,6 +2663,7 @@ enum { static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + HDA_CODEC_MUTE_BEEP(NULL, 0, 0, 0), STAC_MONO_MUX, STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), @@ -2669,7 +2675,8 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { static struct snd_kcontrol_new * stac_control_new(struct sigmatel_spec *spec, struct snd_kcontrol_new *ktemp, - const char *name) + const char *name, + hda_nid_t nid) { struct snd_kcontrol_new *knew; @@ -2685,6 +2692,8 @@ stac_control_new(struct sigmatel_spec *spec, spec->kctls.alloced--; return NULL; } + if (nid) + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; return knew; } @@ -2693,7 +2702,8 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec, int idx, const char *name, unsigned long val) { - struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name); + struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, + get_amp_nid_(val)); if (!knew) return -ENOMEM; knew->index = idx; @@ -2764,7 +2774,7 @@ static int stac92xx_add_input_source(struct sigmatel_spec *spec) if (!spec->num_adcs || imux->num_items <= 1) return 0; /* no need for input source control */ knew = stac_control_new(spec, &stac_input_src_temp, - stac_input_src_temp.name); + stac_input_src_temp.name, 0); if (!knew) return -ENOMEM; knew->count = spec->num_adcs; @@ -3221,12 +3231,15 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, { struct sigmatel_spec *spec = codec->spec; u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); - int err; + int err, type = STAC_CTL_WIDGET_MUTE_BEEP; + + if (spec->anabeep_nid == nid) + type = STAC_CTL_WIDGET_MUTE; /* check for mute support for the the amp */ if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "PC Beep Playback Switch", + err = stac92xx_add_control(spec, type, + "Beep Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -3235,7 +3248,7 @@ static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, /* check to see if there is volume support for the amp */ if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "PC Beep Playback Volume", + "Beep Playback Volume", HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -3258,12 +3271,7 @@ static int stac92xx_dig_beep_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int enabled = !!ucontrol->value.integer.value[0]; - if (codec->beep->enabled != enabled) { - codec->beep->enabled = enabled; - return 1; - } - return 0; + return snd_hda_enable_beep_device(codec, ucontrol->value.integer.value[0]); } static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { @@ -3276,7 +3284,7 @@ static struct snd_kcontrol_new stac92xx_dig_beep_ctrl = { static int stac92xx_beep_switch_ctl(struct hda_codec *codec) { return stac92xx_add_control_temp(codec->spec, &stac92xx_dig_beep_ctrl, - 0, "PC Beep Playback Switch", 0); + 0, "Beep Playback Switch", 0); } #endif @@ -3631,6 +3639,26 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec) } } +static int is_dual_headphones(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i, valid_hps; + + if (spec->autocfg.line_out_type != AUTO_PIN_SPEAKER_OUT || + spec->autocfg.hp_outs <= 1) + return 0; + valid_hps = 0; + for (i = 0; i < spec->autocfg.hp_outs; i++) { + hda_nid_t nid = spec->autocfg.hp_pins[i]; + unsigned int cfg = snd_hda_codec_get_pincfg(codec, nid); + if (get_defcfg_location(cfg) & AC_JACK_LOC_SEPARATE) + continue; + valid_hps++; + } + return (valid_hps > 1); +} + + static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) { struct sigmatel_spec *spec = codec->spec; @@ -3647,8 +3675,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out /* If we have no real line-out pin and multiple hp-outs, HPs should * be set up as multi-channel outputs. */ - if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT && - spec->autocfg.hp_outs > 1) { + if (is_dual_headphones(codec)) { /* Copy hp_outs to line_outs, backup line_outs in * speaker_outs so that the following routines can handle * HP pins as primary outputs. @@ -4329,6 +4356,28 @@ static void stac92xx_free_kctls(struct hda_codec *codec) snd_array_free(&spec->kctls); } +static void stac92xx_shutup(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + hda_nid_t nid; + + /* reset each pin before powering down DAC/ADC to avoid click noise */ + nid = codec->start_nid; + for (i = 0; i < codec->num_nodes; i++, nid++) { + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wcaps); + if (wid_type == AC_WID_PIN) + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } + + if (spec->eapd_mask) + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, spec->gpio_data & + ~spec->eapd_mask); +} + static void stac92xx_free(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -4336,6 +4385,7 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; + stac92xx_shutup(codec); stac92xx_free_jacks(codec); snd_array_free(&spec->events); @@ -4386,12 +4436,16 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, pin_ctl & ~flag); } -static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) +static inline int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) { if (!nid) return 0; - if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0x00) - & (1 << 31)) + /* NOTE: we can't use snd_hda_jack_detect() here because STAC/IDT + * codecs behave wrongly when SET_PIN_SENSE is triggered, although + * the pincap gives TRIG_REQ bit. + */ + if (snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0) & + AC_PINSENSE_PRESENCE) return 1; return 0; } @@ -4791,28 +4845,28 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec, return 0; } -#endif -static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) +static int idt92hd83xxx_hp_check_power_status(struct hda_codec *codec, + hda_nid_t nid) { struct sigmatel_spec *spec = codec->spec; - int i; - hda_nid_t nid; - /* reset each pin before powering down DAC/ADC to avoid click noise */ - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - unsigned int wid_type = get_wcaps_type(wcaps); - if (wid_type == AC_WID_PIN) - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } + if (nid != 0x13) + return 0; + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & HDA_AMP_MUTE) + spec->gpio_data |= spec->gpio_led; /* mute LED on */ + else + spec->gpio_data &= ~spec->gpio_led; /* mute LED off */ + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); - if (spec->eapd_mask) - stac_gpio_set(codec, spec->gpio_mask, - spec->gpio_dir, spec->gpio_data & - ~spec->eapd_mask); + return 0; +} + +#endif + +static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + stac92xx_shutup(codec); return 0; } #endif @@ -4827,6 +4881,7 @@ static struct hda_codec_ops stac92xx_patch_ops = { .suspend = stac92xx_suspend, .resume = stac92xx_resume, #endif + .reboot_notify = stac92xx_shutup, }; static int patch_stac9200(struct hda_codec *codec) @@ -5172,6 +5227,22 @@ again: break; } + codec->patch_ops = stac92xx_patch_ops; + + if (spec->board_config == STAC_92HD83XXX_HP) + spec->gpio_led = 0x01; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (spec->gpio_led) { + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + spec->gpio_data |= spec->gpio_led; + /* register check_power_status callback. */ + codec->patch_ops.check_power_status = + idt92hd83xxx_hp_check_power_status; + } +#endif + err = stac92xx_parse_auto_config(codec, 0x1d, 0); if (!err) { if (spec->board_config < 0) { @@ -5207,8 +5278,6 @@ again: snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, num_dacs); - codec->patch_ops = stac92xx_patch_ops; - codec->proc_widget_hook = stac92hd_proc_hook; return 0; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index ee89db90c9b..b70e26ad263 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1,10 +1,10 @@ /* * Universal Interface for Intel High Definition Audio Codec * - * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec + * HD audio interface patch for VIA VT17xx/VT18xx/VT20xx codec * - * Copyright (c) 2006-2008 Lydia Wang <lydiawang@viatech.com> - * Takashi Iwai <tiwai@suse.de> + * (C) 2006-2009 VIA Technology, Inc. + * (C) 2006-2008 Takashi Iwai <tiwai@suse.de> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -22,21 +22,27 @@ */ /* * * * * * * * * * * * * * Release History * * * * * * * * * * * * * * * * */ -/* */ +/* */ /* 2006-03-03 Lydia Wang Create the basic patch to support VT1708 codec */ -/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ -/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ +/* 2006-03-14 Lydia Wang Modify hard code for some pin widget nid */ +/* 2006-08-02 Lydia Wang Add support to VT1709 codec */ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ -/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ -/* 2007-09-17 Lydia Wang Add VT1708B codec support */ +/* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ +/* 2007-09-17 Lydia Wang Add VT1708B codec support */ /* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */ /* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */ -/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ -/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ -/* 2008-04-09 Lydia Wang Add Independent HP feature */ +/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ +/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ +/* 2008-04-09 Lydia Wang Add Independent HP feature */ /* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ -/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ -/* */ +/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ +/* 2009-02-16 Logan Li Add support for VT1718S */ +/* 2009-03-13 Logan Li Add support for VT1716S */ +/* 2009-04-14 Lydai Wang Add support for VT1828S and VT2020 */ +/* 2009-07-08 Lydia Wang Add support for VT2002P */ +/* 2009-07-21 Lydia Wang Add support for VT1812 */ +/* 2009-09-19 Lydia Wang Add support for VT1818S */ +/* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -76,14 +82,6 @@ #define VT1702_HP_NID 0x17 #define VT1702_DIGOUT_NID 0x11 -#define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) -#define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) -#define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) -#define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) -#define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) -#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397) -#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398) - enum VIA_HDA_CODEC { UNKNOWN = -1, VT1708, @@ -92,12 +90,76 @@ enum VIA_HDA_CODEC { VT1708B_8CH, VT1708B_4CH, VT1708S, + VT1708BCE, VT1702, + VT1718S, + VT1716S, + VT2002P, + VT1812, CODEC_TYPES, }; -static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) +struct via_spec { + /* codec parameterization */ + struct snd_kcontrol_new *mixers[6]; + unsigned int num_mixers; + + struct hda_verb *init_verbs[5]; + unsigned int num_iverbs; + + char *stream_name_analog; + struct hda_pcm_stream *stream_analog_playback; + struct hda_pcm_stream *stream_analog_capture; + + char *stream_name_digital; + struct hda_pcm_stream *stream_digital_playback; + struct hda_pcm_stream *stream_digital_capture; + + /* playback */ + struct hda_multi_out multiout; + hda_nid_t slave_dig_outs[2]; + + /* capture */ + unsigned int num_adc_nids; + hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; + hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; + + /* capture source */ + const struct hda_input_mux *input_mux; + unsigned int cur_mux[3]; + + /* PCM information */ + struct hda_pcm pcm_rec[3]; + + /* dynamic controls, init_verbs and input_mux */ + struct auto_pin_cfg autocfg; + struct snd_array kctls; + struct hda_input_mux private_imux[2]; + hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + + /* HP mode source */ + const struct hda_input_mux *hp_mux; + unsigned int hp_independent_mode; + unsigned int hp_independent_mode_index; + unsigned int smart51_enabled; + unsigned int dmic_enabled; + enum VIA_HDA_CODEC codec_type; + + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work vt1708_hp_work; + int vt1708_jack_detectect; + int vt1708_hp_present; +#ifdef CONFIG_SND_HDA_POWER_SAVE + struct hda_loopback_check loopback; +#endif +}; + +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { + u32 vendor_id = codec->vendor_id; u16 ven_id = vendor_id >> 16; u16 dev_id = vendor_id & 0xffff; enum VIA_HDA_CODEC codec_type; @@ -111,9 +173,11 @@ static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) codec_type = VT1709_10CH; else if (dev_id >= 0xe714 && dev_id <= 0xe717) codec_type = VT1709_6CH; - else if (dev_id >= 0xe720 && dev_id <= 0xe723) + else if (dev_id >= 0xe720 && dev_id <= 0xe723) { codec_type = VT1708B_8CH; - else if (dev_id >= 0xe724 && dev_id <= 0xe727) + if (snd_hda_param_read(codec, 0x16, AC_PAR_CONNLIST_LEN) == 0x7) + codec_type = VT1708BCE; + } else if (dev_id >= 0xe724 && dev_id <= 0xe727) codec_type = VT1708B_4CH; else if ((dev_id & 0xfff) == 0x397 && (dev_id >> 12) < 8) @@ -121,6 +185,19 @@ static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) else if ((dev_id & 0xfff) == 0x398 && (dev_id >> 12) < 8) codec_type = VT1702; + else if ((dev_id & 0xfff) == 0x428 + && (dev_id >> 12) < 8) + codec_type = VT1718S; + else if (dev_id == 0x0433 || dev_id == 0xa721) + codec_type = VT1716S; + else if (dev_id == 0x0441 || dev_id == 0x4441) + codec_type = VT1718S; + else if (dev_id == 0x0438 || dev_id == 0x4438) + codec_type = VT2002P; + else if (dev_id == 0x0448) + codec_type = VT1812; + else if (dev_id == 0x0440) + codec_type = VT1708S; else codec_type = UNKNOWN; return codec_type; @@ -128,10 +205,16 @@ static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 +#define VIA_JACK_EVENT 0x04 +#define VIA_MONO_EVENT 0x08 +#define VIA_SPEAKER_EVENT 0x10 +#define VIA_BIND_HP_EVENT 0x20 enum { VIA_CTL_WIDGET_VOL, VIA_CTL_WIDGET_MUTE, + VIA_CTL_WIDGET_ANALOG_MUTE, + VIA_CTL_WIDGET_BIND_PIN_MUTE, }; enum { @@ -141,99 +224,162 @@ enum { AUTO_SEQ_SIDE }; -/* Some VT1708S based boards gets the micboost setting wrong, so we have - * to apply some brute-force and re-write the TLV's by software. */ -static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, - unsigned int size, unsigned int __user *_tlv) +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle); +static void set_jack_power_state(struct hda_codec *codec); +static int is_aa_path_mute(struct hda_codec *codec); + +static void vt1708_start_hp_work(struct via_spec *spec) { - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + if (!delayed_work_pending(&spec->vt1708_hp_work)) + schedule_delayed_work(&spec->vt1708_hp_work, + msecs_to_jiffies(100)); +} - if (get_codec_type(codec->vendor_id) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - if (size < 4 * sizeof(unsigned int)) - return -ENOMEM; - if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */ - return -EFAULT; - if (put_user(2 * sizeof(unsigned int), _tlv + 1)) - return -EFAULT; - if (put_user(0, _tlv + 2)) /* offset = 0 */ - return -EFAULT; - if (put_user(1000, _tlv + 3)) /* step size = 10 dB */ - return -EFAULT; - } - return 0; +static void vt1708_stop_hp_work(struct via_spec *spec) +{ + if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) + return; + if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 + && !is_aa_path_mute(spec->codec)) + return; + snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, + !spec->vt1708_jack_detectect); + cancel_delayed_work(&spec->vt1708_hp_work); + flush_scheduled_work(); } -static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) + +static int analog_input_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { + int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S - && (nid == 0x1a || nid == 0x1e)) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 3; + set_jack_power_state(codec); + analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); + if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { + if (is_aa_path_mute(codec)) + vt1708_start_hp_work(codec->spec); + else + vt1708_stop_hp_work(codec->spec); } - return 0; + return change; } -static struct snd_kcontrol_new vt1708_control_templates[] = { - HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), -}; - - -struct via_spec { - /* codec parameterization */ - struct snd_kcontrol_new *mixers[3]; - unsigned int num_mixers; +/* modify .put = snd_hda_mixer_amp_switch_put */ +#define ANALOG_INPUT_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = analog_input_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } - struct hda_verb *init_verbs[5]; - unsigned int num_iverbs; +static void via_hp_bind_automute(struct hda_codec *codec); - char *stream_name_analog; - struct hda_pcm_stream *stream_analog_playback; - struct hda_pcm_stream *stream_analog_capture; - - char *stream_name_digital; - struct hda_pcm_stream *stream_digital_playback; - struct hda_pcm_stream *stream_digital_capture; - - /* playback */ - struct hda_multi_out multiout; - hda_nid_t slave_dig_outs[2]; - - /* capture */ - unsigned int num_adc_nids; - hda_nid_t *adc_nids; - hda_nid_t mux_nids[3]; - hda_nid_t dig_in_nid; - hda_nid_t dig_in_pin; +static int bind_pin_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int i; + int change = 0; - /* capture source */ - const struct hda_input_mux *input_mux; - unsigned int cur_mux[3]; + long *valp = ucontrol->value.integer.value; + int lmute, rmute; + if (strstr(kcontrol->id.name, "Switch") == NULL) { + snd_printd("Invalid control!\n"); + return change; + } + change = snd_hda_mixer_amp_switch_put(kcontrol, + ucontrol); + /* Get mute value */ + lmute = *valp ? 0 : HDA_AMP_MUTE; + valp++; + rmute = *valp ? 0 : HDA_AMP_MUTE; + + /* Set hp pins */ + if (!spec->hp_independent_mode) { + for (i = 0; i < spec->autocfg.hp_outs; i++) { + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + snd_hda_codec_amp_update( + codec, spec->autocfg.hp_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } - /* PCM information */ - struct hda_pcm pcm_rec[3]; + if (!lmute && !rmute) { + /* Line Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + /* unmute */ + via_hp_bind_automute(codec); - /* dynamic controls, init_verbs and input_mux */ - struct auto_pin_cfg autocfg; - struct snd_array kctls; - struct hda_input_mux private_imux[2]; - hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + } else { + if (lmute) { + /* Mute all left channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 0, HDA_OUTPUT, 0, HDA_AMP_MUTE, + lmute); + } + if (rmute) { + /* mute all right channels */ + for (i = 1; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.line_out_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_update( + codec, + spec->autocfg.speaker_pins[i], + 1, HDA_OUTPUT, 0, HDA_AMP_MUTE, + rmute); + } + } + return change; +} - /* HP mode source */ - const struct hda_input_mux *hp_mux; - unsigned int hp_independent_mode; +#define BIND_PIN_MUTE \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = NULL, \ + .index = 0, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = bind_pin_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0) } -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct hda_loopback_check loopback; -#endif +static struct snd_kcontrol_new via_control_templates[] = { + HDA_CODEC_VOLUME(NULL, 0, 0, 0), + HDA_CODEC_MUTE(NULL, 0, 0, 0), + ANALOG_INPUT_MUTE, + BIND_PIN_MUTE, }; static hda_nid_t vt1708_adc_nids[2] = { @@ -261,6 +407,27 @@ static hda_nid_t vt1702_adc_nids[3] = { 0x12, 0x20, 0x1F }; +static hda_nid_t vt1718S_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + +static hda_nid_t vt1716S_adc_nids[2] = { + /* ADC1-2 */ + 0x13, 0x14 +}; + +static hda_nid_t vt2002P_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + +static hda_nid_t vt1812_adc_nids[2] = { + /* ADC1-2 */ + 0x10, 0x11 +}; + + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -271,10 +438,12 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, knew = snd_array_new(&spec->kctls); if (!knew) return -ENOMEM; - *knew = vt1708_control_templates[type]; + *knew = via_control_templates[type]; knew->name = kstrdup(name, GFP_KERNEL); if (!knew->name) return -ENOMEM; + if (get_amp_nid_(val)) + knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); knew->private_value = val; return 0; } @@ -293,8 +462,8 @@ static void via_free_kctls(struct hda_codec *codec) } /* create input playback/capture controls for the given pin */ -static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, - const char *ctlname, int idx, int mix_nid) +static int via_new_analog_input(struct via_spec *spec, const char *ctlname, + int idx, int mix_nid) { char name[32]; int err; @@ -305,7 +474,7 @@ static int via_new_analog_input(struct via_spec *spec, hda_nid_t pin, if (err < 0) return err; sprintf(name, "%s Playback Switch", ctlname); - err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + err = via_add_control(spec, VIA_CTL_WIDGET_ANALOG_MUTE, name, HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT)); if (err < 0) return err; @@ -322,7 +491,7 @@ static void via_auto_set_output_and_unmute(struct hda_codec *codec, snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) - snd_hda_codec_write(codec, nid, 0, + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } @@ -343,10 +512,13 @@ static void via_auto_init_hp_out(struct hda_codec *codec) { struct via_spec *spec = codec->spec; hda_nid_t pin; + int i; - pin = spec->autocfg.hp_pins[0]; - if (pin) /* connect to front */ - via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + for (i = 0; i < spec->autocfg.hp_outs; i++) { + pin = spec->autocfg.hp_pins[i]; + if (pin) /* connect to front */ + via_auto_set_output_and_unmute(codec, pin, PIN_HP, 0); + } } static void via_auto_init_analog_input(struct hda_codec *codec) @@ -364,6 +536,502 @@ static void via_auto_init_analog_input(struct hda_codec *codec) } } + +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin); + +static void set_pin_power_state(struct hda_codec *codec, hda_nid_t nid, + unsigned int *affected_parm) +{ + unsigned parm; + unsigned def_conf = snd_hda_codec_get_pincfg(codec, nid); + unsigned no_presence = (def_conf & AC_DEFCFG_MISC) + >> AC_DEFCFG_MISC_SHIFT + & AC_DEFCFG_MISC_NO_PRESENCE; /* do not support pin sense */ + unsigned present = snd_hda_jack_detect(codec, nid); + struct via_spec *spec = codec->spec; + if ((spec->smart51_enabled && is_smart51_pins(spec, nid)) + || ((no_presence || present) + && get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)) { + *affected_parm = AC_PWRST_D0; /* if it's connected */ + parm = AC_PWRST_D0; + } else + parm = AC_PWRST_D3; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, parm); +} + +static void set_jack_power_state(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int imux_is_smixer; + unsigned int parm; + + if (spec->codec_type == VT1702) { + imux_is_smixer = snd_hda_codec_read( + codec, 0x13, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 1/2/5 (14h/15h/18h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x14, &parm); + set_pin_power_state(codec, 0x15, &parm); + set_pin_power_state(codec, 0x18, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; /* SW0 = stereo mixer (idx 3) */ + /* SW0 (13h), AIW 0/1/2 (12h/1fh/20h) */ + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW 3/4 (16h/17h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x16, &parm); + set_pin_power_state(codec, 0x17, &parm); + /* MW0 (1ah), AOW 0/1 (10h/1dh) */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1d, 0, AC_VERB_SET_POWER_STATE, + parm); + } else if (spec->codec_type == VT1708B_8CH + || spec->codec_type == VT1708B_4CH + || spec->codec_type == VT1708S) { + /* SW0 (17h) = stereo mixer */ + int is_8ch = spec->codec_type != VT1708B_4CH; + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) + == ((spec->codec_type == VT1708S) ? 5 : 0); + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW 0/1 (13h/14h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW6 (22h), SW2 (26h), AOW2 (24h) */ + if (is_8ch) { + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x22, &parm); + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x24, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* PW 3/4/7 (1ch/1dh/23h) */ + parm = AC_PWRST_D3; + /* force to D0 for internal Speaker */ + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + if (is_8ch) + set_pin_power_state(codec, 0x23, &parm); + /* MW0 (16h), Sw3 (27h), AOW 0/3 (10h/25h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + if (is_8ch) { + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x27, 0, + AC_VERB_SET_POWER_STATE, parm); + } + } else if (spec->codec_type == VT1718S) { + /* MUX6 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX6/7 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW3 (27h), MW2 (1ah), AOW3 (bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x27, &parm); + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0xb, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW2 (26h), AOW2 (ah) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0xa, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW0/1 (24h/25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + set_pin_power_state(codec, 0x25, &parm); + if (!spec->hp_independent_mode) /* check for redirected HP */ + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x8, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x9, 0, AC_VERB_SET_POWER_STATE, + parm); + /* MW9 (21h), Mw2 (1ah), AOW0 (8h) */ + snd_hda_codec_write(codec, 0x21, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + if (spec->hp_independent_mode) { + /* PW4 (28h), MW3 (1bh), MUX1(34h), AOW4 (ch) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0xc, 0, + AC_VERB_SET_POWER_STATE, parm); + } + } else if (spec->codec_type == VT1716S) { + unsigned int mono_out, present; + /* SW0 (17h) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x17, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 1/2/5 (1ah/1bh/1eh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1a, &parm); + set_pin_power_state(codec, 0x1b, &parm); + set_pin_power_state(codec, 0x1e, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* SW0 (17h), AIW0(13h) */ + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x13, 0, AC_VERB_SET_POWER_STATE, + parm); + + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1e, &parm); + /* PW11 (22h) */ + if (spec->dmic_enabled) + set_pin_power_state(codec, 0x22, &parm); + else + snd_hda_codec_write( + codec, 0x22, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + /* SW2(26h), AIW1(14h) */ + snd_hda_codec_write(codec, 0x26, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* outputs */ + /* PW0 (19h), SW1 (18h), AOW1 (11h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x19, &parm); + /* Smart 5.1 PW2(1bh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1b, &parm); + snd_hda_codec_write(codec, 0x18, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW7 (23h), SW3 (27h), AOW3 (25h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x23, &parm); + /* Smart 5.1 PW1(1ah) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1a, &parm); + snd_hda_codec_write(codec, 0x27, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Smart 5.1 PW5(1eh) */ + if (spec->smart51_enabled) + set_pin_power_state(codec, 0x1e, &parm); + snd_hda_codec_write(codec, 0x25, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* Mono out */ + /* SW4(28h)->MW1(29h)-> PW12 (2ah)*/ + present = snd_hda_jack_detect(codec, 0x1c); + if (present) + mono_out = 0; + else { + present = snd_hda_jack_detect(codec, 0x1d); + if (!spec->hp_independent_mode && present) + mono_out = 0; + else + mono_out = 1; + } + parm = mono_out ? AC_PWRST_D0 : AC_PWRST_D3; + snd_hda_codec_write(codec, 0x28, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x29, 0, AC_VERB_SET_POWER_STATE, + parm); + snd_hda_codec_write(codec, 0x2a, 0, AC_VERB_SET_POWER_STATE, + parm); + + /* PW 3/4 (1ch/1dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x1c, &parm); + set_pin_power_state(codec, 0x1d, &parm); + /* HP Independent Mode, power on AOW3 */ + if (spec->hp_independent_mode) + snd_hda_codec_write(codec, 0x25, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* force to D0 for internal Speaker */ + /* MW0 (16h), AOW0 (10h) */ + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_POWER_STATE, + imux_is_smixer ? AC_PWRST_D0 : parm); + snd_hda_codec_write(codec, 0x10, 0, AC_VERB_SET_POWER_STATE, + mono_out ? AC_PWRST_D0 : parm); + } else if (spec->codec_type == VT2002P) { + unsigned int present; + /* MUX9 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 3; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX9/10 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (26h), MW4 (1ch), MUX4(37h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x26, &parm); + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x37, + 0, AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (19h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Class-D */ + /* PW0 (24h), MW0(18h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* Mono Out */ + /* PW15 (31h), MW8(17h), MUX8(3bh) */ + present = snd_hda_jack_detect(codec, 0x26); + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else { + snd_hda_codec_write( + codec, 0x17, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + snd_hda_codec_write( + codec, 0x3b, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } else if (spec->codec_type == VT1812) { + unsigned int present; + /* MUX10 (1eh) = stereo mixer */ + imux_is_smixer = snd_hda_codec_read( + codec, 0x1e, 0, AC_VERB_GET_CONNECT_SEL, 0x00) == 5; + /* inputs */ + /* PW 5/6/7 (29h/2ah/2bh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x29, &parm); + set_pin_power_state(codec, 0x2a, &parm); + set_pin_power_state(codec, 0x2b, &parm); + if (imux_is_smixer) + parm = AC_PWRST_D0; + /* MUX10/11 (1eh/1fh), AIW 0/1 (10h/11h) */ + snd_hda_codec_write(codec, 0x1e, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x1f, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x11, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* outputs */ + /* AOW0 (8h)*/ + snd_hda_codec_write(codec, 0x8, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + + /* PW4 (28h), MW4 (18h), MUX4(38h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x28, &parm); + snd_hda_codec_write(codec, 0x18, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x38, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* PW1 (25h), MW1 (15h), MUX1(35h), AOW1 (9h) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x25, &parm); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x35, 0, + AC_VERB_SET_POWER_STATE, parm); + if (spec->hp_independent_mode) { + snd_hda_codec_write(codec, 0x9, 0, + AC_VERB_SET_POWER_STATE, parm); + } + + /* Internal Speaker */ + /* PW0 (24h), MW0(14h), MUX0(34h) */ + present = snd_hda_jack_detect(codec, 0x25); + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x24, &parm); + if (present) { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x34, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + /* Mono Out */ + /* PW13 (31h), MW13(1ch), MUX13(3ch), MW14(3eh) */ + present = snd_hda_jack_detect(codec, 0x28); + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x31, &parm); + if (present) { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D3); + } else { + snd_hda_codec_write(codec, 0x1c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3c, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + snd_hda_codec_write(codec, 0x3e, 0, + AC_VERB_SET_POWER_STATE, + AC_PWRST_D0); + } + + /* PW15 (33h), MW15 (1dh), MUX15(3dh) */ + parm = AC_PWRST_D3; + set_pin_power_state(codec, 0x33, &parm); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_POWER_STATE, parm); + snd_hda_codec_write(codec, 0x3d, 0, + AC_VERB_SET_POWER_STATE, parm); + + /* MW9 (21h) */ + if (imux_is_smixer || !is_aa_path_mute(codec)) + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + else + snd_hda_codec_write( + codec, 0x21, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + } +} + /* * input MUX handling */ @@ -395,6 +1063,14 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, if (!spec->mux_nids[adc_idx]) return -EINVAL; + /* switch to D0 beofre change index */ + if (snd_hda_codec_read(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_GET_POWER_STATE, 0x00) != AC_PWRST_D0) + snd_hda_codec_write(codec, spec->mux_nids[adc_idx], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + /* update jack power state */ + set_jack_power_state(codec); + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); @@ -413,16 +1089,74 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.hp_pins[0]; - unsigned int pinsel = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_SEL, - 0x00); - + hda_nid_t nid; + unsigned int pinsel; + + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + break; + case VT2002P: + nid = 0x35; + break; + case VT1812: + nid = 0x3d; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + /* use !! to translate conn sel 2 for VT1718S */ + pinsel = !!snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_SEL, + 0x00); ucontrol->value.enumerated.item[0] = pinsel; return 0; } +static void activate_ctl(struct hda_codec *codec, const char *name, int active) +{ + struct snd_kcontrol *ctl = snd_hda_find_mixer_ctl(codec, name); + if (ctl) { + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + ctl->vd[0].access |= active + ? 0 : SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(codec->bus->card, + SNDRV_CTL_EVENT_MASK_VALUE, &ctl->id); + } +} + +static int update_side_mute_status(struct hda_codec *codec) +{ + /* mute side channel */ + struct via_spec *spec = codec->spec; + unsigned int parm = spec->hp_independent_mode + ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + hda_nid_t sw3; + + switch (spec->codec_type) { + case VT1708: + sw3 = 0x1b; + break; + case VT1709_10CH: + sw3 = 0x29; + break; + case VT1708B_8CH: + case VT1708S: + sw3 = 0x27; + break; + default: + sw3 = 0; + break; + } + + if (sw3) + snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, + parm); + return 0; +} + static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -430,47 +1164,46 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; hda_nid_t nid = spec->autocfg.hp_pins[0]; unsigned int pinsel = ucontrol->value.enumerated.item[0]; - unsigned int con_nid = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - - if (con_nid == spec->multiout.hp_nid) { - if (pinsel == 0) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } else if (pinsel == 1) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } - } else { - if (pinsel == 0) { - if (spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs += 1; - spec->hp_independent_mode = 0; - } - } else if (pinsel == 1) { - if (!spec->hp_independent_mode) { - if (spec->multiout.num_dacs > 1) - spec->multiout.num_dacs -= 1; - spec->hp_independent_mode = 1; - } - } + /* Get Independent Mode index of headphone pin widget */ + spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel + ? 1 : 0; + + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ + spec->multiout.num_dacs = 4; + break; + case VT2002P: + nid = 0x35; + break; + case VT1812: + nid = 0x3d; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); + + if (spec->multiout.hp_nid && spec->multiout.hp_nid + != spec->multiout.dac_nids[HDA_FRONT]) + snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, + 0, 0, 0); + + update_side_mute_status(codec); + /* update HP volume/swtich active state */ + if (spec->codec_type == VT1708S + || spec->codec_type == VT1702 + || spec->codec_type == VT1718S + || spec->codec_type == VT1716S + || spec->codec_type == VT2002P + || spec->codec_type == VT1812) { + activate_ctl(codec, "Headphone Playback Volume", + spec->hp_independent_mode); + activate_ctl(codec, "Headphone Playback Switch", + spec->hp_independent_mode); } - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - pinsel); - - if (spec->multiout.hp_nid && - spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) - snd_hda_codec_setup_stream(codec, - spec->multiout.hp_nid, - 0, 0, 0); - return 0; } @@ -486,6 +1219,175 @@ static struct snd_kcontrol_new via_hp_mixer[] = { { } /* end */ }; +static void notify_aa_path_ctls(struct hda_codec *codec) +{ + int i; + struct snd_ctl_elem_id id; + const char *labels[] = {"Mic", "Front Mic", "Line"}; + + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + for (i = 0; i < ARRAY_SIZE(labels); i++) { + sprintf(id.name, "%s Playback Volume", labels[i]); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +static void mute_aa_path(struct hda_codec *codec, int mute) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708: + nid_mixer = 0x17; + start_idx = 2; + end_idx = 4; + break; + case VT1709_10CH: + case VT1709_6CH: + nid_mixer = 0x18; + start_idx = 2; + end_idx = 4; + break; + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + case VT1716S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + default: + return; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + int val = mute ? HDA_AMP_MUTE : HDA_AMP_UNMUTE; + snd_hda_codec_amp_stereo(codec, nid_mixer, HDA_INPUT, i, + HDA_AMP_MUTE, val); + } +} +static int is_smart51_pins(struct via_spec *spec, hda_nid_t pin) +{ + int res = 0; + int index; + for (index = AUTO_PIN_MIC; index < AUTO_PIN_FRONT_LINE; index++) { + if (pin == spec->autocfg.input_pins[index]) { + res = 1; + break; + } + } + return res; +} + +static int via_smart51_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int via_smart51_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int on = 1; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + int ctl = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, + 0); + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode + && spec->codec_type != VT1718S) + continue; /* ignore FMic for independent HP */ + if (ctl & AC_PINCTL_IN_EN + && !(ctl & AC_PINCTL_OUT_EN)) + on = 0; + } + } + *ucontrol->value.integer.value = on; + return 0; +} + +static int via_smart51_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int out_in = *ucontrol->value.integer.value + ? AC_PINCTL_OUT_EN : AC_PINCTL_IN_EN; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + int i; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + hda_nid_t nid = spec->autocfg.input_pins[index[i]]; + if (i == AUTO_PIN_FRONT_MIC + && spec->hp_independent_mode + && spec->codec_type != VT1718S) + continue; /* don't retask FMic for independent HP */ + if (nid) { + unsigned int parm = snd_hda_codec_read( + codec, nid, 0, + AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + parm &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); + parm |= out_in; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + parm); + if (out_in == AC_PINCTL_OUT_EN) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + if (spec->codec_type == VT1718S) + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, 0, HDA_AMP_MUTE, + HDA_AMP_UNMUTE); + } + if (i == AUTO_PIN_FRONT_MIC) { + if (spec->codec_type == VT1708S + || spec->codec_type == VT1716S) { + /* input = index 1 (AOW3) */ + snd_hda_codec_write( + codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, 1); + snd_hda_codec_amp_stereo( + codec, nid, HDA_OUTPUT, + 0, HDA_AMP_MUTE, HDA_AMP_UNMUTE); + } + } + } + spec->smart51_enabled = *ucontrol->value.integer.value; + set_jack_power_state(codec); + return 1; +} + +static struct snd_kcontrol_new via_smart51_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Smart 5.1", + .count = 1, + .info = via_smart51_info, + .get = via_smart51_get, + .put = via_smart51_put, + }, + {} /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -506,6 +1408,112 @@ static struct snd_kcontrol_new vt1708_capture_mixer[] = { }, { } /* end */ }; + +/* check AA path's mute statue */ +static int is_aa_path_mute(struct hda_codec *codec) +{ + int mute = 1; + hda_nid_t nid_mixer; + int start_idx; + int end_idx; + int i; + struct via_spec *spec = codec->spec; + /* get nid of MW0 and start & end index */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + case VT1708S: + case VT1716S: + nid_mixer = 0x16; + start_idx = 2; + end_idx = 4; + break; + case VT1702: + nid_mixer = 0x1a; + start_idx = 1; + end_idx = 3; + break; + case VT1718S: + nid_mixer = 0x21; + start_idx = 1; + end_idx = 3; + break; + case VT2002P: + case VT1812: + nid_mixer = 0x21; + start_idx = 0; + end_idx = 2; + break; + default: + return 0; + } + /* check AA path's mute status */ + for (i = start_idx; i <= end_idx; i++) { + unsigned int con_list = snd_hda_codec_read( + codec, nid_mixer, 0, AC_VERB_GET_CONNECT_LIST, i/4*4); + int shift = 8 * (i % 4); + hda_nid_t nid_pin = (con_list & (0xff << shift)) >> shift; + unsigned int defconf = snd_hda_codec_get_pincfg(codec, nid_pin); + if (get_defcfg_connect(defconf) == AC_JACK_PORT_COMPLEX) { + /* check mute status while the pin is connected */ + int mute_l = snd_hda_codec_amp_read(codec, nid_mixer, 0, + HDA_INPUT, i) >> 7; + int mute_r = snd_hda_codec_amp_read(codec, nid_mixer, 1, + HDA_INPUT, i) >> 7; + if (!mute_l || !mute_r) { + mute = 0; + break; + } + } + } + return mute; +} + +/* enter/exit analog low-current mode */ +static void analog_low_current_mode(struct hda_codec *codec, int stream_idle) +{ + struct via_spec *spec = codec->spec; + static int saved_stream_idle = 1; /* saved stream idle status */ + int enable = is_aa_path_mute(codec); + unsigned int verb = 0; + unsigned int parm = 0; + + if (stream_idle == -1) /* stream status did not change */ + enable = enable && saved_stream_idle; + else { + enable = enable && stream_idle; + saved_stream_idle = stream_idle; + } + + /* decide low current mode's verb & parameter */ + switch (spec->codec_type) { + case VT1708B_8CH: + case VT1708B_4CH: + verb = 0xf70; + parm = enable ? 0x02 : 0x00; /* 0x02: 2/3x, 0x00: 1x */ + break; + case VT1708S: + case VT1718S: + case VT1716S: + verb = 0xf73; + parm = enable ? 0x51 : 0xe1; /* 0x51: 4/28x, 0xe1: 1x */ + break; + case VT1702: + verb = 0xf73; + parm = enable ? 0x01 : 0x1d; /* 0x01: 4/40x, 0x1d: 1x */ + break; + case VT2002P: + case VT1812: + verb = 0xf93; + parm = enable ? 0x00 : 0xe0; /* 0x00: 4/40x, 0xe0: 1x */ + break; + default: + return; /* other codecs are not supported */ + } + /* send verb */ + snd_hda_codec_write(codec, codec->afg, 0, verb, parm); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -534,9 +1542,9 @@ static struct hda_verb vt1708_volume_init_verbs[] = { {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - - /* Setup default input to PW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* Setup default input MW0 to PW4 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -547,30 +1555,13 @@ static int via_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + analog_low_current_mode(codec, idle); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } -static int via_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, - stream_tag, format, substream); -} - -static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct via_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - - static void playback_multi_pcm_prep_0(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, @@ -615,8 +1606,8 @@ static void playback_multi_pcm_prep_0(struct hda_codec *codec, snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); - if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && - !spec->hp_independent_mode) + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] + && !spec->hp_independent_mode) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); @@ -658,7 +1649,7 @@ static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo, snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); } - + vt1708_start_hp_work(spec); return 0; } @@ -698,7 +1689,7 @@ static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo, snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); } - + vt1708_stop_hp_work(spec); return 0; } @@ -779,7 +1770,7 @@ static struct hda_pcm_stream vt1708_pcm_analog_playback = { }; static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ @@ -790,8 +1781,8 @@ static struct hda_pcm_stream vt1708_pcm_analog_s16_playback = { .formats = SNDRV_PCM_FMTBIT_S16_LE, .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -853,6 +1844,11 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* init power states */ + set_jack_power_state(codec); + analog_low_current_mode(codec, 1); + via_free_kctls(codec); /* no longer needed */ return 0; } @@ -866,8 +1862,10 @@ static int via_build_pcms(struct hda_codec *codec) codec->pcm_info = info; info->name = spec->stream_name_analog; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = + *(spec->stream_analog_playback); + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; @@ -904,20 +1902,58 @@ static void via_free(struct hda_codec *codec) return; via_free_kctls(codec); + vt1708_stop_hp_work(spec); kfree(codec->spec); } /* mute internal speaker if HP is plugged */ static void via_hp_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = 0; struct via_spec *spec = codec->spec; - present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], - HDA_OUTPUT, 0, HDA_AMP_MUTE, - present ? HDA_AMP_MUTE : 0); + present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + /* auto mute */ + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Front Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +/* mute mono out if HP or Line out is plugged */ +static void via_mono_automute(struct hda_codec *codec) +{ + unsigned int hp_present, lineout_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1716S) + return; + + lineout_present = snd_hda_jack_detect(codec, + spec->autocfg.line_out_pins[0]); + + /* Mute Mono Out if Line Out is plugged */ + if (lineout_present) { + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, HDA_AMP_MUTE); + return; + } + + hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); + + if (!spec->hp_independent_mode) + snd_hda_codec_amp_stereo( + codec, 0x2A, HDA_OUTPUT, 0, HDA_AMP_MUTE, + hp_present ? HDA_AMP_MUTE : 0); } static void via_gpio_control(struct hda_codec *codec) @@ -968,15 +2004,83 @@ static void via_gpio_control(struct hda_codec *codec) } } +/* mute Internal-Speaker if HP is plugged */ +static void via_speaker_automute(struct hda_codec *codec) +{ + unsigned int hp_present; + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT2002P && spec->codec_type != VT1812) + return; + + hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); + + if (!spec->hp_independent_mode) { + struct snd_ctl_elem_id id; + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + /* notify change */ + memset(&id, 0, sizeof(id)); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + strcpy(id.name, "Speaker Playback Switch"); + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, + &id); + } +} + +/* mute line-out and internal speaker if HP is plugged */ +static void via_hp_bind_automute(struct hda_codec *codec) +{ + /* use long instead of int below just to avoid an internal compiler + * error with gcc 4.0.x + */ + unsigned long hp_present, present = 0; + struct via_spec *spec = codec->spec; + int i; + + if (!spec->autocfg.hp_pins[0] || !spec->autocfg.line_out_pins[0]) + return; + + hp_present = snd_hda_jack_detect(codec, spec->autocfg.hp_pins[0]); + + present = snd_hda_jack_detect(codec, spec->autocfg.line_out_pins[0]); + + if (!spec->hp_independent_mode) { + /* Mute Line-Outs */ + for (i = 0; i < spec->autocfg.line_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[i], + HDA_OUTPUT, 0, + HDA_AMP_MUTE, hp_present ? HDA_AMP_MUTE : 0); + if (hp_present) + present = hp_present; + } + /* Speakers */ + for (i = 0; i < spec->autocfg.speaker_outs; i++) + snd_hda_codec_amp_stereo( + codec, spec->autocfg.speaker_pins[i], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + + /* unsolicited event for jack sensing */ static void via_unsol_event(struct hda_codec *codec, unsigned int res) { res >>= 26; - if (res == VIA_HP_EVENT) + if (res & VIA_HP_EVENT) via_hp_automute(codec); - else if (res == VIA_GPIO_EVENT) + if (res & VIA_GPIO_EVENT) via_gpio_control(codec); + if (res & VIA_JACK_EVENT) + set_jack_power_state(codec); + if (res & VIA_MONO_EVENT) + via_mono_automute(codec); + if (res & VIA_SPEAKER_EVENT) + via_speaker_automute(codec); + if (res & VIA_BIND_HP_EVENT) + via_hp_bind_automute(codec); } static int via_init(struct hda_codec *codec) @@ -986,6 +2090,10 @@ static int via_init(struct hda_codec *codec) for (i = 0; i < spec->num_iverbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + spec->codec_type = get_codec_type(codec); + if (spec->codec_type == VT1708BCE) + spec->codec_type = VT1708S; /* VT1708BCE & VT1708S are almost + same */ /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (spec->dig_in_pin) { @@ -1003,8 +2111,17 @@ static int via_init(struct hda_codec *codec) if (spec->slave_dig_outs[0]) codec->slave_dig_outs = spec->slave_dig_outs; - return 0; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +static int via_suspend(struct hda_codec *codec, pm_message_t state) +{ + struct via_spec *spec = codec->spec; + vt1708_stop_hp_work(spec); + return 0; } +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE static int via_check_power_status(struct hda_codec *codec, hda_nid_t nid) @@ -1021,6 +2138,9 @@ static struct hda_codec_ops via_patch_ops = { .build_pcms = via_build_pcms, .init = via_init, .free = via_free, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = via_suspend, +#endif #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = via_check_power_status, #endif @@ -1036,8 +2156,8 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.num_dacs = cfg->line_outs; spec->multiout.dac_nids = spec->private_dac_nids; - - for(i = 0; i < 4; i++) { + + for (i = 0; i < 4; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ @@ -1067,7 +2187,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid, nid_vol = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x17, 0x19, 0x1a, 0x1b}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { @@ -1075,9 +2195,8 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, if (!nid) continue; - - if (i != AUTO_SEQ_FRONT) - nid_vol = 0x18 + i; + + nid_vol = nid_vols[i]; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -1105,21 +2224,21 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ + } else if (i == AUTO_SEQ_FRONT) { /* add control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x17, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -1178,6 +2297,7 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) return 0; spec->multiout.hp_nid = VT1708_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -1218,7 +2338,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -1231,8 +2351,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x17); + err = via_new_analog_input(spec, labels[i], idx, 0x17); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -1260,16 +2379,60 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { - if (seqassoc == 0xff) { - def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_set_pincfg(codec, nid, def_conf); - } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE + && (seqassoc == 0xf0 || seqassoc == 0xff)) { + def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1297,6 +2460,10 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); + if (err < 0) + return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; @@ -1316,19 +2483,44 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } /* init callback for auto-configuration model -- overriding the default init */ static int via_auto_init(struct hda_codec *codec) { + struct via_spec *spec = codec->spec; + via_init(codec); via_auto_init_multi_out(codec); via_auto_init_hp_out(codec); via_auto_init_analog_input(codec); + if (spec->codec_type == VT2002P || spec->codec_type == VT1812) { + via_hp_bind_automute(codec); + } else { + via_hp_automute(codec); + via_speaker_automute(codec); + } + return 0; } +static void vt1708_update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + vt1708_hp_work.work); + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (spec->vt1708_hp_present + != snd_hda_jack_detect(spec->codec, spec->autocfg.hp_pins[0])) { + spec->vt1708_hp_present ^= 1; + via_hp_automute(spec->codec); + } + vt1708_start_hp_work(spec); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1378,7 +2570,7 @@ static int patch_vt1708(struct hda_codec *codec) "from BIOS. Using genenic mode...\n"); } - + spec->stream_name_analog = "VT1708 Analog"; spec->stream_analog_playback = &vt1708_pcm_analog_playback; /* disable 32bit format on VT1708 */ @@ -1390,7 +2582,7 @@ static int patch_vt1708(struct hda_codec *codec) spec->stream_digital_playback = &vt1708_pcm_digital_playback; spec->stream_digital_capture = &vt1708_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708_adc_nids); @@ -1405,7 +2597,8 @@ static int patch_vt1708(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; } @@ -1433,7 +2626,8 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = { }; static struct hda_verb vt1709_uniwill_init_verbs[] = { - {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, { } }; @@ -1473,8 +2667,8 @@ static struct hda_verb vt1709_10ch_volume_init_verbs[] = { {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Set input of PW4 as AOW4 */ - {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Set input of PW4 as MW0 */ + {0x20, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, { } @@ -1487,8 +2681,8 @@ static struct hda_pcm_stream vt1709_10ch_pcm_analog_playback = { .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -1499,8 +2693,8 @@ static struct hda_pcm_stream vt1709_6ch_pcm_analog_playback = { .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, }, }; @@ -1575,11 +2769,11 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.dac_nids[cfg->line_outs] = 0x28; /* AOW4 */ } else if (cfg->line_outs == 3) { /* 6 channels */ - for(i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < cfg->line_outs; i++) { nid = cfg->line_out_pins[i]; if (nid) { /* config dac list */ - switch(i) { + switch (i) { case AUTO_SEQ_FRONT: /* AOW0 */ spec->multiout.dac_nids[i] = 0x10; @@ -1608,56 +2802,58 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid = 0; + hda_nid_t nid, nid_vol, nid_vols[] = {0x18, 0x1a, 0x1b, 0x29}; int i, err; for (i = 0; i <= AUTO_SEQ_SIDE; i++) { nid = cfg->line_out_pins[i]; - if (!nid) + if (!nid) continue; + nid_vol = nid_vols[i]; + if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Center Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "LFE Playback Volume", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 1, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(0x1b, 2, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); if (err < 0) return err; - } else if (i == AUTO_SEQ_FRONT){ - /* add control to mixer index 0 */ + } else if (i == AUTO_SEQ_FRONT) { + /* ADD control to mixer index 0 */ err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Master Front Playback Volume", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, "Master Front Playback Switch", - HDA_COMPOSE_AMP_VAL(0x18, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_INPUT)); if (err < 0) return err; - + /* add control to PW3 */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, @@ -1674,26 +2870,26 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, } else if (i == AUTO_SEQ_SURROUND) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SIDE) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -1714,6 +2910,7 @@ static int vt1709_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) spec->multiout.hp_nid = VT1709_HP_DAC_NID; else if (spec->multiout.num_dacs == 3) /* 6 channels */ spec->multiout.hp_nid = 0; + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -1752,7 +2949,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, case 0x1d: /* Mic */ idx = 2; break; - + case 0x1e: /* Line In */ idx = 3; break; @@ -1765,8 +2962,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x18); + err = via_new_analog_input(spec, labels[i], idx, 0x18); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -1816,6 +3012,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -1861,7 +3058,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -1955,7 +3152,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) spec->stream_digital_playback = &vt1709_pcm_digital_playback; spec->stream_digital_capture = &vt1709_pcm_digital_capture; - + if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); @@ -2024,7 +3221,7 @@ static struct hda_verb vt1708B_8ch_volume_init_verbs[] = { {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Setup default input to PW4 */ - {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, + {0x1d, AC_VERB_SET_CONNECT_SEL, 0}, /* PW9 Output enable */ {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* PW10 Input enable */ @@ -2068,10 +3265,29 @@ static struct hda_verb vt1708B_4ch_volume_init_verbs[] = { }; static struct hda_verb vt1708B_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; +static int via_pcm_open_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + int idle = substream->pstr->substream_opened == 1 + && substream->ref_count == 0; + + analog_low_current_mode(codec, idle); + return 0; +} + static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .substreams = 2, .channels_min = 2, @@ -2080,7 +3296,8 @@ static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2102,8 +3319,10 @@ static struct hda_pcm_stream vt1708B_pcm_analog_capture = { .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2260,6 +3479,7 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) return 0; spec->multiout.hp_nid = VT1708B_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -2313,8 +3533,7 @@ static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2364,6 +3583,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2376,12 +3596,14 @@ static struct hda_amp_list vt1708B_loopbacks[] = { { } /* end */ }; #endif - +static int patch_vt1708S(struct hda_codec *codec); static int patch_vt1708B_8ch(struct hda_codec *codec) { struct via_spec *spec; int err; + if (get_codec_type(codec) == VT1708BCE) + return patch_vt1708S(codec); /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2483,29 +3705,15 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) /* Patch for VT1708S */ -/* VT1708S software backdoor based override for buggy hardware micboost - * setting */ -#define MIC_BOOST_VOLUME(xname, nid) { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = xname, \ - .index = 0, \ - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ - SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ - .info = mic_boost_volume_info, \ - .get = snd_hda_mixer_amp_volume_get, \ - .put = snd_hda_mixer_amp_volume_put, \ - .tlv = { .c = mic_boost_tlv }, \ - .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) } - /* capture mixer elements */ static struct snd_kcontrol_new vt1708S_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), - MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A), - MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, /* The multiple "Capture Source" controls confuse alsamixer @@ -2542,11 +3750,21 @@ static struct hda_verb vt1708S_volume_init_verbs[] = { {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* Enable Mic Boost Volume backdoor */ {0x1, 0xf98, 0x1}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, { } }; static struct hda_verb vt1708S_uniwill_init_verbs[] = { - {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x22, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -2557,8 +3775,9 @@ static struct hda_pcm_stream vt1708S_pcm_analog_playback = { .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2568,8 +3787,10 @@ static struct hda_pcm_stream vt1708S_pcm_analog_capture = { .channels_max = 2, .nid = 0x13, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2726,6 +3947,7 @@ static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) return 0; spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */ + spec->hp_independent_mode_index = 1; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -2780,8 +4002,7 @@ static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec, idx = 1; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], - idx, 0x16); + err = via_new_analog_input(spec, labels[i], idx, 0x16); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -2852,6 +4073,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) if (spec->hp_mux) spec->mixers[spec->num_mixers++] = via_hp_mixer; + spec->mixers[spec->num_mixers++] = via_smart51_mixer; return 1; } @@ -2865,6 +4087,16 @@ static struct hda_amp_list vt1708S_loopbacks[] = { }; #endif +static void override_mic_boost(struct hda_codec *codec, hda_nid_t pin, + int offset, int num_steps, int step_size) +{ + snd_hda_override_amp_caps(codec, pin, HDA_INPUT, + (offset << AC_AMPCAP_OFFSET_SHIFT) | + (num_steps << AC_AMPCAP_NUM_STEPS_SHIFT) | + (step_size << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + static int patch_vt1708S(struct hda_codec *codec) { struct via_spec *spec; @@ -2890,17 +4122,25 @@ static int patch_vt1708S(struct hda_codec *codec) spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs; spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs; - spec->stream_name_analog = "VT1708S Analog"; + if (codec->vendor_id == 0x11060440) + spec->stream_name_analog = "VT1818S Analog"; + else + spec->stream_name_analog = "VT1708S Analog"; spec->stream_analog_playback = &vt1708S_pcm_analog_playback; spec->stream_analog_capture = &vt1708S_pcm_analog_capture; - spec->stream_name_digital = "VT1708S Digital"; + if (codec->vendor_id == 0x11060440) + spec->stream_name_digital = "VT1818S Digital"; + else + spec->stream_name_digital = "VT1708S Digital"; spec->stream_digital_playback = &vt1708S_pcm_digital_playback; if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; spec->num_mixers++; } @@ -2913,6 +4153,16 @@ static int patch_vt1708S(struct hda_codec *codec) spec->loopback.amplist = vt1708S_loopbacks; #endif + /* correct names for VT1708BCE */ + if (get_codec_type(codec) == VT1708BCE) { + kfree(codec->chip_name); + codec->chip_name = kstrdup("VT1708BCE", GFP_KERNEL); + snprintf(codec->bus->card->mixername, + sizeof(codec->bus->card->mixername), + "%s %s", codec->vendor_name, codec->chip_name); + spec->stream_name_analog = "VT1708BCE Analog"; + spec->stream_name_digital = "VT1708BCE Digital"; + } return 0; } @@ -2967,12 +4217,20 @@ static struct hda_verb vt1702_volume_init_verbs[] = { /* PW6 PW7 Output enable */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* mixer enable */ + {0x1, 0xF88, 0x3}, + /* GPIO 0~2 */ + {0x1, 0xF82, 0x3F}, { } }; static struct hda_verb vt1702_uniwill_init_verbs[] = { - {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT}, - {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, { } }; @@ -2984,7 +4242,8 @@ static struct hda_pcm_stream vt1702_pcm_analog_playback = { .ops = { .open = via_playback_pcm_open, .prepare = via_playback_multi_pcm_prepare, - .cleanup = via_playback_multi_pcm_cleanup + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -2994,8 +4253,10 @@ static struct hda_pcm_stream vt1702_pcm_analog_capture = { .channels_max = 2, .nid = 0x12, /* NID to query formats and rates */ .ops = { + .open = via_pcm_open_close, .prepare = via_capture_pcm_prepare, - .cleanup = via_capture_pcm_cleanup + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close }, }; @@ -3065,12 +4326,13 @@ static int vt1702_auto_create_line_out_ctls(struct via_spec *spec, static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { - int err; - + int err, i; + struct hda_input_mux *imux; + static const char *texts[] = { "ON", "OFF", NULL}; if (!pin) return 0; - spec->multiout.hp_nid = 0x1D; + spec->hp_independent_mode_index = 0; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, "Headphone Playback Volume", @@ -3084,8 +4346,18 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) if (err < 0) return err; - create_hp_imux(spec); + imux = &spec->private_imux[1]; + /* for hp mode select */ + i = 0; + while (texts[i] != NULL) { + imux->items[imux->num_items].label = texts[i]; + imux->items[imux->num_items].index = i; + imux->num_items++; + i++; + } + + spec->hp_mux = &spec->private_imux[1]; return 0; } @@ -3121,8 +4393,7 @@ static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec, idx = 3; break; } - err = via_new_analog_input(spec, cfg->input_pins[i], - labels[i], idx, 0x1A); + err = via_new_analog_input(spec, labels[i], idx, 0x1A); if (err < 0) return err; imux->items[imux->num_items].label = labels[i]; @@ -3152,6 +4423,12 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); if (err < 0) return err; + /* limit AA path volume to 0 dB */ + snd_hda_override_amp_caps(codec, 0x1A, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x5 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -3185,8 +4462,6 @@ static int patch_vt1702(struct hda_codec *codec) { struct via_spec *spec; int err; - unsigned int response; - unsigned char control; /* create a codec specific record */ spec = kzalloc(sizeof(*spec), GFP_KERNEL); @@ -3231,17 +4506,1638 @@ static int patch_vt1702(struct hda_codec *codec) spec->loopback.amplist = vt1702_loopbacks; #endif - /* Open backdoor */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0); - control = (unsigned char)(response & 0xff); - control |= 0x3; - snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control); + return 0; +} + +/* Patch for VT1718S */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1718S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1718S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + + /* Setup default input of Front HP to MW9 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* PW9 PW10 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + /* PW11 Input enable */ + {0x2f, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_IN_EN}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf88, 0x8}, + /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Unmute MW4's index 0 */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + + +static struct hda_verb vt1718S_uniwill_init_verbs[] = { + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x24, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x27, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 10, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1718S_pcm_digital_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1718S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 4; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x8; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0xa; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x9; + break; + case AUTO_SEQ_SIDE: + spec->multiout.dac_nids[i] = 0xb; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + hda_nid_t nid_vols[] = {0x8, 0x9, 0xa, 0xb}; + hda_nid_t nid_mutes[] = {0x24, 0x25, 0x26, 0x27}; + hda_nid_t nid, nid_vol, nid_mute = 0; + int i, err; + + for (i = 0; i <= AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* Front */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1718S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0xc; /* AOW4 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0xc, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1718S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 1; + break; + + case 0x2a: /* Line In */ + idx = 2; + break; + + case 0x29: /* Front Mic */ + idx = 3; + break; + + case 0x2c: /* CD */ + idx = 0; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + return 0; +} + +static int vt1718S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + + if (err < 0) + return err; + err = vt1718S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1718S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1718S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1718S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->autocfg.dig_in_pin && codec->vendor_id == 0x11060428) + spec->dig_in_nid = 0x13; + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1718S_loopbacks[] = { + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { 0x21, HDA_INPUT, 3 }, + { 0x21, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1718S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; - /* Enable GPIO 0&1 for volume&mute control */ - /* Enable GPIO 2 for DMIC-DATA */ - response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0); - control = (unsigned char)((response >> 16) & 0x3f); - snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control); + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1718S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1718S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1718S_uniwill_init_verbs; + + if (codec->vendor_id == 0x11060441) + spec->stream_name_analog = "VT2020 Analog"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_analog = "VT1828S Analog"; + else + spec->stream_name_analog = "VT1718S Analog"; + spec->stream_analog_playback = &vt1718S_pcm_analog_playback; + spec->stream_analog_capture = &vt1718S_pcm_analog_capture; + + if (codec->vendor_id == 0x11060441) + spec->stream_name_digital = "VT2020 Digital"; + else if (codec->vendor_id == 0x11064441) + spec->stream_name_digital = "VT1828S Digital"; + else + spec->stream_name_digital = "VT1718S Digital"; + spec->stream_digital_playback = &vt1718S_pcm_digital_playback; + if (codec->vendor_id == 0x11060428 || codec->vendor_id == 0x11060441) + spec->stream_digital_capture = &vt1718S_pcm_digital_capture; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1718S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1718S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1718S_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1718S_loopbacks; +#endif + + return 0; +} + +/* Patch for VT1716S */ + +static int vt1716s_dmic_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int vt1716s_dmic_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + int index = 0; + + index = snd_hda_codec_read(codec, 0x26, 0, + AC_VERB_GET_CONNECT_SEL, 0); + if (index != -1) + *ucontrol->value.integer.value = index; + + return 0; +} + +static int vt1716s_dmic_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int index = *ucontrol->value.integer.value; + + snd_hda_codec_write(codec, 0x26, 0, + AC_VERB_SET_CONNECT_SEL, index); + spec->dmic_enabled = index; + set_jack_power_state(codec); + + return 1; +} + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1716S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x1A, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x22, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Mic Capture Switch", + .count = 1, + .info = vt1716s_dmic_info, + .get = vt1716s_dmic_get, + .put = vt1716s_dmic_put, + }, + {} /* end */ +}; + + +/* mono-out mixer elements */ +static struct snd_kcontrol_new vt1716S_mono_out_mixer[] = { + HDA_CODEC_MUTE("Mono Playback Switch", 0x2a, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb vt1716S_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Stereo Mixer = 5 */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x5}, + + /* Setup default input of PW4 to MW0 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0}, + + /* Setup default input of SW1 as MW0 */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* Setup default input of SW4 as AOW0 */ + {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* PW9 PW10 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + + /* Unmute SW1, PW12 */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* PW12 Output enable */ + {0x2a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Enable Boost Volume backdoor */ + {0x1, 0xf8a, 0x80}, + /* don't bybass mixer */ + {0x1, 0xf88, 0xc0}, + /* Enable mono output */ + {0x1, 0xf90, 0x08}, + { } +}; + + +static struct hda_verb vt1716S_uniwill_init_verbs[] = { + {0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_HP_EVENT | VIA_JACK_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_MONO_EVENT | VIA_JACK_EVENT}, + {0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x23, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 6, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x13, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1716S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1716S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 3; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x10; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0x25; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x11; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1716S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[3] = { "Front", "Surround", "C/LFE" }; + hda_nid_t nid_vols[] = {0x10, 0x11, 0x25}; + hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x27}; + hda_nid_t nid, nid_vol, nid_mute; + int i, err; + + for (i = 0; i <= AUTO_SEQ_CENLFE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control( + spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +} + +static int vt1716S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x25; /* AOW3 */ + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1716S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x1a: /* Mic */ + idx = 2; + break; + + case 0x1b: /* Line In */ + idx = 3; + break; + + case 0x1e: /* Front Mic */ + idx = 4; + break; + + case 0x1f: /* CD */ + idx = 1; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x16); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx-1; + imux->num_items++; + } + return 0; +} + +static int vt1716S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = vt1716S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1716S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1716S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1716S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + spec->mixers[spec->num_mixers++] = via_smart51_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1716S_loopbacks[] = { + { 0x16, HDA_INPUT, 1 }, + { 0x16, HDA_INPUT, 2 }, + { 0x16, HDA_INPUT, 3 }, + { 0x16, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1716S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1716S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1716S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1716S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1716S Analog"; + spec->stream_analog_playback = &vt1716S_pcm_analog_playback; + spec->stream_analog_capture = &vt1716S_pcm_analog_capture; + + spec->stream_name_digital = "VT1716S Digital"; + spec->stream_digital_playback = &vt1716S_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1716S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1716S_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x1a, 0, 3, 40); + override_mic_boost(codec, 0x1e, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1716S_capture_mixer; + spec->num_mixers++; + } + + spec->mixers[spec->num_mixers] = vt1716s_dmic_mixer; + spec->num_mixers++; + + spec->mixers[spec->num_mixers++] = vt1716S_mono_out_mixer; + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1716S_loopbacks; +#endif + + return 0; +} + +/* for vt2002P */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt2002P_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt2002P_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/8: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/8 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x37, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3b, AC_VERB_SET_CONNECT_SEL, 0}, + + /* set PW0 index=0 (MW0) */ + {0x24, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0x88}, + { } +}; + + +static struct hda_verb vt2002P_uniwill_init_verbs[] = { + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x26, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt2002P_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt2002P_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt2002P_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x26, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt2002P_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt2002P_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 3, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 3; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 4; + imux->num_items++; + + return 0; +} + +static int vt2002P_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + + err = vt2002P_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt2002P_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt2002P_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt2002P_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt2002P_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt2002P */ +static int patch_vt2002P(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt2002P_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt2002P_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt2002P_uniwill_init_verbs; + + spec->stream_name_analog = "VT2002P Analog"; + spec->stream_analog_playback = &vt2002P_pcm_analog_playback; + spec->stream_analog_capture = &vt2002P_pcm_analog_capture; + + spec->stream_name_digital = "VT2002P Digital"; + spec->stream_digital_playback = &vt2002P_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt2002P_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt2002P_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt2002P_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt2002P_loopbacks; +#endif + + return 0; +} + +/* for vt1812 */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1812_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x10, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x11, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Boost Capture Volume", 0x2b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Boost Capture Volume", 0x29, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + .name = "Input Source", + .count = 2, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1812_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x8, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x9, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Mute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* MUX Indices: Mic = 0 */ + {0x1e, AC_VERB_SET_CONNECT_SEL, 0}, + {0x1f, AC_VERB_SET_CONNECT_SEL, 0}, + + /* PW9 Output enable */ + {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, + + /* Enable Boost Volume backdoor */ + {0x1, 0xfb9, 0x24}, + + /* MW0/1/4/13/15: un-mute index 0 (MUXx), un-mute index 1 (MW9) */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* set MUX0/1/4/13/15 = 0 (AOW0) */ + {0x34, AC_VERB_SET_CONNECT_SEL, 0}, + {0x35, AC_VERB_SET_CONNECT_SEL, 0}, + {0x38, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x3d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Enable AOW0 to MW9 */ + {0x1, 0xfb8, 0xa8}, + { } +}; + + +static struct hda_verb vt1812_uniwill_init_verbs[] = { + {0x33, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x25, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT }, + {0x28, AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | VIA_JACK_EVENT | VIA_BIND_HP_EVENT}, + {0x29, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + {0x2b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_JACK_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1812_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x8, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_pcm_open_close, + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup, + .close = via_pcm_open_close, + }, +}; + +static struct hda_pcm_stream vt1812_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare, + .cleanup = via_dig_playback_pcm_cleanup + }, +}; +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1812_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + if (cfg->line_out_pins[0]) + spec->multiout.dac_nids[0] = 0x8; + return 0; +} + + +/* add playback controls from the parsed DAC table */ +static int vt1812_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + /* Line-Out: PortE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x8, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_BIND_PIN_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x28, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt1812_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x9; + spec->hp_independent_mode_index = 1; + + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL( + spec->multiout.hp_nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1812_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x2b: /* Mic */ + idx = 0; + break; + + case 0x2a: /* Line In */ + idx = 1; + break; + + case 0x29: /* Front Mic */ + idx = 2; + break; + } + err = via_new_analog_input(spec, labels[i], idx, 0x21); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } + /* build volume/mute control of loopback */ + err = via_new_analog_input(spec, "Stereo Mixer", 5, 0x21); + if (err < 0) + return err; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + /* for digital mic select */ + imux->items[imux->num_items].label = "Digital Mic"; + imux->items[imux->num_items].index = 6; + imux->num_items++; + + return 0; +} + +static int vt1812_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + fill_dig_outs(codec); + err = vt1812_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + + if (!spec->autocfg.line_outs && !spec->autocfg.hp_outs) + return 0; /* can't find valid BIOS pin config */ + + err = vt1812_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1812_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1812_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + fill_dig_outs(codec); + + if (spec->kctls.list) + spec->mixers[spec->num_mixers++] = spec->kctls.list; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1812_loopbacks[] = { + { 0x21, HDA_INPUT, 0 }, + { 0x21, HDA_INPUT, 1 }, + { 0x21, HDA_INPUT, 2 }, + { } /* end */ +}; +#endif + + +/* patch for vt1812 */ +static int patch_vt1812(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1812_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + + spec->init_verbs[spec->num_iverbs++] = vt1812_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1812_uniwill_init_verbs; + + spec->stream_name_analog = "VT1812 Analog"; + spec->stream_analog_playback = &vt1812_pcm_analog_playback; + spec->stream_analog_capture = &vt1812_pcm_analog_capture; + + spec->stream_name_digital = "VT1812 Digital"; + spec->stream_digital_playback = &vt1812_pcm_digital_playback; + + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1812_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1812_adc_nids); + get_mux_nids(codec); + override_mic_boost(codec, 0x2b, 0, 3, 40); + override_mic_boost(codec, 0x29, 0, 3, 40); + spec->mixers[spec->num_mixers] = vt1812_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; + +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1812_loopbacks; +#endif return 0; } @@ -3318,6 +6214,23 @@ static struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1702}, { .id = 0x11067398, .name = "VT1702", .patch = patch_vt1702}, + { .id = 0x11060428, .name = "VT1718S", + .patch = patch_vt1718S}, + { .id = 0x11064428, .name = "VT1718S", + .patch = patch_vt1718S}, + { .id = 0x11060441, .name = "VT2020", + .patch = patch_vt1718S}, + { .id = 0x11064441, .name = "VT1828S", + .patch = patch_vt1718S}, + { .id = 0x11060433, .name = "VT1716S", + .patch = patch_vt1716S}, + { .id = 0x1106a721, .name = "VT1716S", + .patch = patch_vt1716S}, + { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P}, + { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812}, + { .id = 0x11060440, .name = "VT1818S", + .patch = patch_vt1708S}, {} /* terminator */ }; diff --git a/sound/pci/ice1712/Makefile b/sound/pci/ice1712/Makefile index 536eae2ccf9..f7ce33f00ea 100644 --- a/sound/pci/ice1712/Makefile +++ b/sound/pci/ice1712/Makefile @@ -5,7 +5,7 @@ snd-ice17xx-ak4xxx-objs := ak4xxx.o snd-ice1712-objs := ice1712.o delta.o hoontech.o ews.o -snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o +snd-ice1724-objs := ice1724.o amp.o revo.o aureon.o vt1720_mobo.o pontis.o prodigy192.o prodigy_hifi.o juli.o phase.o wtm.o se.o maya44.o quartet.o # Toplevel Module Dependency obj-$(CONFIG_SND_ICE1712) += snd-ice1712.o snd-ice17xx-ak4xxx.o diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index d74033a2cfb..c7cff6f8168 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -298,6 +298,16 @@ static void snd_ice1712_set_gpio_dir(struct snd_ice1712 *ice, unsigned int data) inb(ICEREG(ice, DATA)); /* dummy read for pci-posting */ } +static unsigned int snd_ice1712_get_gpio_dir(struct snd_ice1712 *ice) +{ + return snd_ice1712_read(ice, ICE1712_IREG_GPIO_DIRECTION); +} + +static unsigned int snd_ice1712_get_gpio_mask(struct snd_ice1712 *ice) +{ + return snd_ice1712_read(ice, ICE1712_IREG_GPIO_WRITE_MASK); +} + static void snd_ice1712_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) { snd_ice1712_write(ice, ICE1712_IREG_GPIO_WRITE_MASK, data); @@ -2557,7 +2567,9 @@ static int __devinit snd_ice1712_create(struct snd_card *card, mutex_init(&ice->i2c_mutex); mutex_init(&ice->open_mutex); ice->gpio.set_mask = snd_ice1712_set_gpio_mask; + ice->gpio.get_mask = snd_ice1712_get_gpio_mask; ice->gpio.set_dir = snd_ice1712_set_gpio_dir; + ice->gpio.get_dir = snd_ice1712_get_gpio_dir; ice->gpio.set_data = snd_ice1712_set_gpio_data; ice->gpio.get_data = snd_ice1712_get_gpio_data; diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index d063149e704..0da778a69ef 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -359,7 +359,9 @@ struct snd_ice1712 { unsigned int saved[2]; /* for ewx_i2c */ /* operators */ void (*set_mask)(struct snd_ice1712 *ice, unsigned int data); + unsigned int (*get_mask)(struct snd_ice1712 *ice); void (*set_dir)(struct snd_ice1712 *ice, unsigned int data); + unsigned int (*get_dir)(struct snd_ice1712 *ice); void (*set_data)(struct snd_ice1712 *ice, unsigned int data); unsigned int (*get_data)(struct snd_ice1712 *ice); /* misc operators - move to another place? */ @@ -377,8 +379,11 @@ struct snd_ice1712 { unsigned int (*get_rate)(struct snd_ice1712 *ice); void (*set_rate)(struct snd_ice1712 *ice, unsigned int rate); unsigned char (*set_mclk)(struct snd_ice1712 *ice, unsigned int rate); - void (*set_spdif_clock)(struct snd_ice1712 *ice); - + int (*set_spdif_clock)(struct snd_ice1712 *ice, int type); + int (*get_spdif_master_type)(struct snd_ice1712 *ice); + char **ext_clock_names; + int ext_clock_count; + void (*pro_open)(struct snd_ice1712 *, struct snd_pcm_substream *); #ifdef CONFIG_PM int (*pm_suspend)(struct snd_ice1712 *); int (*pm_resume)(struct snd_ice1712 *); @@ -399,6 +404,11 @@ static inline void snd_ice1712_gpio_set_dir(struct snd_ice1712 *ice, unsigned in ice->gpio.set_dir(ice, bits); } +static inline unsigned int snd_ice1712_gpio_get_dir(struct snd_ice1712 *ice) +{ + return ice->gpio.get_dir(ice); +} + static inline void snd_ice1712_gpio_set_mask(struct snd_ice1712 *ice, unsigned int bits) { ice->gpio.set_mask(ice, bits); diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 10fc92c0557..ae29073eea9 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -53,6 +53,7 @@ #include "phase.h" #include "wtm.h" #include "se.h" +#include "quartet.h" MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("VIA ICEnsemble ICE1724/1720 (Envy24HT/PT)"); @@ -70,6 +71,7 @@ MODULE_SUPPORTED_DEVICE("{" PHASE_DEVICE_DESC WTM_DEVICE_DESC SE_DEVICE_DESC + QTET_DEVICE_DESC "{VIA,VT1720}," "{VIA,VT1724}," "{ICEnsemble,Generic ICE1724}," @@ -104,6 +106,8 @@ static int PRO_RATE_LOCKED; static int PRO_RATE_RESET = 1; static unsigned int PRO_RATE_DEFAULT = 44100; +static char *ext_clock_names[1] = { "IEC958 In" }; + /* * Basic I/O */ @@ -118,9 +122,12 @@ static inline int stdclock_is_spdif_master(struct snd_ice1712 *ice) return (inb(ICEMT1724(ice, RATE)) & VT1724_SPDIF_MASTER) ? 1 : 0; } +/* + * locking rate makes sense only for internal clock mode + */ static inline int is_pro_rate_locked(struct snd_ice1712 *ice) { - return ice->is_spdif_master(ice) || PRO_RATE_LOCKED; + return (!ice->is_spdif_master(ice)) && PRO_RATE_LOCKED; } /* @@ -196,6 +203,12 @@ static void snd_vt1724_set_gpio_dir(struct snd_ice1712 *ice, unsigned int data) inw(ICEREG1724(ice, GPIO_DIRECTION)); /* dummy read for pci-posting */ } +/* get gpio direction 0 = read, 1 = write */ +static unsigned int snd_vt1724_get_gpio_dir(struct snd_ice1712 *ice) +{ + return inl(ICEREG1724(ice, GPIO_DIRECTION)); +} + /* set the gpio mask (0 = writable) */ static void snd_vt1724_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) { @@ -205,6 +218,17 @@ static void snd_vt1724_set_gpio_mask(struct snd_ice1712 *ice, unsigned int data) inw(ICEREG1724(ice, GPIO_WRITE_MASK)); /* dummy read for pci-posting */ } +static unsigned int snd_vt1724_get_gpio_mask(struct snd_ice1712 *ice) +{ + unsigned int mask; + if (!ice->vt1720) + mask = (unsigned int)inb(ICEREG1724(ice, GPIO_WRITE_MASK_22)); + else + mask = 0; + mask = (mask << 16) | inw(ICEREG1724(ice, GPIO_WRITE_MASK)); + return mask; +} + static void snd_vt1724_set_gpio_data(struct snd_ice1712 *ice, unsigned int data) { outw(data, ICEREG1724(ice, GPIO_DATA)); @@ -651,16 +675,22 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY; } if (!force && is_pro_rate_locked(ice)) { + /* comparing required and current rate - makes sense for + * internal clock only */ spin_unlock_irqrestore(&ice->reg_lock, flags); return (rate == ice->cur_rate) ? 0 : -EBUSY; } - old_rate = ice->get_rate(ice); - if (force || (old_rate != rate)) - ice->set_rate(ice, rate); - else if (rate == ice->cur_rate) { - spin_unlock_irqrestore(&ice->reg_lock, flags); - return 0; + if (force || !ice->is_spdif_master(ice)) { + /* force means the rate was switched by ucontrol, otherwise + * setting clock rate for internal clock mode */ + old_rate = ice->get_rate(ice); + if (force || (old_rate != rate)) + ice->set_rate(ice, rate); + else if (rate == ice->cur_rate) { + spin_unlock_irqrestore(&ice->reg_lock, flags); + return 0; + } } ice->cur_rate = rate; @@ -1016,6 +1046,8 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->pro_open) + ice->pro_open(ice, substream); return 0; } @@ -1034,6 +1066,8 @@ static int snd_vt1724_capture_pro_open(struct snd_pcm_substream *substream) VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, VT1724_BUFFER_ALIGN); + if (ice->pro_open) + ice->pro_open(ice, substream); return 0; } @@ -1787,15 +1821,21 @@ static int snd_vt1724_pro_internal_clock_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - + int hw_rates_count = ice->hw_rates->count; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = ice->hw_rates->count + 1; + + uinfo->value.enumerated.items = hw_rates_count + ice->ext_clock_count; + /* upper limit - keep at top */ if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; - if (uinfo->value.enumerated.item == uinfo->value.enumerated.items - 1) - strcpy(uinfo->value.enumerated.name, "IEC958 Input"); + if (uinfo->value.enumerated.item >= hw_rates_count) + /* ext_clock items */ + strcpy(uinfo->value.enumerated.name, + ice->ext_clock_names[ + uinfo->value.enumerated.item - hw_rates_count]); else + /* int clock items */ sprintf(uinfo->value.enumerated.name, "%d", ice->hw_rates->list[uinfo->value.enumerated.item]); return 0; @@ -1809,7 +1849,8 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol, spin_lock_irq(&ice->reg_lock); if (ice->is_spdif_master(ice)) { - ucontrol->value.enumerated.item[0] = ice->hw_rates->count; + ucontrol->value.enumerated.item[0] = ice->hw_rates->count + + ice->get_spdif_master_type(ice); } else { rate = ice->get_rate(ice); ucontrol->value.enumerated.item[0] = 0; @@ -1824,8 +1865,14 @@ static int snd_vt1724_pro_internal_clock_get(struct snd_kcontrol *kcontrol, return 0; } +static int stdclock_get_spdif_master_type(struct snd_ice1712 *ice) +{ + /* standard external clock - only single type - SPDIF IN */ + return 0; +} + /* setting clock to external - SPDIF */ -static void stdclock_set_spdif_clock(struct snd_ice1712 *ice) +static int stdclock_set_spdif_clock(struct snd_ice1712 *ice, int type) { unsigned char oval; unsigned char i2s_oval; @@ -1834,27 +1881,30 @@ static void stdclock_set_spdif_clock(struct snd_ice1712 *ice) /* setting 256fs */ i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); outb(i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); + return 0; } + static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int old_rate, new_rate; unsigned int item = ucontrol->value.enumerated.item[0]; - unsigned int spdif = ice->hw_rates->count; + unsigned int first_ext_clock = ice->hw_rates->count; - if (item > spdif) + if (item > first_ext_clock + ice->ext_clock_count - 1) return -EINVAL; + /* if rate = 0 => external clock */ spin_lock_irq(&ice->reg_lock); if (ice->is_spdif_master(ice)) old_rate = 0; else old_rate = ice->get_rate(ice); - if (item == spdif) { - /* switching to external clock via SPDIF */ - ice->set_spdif_clock(ice); + if (item >= first_ext_clock) { + /* switching to external clock */ + ice->set_spdif_clock(ice, item - first_ext_clock); new_rate = 0; } else { /* internal on-card clock */ @@ -1866,7 +1916,7 @@ static int snd_vt1724_pro_internal_clock_put(struct snd_kcontrol *kcontrol, } spin_unlock_irq(&ice->reg_lock); - /* the first reset to the SPDIF master mode? */ + /* the first switch to the ext. clock mode? */ if (old_rate != new_rate && !new_rate) { /* notify akm chips as well */ unsigned int i; @@ -2136,6 +2186,7 @@ static struct snd_ice1712_card_info *card_tables[] __devinitdata = { snd_vt1724_phase_cards, snd_vt1724_wtm_cards, snd_vt1724_se_cards, + snd_vt1724_qtet_cards, NULL, }; @@ -2434,7 +2485,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card, mutex_init(&ice->open_mutex); mutex_init(&ice->i2c_mutex); ice->gpio.set_mask = snd_vt1724_set_gpio_mask; + ice->gpio.get_mask = snd_vt1724_get_gpio_mask; ice->gpio.set_dir = snd_vt1724_set_gpio_dir; + ice->gpio.get_dir = snd_vt1724_get_gpio_dir; ice->gpio.set_data = snd_vt1724_set_gpio_data; ice->gpio.get_data = snd_vt1724_get_gpio_data; ice->card = card; @@ -2522,6 +2575,9 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, return err; } + /* field init before calling chip_init */ + ice->ext_clock_count = 0; + for (tbl = card_tables; *tbl; tbl++) { for (c = *tbl; c->subvendor; c++) { if (c->subvendor == ice->eeprom.subvendor) { @@ -2560,6 +2616,13 @@ __found: ice->set_mclk = stdclock_set_mclk; if (!ice->set_spdif_clock) ice->set_spdif_clock = stdclock_set_spdif_clock; + if (!ice->get_spdif_master_type) + ice->get_spdif_master_type = stdclock_get_spdif_master_type; + if (!ice->ext_clock_names) + ice->ext_clock_names = ext_clock_names; + if (!ice->ext_clock_count) + ice->ext_clock_count = ARRAY_SIZE(ext_clock_names); + if (!ice->hw_rates) set_std_hw_rates(ice); @@ -2719,7 +2782,7 @@ static int snd_vt1724_resume(struct pci_dev *pci) if (ice->pm_saved_is_spdif_master) { /* switching to external clock via SPDIF */ - ice->set_spdif_clock(ice); + ice->set_spdif_clock(ice, 0); } else { /* internal on-card clock */ snd_vt1724_set_pro_rate(ice, ice->pro_rate_default, 1); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index fd948bfd9ae..0c9413d5341 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -412,25 +412,6 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = { }, }; - -static void ak4358_proc_regs_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; - int reg, val; - for (reg = 0; reg <= 0xf; reg++) { - val = snd_akm4xxx_get(ice->akm, 0, reg); - snd_iprintf(buffer, "0x%02x = 0x%02x\n", reg, val); - } -} - -static void ak4358_proc_init(struct snd_ice1712 *ice) -{ - struct snd_info_entry *entry; - if (!snd_card_proc_new(ice->card, "ak4358_codec", &entry)) - snd_info_set_text_ops(entry, ice, ak4358_proc_regs_read); -} - static char *slave_vols[] __devinitdata = { PCM_VOLUME, MONITOR_AN_IN_VOLUME, @@ -496,14 +477,37 @@ static int __devinit juli_add_controls(struct snd_ice1712 *ice) /* only capture SPDIF over AK4114 */ err = snd_ak4114_build(spec->ak4114, NULL, ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - - ak4358_proc_init(ice); if (err < 0) return err; return 0; } /* + * suspend/resume + * */ + +#ifdef CONFIG_PM +static int juli_resume(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + struct juli_spec *spec = ice->spec; + /* akm4358 un-reset, un-mute */ + snd_akm4xxx_reset(ak, 0); + /* reinit ak4114 */ + snd_ak4114_reinit(spec->ak4114); + return 0; +} + +static int juli_suspend(struct snd_ice1712 *ice) +{ + struct snd_akm4xxx *ak = ice->akm; + /* akm4358 reset and soft-mute */ + snd_akm4xxx_reset(ak, 1); + return 0; +} +#endif + +/* * initialize the chip */ @@ -550,13 +554,14 @@ static inline unsigned char juli_set_mclk(struct snd_ice1712 *ice, } /* setting clock to external - SPDIF */ -static void juli_set_spdif_clock(struct snd_ice1712 *ice) +static int juli_set_spdif_clock(struct snd_ice1712 *ice, int type) { unsigned int old; old = ice->gpio.get_data(ice); /* external clock (= 0), multiply 1x, 48kHz */ ice->gpio.set_data(ice, (old & ~GPIO_RATE_MASK) | GPIO_MULTI_1X | GPIO_FREQ_48KHZ); + return 0; } /* Called when ak4114 detects change in the input SPDIF stream */ @@ -646,6 +651,13 @@ static int __devinit juli_init(struct snd_ice1712 *ice) ice->set_spdif_clock = juli_set_spdif_clock; ice->spdif.ops.open = juli_spdif_in_open; + +#ifdef CONFIG_PM + ice->pm_resume = juli_resume; + ice->pm_suspend = juli_suspend; + ice->pm_suspend_enabled = 1; +#endif + return 0; } diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c new file mode 100644 index 00000000000..1948632787e --- /dev/null +++ b/sound/pci/ice1712/quartet.c @@ -0,0 +1,1130 @@ +/* + * ALSA driver for ICEnsemble VT1724 (Envy24HT) + * + * Lowlevel functions for Infrasonic Quartet + * + * Copyright (c) 2009 Pavel Hofman <pavel.hofman@ivitera.com> + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <asm/io.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/tlv.h> +#include <sound/info.h> + +#include "ice1712.h" +#include "envy24ht.h" +#include <sound/ak4113.h> +#include "quartet.h" + +struct qtet_spec { + struct ak4113 *ak4113; + unsigned int scr; /* system control register */ + unsigned int mcr; /* monitoring control register */ + unsigned int cpld; /* cpld register */ +}; + +struct qtet_kcontrol_private { + unsigned int bit; + void (*set_register)(struct snd_ice1712 *ice, unsigned int val); + unsigned int (*get_register)(struct snd_ice1712 *ice); + unsigned char *texts[2]; +}; + +enum { + IN12_SEL = 0, + IN34_SEL, + AIN34_SEL, + COAX_OUT, + IN12_MON12, + IN12_MON34, + IN34_MON12, + IN34_MON34, + OUT12_MON34, + OUT34_MON12, +}; + +static char *ext_clock_names[3] = {"IEC958 In", "Word Clock 1xFS", + "Word Clock 256xFS"}; + +/* chip address on I2C bus */ +#define AK4113_ADDR 0x26 /* S/PDIF receiver */ + +/* chip address on SPI bus */ +#define AK4620_ADDR 0x02 /* ADC/DAC */ + + +/* + * GPIO pins + */ + +/* GPIO0 - O - DATA0, def. 0 */ +#define GPIO_D0 (1<<0) +/* GPIO1 - I/O - DATA1, Jack Detect Input0 (0:present, 1:missing), def. 1 */ +#define GPIO_D1_JACKDTC0 (1<<1) +/* GPIO2 - I/O - DATA2, Jack Detect Input1 (0:present, 1:missing), def. 1 */ +#define GPIO_D2_JACKDTC1 (1<<2) +/* GPIO3 - I/O - DATA3, def. 1 */ +#define GPIO_D3 (1<<3) +/* GPIO4 - I/O - DATA4, SPI CDTO, def. 1 */ +#define GPIO_D4_SPI_CDTO (1<<4) +/* GPIO5 - I/O - DATA5, SPI CCLK, def. 1 */ +#define GPIO_D5_SPI_CCLK (1<<5) +/* GPIO6 - I/O - DATA6, Cable Detect Input (0:detected, 1:not detected */ +#define GPIO_D6_CD (1<<6) +/* GPIO7 - I/O - DATA7, Device Detect Input (0:detected, 1:not detected */ +#define GPIO_D7_DD (1<<7) +/* GPIO8 - O - CPLD Chip Select, def. 1 */ +#define GPIO_CPLD_CSN (1<<8) +/* GPIO9 - O - CPLD register read/write (0:write, 1:read), def. 0 */ +#define GPIO_CPLD_RW (1<<9) +/* GPIO10 - O - SPI Chip Select for CODEC#0, def. 1 */ +#define GPIO_SPI_CSN0 (1<<10) +/* GPIO11 - O - SPI Chip Select for CODEC#1, def. 1 */ +#define GPIO_SPI_CSN1 (1<<11) +/* GPIO12 - O - Ex. Register Output Enable (0:enable, 1:disable), def. 1, + * init 0 */ +#define GPIO_EX_GPIOE (1<<12) +/* GPIO13 - O - Ex. Register0 Chip Select for System Control Register, + * def. 1 */ +#define GPIO_SCR (1<<13) +/* GPIO14 - O - Ex. Register1 Chip Select for Monitor Control Register, + * def. 1 */ +#define GPIO_MCR (1<<14) + +#define GPIO_SPI_ALL (GPIO_D4_SPI_CDTO | GPIO_D5_SPI_CCLK |\ + GPIO_SPI_CSN0 | GPIO_SPI_CSN1) + +#define GPIO_DATA_MASK (GPIO_D0 | GPIO_D1_JACKDTC0 | \ + GPIO_D2_JACKDTC1 | GPIO_D3 | \ + GPIO_D4_SPI_CDTO | GPIO_D5_SPI_CCLK | \ + GPIO_D6_CD | GPIO_D7_DD) + +/* System Control Register GPIO_SCR data bits */ +/* Mic/Line select relay (0:line, 1:mic) */ +#define SCR_RELAY GPIO_D0 +/* Phantom power drive control (0:5V, 1:48V) */ +#define SCR_PHP_V GPIO_D1_JACKDTC0 +/* H/W mute control (0:Normal, 1:Mute) */ +#define SCR_MUTE GPIO_D2_JACKDTC1 +/* Phantom power control (0:Phantom on, 1:off) */ +#define SCR_PHP GPIO_D3 +/* Analog input 1/2 Source Select */ +#define SCR_AIN12_SEL0 GPIO_D4_SPI_CDTO +#define SCR_AIN12_SEL1 GPIO_D5_SPI_CCLK +/* Analog input 3/4 Source Select (0:line, 1:hi-z) */ +#define SCR_AIN34_SEL GPIO_D6_CD +/* Codec Power Down (0:power down, 1:normal) */ +#define SCR_CODEC_PDN GPIO_D7_DD + +#define SCR_AIN12_LINE (0) +#define SCR_AIN12_MIC (SCR_AIN12_SEL0) +#define SCR_AIN12_LOWCUT (SCR_AIN12_SEL1 | SCR_AIN12_SEL0) + +/* Monitor Control Register GPIO_MCR data bits */ +/* Input 1/2 to Monitor 1/2 (0:off, 1:on) */ +#define MCR_IN12_MON12 GPIO_D0 +/* Input 1/2 to Monitor 3/4 (0:off, 1:on) */ +#define MCR_IN12_MON34 GPIO_D1_JACKDTC0 +/* Input 3/4 to Monitor 1/2 (0:off, 1:on) */ +#define MCR_IN34_MON12 GPIO_D2_JACKDTC1 +/* Input 3/4 to Monitor 3/4 (0:off, 1:on) */ +#define MCR_IN34_MON34 GPIO_D3 +/* Output to Monitor 1/2 (0:off, 1:on) */ +#define MCR_OUT34_MON12 GPIO_D4_SPI_CDTO +/* Output to Monitor 3/4 (0:off, 1:on) */ +#define MCR_OUT12_MON34 GPIO_D5_SPI_CCLK + +/* CPLD Register DATA bits */ +/* Clock Rate Select */ +#define CPLD_CKS0 GPIO_D0 +#define CPLD_CKS1 GPIO_D1_JACKDTC0 +#define CPLD_CKS2 GPIO_D2_JACKDTC1 +/* Sync Source Select (0:Internal, 1:External) */ +#define CPLD_SYNC_SEL GPIO_D3 +/* Word Clock FS Select (0:FS, 1:256FS) */ +#define CPLD_WORD_SEL GPIO_D4_SPI_CDTO +/* Coaxial Output Source (IS-Link) (0:SPDIF, 1:I2S) */ +#define CPLD_COAX_OUT GPIO_D5_SPI_CCLK +/* Input 1/2 Source Select (0:Analog12, 1:An34) */ +#define CPLD_IN12_SEL GPIO_D6_CD +/* Input 3/4 Source Select (0:Analog34, 1:Digital In) */ +#define CPLD_IN34_SEL GPIO_D7_DD + +/* internal clock (CPLD_SYNC_SEL = 0) options */ +#define CPLD_CKS_44100HZ (0) +#define CPLD_CKS_48000HZ (CPLD_CKS0) +#define CPLD_CKS_88200HZ (CPLD_CKS1) +#define CPLD_CKS_96000HZ (CPLD_CKS1 | CPLD_CKS0) +#define CPLD_CKS_176400HZ (CPLD_CKS2) +#define CPLD_CKS_192000HZ (CPLD_CKS2 | CPLD_CKS0) + +#define CPLD_CKS_MASK (CPLD_CKS0 | CPLD_CKS1 | CPLD_CKS2) + +/* external clock (CPLD_SYNC_SEL = 1) options */ +/* external clock - SPDIF */ +#define CPLD_EXT_SPDIF (0 | CPLD_SYNC_SEL) +/* external clock - WordClock 1xfs */ +#define CPLD_EXT_WORDCLOCK_1FS (CPLD_CKS1 | CPLD_SYNC_SEL) +/* external clock - WordClock 256xfs */ +#define CPLD_EXT_WORDCLOCK_256FS (CPLD_CKS1 | CPLD_WORD_SEL |\ + CPLD_SYNC_SEL) + +#define EXT_SPDIF_TYPE 0 +#define EXT_WORDCLOCK_1FS_TYPE 1 +#define EXT_WORDCLOCK_256FS_TYPE 2 + +#define AK4620_DFS0 (1<<0) +#define AK4620_DFS1 (1<<1) +#define AK4620_CKS0 (1<<2) +#define AK4620_CKS1 (1<<3) +/* Clock and Format Control register */ +#define AK4620_DFS_REG 0x02 + +/* Deem and Volume Control register */ +#define AK4620_DEEMVOL_REG 0x03 +#define AK4620_SMUTE (1<<7) + +/* + * Conversion from int value to its binary form. Used for debugging. + * The output buffer must be allocated prior to calling the function. + */ +static char *get_binary(char *buffer, int value) +{ + int i, j, pos; + pos = 0; + for (i = 0; i < 4; ++i) { + for (j = 0; j < 8; ++j) { + if (value & (1 << (31-(i*8 + j)))) + buffer[pos] = '1'; + else + buffer[pos] = '0'; + pos++; + } + if (i < 3) { + buffer[pos] = ' '; + pos++; + } + } + buffer[pos] = '\0'; + return buffer; +} + +/* + * Initial setup of the conversion array GPIO <-> rate + */ +static unsigned int qtet_rates[] = { + 44100, 48000, 88200, + 96000, 176400, 192000, +}; + +static unsigned int cks_vals[] = { + CPLD_CKS_44100HZ, CPLD_CKS_48000HZ, CPLD_CKS_88200HZ, + CPLD_CKS_96000HZ, CPLD_CKS_176400HZ, CPLD_CKS_192000HZ, +}; + +static struct snd_pcm_hw_constraint_list qtet_rates_info = { + .count = ARRAY_SIZE(qtet_rates), + .list = qtet_rates, + .mask = 0, +}; + +static void qtet_ak4113_write(void *private_data, unsigned char reg, + unsigned char val) +{ + snd_vt1724_write_i2c((struct snd_ice1712 *)private_data, AK4113_ADDR, + reg, val); +} + +static unsigned char qtet_ak4113_read(void *private_data, unsigned char reg) +{ + return snd_vt1724_read_i2c((struct snd_ice1712 *)private_data, + AK4113_ADDR, reg); +} + + +/* + * AK4620 section + */ + +/* + * Write data to addr register of ak4620 + */ +static void qtet_akm_write(struct snd_akm4xxx *ak, int chip, + unsigned char addr, unsigned char data) +{ + unsigned int tmp, orig_dir; + int idx; + unsigned int addrdata; + struct snd_ice1712 *ice = ak->private_data[0]; + + if (snd_BUG_ON(chip < 0 || chip >= 4)) + return; + /*printk(KERN_DEBUG "Writing to AK4620: chip=%d, addr=0x%x, + data=0x%x\n", chip, addr, data);*/ + orig_dir = ice->gpio.get_dir(ice); + ice->gpio.set_dir(ice, orig_dir | GPIO_SPI_ALL); + /* set mask - only SPI bits */ + ice->gpio.set_mask(ice, ~GPIO_SPI_ALL); + + tmp = ice->gpio.get_data(ice); + /* high all */ + tmp |= GPIO_SPI_ALL; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* drop chip select */ + if (chip) + /* CODEC 1 */ + tmp &= ~GPIO_SPI_CSN1; + else + tmp &= ~GPIO_SPI_CSN0; + ice->gpio.set_data(ice, tmp); + udelay(100); + + /* build I2C address + data byte */ + addrdata = (AK4620_ADDR << 6) | 0x20 | (addr & 0x1f); + addrdata = (addrdata << 8) | data; + for (idx = 15; idx >= 0; idx--) { + /* drop clock */ + tmp &= ~GPIO_D5_SPI_CCLK; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* set data */ + if (addrdata & (1 << idx)) + tmp |= GPIO_D4_SPI_CDTO; + else + tmp &= ~GPIO_D4_SPI_CDTO; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* raise clock */ + tmp |= GPIO_D5_SPI_CCLK; + ice->gpio.set_data(ice, tmp); + udelay(100); + } + /* all back to 1 */ + tmp |= GPIO_SPI_ALL; + ice->gpio.set_data(ice, tmp); + udelay(100); + + /* return all gpios to non-writable */ + ice->gpio.set_mask(ice, 0xffffff); + /* restore GPIOs direction */ + ice->gpio.set_dir(ice, orig_dir); +} + +static void qtet_akm_set_regs(struct snd_akm4xxx *ak, unsigned char addr, + unsigned char mask, unsigned char value) +{ + unsigned char tmp; + int chip; + for (chip = 0; chip < ak->num_chips; chip++) { + tmp = snd_akm4xxx_get(ak, chip, addr); + /* clear the bits */ + tmp &= ~mask; + /* set the new bits */ + tmp |= value; + snd_akm4xxx_write(ak, chip, addr, tmp); + } +} + +/* + * change the rate of AK4620 + */ +static void qtet_akm_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) +{ + unsigned char ak4620_dfs; + + if (rate == 0) /* no hint - S/PDIF input is master or the new spdif + input rate undetected, simply return */ + return; + + /* adjust DFS on codecs - see datasheet */ + if (rate > 108000) + ak4620_dfs = AK4620_DFS1 | AK4620_CKS1; + else if (rate > 54000) + ak4620_dfs = AK4620_DFS0 | AK4620_CKS0; + else + ak4620_dfs = 0; + + /* set new value */ + qtet_akm_set_regs(ak, AK4620_DFS_REG, AK4620_DFS0 | AK4620_DFS1 | + AK4620_CKS0 | AK4620_CKS1, ak4620_dfs); +} + +#define AK_CONTROL(xname, xch) { .name = xname, .num_channels = xch } + +#define PCM_12_PLAYBACK_VOLUME "PCM 1/2 Playback Volume" +#define PCM_34_PLAYBACK_VOLUME "PCM 3/4 Playback Volume" +#define PCM_12_CAPTURE_VOLUME "PCM 1/2 Capture Volume" +#define PCM_34_CAPTURE_VOLUME "PCM 3/4 Capture Volume" + +static const struct snd_akm4xxx_dac_channel qtet_dac[] = { + AK_CONTROL(PCM_12_PLAYBACK_VOLUME, 2), + AK_CONTROL(PCM_34_PLAYBACK_VOLUME, 2), +}; + +static const struct snd_akm4xxx_adc_channel qtet_adc[] = { + AK_CONTROL(PCM_12_CAPTURE_VOLUME, 2), + AK_CONTROL(PCM_34_CAPTURE_VOLUME, 2), +}; + +static struct snd_akm4xxx akm_qtet_dac __devinitdata = { + .type = SND_AK4620, + .num_dacs = 4, /* DAC1 - Output 12 + */ + .num_adcs = 4, /* ADC1 - Input 12 + */ + .ops = { + .write = qtet_akm_write, + .set_rate_val = qtet_akm_set_rate_val, + }, + .dac_info = qtet_dac, + .adc_info = qtet_adc, +}; + +/* Communication routines with the CPLD */ + + +/* Writes data to external register reg, both reg and data are + * GPIO representations */ +static void reg_write(struct snd_ice1712 *ice, unsigned int reg, + unsigned int data) +{ + unsigned int tmp; + + mutex_lock(&ice->gpio_mutex); + /* set direction of used GPIOs*/ + /* all outputs */ + tmp = 0x00ffff; + ice->gpio.set_dir(ice, tmp); + /* mask - writable bits */ + ice->gpio.set_mask(ice, ~(tmp)); + /* write the data */ + tmp = ice->gpio.get_data(ice); + tmp &= ~GPIO_DATA_MASK; + tmp |= data; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* drop output enable */ + tmp &= ~GPIO_EX_GPIOE; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* drop the register gpio */ + tmp &= ~reg; + ice->gpio.set_data(ice, tmp); + udelay(100); + /* raise the register GPIO */ + tmp |= reg; + ice->gpio.set_data(ice, tmp); + udelay(100); + + /* raise all data gpios */ + tmp |= GPIO_DATA_MASK; + ice->gpio.set_data(ice, tmp); + /* mask - immutable bits */ + ice->gpio.set_mask(ice, 0xffffff); + /* outputs only 8-15 */ + ice->gpio.set_dir(ice, 0x00ff00); + mutex_unlock(&ice->gpio_mutex); +} + +static unsigned int get_scr(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + return spec->scr; +} + +static unsigned int get_mcr(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + return spec->mcr; +} + +static unsigned int get_cpld(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + return spec->cpld; +} + +static void set_scr(struct snd_ice1712 *ice, unsigned int val) +{ + struct qtet_spec *spec = ice->spec; + reg_write(ice, GPIO_SCR, val); + spec->scr = val; +} + +static void set_mcr(struct snd_ice1712 *ice, unsigned int val) +{ + struct qtet_spec *spec = ice->spec; + reg_write(ice, GPIO_MCR, val); + spec->mcr = val; +} + +static void set_cpld(struct snd_ice1712 *ice, unsigned int val) +{ + struct qtet_spec *spec = ice->spec; + reg_write(ice, GPIO_CPLD_CSN, val); + spec->cpld = val; +} +#ifdef CONFIG_PROC_FS +static void proc_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ice1712 *ice = entry->private_data; + char bin_buffer[36]; + + snd_iprintf(buffer, "SCR: %s\n", get_binary(bin_buffer, + get_scr(ice))); + snd_iprintf(buffer, "MCR: %s\n", get_binary(bin_buffer, + get_mcr(ice))); + snd_iprintf(buffer, "CPLD: %s\n", get_binary(bin_buffer, + get_cpld(ice))); +} + +static void proc_init(struct snd_ice1712 *ice) +{ + struct snd_info_entry *entry; + if (!snd_card_proc_new(ice->card, "quartet", &entry)) + snd_info_set_text_ops(entry, ice, proc_regs_read); +} +#else /* !CONFIG_PROC_FS */ +static void proc_init(struct snd_ice1712 *ice) {} +#endif + +static int qtet_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val; + val = get_scr(ice) & SCR_MUTE; + ucontrol->value.integer.value[0] = (val) ? 0 : 1; + return 0; +} + +static int qtet_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new, smute; + old = get_scr(ice) & SCR_MUTE; + if (ucontrol->value.integer.value[0]) { + /* unmute */ + new = 0; + /* un-smuting DAC */ + smute = 0; + } else { + /* mute */ + new = SCR_MUTE; + /* smuting DAC */ + smute = AK4620_SMUTE; + } + if (old != new) { + struct snd_akm4xxx *ak = ice->akm; + set_scr(ice, (get_scr(ice) & ~SCR_MUTE) | new); + /* set smute */ + qtet_akm_set_regs(ak, AK4620_DEEMVOL_REG, AK4620_SMUTE, smute); + return 1; + } + /* no change */ + return 0; +} + +static int qtet_ain12_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[3] = {"Line In 1/2", "Mic", "Mic + Low-cut"}; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = ARRAY_SIZE(texts); + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int qtet_ain12_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val, result; + val = get_scr(ice) & (SCR_AIN12_SEL1 | SCR_AIN12_SEL0); + switch (val) { + case SCR_AIN12_LINE: + result = 0; + break; + case SCR_AIN12_MIC: + result = 1; + break; + case SCR_AIN12_LOWCUT: + result = 2; + break; + default: + /* BUG - no other combinations allowed */ + snd_BUG(); + result = 0; + } + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int qtet_ain12_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new, tmp, masked_old; + old = new = get_scr(ice); + masked_old = old & (SCR_AIN12_SEL1 | SCR_AIN12_SEL0); + tmp = ucontrol->value.integer.value[0]; + if (tmp == 2) + tmp = 3; /* binary 10 is not supported */ + tmp <<= 4; /* shifting to SCR_AIN12_SEL0 */ + if (tmp != masked_old) { + /* change requested */ + switch (tmp) { + case SCR_AIN12_LINE: + new = old & ~(SCR_AIN12_SEL1 | SCR_AIN12_SEL0); + set_scr(ice, new); + /* turn off relay */ + new &= ~SCR_RELAY; + set_scr(ice, new); + break; + case SCR_AIN12_MIC: + /* turn on relay */ + new = old | SCR_RELAY; + set_scr(ice, new); + new = (new & ~SCR_AIN12_SEL1) | SCR_AIN12_SEL0; + set_scr(ice, new); + break; + case SCR_AIN12_LOWCUT: + /* turn on relay */ + new = old | SCR_RELAY; + set_scr(ice, new); + new |= SCR_AIN12_SEL1 | SCR_AIN12_SEL0; + set_scr(ice, new); + break; + default: + snd_BUG(); + } + return 1; + } + /* no change */ + return 0; +} + +static int qtet_php_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int val; + /* if phantom voltage =48V, phantom on */ + val = get_scr(ice) & SCR_PHP_V; + ucontrol->value.integer.value[0] = val ? 1 : 0; + return 0; +} + +static int qtet_php_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new; + old = new = get_scr(ice); + if (ucontrol->value.integer.value[0] /* phantom on requested */ + && (~old & SCR_PHP_V)) /* 0 = voltage 5V */ { + /* is off, turn on */ + /* turn voltage on first, = 1 */ + new = old | SCR_PHP_V; + set_scr(ice, new); + /* turn phantom on, = 0 */ + new &= ~SCR_PHP; + set_scr(ice, new); + } else if (!ucontrol->value.integer.value[0] && (old & SCR_PHP_V)) { + /* phantom off requested and 1 = voltage 48V */ + /* is on, turn off */ + /* turn voltage off first, = 0 */ + new = old & ~SCR_PHP_V; + set_scr(ice, new); + /* turn phantom off, = 1 */ + new |= SCR_PHP; + set_scr(ice, new); + } + if (old != new) + return 1; + /* no change */ + return 0; +} + +#define PRIV_SW(xid, xbit, xreg) [xid] = {.bit = xbit,\ + .set_register = set_##xreg,\ + .get_register = get_##xreg, } + + +#define PRIV_ENUM2(xid, xbit, xreg, xtext1, xtext2) [xid] = {.bit = xbit,\ + .set_register = set_##xreg,\ + .get_register = get_##xreg,\ + .texts = {xtext1, xtext2} } + +static struct qtet_kcontrol_private qtet_privates[] = { + PRIV_ENUM2(IN12_SEL, CPLD_IN12_SEL, cpld, "An In 1/2", "An In 3/4"), + PRIV_ENUM2(IN34_SEL, CPLD_IN34_SEL, cpld, "An In 3/4", "IEC958 In"), + PRIV_ENUM2(AIN34_SEL, SCR_AIN34_SEL, scr, "Line In 3/4", "Hi-Z"), + PRIV_ENUM2(COAX_OUT, CPLD_COAX_OUT, cpld, "IEC958", "I2S"), + PRIV_SW(IN12_MON12, MCR_IN12_MON12, mcr), + PRIV_SW(IN12_MON34, MCR_IN12_MON34, mcr), + PRIV_SW(IN34_MON12, MCR_IN34_MON12, mcr), + PRIV_SW(IN34_MON34, MCR_IN34_MON34, mcr), + PRIV_SW(OUT12_MON34, MCR_OUT12_MON34, mcr), + PRIV_SW(OUT34_MON12, MCR_OUT34_MON12, mcr), +}; + +static int qtet_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct qtet_kcontrol_private private = + qtet_privates[kcontrol->private_value]; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = ARRAY_SIZE(private.texts); + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + private.texts[uinfo->value.enumerated.item]); + + return 0; +} + +static int qtet_sw_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct qtet_kcontrol_private private = + qtet_privates[kcontrol->private_value]; + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = + (private.get_register(ice) & private.bit) ? 1 : 0; + return 0; +} + +static int qtet_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct qtet_kcontrol_private private = + qtet_privates[kcontrol->private_value]; + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + unsigned int old, new; + old = private.get_register(ice); + if (ucontrol->value.integer.value[0]) + new = old | private.bit; + else + new = old & ~private.bit; + if (old != new) { + private.set_register(ice, new); + return 1; + } + /* no change */ + return 0; +} + +#define qtet_sw_info snd_ctl_boolean_mono_info + +#define QTET_CONTROL(xname, xtype, xpriv) \ + {.iface = SNDRV_CTL_ELEM_IFACE_MIXER,\ + .name = xname,\ + .info = qtet_##xtype##_info,\ + .get = qtet_sw_get,\ + .put = qtet_sw_put,\ + .private_value = xpriv } + +static struct snd_kcontrol_new qtet_controls[] __devinitdata = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = qtet_sw_info, + .get = qtet_mute_get, + .put = qtet_mute_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Phantom Power", + .info = qtet_sw_info, + .get = qtet_php_get, + .put = qtet_php_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog In 1/2 Capture Switch", + .info = qtet_ain12_enum_info, + .get = qtet_ain12_sw_get, + .put = qtet_ain12_sw_put, + .private_value = 0 + }, + QTET_CONTROL("Analog In 3/4 Capture Switch", enum, AIN34_SEL), + QTET_CONTROL("PCM In 1/2 Capture Switch", enum, IN12_SEL), + QTET_CONTROL("PCM In 3/4 Capture Switch", enum, IN34_SEL), + QTET_CONTROL("Coax Output Source", enum, COAX_OUT), + QTET_CONTROL("Analog In 1/2 to Monitor 1/2", sw, IN12_MON12), + QTET_CONTROL("Analog In 1/2 to Monitor 3/4", sw, IN12_MON34), + QTET_CONTROL("Analog In 3/4 to Monitor 1/2", sw, IN34_MON12), + QTET_CONTROL("Analog In 3/4 to Monitor 3/4", sw, IN34_MON34), + QTET_CONTROL("Output 1/2 to Monitor 3/4", sw, OUT12_MON34), + QTET_CONTROL("Output 3/4 to Monitor 1/2", sw, OUT34_MON12), +}; + +static char *slave_vols[] __devinitdata = { + PCM_12_PLAYBACK_VOLUME, + PCM_34_PLAYBACK_VOLUME, + NULL +}; + +static __devinitdata +DECLARE_TLV_DB_SCALE(qtet_master_db_scale, -6350, 50, 1); + +static struct snd_kcontrol __devinit *ctl_find(struct snd_card *card, + const char *name) +{ + struct snd_ctl_elem_id sid; + memset(&sid, 0, sizeof(sid)); + /* FIXME: strcpy is bad. */ + strcpy(sid.name, name); + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_find_id(card, &sid); +} + +static void __devinit add_slaves(struct snd_card *card, + struct snd_kcontrol *master, char **list) +{ + for (; *list; list++) { + struct snd_kcontrol *slave = ctl_find(card, *list); + if (slave) + snd_ctl_add_slave(master, slave); + } +} + +static int __devinit qtet_add_controls(struct snd_ice1712 *ice) +{ + struct qtet_spec *spec = ice->spec; + int err, i; + struct snd_kcontrol *vmaster; + err = snd_ice1712_akm4xxx_build_controls(ice); + if (err < 0) + return err; + for (i = 0; i < ARRAY_SIZE(qtet_controls); i++) { + err = snd_ctl_add(ice->card, + snd_ctl_new1(&qtet_controls[i], ice)); + if (err < 0) + return err; + } + + /* Create virtual master control */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", + qtet_master_db_scale); + if (!vmaster) + return -ENOMEM; + add_slaves(ice->card, vmaster, slave_vols); + err = snd_ctl_add(ice->card, vmaster); + if (err < 0) + return err; + /* only capture SPDIF over AK4113 */ + err = snd_ak4113_build(spec->ak4113, + ice->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); + if (err < 0) + return err; + return 0; +} + +static inline int qtet_is_spdif_master(struct snd_ice1712 *ice) +{ + /* CPLD_SYNC_SEL: 0 = internal, 1 = external (i.e. spdif master) */ + return (get_cpld(ice) & CPLD_SYNC_SEL) ? 1 : 0; +} + +static unsigned int qtet_get_rate(struct snd_ice1712 *ice) +{ + int i; + unsigned char result; + + result = get_cpld(ice) & CPLD_CKS_MASK; + for (i = 0; i < ARRAY_SIZE(cks_vals); i++) + if (cks_vals[i] == result) + return qtet_rates[i]; + return 0; +} + +static int get_cks_val(int rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(qtet_rates); i++) + if (qtet_rates[i] == rate) + return cks_vals[i]; + return 0; +} + +/* setting new rate */ +static void qtet_set_rate(struct snd_ice1712 *ice, unsigned int rate) +{ + unsigned int new; + unsigned char val; + /* switching ice1724 to external clock - supplied by ext. circuits */ + val = inb(ICEMT1724(ice, RATE)); + outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); + + new = (get_cpld(ice) & ~CPLD_CKS_MASK) | get_cks_val(rate); + /* switch to internal clock, drop CPLD_SYNC_SEL */ + new &= ~CPLD_SYNC_SEL; + /* printk(KERN_DEBUG "QT - set_rate: old %x, new %x\n", + get_cpld(ice), new); */ + set_cpld(ice, new); +} + +static inline unsigned char qtet_set_mclk(struct snd_ice1712 *ice, + unsigned int rate) +{ + /* no change in master clock */ + return 0; +} + +/* setting clock to external - SPDIF */ +static int qtet_set_spdif_clock(struct snd_ice1712 *ice, int type) +{ + unsigned int old, new; + + old = new = get_cpld(ice); + new &= ~(CPLD_CKS_MASK | CPLD_WORD_SEL); + switch (type) { + case EXT_SPDIF_TYPE: + new |= CPLD_EXT_SPDIF; + break; + case EXT_WORDCLOCK_1FS_TYPE: + new |= CPLD_EXT_WORDCLOCK_1FS; + break; + case EXT_WORDCLOCK_256FS_TYPE: + new |= CPLD_EXT_WORDCLOCK_256FS; + break; + default: + snd_BUG(); + } + if (old != new) { + set_cpld(ice, new); + /* changed */ + return 1; + } + return 0; +} + +static int qtet_get_spdif_master_type(struct snd_ice1712 *ice) +{ + unsigned int val; + int result; + val = get_cpld(ice); + /* checking only rate/clock-related bits */ + val &= (CPLD_CKS_MASK | CPLD_WORD_SEL | CPLD_SYNC_SEL); + if (!(val & CPLD_SYNC_SEL)) { + /* switched to internal clock, is not any external type */ + result = -1; + } else { + switch (val) { + case (CPLD_EXT_SPDIF): + result = EXT_SPDIF_TYPE; + break; + case (CPLD_EXT_WORDCLOCK_1FS): + result = EXT_WORDCLOCK_1FS_TYPE; + break; + case (CPLD_EXT_WORDCLOCK_256FS): + result = EXT_WORDCLOCK_256FS_TYPE; + break; + default: + /* undefined combination of external clock setup */ + snd_BUG(); + result = 0; + } + } + return result; +} + +/* Called when ak4113 detects change in the input SPDIF stream */ +static void qtet_ak4113_change(struct ak4113 *ak4113, unsigned char c0, + unsigned char c1) +{ + struct snd_ice1712 *ice = ak4113->change_callback_private; + int rate; + if ((qtet_get_spdif_master_type(ice) == EXT_SPDIF_TYPE) && + c1) { + /* only for SPDIF master mode, rate was changed */ + rate = snd_ak4113_external_rate(ak4113); + /* printk(KERN_DEBUG "ak4113 - input rate changed to %d\n", + rate); */ + qtet_akm_set_rate_val(ice->akm, rate); + } +} + +/* + * If clock slaved to SPDIF-IN, setting runtime rate + * to the detected external rate + */ +static void qtet_spdif_in_open(struct snd_ice1712 *ice, + struct snd_pcm_substream *substream) +{ + struct qtet_spec *spec = ice->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int rate; + + if (qtet_get_spdif_master_type(ice) != EXT_SPDIF_TYPE) + /* not external SPDIF, no rate limitation */ + return; + /* only external SPDIF can detect incoming sample rate */ + rate = snd_ak4113_external_rate(spec->ak4113); + if (rate >= runtime->hw.rate_min && rate <= runtime->hw.rate_max) { + runtime->hw.rate_min = rate; + runtime->hw.rate_max = rate; + } +} + +/* + * initialize the chip + */ +static int __devinit qtet_init(struct snd_ice1712 *ice) +{ + static const unsigned char ak4113_init_vals[] = { + /* AK4113_REG_PWRDN */ AK4113_RST | AK4113_PWN | + AK4113_OCKS0 | AK4113_OCKS1, + /* AK4113_REQ_FORMAT */ AK4113_DIF_I24I2S | AK4113_VTX | + AK4113_DEM_OFF | AK4113_DEAU, + /* AK4113_REG_IO0 */ AK4113_OPS2 | AK4113_TXE | + AK4113_XTL_24_576M, + /* AK4113_REG_IO1 */ AK4113_EFH_1024LRCLK | AK4113_IPS(0), + /* AK4113_REG_INT0_MASK */ 0, + /* AK4113_REG_INT1_MASK */ 0, + /* AK4113_REG_DATDTS */ 0, + }; + int err; + struct qtet_spec *spec; + struct snd_akm4xxx *ak; + unsigned char val; + + /* switching ice1724 to external clock - supplied by ext. circuits */ + val = inb(ICEMT1724(ice, RATE)); + outb(val | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + /* qtet is clocked by Xilinx array */ + ice->hw_rates = &qtet_rates_info; + ice->is_spdif_master = qtet_is_spdif_master; + ice->get_rate = qtet_get_rate; + ice->set_rate = qtet_set_rate; + ice->set_mclk = qtet_set_mclk; + ice->set_spdif_clock = qtet_set_spdif_clock; + ice->get_spdif_master_type = qtet_get_spdif_master_type; + ice->ext_clock_names = ext_clock_names; + ice->ext_clock_count = ARRAY_SIZE(ext_clock_names); + /* since Qtet can detect correct SPDIF-in rate, all streams can be + * limited to this specific rate */ + ice->spdif.ops.open = ice->pro_open = qtet_spdif_in_open; + ice->spec = spec; + + /* Mute Off */ + /* SCR Initialize*/ + /* keep codec power down first */ + set_scr(ice, SCR_PHP); + udelay(1); + /* codec power up */ + set_scr(ice, SCR_PHP | SCR_CODEC_PDN); + + /* MCR Initialize */ + set_mcr(ice, 0); + + /* CPLD Initialize */ + set_cpld(ice, 0); + + + ice->num_total_dacs = 2; + ice->num_total_adcs = 2; + + ice->akm = kcalloc(2, sizeof(struct snd_akm4xxx), GFP_KERNEL); + ak = ice->akm; + if (!ak) + return -ENOMEM; + /* only one codec with two chips */ + ice->akm_codecs = 1; + err = snd_ice1712_akm4xxx_init(ak, &akm_qtet_dac, NULL, ice); + if (err < 0) + return err; + err = snd_ak4113_create(ice->card, + qtet_ak4113_read, + qtet_ak4113_write, + ak4113_init_vals, + ice, &spec->ak4113); + if (err < 0) + return err; + /* callback for codecs rate setting */ + spec->ak4113->change_callback = qtet_ak4113_change; + spec->ak4113->change_callback_private = ice; + /* AK41143 in Quartet can detect external rate correctly + * (i.e. check_flags = 0) */ + spec->ak4113->check_flags = 0; + + proc_init(ice); + + qtet_set_rate(ice, 44100); + return 0; +} + +static unsigned char qtet_eeprom[] __devinitdata = { + [ICE_EEP2_SYSCONF] = 0x28, /* clock 256(24MHz), mpu401, 1xADC, + 1xDACs, SPDIF in */ + [ICE_EEP2_ACLINK] = 0x80, /* I2S */ + [ICE_EEP2_I2S] = 0x78, /* 96k, 24bit, 192k */ + [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, in, out-ext */ + [ICE_EEP2_GPIO_DIR] = 0x00, /* 0-7 inputs, switched to output + only during output operations */ + [ICE_EEP2_GPIO_DIR1] = 0xff, /* 8-15 outputs */ + [ICE_EEP2_GPIO_DIR2] = 0x00, + [ICE_EEP2_GPIO_MASK] = 0xff, /* changed only for OUT operations */ + [ICE_EEP2_GPIO_MASK1] = 0x00, + [ICE_EEP2_GPIO_MASK2] = 0xff, + + [ICE_EEP2_GPIO_STATE] = 0x00, /* inputs */ + [ICE_EEP2_GPIO_STATE1] = 0x7d, /* all 1, but GPIO_CPLD_RW + and GPIO15 always zero */ + [ICE_EEP2_GPIO_STATE2] = 0x00, /* inputs */ +}; + +/* entry point */ +struct snd_ice1712_card_info snd_vt1724_qtet_cards[] __devinitdata = { + { + .subvendor = VT1724_SUBDEVICE_QTET, + .name = "Infrasonic Quartet", + .model = "quartet", + .chip_init = qtet_init, + .build_controls = qtet_add_controls, + .eeprom_size = sizeof(qtet_eeprom), + .eeprom_data = qtet_eeprom, + }, + { } /* terminator */ +}; diff --git a/sound/pci/ice1712/quartet.h b/sound/pci/ice1712/quartet.h new file mode 100644 index 00000000000..80809b72439 --- /dev/null +++ b/sound/pci/ice1712/quartet.h @@ -0,0 +1,10 @@ +#ifndef __SOUND_QTET_H +#define __SOUND_QTET_H + +#define QTET_DEVICE_DESC "{Infrasonic,Quartet}," + +#define VT1724_SUBDEVICE_QTET 0x30305349 /* Infrasonic Quartet */ + +extern struct snd_ice1712_card_info snd_vt1724_qtet_cards[]; + +#endif /* __SOUND_QTET_H */ diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index aac20fb4aad..b990143636f 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2063,6 +2063,12 @@ static struct ac97_quirk ac97_quirks[] __devinitdata = { .type = AC97_TUNE_HP_ONLY }, { + .subvendor = 0x161f, + .subdevice = 0x203a, + .name = "Gateway 4525GZ", /* AD1981B */ + .type = AC97_TUNE_INV_EAPD + }, + { .subvendor = 0x1734, .subdevice = 0x0088, .name = "Fujitsu-Siemens D1522", /* AD1981 */ diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 4ba07d42fd1..389941cf610 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -1,7 +1,8 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o snd-hifier-objs := hifier.o snd-oxygen-objs := oxygen.o -snd-virtuoso-objs := virtuoso.o +snd-virtuoso-objs := virtuoso.o xonar_lib.o \ + xonar_pcm179x.o xonar_cs43xx.o xonar_hdmi.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o obj-$(CONFIG_SND_HIFIER) += snd-hifier.o diff --git a/sound/pci/oxygen/cs2000.h b/sound/pci/oxygen/cs2000.h new file mode 100644 index 00000000000..c3501bdb5ed --- /dev/null +++ b/sound/pci/oxygen/cs2000.h @@ -0,0 +1,83 @@ +#ifndef CS2000_H_INCLUDED +#define CS2000_H_INCLUDED + +#define CS2000_DEV_ID 0x01 +#define CS2000_DEV_CTRL 0x02 +#define CS2000_DEV_CFG_1 0x03 +#define CS2000_DEV_CFG_2 0x04 +#define CS2000_GLOBAL_CFG 0x05 +#define CS2000_RATIO_0 0x06 /* 32 bits, big endian */ +#define CS2000_RATIO_1 0x0a +#define CS2000_RATIO_2 0x0e +#define CS2000_RATIO_3 0x12 +#define CS2000_FUN_CFG_1 0x16 +#define CS2000_FUN_CFG_2 0x17 +#define CS2000_FUN_CFG_3 0x1e + +/* DEV_ID */ +#define CS2000_DEVICE_MASK 0xf8 +#define CS2000_REVISION_MASK 0x07 + +/* DEV_CTRL */ +#define CS2000_UNLOCK 0x80 +#define CS2000_AUX_OUT_DIS 0x02 +#define CS2000_CLK_OUT_DIS 0x01 + +/* DEV_CFG_1 */ +#define CS2000_R_MOD_SEL_MASK 0xe0 +#define CS2000_R_MOD_SEL_1 0x00 +#define CS2000_R_MOD_SEL_2 0x20 +#define CS2000_R_MOD_SEL_4 0x40 +#define CS2000_R_MOD_SEL_8 0x60 +#define CS2000_R_MOD_SEL_1_2 0x80 +#define CS2000_R_MOD_SEL_1_4 0xa0 +#define CS2000_R_MOD_SEL_1_8 0xc0 +#define CS2000_R_MOD_SEL_1_16 0xe0 +#define CS2000_R_SEL_MASK 0x18 +#define CS2000_R_SEL_SHIFT 3 +#define CS2000_AUX_OUT_SRC_MASK 0x06 +#define CS2000_AUX_OUT_SRC_REF_CLK 0x00 +#define CS2000_AUX_OUT_SRC_CLK_IN 0x02 +#define CS2000_AUX_OUT_SRC_CLK_OUT 0x04 +#define CS2000_AUX_OUT_SRC_PLL_LOCK 0x06 +#define CS2000_EN_DEV_CFG_1 0x01 + +/* DEV_CFG_2 */ +#define CS2000_LOCK_CLK_MASK 0x06 +#define CS2000_LOCK_CLK_SHIFT 1 +#define CS2000_FRAC_N_SRC_MASK 0x01 +#define CS2000_FRAC_N_SRC_STATIC 0x00 +#define CS2000_FRAC_N_SRC_DYNAMIC 0x01 + +/* GLOBAL_CFG */ +#define CS2000_FREEZE 0x08 +#define CS2000_EN_DEV_CFG_2 0x01 + +/* FUN_CFG_1 */ +#define CS2000_CLK_SKIP_EN 0x80 +#define CS2000_AUX_LOCK_CFG_MASK 0x40 +#define CS2000_AUX_LOCK_CFG_PP_HIGH 0x00 +#define CS2000_AUX_LOCK_CFG_OD_LOW 0x40 +#define CS2000_REF_CLK_DIV_MASK 0x18 +#define CS2000_REF_CLK_DIV_4 0x00 +#define CS2000_REF_CLK_DIV_2 0x08 +#define CS2000_REF_CLK_DIV_1 0x10 + +/* FUN_CFG_2 */ +#define CS2000_CLK_OUT_UNL 0x10 +#define CS2000_L_F_RATIO_CFG_MASK 0x08 +#define CS2000_L_F_RATIO_CFG_20_12 0x00 +#define CS2000_L_F_RATIO_CFG_12_20 0x08 + +/* FUN_CFG_3 */ +#define CS2000_CLK_IN_BW_MASK 0x70 +#define CS2000_CLK_IN_BW_1 0x00 +#define CS2000_CLK_IN_BW_2 0x10 +#define CS2000_CLK_IN_BW_4 0x20 +#define CS2000_CLK_IN_BW_8 0x30 +#define CS2000_CLK_IN_BW_16 0x40 +#define CS2000_CLK_IN_BW_32 0x50 +#define CS2000_CLK_IN_BW_64 0x60 +#define CS2000_CLK_IN_BW_128 0x70 + +#endif diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 84ef1318341..e3c229b6331 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -17,6 +17,12 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +/* + * CMI8788: + * + * SPI 0 -> AK4396 + */ + #include <linux/delay.h> #include <linux/pci.h> #include <sound/control.h> @@ -51,23 +57,28 @@ static struct pci_device_id hifier_ids[] __devinitdata = { MODULE_DEVICE_TABLE(pci, hifier_ids); struct hifier_data { - u8 ak4396_ctl2; + u8 ak4396_regs[5]; }; static void ak4396_write(struct oxygen *chip, u8 reg, u8 value) { + struct hifier_data *data = chip->model_data; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (0 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); + data->ak4396_regs[reg] = value; } -static void update_ak4396_volume(struct oxygen *chip) +static void ak4396_write_cached(struct oxygen *chip, u8 reg, u8 value) { - ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); - ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); + struct hifier_data *data = chip->model_data; + + if (value != data->ak4396_regs[reg]) + ak4396_write(chip, reg, value); } static void hifier_registers_init(struct oxygen *chip) @@ -75,16 +86,19 @@ static void hifier_registers_init(struct oxygen *chip) struct hifier_data *data = chip->model_data; ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, AK4396_CONTROL_2, data->ak4396_ctl2); + ak4396_write(chip, AK4396_CONTROL_2, + data->ak4396_regs[AK4396_CONTROL_2]); ak4396_write(chip, AK4396_CONTROL_3, AK4396_PCM); - update_ak4396_volume(chip); + ak4396_write(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write(chip, AK4396_RCH_ATT, chip->dac_volume[1]); } static void hifier_init(struct oxygen *chip) { struct hifier_data *data = chip->model_data; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + data->ak4396_regs[AK4396_CONTROL_2] = + AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; hifier_registers_init(chip); snd_component_add(chip->card, "AK4396"); @@ -106,20 +120,29 @@ static void set_ak4396_params(struct oxygen *chip, struct hifier_data *data = chip->model_data; u8 value; - value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; + value = data->ak4396_regs[AK4396_CONTROL_2] & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; else if (params_rate(params) <= 108000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; - data->ak4396_ctl2 = value; msleep(1); /* wait for the new MCLK to become stable */ - ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB); - ak4396_write(chip, AK4396_CONTROL_2, value); - ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); + if (value != data->ak4396_regs[AK4396_CONTROL_2]) { + ak4396_write(chip, AK4396_CONTROL_1, + AK4396_DIF_24_MSB); + ak4396_write(chip, AK4396_CONTROL_2, value); + ak4396_write(chip, AK4396_CONTROL_1, + AK4396_DIF_24_MSB | AK4396_RSTN); + } +} + +static void update_ak4396_volume(struct oxygen *chip) +{ + ak4396_write_cached(chip, AK4396_LCH_ATT, chip->dac_volume[0]); + ak4396_write_cached(chip, AK4396_RCH_ATT, chip->dac_volume[1]); } static void update_ak4396_mute(struct oxygen *chip) @@ -127,11 +150,10 @@ static void update_ak4396_mute(struct oxygen *chip) struct hifier_data *data = chip->model_data; u8 value; - value = data->ak4396_ctl2 & ~AK4396_SMUTE; + value = data->ak4396_regs[AK4396_CONTROL_2] & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; - data->ak4396_ctl2 = value; - ak4396_write(chip, AK4396_CONTROL_2, value); + ak4396_write_cached(chip, AK4396_CONTROL_2, value); } static void set_cs5340_params(struct oxygen *chip, @@ -141,21 +163,14 @@ static void set_cs5340_params(struct oxygen *chip, static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); -static int hifier_control_filter(struct snd_kcontrol_new *template) -{ - if (!strcmp(template->name, "Stereo Upmixing")) - return 1; /* stereo only - we don't need upmixing */ - return 0; -} - static const struct oxygen_model model_hifier = { .shortname = "C-Media CMI8787", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", .init = hifier_init, - .control_filter = hifier_control_filter, .cleanup = hifier_cleanup, .resume = hifier_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_ak4396_params, .set_adc_params = set_cs5340_params, .update_dac_volume = update_ak4396_volume, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 72db4c39007..acbedebcffd 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -18,6 +18,8 @@ */ /* + * CMI8788: + * * SPI 0 -> 1st AK4396 (front) * SPI 1 -> 2nd AK4396 (surround) * SPI 2 -> 3rd AK4396 (center/LFE) @@ -27,6 +29,10 @@ * GPIO 0 -> DFS0 of AK5385 * GPIO 1 -> DFS1 of AK5385 * GPIO 8 -> enable headphone amplifier on HT-Omega models + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to ADC input */ #include <linux/delay.h> @@ -91,8 +97,8 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_CLARO_HP 0x0100 struct generic_data { - u8 ak4396_ctl2; - u16 saved_wm8785_registers[2]; + u8 ak4396_regs[4][5]; + u16 wm8785_regs[3]; }; static void ak4396_write(struct oxygen *chip, unsigned int codec, @@ -102,12 +108,24 @@ static void ak4396_write(struct oxygen *chip, unsigned int codec, static const u8 codec_spi_map[4] = { 0, 1, 2, 4 }; + struct generic_data *data = chip->model_data; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | OXYGEN_SPI_DATA_LENGTH_2 | OXYGEN_SPI_CLOCK_160 | (codec_spi_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_HI, AK4396_WRITE | (reg << 8) | value); + data->ak4396_regs[codec][reg] = value; +} + +static void ak4396_write_cached(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + struct generic_data *data = chip->model_data; + + if (value != data->ak4396_regs[codec][reg]) + ak4396_write(chip, codec, reg, value); } static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) @@ -120,20 +138,8 @@ static void wm8785_write(struct oxygen *chip, u8 reg, unsigned int value) (3 << OXYGEN_SPI_CODEC_SHIFT) | OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); - if (reg < ARRAY_SIZE(data->saved_wm8785_registers)) - data->saved_wm8785_registers[reg] = value; -} - -static void update_ak4396_volume(struct oxygen *chip) -{ - unsigned int i; - - for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_LCH_ATT, chip->dac_volume[i * 2]); - ak4396_write(chip, i, - AK4396_RCH_ATT, chip->dac_volume[i * 2 + 1]); - } + if (reg < ARRAY_SIZE(data->wm8785_regs)) + data->wm8785_regs[reg] = value; } static void ak4396_registers_init(struct oxygen *chip) @@ -142,21 +148,25 @@ static void ak4396_registers_init(struct oxygen *chip) unsigned int i; for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); - ak4396_write(chip, i, - AK4396_CONTROL_2, data->ak4396_ctl2); - ak4396_write(chip, i, - AK4396_CONTROL_3, AK4396_PCM); + ak4396_write(chip, i, AK4396_CONTROL_1, + AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write(chip, i, AK4396_CONTROL_2, + data->ak4396_regs[0][AK4396_CONTROL_2]); + ak4396_write(chip, i, AK4396_CONTROL_3, + AK4396_PCM); + ak4396_write(chip, i, AK4396_LCH_ATT, + chip->dac_volume[i * 2]); + ak4396_write(chip, i, AK4396_RCH_ATT, + chip->dac_volume[i * 2 + 1]); } - update_ak4396_volume(chip); } static void ak4396_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; - data->ak4396_ctl2 = AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; + data->ak4396_regs[0][AK4396_CONTROL_2] = + AK4396_SMUTE | AK4396_DEM_OFF | AK4396_DFS_NORMAL; ak4396_registers_init(chip); snd_component_add(chip->card, "AK4396"); } @@ -173,17 +183,17 @@ static void wm8785_registers_init(struct oxygen *chip) struct generic_data *data = chip->model_data; wm8785_write(chip, WM8785_R7, 0); - wm8785_write(chip, WM8785_R0, data->saved_wm8785_registers[0]); - wm8785_write(chip, WM8785_R1, data->saved_wm8785_registers[1]); + wm8785_write(chip, WM8785_R0, data->wm8785_regs[0]); + wm8785_write(chip, WM8785_R2, data->wm8785_regs[2]); } static void wm8785_init(struct oxygen *chip) { struct generic_data *data = chip->model_data; - data->saved_wm8785_registers[0] = WM8785_MCR_SLAVE | - WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; - data->saved_wm8785_registers[1] = WM8785_WL_24; + data->wm8785_regs[0] = + WM8785_MCR_SLAVE | WM8785_OSR_SINGLE | WM8785_FORMAT_LJUST; + data->wm8785_regs[2] = WM8785_HPFR | WM8785_HPFL; wm8785_registers_init(chip); snd_component_add(chip->card, "WM8785"); } @@ -264,24 +274,36 @@ static void set_ak4396_params(struct oxygen *chip, unsigned int i; u8 value; - value = data->ak4396_ctl2 & ~AK4396_DFS_MASK; + value = data->ak4396_regs[0][AK4396_CONTROL_2] & ~AK4396_DFS_MASK; if (params_rate(params) <= 54000) value |= AK4396_DFS_NORMAL; else if (params_rate(params) <= 108000) value |= AK4396_DFS_DOUBLE; else value |= AK4396_DFS_QUAD; - data->ak4396_ctl2 = value; msleep(1); /* wait for the new MCLK to become stable */ + if (value != data->ak4396_regs[0][AK4396_CONTROL_2]) { + for (i = 0; i < 4; ++i) { + ak4396_write(chip, i, AK4396_CONTROL_1, + AK4396_DIF_24_MSB); + ak4396_write(chip, i, AK4396_CONTROL_2, value); + ak4396_write(chip, i, AK4396_CONTROL_1, + AK4396_DIF_24_MSB | AK4396_RSTN); + } + } +} + +static void update_ak4396_volume(struct oxygen *chip) +{ + unsigned int i; + for (i = 0; i < 4; ++i) { - ak4396_write(chip, i, - AK4396_CONTROL_1, AK4396_DIF_24_MSB); - ak4396_write(chip, i, - AK4396_CONTROL_2, value); - ak4396_write(chip, i, - AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); + ak4396_write_cached(chip, i, AK4396_LCH_ATT, + chip->dac_volume[i * 2]); + ak4396_write_cached(chip, i, AK4396_RCH_ATT, + chip->dac_volume[i * 2 + 1]); } } @@ -291,21 +313,19 @@ static void update_ak4396_mute(struct oxygen *chip) unsigned int i; u8 value; - value = data->ak4396_ctl2 & ~AK4396_SMUTE; + value = data->ak4396_regs[0][AK4396_CONTROL_2] & ~AK4396_SMUTE; if (chip->dac_mute) value |= AK4396_SMUTE; - data->ak4396_ctl2 = value; for (i = 0; i < 4; ++i) - ak4396_write(chip, i, AK4396_CONTROL_2, value); + ak4396_write_cached(chip, i, AK4396_CONTROL_2, value); } static void set_wm8785_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { + struct generic_data *data = chip->model_data; unsigned int value; - wm8785_write(chip, WM8785_R7, 0); - value = WM8785_MCR_SLAVE | WM8785_FORMAT_LJUST; if (params_rate(params) <= 48000) value |= WM8785_OSR_SINGLE; @@ -313,13 +333,11 @@ static void set_wm8785_params(struct oxygen *chip, value |= WM8785_OSR_DOUBLE; else value |= WM8785_OSR_QUAD; - wm8785_write(chip, WM8785_R0, value); - - if (snd_pcm_format_width(params_format(params)) <= 16) - value = WM8785_WL_16; - else - value = WM8785_WL_24; - wm8785_write(chip, WM8785_R1, value); + if (value != data->wm8785_regs[0]) { + wm8785_write(chip, WM8785_R7, 0); + wm8785_write(chip, WM8785_R0, value); + wm8785_write(chip, WM8785_R2, data->wm8785_regs[2]); + } } static void set_ak5385_params(struct oxygen *chip, @@ -337,6 +355,134 @@ static void set_ak5385_params(struct oxygen *chip, value, GPIO_AK5385_DFS_MASK); } +static int rolloff_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "Sharp Roll-off", "Slow Roll-off" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int rolloff_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->ak4396_regs[0][AK4396_CONTROL_2] & AK4396_SLOW) != 0; + return 0; +} + +static int rolloff_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + unsigned int i; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->ak4396_regs[0][AK4396_CONTROL_2]; + if (value->value.enumerated.item[0]) + reg |= AK4396_SLOW; + else + reg &= ~AK4396_SLOW; + changed = reg != data->ak4396_regs[0][AK4396_CONTROL_2]; + if (changed) { + for (i = 0; i < 4; ++i) + ak4396_write(chip, i, AK4396_CONTROL_2, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new rolloff_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Filter Playback Enum", + .info = rolloff_info, + .get = rolloff_get, + .put = rolloff_put, +}; + +static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "None", "High-pass Filter" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->wm8785_regs[WM8785_R2] & WM8785_HPFR) != 0; + return 0; +} + +static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct generic_data *data = chip->model_data; + unsigned int reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8785_regs[WM8785_R2] & ~(WM8785_HPFR | WM8785_HPFL); + if (value->value.enumerated.item[0]) + reg |= WM8785_HPFR | WM8785_HPFL; + changed = reg != data->wm8785_regs[WM8785_R2]; + if (changed) + wm8785_write(chip, WM8785_R2, reg); + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new hpf_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, +}; + +static int generic_mixer_init(struct oxygen *chip) +{ + return snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); +} + +static int generic_wm8785_mixer_init(struct oxygen *chip) +{ + int err; + + err = generic_mixer_init(chip); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&hpf_control, chip)); + if (err < 0) + return err; + return 0; +} + static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); static const struct oxygen_model model_generic = { @@ -344,8 +490,10 @@ static const struct oxygen_model model_generic = { .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", .init = generic_init, + .mixer_init = generic_wm8785_mixer_init, .cleanup = generic_cleanup, .resume = generic_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, .set_dac_params = set_ak4396_params, .set_adc_params = set_wm8785_params, .update_dac_volume = update_ak4396_volume, @@ -374,6 +522,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, switch (id->driver_data) { case MODEL_MERIDIAN: chip->model.init = meridian_init; + chip->model.mixer_init = generic_mixer_init; chip->model.resume = meridian_resume; chip->model.set_adc_params = set_ak5385_params; chip->model.device_config = PLAYBACK_0_TO_I2S | @@ -389,6 +538,7 @@ static int __devinit get_oxygen_model(struct oxygen *chip, break; case MODEL_CLARO_HALO: chip->model.init = claro_halo_init; + chip->model.mixer_init = generic_mixer_init; chip->model.cleanup = claro_cleanup; chip->model.suspend = claro_suspend; chip->model.resume = claro_resume; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index bd615dbffad..6147216af74 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -78,12 +78,15 @@ struct oxygen_model { void (*resume)(struct oxygen *chip); void (*pcm_hardware_filter)(unsigned int channel, struct snd_pcm_hardware *hardware); + unsigned int (*get_i2s_mclk)(struct oxygen *chip, unsigned int channel, + struct snd_pcm_hw_params *hw_params); void (*set_dac_params)(struct oxygen *chip, struct snd_pcm_hw_params *params); void (*set_adc_params)(struct oxygen *chip, struct snd_pcm_hw_params *params); void (*update_dac_volume)(struct oxygen *chip); void (*update_dac_mute)(struct oxygen *chip); + void (*update_center_lfe_mix)(struct oxygen *chip, bool mixed); void (*gpio_changed)(struct oxygen *chip); void (*uart_input)(struct oxygen *chip); void (*ac97_switch)(struct oxygen *chip, @@ -162,6 +165,8 @@ void oxygen_update_spdif_source(struct oxygen *chip); /* oxygen_pcm.c */ int oxygen_pcm_init(struct oxygen *chip); +unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, unsigned int channel, + struct snd_pcm_hw_params *hw_params); /* oxygen_io.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9a8936e2074..9c5e6450eeb 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -278,7 +278,11 @@ oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) static void oxygen_restore_eeprom(struct oxygen *chip, const struct pci_device_id *id) { - if (oxygen_read_eeprom(chip, 0) != OXYGEN_EEPROM_ID) { + u16 eeprom_id; + + eeprom_id = oxygen_read_eeprom(chip, 0); + if (eeprom_id != OXYGEN_EEPROM_ID && + (eeprom_id != 0xffff || id->subdevice != 0x8788)) { /* * This function gets called only when a known card model has * been detected, i.e., we know there is a valid subsystem @@ -303,6 +307,28 @@ static void oxygen_restore_eeprom(struct oxygen *chip, } } +static void pci_bridge_magic(void) +{ + struct pci_dev *pci = NULL; + u32 tmp; + + for (;;) { + /* If there is any Pericom PI7C9X110 PCI-E/PCI bridge ... */ + pci = pci_get_device(0x12d8, 0xe110, pci); + if (!pci) + break; + /* + * ... configure its secondary internal arbiter to park to + * the secondary port, instead of to the last master. + */ + if (!pci_read_config_dword(pci, 0x40, &tmp)) { + tmp |= 1; + pci_write_config_dword(pci, 0x40, tmp); + } + /* Why? Try asking C-Media. */ + } +} + static void oxygen_init(struct oxygen *chip) { unsigned int i; @@ -581,6 +607,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, snd_card_set_dev(card, &pci->dev); card->private_free = oxygen_card_free; + pci_bridge_magic(); oxygen_init(chip); chip->model.init(chip); diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 5401c547c4e..f375b8a2786 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -99,11 +99,15 @@ static int dac_mute_put(struct snd_kcontrol *ctl, static int upmix_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { - static const char *const names[3] = { - "Front", "Front+Surround", "Front+Surround+Back" + static const char *const names[5] = { + "Front", + "Front+Surround", + "Front+Surround+Back", + "Front+Surround+Center/LFE", + "Front+Surround+Center/LFE+Back", }; struct oxygen *chip = ctl->private_data; - unsigned int count = 2 + (chip->model.dac_channels == 8); + unsigned int count = chip->model.update_center_lfe_mix ? 5 : 3; info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; info->count = 1; @@ -127,7 +131,7 @@ static int upmix_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) void oxygen_update_dac_routing(struct oxygen *chip) { /* DAC 0: front, DAC 1: surround, DAC 2: center/LFE, DAC 3: back */ - static const unsigned int reg_values[3] = { + static const unsigned int reg_values[5] = { /* stereo -> front */ (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | (1 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | @@ -143,6 +147,16 @@ void oxygen_update_dac_routing(struct oxygen *chip) (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | (2 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), + /* stereo -> front+surround+center/LFE */ + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (3 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), + /* stereo -> front+surround+center/LFE+back */ + (0 << OXYGEN_PLAY_DAC0_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC1_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC2_SOURCE_SHIFT) | + (0 << OXYGEN_PLAY_DAC3_SOURCE_SHIFT), }; u8 channels; unsigned int reg_value; @@ -167,22 +181,23 @@ void oxygen_update_dac_routing(struct oxygen *chip) OXYGEN_PLAY_DAC1_SOURCE_MASK | OXYGEN_PLAY_DAC2_SOURCE_MASK | OXYGEN_PLAY_DAC3_SOURCE_MASK); + if (chip->model.update_center_lfe_mix) + chip->model.update_center_lfe_mix(chip, chip->dac_routing > 2); } static int upmix_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) { struct oxygen *chip = ctl->private_data; - unsigned int count = 2 + (chip->model.dac_channels == 8); + unsigned int count = chip->model.update_center_lfe_mix ? 5 : 3; int changed; + if (value->value.enumerated.item[0] >= count) + return -EINVAL; mutex_lock(&chip->mutex); changed = value->value.enumerated.item[0] != chip->dac_routing; if (changed) { - chip->dac_routing = min(value->value.enumerated.item[0], - count - 1); - spin_lock_irq(&chip->reg_lock); + chip->dac_routing = value->value.enumerated.item[0]; oxygen_update_dac_routing(chip); - spin_unlock_irq(&chip->reg_lock); } mutex_unlock(&chip->mutex); return changed; @@ -790,7 +805,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", + .name = "Analog Input Monitor Playback Switch", .info = snd_ctl_boolean_mono_info, .get = monitor_get, .put = monitor_put, @@ -798,7 +813,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", + .name = "Analog Input Monitor Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = monitor_volume_info, @@ -815,7 +830,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", + .name = "Analog Input Monitor Playback Switch", .info = snd_ctl_boolean_mono_info, .get = monitor_get, .put = monitor_put, @@ -823,7 +838,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", + .name = "Analog Input Monitor Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = monitor_volume_info, @@ -840,7 +855,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Switch", + .name = "Analog Input Monitor Playback Switch", .index = 1, .info = snd_ctl_boolean_mono_info, .get = monitor_get, @@ -849,7 +864,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Input Monitor Volume", + .name = "Analog Input Monitor Playback Volume", .index = 1, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, @@ -867,7 +882,7 @@ static const struct { .controls = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Monitor Switch", + .name = "Digital Input Monitor Playback Switch", .info = snd_ctl_boolean_mono_info, .get = monitor_get, .put = monitor_put, @@ -875,7 +890,7 @@ static const struct { }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Monitor Volume", + .name = "Digital Input Monitor Playback Volume", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, .info = monitor_volume_info, @@ -954,6 +969,9 @@ static int add_controls(struct oxygen *chip, if (err == 1) continue; } + if (!strcmp(template.name, "Stereo Upmixing") && + chip->model.dac_channels == 2) + continue; if (!strcmp(template.name, "Master Playback Volume") && chip->model.dac_tlv) { template.tlv.p = chip->model.dac_tlv; diff --git a/sound/pci/oxygen/oxygen_pcm.c b/sound/pci/oxygen/oxygen_pcm.c index ef2345d82b8..9dff6954c39 100644 --- a/sound/pci/oxygen/oxygen_pcm.c +++ b/sound/pci/oxygen/oxygen_pcm.c @@ -271,13 +271,16 @@ static unsigned int oxygen_rate(struct snd_pcm_hw_params *hw_params) } } -static unsigned int oxygen_i2s_mclk(struct snd_pcm_hw_params *hw_params) +unsigned int oxygen_default_i2s_mclk(struct oxygen *chip, + unsigned int channel, + struct snd_pcm_hw_params *hw_params) { if (params_rate(hw_params) <= 96000) return OXYGEN_I2S_MCLK_256; else return OXYGEN_I2S_MCLK_128; } +EXPORT_SYMBOL(oxygen_default_i2s_mclk); static unsigned int oxygen_i2s_bits(struct snd_pcm_hw_params *hw_params) { @@ -354,7 +357,7 @@ static int oxygen_rec_a_hw_params(struct snd_pcm_substream *substream, OXYGEN_REC_FORMAT_A_MASK); oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, oxygen_rate(hw_params) | - oxygen_i2s_mclk(hw_params) | + chip->model.get_i2s_mclk(chip, PCM_A, hw_params) | chip->model.adc_i2s_format | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | @@ -390,7 +393,8 @@ static int oxygen_rec_b_hw_params(struct snd_pcm_substream *substream, if (!is_ac97) oxygen_write16_masked(chip, OXYGEN_I2S_B_FORMAT, oxygen_rate(hw_params) | - oxygen_i2s_mclk(hw_params) | + chip->model.get_i2s_mclk(chip, PCM_B, + hw_params) | chip->model.adc_i2s_format | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | @@ -435,6 +439,7 @@ static int oxygen_spdif_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; + mutex_lock(&chip->mutex); spin_lock_irq(&chip->reg_lock); oxygen_clear_bits32(chip, OXYGEN_SPDIF_CONTROL, OXYGEN_SPDIF_OUT_ENABLE); @@ -446,6 +451,7 @@ static int oxygen_spdif_hw_params(struct snd_pcm_substream *substream, OXYGEN_SPDIF_OUT_RATE_MASK); oxygen_update_spdif_source(chip); spin_unlock_irq(&chip->reg_lock); + mutex_unlock(&chip->mutex); return 0; } @@ -459,6 +465,7 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; + mutex_lock(&chip->mutex); spin_lock_irq(&chip->reg_lock); oxygen_write8_masked(chip, OXYGEN_PLAY_CHANNELS, oxygen_play_channels(hw_params), @@ -469,18 +476,18 @@ static int oxygen_multich_hw_params(struct snd_pcm_substream *substream, oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, oxygen_rate(hw_params) | chip->model.dac_i2s_format | - oxygen_i2s_mclk(hw_params) | + chip->model.get_i2s_mclk(chip, PCM_MULTICH, + hw_params) | oxygen_i2s_bits(hw_params), OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_FORMAT_MASK | OXYGEN_I2S_MCLK_MASK | OXYGEN_I2S_BITS_MASK); - oxygen_update_dac_routing(chip); oxygen_update_spdif_source(chip); spin_unlock_irq(&chip->reg_lock); - mutex_lock(&chip->mutex); chip->model.set_dac_params(chip, hw_params); + oxygen_update_dac_routing(chip); mutex_unlock(&chip->mutex); return 0; } diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6ebcb6bdd71..6accaf9580b 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -17,145 +17,12 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -/* - * Xonar D2/D2X - * ------------ - * - * CMI8788: - * - * SPI 0 -> 1st PCM1796 (front) - * SPI 1 -> 2nd PCM1796 (surround) - * SPI 2 -> 3rd PCM1796 (center/LFE) - * SPI 4 -> 4th PCM1796 (back) - * - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 5 <- external power present (D2X only) - * GPIO 7 -> ALT - * GPIO 8 -> enable output to speakers - */ - -/* - * Xonar D1/DX - * ----------- - * - * CMI8788: - * - * I²C <-> CS4398 (front) - * <-> CS4362A (surround, center/LFE, back) - * - * GPI 0 <- external power present (DX only) - * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> enable front panel I/O - * GPIO 2 -> M0 of CS5361 - * GPIO 3 -> M1 of CS5361 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) - * - * CS4398: - * - * AD0 <- 1 - * AD1 <- 1 - * - * CS4362A: - * - * AD0 <- 0 - */ - -/* - * Xonar HDAV1.3 (Deluxe) - * ---------------------- - * - * CMI8788: - * - * I²C <-> PCM1796 (front) - * - * GPI 0 <- external power present - * - * GPIO 0 -> enable output to speakers - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) - * - * TXD -> HDMI controller - * RXD <- HDMI controller - * - * PCM1796 front: AD1,0 <- 0,0 - * - * no daughterboard - * ---------------- - * - * GPIO 4 <- 1 - * - * H6 daughterboard - * ---------------- - * - * GPIO 4 <- 0 - * GPIO 5 <- 0 - * - * I²C <-> PCM1796 (surround) - * <-> PCM1796 (center/LFE) - * <-> PCM1796 (back) - * - * PCM1796 surround: AD1,0 <- 0,1 - * PCM1796 center/LFE: AD1,0 <- 1,0 - * PCM1796 back: AD1,0 <- 1,1 - * - * unknown daughterboard - * --------------------- - * - * GPIO 4 <- 0 - * GPIO 5 <- 1 - * - * I²C <-> CS4362A (surround, center/LFE, back) - * - * CS4362A: AD0 <- 0 - */ - -/* - * Xonar Essence ST (Deluxe)/STX - * ----------------------------- - * - * CMI8788: - * - * I²C <-> PCM1792A - * - * GPI 0 <- external power present - * - * GPIO 0 -> enable output to speakers - * GPIO 1 -> route HP to front panel (0) or rear jack (1) - * GPIO 2 -> M0 of CS5381 - * GPIO 3 -> M1 of CS5381 - * GPIO 7 -> route output to speaker jacks (0) or HP (1) - * GPIO 8 -> route input jack to line-in (0) or mic-in (1) - * - * PCM1792A: - * - * AD0 <- 0 - * - * H6 daughterboard - * ---------------- - * - * GPIO 4 <- 0 - * GPIO 5 <- 0 - */ - #include <linux/pci.h> #include <linux/delay.h> -#include <linux/mutex.h> -#include <sound/ac97_codec.h> -#include <sound/asoundef.h> -#include <sound/control.h> #include <sound/core.h> #include <sound/initval.h> #include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/tlv.h> -#include "oxygen.h" -#include "cm9780.h" -#include "pcm1796.h" -#include "cs4398.h" -#include "cs4362a.h" +#include "xonar.h" MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("Asus AVx00 driver"); @@ -173,972 +40,28 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -enum { - MODEL_D2, - MODEL_D2X, - MODEL_D1, - MODEL_DX, - MODEL_HDAV, /* without daughterboard */ - MODEL_HDAV_H6, /* with H6 daughterboard */ - MODEL_ST, - MODEL_ST_H6, - MODEL_STX, -}; - static struct pci_device_id xonar_ids[] __devinitdata = { - { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 }, - { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX }, - { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, - { OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV }, - { OXYGEN_PCI_SUBID(0x1043, 0x8327), .driver_data = MODEL_DX }, - { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, - { OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX }, - { OXYGEN_PCI_SUBID(0x1043, 0x835d), .driver_data = MODEL_ST }, + { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, + { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8314) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8327) }, + { OXYGEN_PCI_SUBID(0x1043, 0x834f) }, + { OXYGEN_PCI_SUBID(0x1043, 0x835c) }, + { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; MODULE_DEVICE_TABLE(pci, xonar_ids); - -#define GPIO_CS53x1_M_MASK 0x000c -#define GPIO_CS53x1_M_SINGLE 0x0000 -#define GPIO_CS53x1_M_DOUBLE 0x0004 -#define GPIO_CS53x1_M_QUAD 0x0008 - -#define GPIO_D2X_EXT_POWER 0x0020 -#define GPIO_D2_ALT 0x0080 -#define GPIO_D2_OUTPUT_ENABLE 0x0100 - -#define GPI_DX_EXT_POWER 0x01 -#define GPIO_DX_OUTPUT_ENABLE 0x0001 -#define GPIO_DX_FRONT_PANEL 0x0002 -#define GPIO_DX_INPUT_ROUTE 0x0100 - -#define GPIO_DB_MASK 0x0030 -#define GPIO_DB_H6 0x0000 -#define GPIO_DB_XX 0x0020 - -#define GPIO_ST_HP_REAR 0x0002 -#define GPIO_ST_HP 0x0080 - -#define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ADx=i, /W=0 */ -#define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ -#define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ - -struct xonar_data { - unsigned int anti_pop_delay; - unsigned int dacs; - u16 output_enable_bit; - u8 ext_power_reg; - u8 ext_power_int_reg; - u8 ext_power_bit; - u8 has_power; - u8 pcm1796_oversampling; - u8 cs4398_fm; - u8 cs4362a_fm; - u8 hdmi_params[5]; -}; - -static void xonar_gpio_changed(struct oxygen *chip); - -static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, - u8 reg, u8 value) -{ - /* maps ALSA channel pair number to SPI output */ - static const u8 codec_map[4] = { - 0, 1, 2, 4 - }; - oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | - OXYGEN_SPI_DATA_LENGTH_2 | - OXYGEN_SPI_CLOCK_160 | - (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | - OXYGEN_SPI_CEN_LATCH_CLOCK_HI, - (reg << 8) | value); -} - -static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec, - u8 reg, u8 value) -{ - oxygen_write_i2c(chip, I2C_DEVICE_PCM1796(codec), reg, value); -} - -static void pcm1796_write(struct oxygen *chip, unsigned int codec, - u8 reg, u8 value) -{ - if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) == - OXYGEN_FUNCTION_SPI) - pcm1796_write_spi(chip, codec, reg, value); - else - pcm1796_write_i2c(chip, codec, reg, value); -} - -static void cs4398_write(struct oxygen *chip, u8 reg, u8 value) -{ - oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value); -} - -static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) -{ - oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); -} - -static void hdmi_write_command(struct oxygen *chip, u8 command, - unsigned int count, const u8 *params) -{ - unsigned int i; - u8 checksum; - - oxygen_write_uart(chip, 0xfb); - oxygen_write_uart(chip, 0xef); - oxygen_write_uart(chip, command); - oxygen_write_uart(chip, count); - for (i = 0; i < count; ++i) - oxygen_write_uart(chip, params[i]); - checksum = 0xfb + 0xef + command + count; - for (i = 0; i < count; ++i) - checksum += params[i]; - oxygen_write_uart(chip, checksum); -} - -static void xonar_enable_output(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - msleep(data->anti_pop_delay); - oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); -} - -static void xonar_common_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - if (data->ext_power_reg) { - oxygen_set_bits8(chip, data->ext_power_int_reg, - data->ext_power_bit); - chip->interrupt_mask |= OXYGEN_INT_GPIO; - chip->model.gpio_changed = xonar_gpio_changed; - data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) - & data->ext_power_bit); - } - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_CS53x1_M_MASK | data->output_enable_bit); - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); - oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); - xonar_enable_output(chip); -} - -static void update_pcm1796_volume(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - - for (i = 0; i < data->dacs; ++i) { - pcm1796_write(chip, i, 16, chip->dac_volume[i * 2]); - pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1]); - } -} - -static void update_pcm1796_mute(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - u8 value; - - value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; - if (chip->dac_mute) - value |= PCM1796_MUTE; - for (i = 0; i < data->dacs; ++i) - pcm1796_write(chip, i, 18, value); -} - -static void pcm1796_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - - for (i = 0; i < data->dacs; ++i) { - pcm1796_write(chip, i, 19, PCM1796_FLT_SHARP | PCM1796_ATS_1); - pcm1796_write(chip, i, 20, data->pcm1796_oversampling); - pcm1796_write(chip, i, 21, 0); - } - update_pcm1796_mute(chip); /* set ATLD before ATL/ATR */ - update_pcm1796_volume(chip); -} - -static void xonar_d2_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->anti_pop_delay = 300; - data->dacs = 4; - data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; - data->pcm1796_oversampling = PCM1796_OS_64; - - pcm1796_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); - - xonar_common_init(chip); - - snd_component_add(chip->card, "PCM1796"); - snd_component_add(chip->card, "CS5381"); -} - -static void xonar_d2x_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->ext_power_reg = OXYGEN_GPIO_DATA; - data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; - data->ext_power_bit = GPIO_D2X_EXT_POWER; - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER); - - xonar_d2_init(chip); -} - -static void update_cs4362a_volumes(struct oxygen *chip) -{ - u8 mute; - - mute = chip->dac_mute ? CS4362A_MUTE : 0; - cs4362a_write(chip, 7, (127 - chip->dac_volume[2]) | mute); - cs4362a_write(chip, 8, (127 - chip->dac_volume[3]) | mute); - cs4362a_write(chip, 10, (127 - chip->dac_volume[4]) | mute); - cs4362a_write(chip, 11, (127 - chip->dac_volume[5]) | mute); - cs4362a_write(chip, 13, (127 - chip->dac_volume[6]) | mute); - cs4362a_write(chip, 14, (127 - chip->dac_volume[7]) | mute); -} - -static void update_cs43xx_volume(struct oxygen *chip) -{ - cs4398_write(chip, 5, (127 - chip->dac_volume[0]) * 2); - cs4398_write(chip, 6, (127 - chip->dac_volume[1]) * 2); - update_cs4362a_volumes(chip); -} - -static void update_cs43xx_mute(struct oxygen *chip) -{ - u8 reg; - - reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; - if (chip->dac_mute) - reg |= CS4398_MUTE_B | CS4398_MUTE_A; - cs4398_write(chip, 4, reg); - update_cs4362a_volumes(chip); -} - -static void cs43xx_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - /* set CPEN (control port mode) and power down */ - cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); - cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); - /* configure */ - cs4398_write(chip, 2, data->cs4398_fm); - cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); - cs4398_write(chip, 7, CS4398_RMP_DN | CS4398_RMP_UP | - CS4398_ZERO_CROSS | CS4398_SOFT_RAMP); - cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); - cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE | - CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); - cs4362a_write(chip, 0x04, CS4362A_RMP_DN | CS4362A_DEM_NONE); - cs4362a_write(chip, 0x05, 0); - cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm); - cs4362a_write(chip, 0x0c, data->cs4362a_fm); - update_cs43xx_volume(chip); - update_cs43xx_mute(chip); - /* clear power down */ - cs4398_write(chip, 8, CS4398_CPEN); - cs4362a_write(chip, 0x01, CS4362A_CPEN); -} - -static void xonar_d1_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->anti_pop_delay = 800; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_FM_SINGLE | - CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - - cs43xx_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, - GPIO_DX_FRONT_PANEL | GPIO_DX_INPUT_ROUTE); - - xonar_common_init(chip); - - snd_component_add(chip->card, "CS4398"); - snd_component_add(chip->card, "CS4362A"); - snd_component_add(chip->card, "CS5361"); -} - -static void xonar_dx_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - - xonar_d1_init(chip); -} - -static void xonar_hdav_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - u8 param; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - - data->anti_pop_delay = 100; - data->dacs = chip->model.private_data == MODEL_HDAV_H6 ? 4 : 1; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - data->pcm1796_oversampling = PCM1796_OS_64; - - pcm1796_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DX_INPUT_ROUTE); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DX_INPUT_ROUTE); - - oxygen_reset_uart(chip); - param = 0; - hdmi_write_command(chip, 0x61, 1, ¶m); - param = 1; - hdmi_write_command(chip, 0x74, 1, ¶m); - data->hdmi_params[1] = IEC958_AES3_CON_FS_48000; - data->hdmi_params[4] = 1; - hdmi_write_command(chip, 0x54, 5, data->hdmi_params); - - xonar_common_init(chip); - - snd_component_add(chip->card, "PCM1796"); - snd_component_add(chip->card, "CS5381"); -} - -static void xonar_st_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, - OXYGEN_2WIRE_LENGTH_8 | - OXYGEN_2WIRE_INTERRUPT_MASK | - OXYGEN_2WIRE_SPEED_FAST); - - if (chip->model.private_data == MODEL_ST_H6) - chip->model.dac_channels = 8; - data->anti_pop_delay = 100; - data->dacs = chip->model.private_data == MODEL_ST_H6 ? 4 : 1; - data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->pcm1796_oversampling = PCM1796_OS_64; - - pcm1796_init(chip); - - oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, - GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); - - xonar_common_init(chip); - - snd_component_add(chip->card, "PCM1792A"); - snd_component_add(chip->card, "CS5381"); -} - -static void xonar_stx_init(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - - xonar_st_init(chip); -} - -static void xonar_disable_output(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - - oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); -} - -static void xonar_d2_cleanup(struct oxygen *chip) -{ - xonar_disable_output(chip); -} - -static void xonar_d1_cleanup(struct oxygen *chip) -{ - xonar_disable_output(chip); - cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); - oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); -} - -static void xonar_hdav_cleanup(struct oxygen *chip) -{ - u8 param = 0; - - hdmi_write_command(chip, 0x74, 1, ¶m); - xonar_disable_output(chip); -} - -static void xonar_st_cleanup(struct oxygen *chip) -{ - xonar_disable_output(chip); -} - -static void xonar_d2_suspend(struct oxygen *chip) -{ - xonar_d2_cleanup(chip); -} - -static void xonar_d1_suspend(struct oxygen *chip) -{ - xonar_d1_cleanup(chip); -} - -static void xonar_hdav_suspend(struct oxygen *chip) -{ - xonar_hdav_cleanup(chip); - msleep(2); -} - -static void xonar_st_suspend(struct oxygen *chip) -{ - xonar_st_cleanup(chip); -} - -static void xonar_d2_resume(struct oxygen *chip) -{ - pcm1796_init(chip); - xonar_enable_output(chip); -} - -static void xonar_d1_resume(struct oxygen *chip) -{ - oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); - msleep(1); - cs43xx_init(chip); - xonar_enable_output(chip); -} - -static void xonar_hdav_resume(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - u8 param; - - oxygen_reset_uart(chip); - param = 0; - hdmi_write_command(chip, 0x61, 1, ¶m); - param = 1; - hdmi_write_command(chip, 0x74, 1, ¶m); - hdmi_write_command(chip, 0x54, 5, data->hdmi_params); - pcm1796_init(chip); - xonar_enable_output(chip); -} - -static void xonar_st_resume(struct oxygen *chip) -{ - pcm1796_init(chip); - xonar_enable_output(chip); -} - -static void xonar_hdav_pcm_hardware_filter(unsigned int channel, - struct snd_pcm_hardware *hardware) -{ - if (channel == PCM_MULTICH) { - hardware->rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_96000 | - SNDRV_PCM_RATE_192000; - hardware->rate_min = 44100; - } -} - -static void set_pcm1796_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct xonar_data *data = chip->model_data; - unsigned int i; - - data->pcm1796_oversampling = - params_rate(params) >= 96000 ? PCM1796_OS_32 : PCM1796_OS_64; - for (i = 0; i < data->dacs; ++i) - pcm1796_write(chip, i, 20, data->pcm1796_oversampling); -} - -static void set_cs53x1_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - unsigned int value; - - if (params_rate(params) <= 54000) - value = GPIO_CS53x1_M_SINGLE; - else if (params_rate(params) <= 108000) - value = GPIO_CS53x1_M_DOUBLE; - else - value = GPIO_CS53x1_M_QUAD; - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - value, GPIO_CS53x1_M_MASK); -} - -static void set_cs43xx_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct xonar_data *data = chip->model_data; - - data->cs4398_fm = CS4398_DEM_NONE | CS4398_DIF_LJUST; - data->cs4362a_fm = CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - if (params_rate(params) <= 50000) { - data->cs4398_fm |= CS4398_FM_SINGLE; - data->cs4362a_fm |= CS4362A_FM_SINGLE; - } else if (params_rate(params) <= 100000) { - data->cs4398_fm |= CS4398_FM_DOUBLE; - data->cs4362a_fm |= CS4362A_FM_DOUBLE; - } else { - data->cs4398_fm |= CS4398_FM_QUAD; - data->cs4362a_fm |= CS4362A_FM_QUAD; - } - cs4398_write(chip, 2, data->cs4398_fm); - cs4362a_write(chip, 0x06, data->cs4362a_fm); - cs4362a_write(chip, 0x09, data->cs4362a_fm); - cs4362a_write(chip, 0x0c, data->cs4362a_fm); -} - -static void set_hdmi_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - struct xonar_data *data = chip->model_data; - - data->hdmi_params[0] = 0; /* 1 = non-audio */ - switch (params_rate(params)) { - case 44100: - data->hdmi_params[1] = IEC958_AES3_CON_FS_44100; - break; - case 48000: - data->hdmi_params[1] = IEC958_AES3_CON_FS_48000; - break; - default: /* 96000 */ - data->hdmi_params[1] = IEC958_AES3_CON_FS_96000; - break; - case 192000: - data->hdmi_params[1] = IEC958_AES3_CON_FS_192000; - break; - } - data->hdmi_params[2] = params_channels(params) / 2 - 1; - if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE) - data->hdmi_params[3] = 0; - else - data->hdmi_params[3] = 0xc0; - data->hdmi_params[4] = 1; /* ? */ - hdmi_write_command(chip, 0x54, 5, data->hdmi_params); -} - -static void set_hdav_params(struct oxygen *chip, - struct snd_pcm_hw_params *params) -{ - set_pcm1796_params(chip, params); - set_hdmi_params(chip, params); -} - -static void xonar_gpio_changed(struct oxygen *chip) -{ - struct xonar_data *data = chip->model_data; - u8 has_power; - - has_power = !!(oxygen_read8(chip, data->ext_power_reg) - & data->ext_power_bit); - if (has_power != data->has_power) { - data->has_power = has_power; - if (has_power) { - snd_printk(KERN_NOTICE "power restored\n"); - } else { - snd_printk(KERN_CRIT - "Hey! Don't unplug the power cable!\n"); - /* TODO: stop PCMs */ - } - } -} - -static void xonar_hdav_uart_input(struct oxygen *chip) -{ - if (chip->uart_input_count >= 2 && - chip->uart_input[chip->uart_input_count - 2] == 'O' && - chip->uart_input[chip->uart_input_count - 1] == 'K') { - printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n"); - print_hex_dump_bytes("", DUMP_PREFIX_OFFSET, - chip->uart_input, chip->uart_input_count); - chip->uart_input_count = 0; - } -} - -static int gpio_bit_switch_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 bit = ctl->private_value; - - value->value.integer.value[0] = - !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit); - return 0; -} - -static int gpio_bit_switch_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 bit = ctl->private_value; - u16 old_bits, new_bits; - int changed; - - spin_lock_irq(&chip->reg_lock); - old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); - if (value->value.integer.value[0]) - new_bits = old_bits | bit; - else - new_bits = old_bits & ~bit; - changed = new_bits != old_bits; - if (changed) - oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits); - spin_unlock_irq(&chip->reg_lock); - return changed; -} - -static const struct snd_kcontrol_new alt_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Loopback Switch", - .info = snd_ctl_boolean_mono_info, - .get = gpio_bit_switch_get, - .put = gpio_bit_switch_put, - .private_value = GPIO_D2_ALT, -}; - -static const struct snd_kcontrol_new front_panel_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Front Panel Switch", - .info = snd_ctl_boolean_mono_info, - .get = gpio_bit_switch_get, - .put = gpio_bit_switch_put, - .private_value = GPIO_DX_FRONT_PANEL, -}; - -static int st_output_switch_info(struct snd_kcontrol *ctl, - struct snd_ctl_elem_info *info) -{ - static const char *const names[3] = { - "Speakers", "Headphones", "FP Headphones" - }; - - info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - info->count = 1; - info->value.enumerated.items = 3; - if (info->value.enumerated.item >= 3) - info->value.enumerated.item = 2; - strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); - return 0; -} - -static int st_output_switch_get(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 gpio; - - gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA); - if (!(gpio & GPIO_ST_HP)) - value->value.enumerated.item[0] = 0; - else if (gpio & GPIO_ST_HP_REAR) - value->value.enumerated.item[0] = 1; - else - value->value.enumerated.item[0] = 2; - return 0; -} - - -static int st_output_switch_put(struct snd_kcontrol *ctl, - struct snd_ctl_elem_value *value) -{ - struct oxygen *chip = ctl->private_data; - u16 gpio_old, gpio; - - mutex_lock(&chip->mutex); - gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA); - gpio = gpio_old; - switch (value->value.enumerated.item[0]) { - case 0: - gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR); - break; - case 1: - gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR; - break; - case 2: - gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR; - break; - } - oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); - mutex_unlock(&chip->mutex); - return gpio != gpio_old; -} - -static const struct snd_kcontrol_new st_output_switch = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Analog Output", - .info = st_output_switch_info, - .get = st_output_switch_get, - .put = st_output_switch_put, -}; - -static void xonar_line_mic_ac97_switch(struct oxygen *chip, - unsigned int reg, unsigned int mute) -{ - if (reg == AC97_LINE) { - spin_lock_irq(&chip->reg_lock); - oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, - mute ? GPIO_DX_INPUT_ROUTE : 0, - GPIO_DX_INPUT_ROUTE); - spin_unlock_irq(&chip->reg_lock); - } -} - -static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0); -static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); - -static int xonar_d2_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - /* CD in is actually connected to the video in pin */ - template->private_value ^= AC97_CD ^ AC97_VIDEO; - return 0; -} - -static int xonar_d1_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - return 0; -} - -static int xonar_st_control_filter(struct snd_kcontrol_new *template) -{ - if (!strncmp(template->name, "CD Capture ", 11)) - return 1; /* no CD input */ - if (!strcmp(template->name, "Stereo Upmixing")) - return 1; /* stereo only - we don't need upmixing */ - return 0; -} - -static int xonar_d2_mixer_init(struct oxygen *chip) -{ - return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); -} - -static int xonar_d1_mixer_init(struct oxygen *chip) -{ - return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); -} - -static int xonar_st_mixer_init(struct oxygen *chip) -{ - return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip)); -} - -static const struct oxygen_model model_xonar_d2 = { - .longname = "Asus Virtuoso 200", - .chip = "AV200", - .init = xonar_d2_init, - .control_filter = xonar_d2_control_filter, - .mixer_init = xonar_d2_mixer_init, - .cleanup = xonar_d2_cleanup, - .suspend = xonar_d2_suspend, - .resume = xonar_d2_resume, - .set_dac_params = set_pcm1796_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_pcm1796_volume, - .update_dac_mute = update_pcm1796_mute, - .dac_tlv = pcm1796_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF | - MIDI_OUTPUT | - MIDI_INPUT, - .dac_channels = 8, - .dac_volume_min = 255 - 2*60, - .dac_volume_max = 255, - .misc_flags = OXYGEN_MISC_MIDI, - .function_flags = OXYGEN_FUNCTION_SPI | - OXYGEN_FUNCTION_ENABLE_SPI_4_5, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static const struct oxygen_model model_xonar_d1 = { - .longname = "Asus Virtuoso 100", - .chip = "AV200", - .init = xonar_d1_init, - .control_filter = xonar_d1_control_filter, - .mixer_init = xonar_d1_mixer_init, - .cleanup = xonar_d1_cleanup, - .suspend = xonar_d1_suspend, - .resume = xonar_d1_resume, - .set_dac_params = set_cs43xx_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_cs43xx_volume, - .update_dac_mute = update_cs43xx_mute, - .ac97_switch = xonar_line_mic_ac97_switch, - .dac_tlv = cs4362a_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, - .dac_channels = 8, - .dac_volume_min = 127 - 60, - .dac_volume_max = 127, - .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static const struct oxygen_model model_xonar_hdav = { - .longname = "Asus Virtuoso 200", - .chip = "AV200", - .init = xonar_hdav_init, - .cleanup = xonar_hdav_cleanup, - .suspend = xonar_hdav_suspend, - .resume = xonar_hdav_resume, - .pcm_hardware_filter = xonar_hdav_pcm_hardware_filter, - .set_dac_params = set_hdav_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_pcm1796_volume, - .update_dac_mute = update_pcm1796_mute, - .uart_input = xonar_hdav_uart_input, - .ac97_switch = xonar_line_mic_ac97_switch, - .dac_tlv = pcm1796_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF, - .dac_channels = 8, - .dac_volume_min = 255 - 2*60, - .dac_volume_max = 255, - .misc_flags = OXYGEN_MISC_MIDI, - .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - -static const struct oxygen_model model_xonar_st = { - .longname = "Asus Virtuoso 100", - .chip = "AV200", - .init = xonar_st_init, - .control_filter = xonar_st_control_filter, - .mixer_init = xonar_st_mixer_init, - .cleanup = xonar_st_cleanup, - .suspend = xonar_st_suspend, - .resume = xonar_st_resume, - .set_dac_params = set_pcm1796_params, - .set_adc_params = set_cs53x1_params, - .update_dac_volume = update_pcm1796_volume, - .update_dac_mute = update_pcm1796_mute, - .ac97_switch = xonar_line_mic_ac97_switch, - .dac_tlv = pcm1796_db_scale, - .model_data_size = sizeof(struct xonar_data), - .device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2, - .dac_channels = 2, - .dac_volume_min = 255 - 2*60, - .dac_volume_max = 255, - .function_flags = OXYGEN_FUNCTION_2WIRE, - .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, - .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, -}; - static int __devinit get_xonar_model(struct oxygen *chip, const struct pci_device_id *id) { - static const struct oxygen_model *const models[] = { - [MODEL_D1] = &model_xonar_d1, - [MODEL_DX] = &model_xonar_d1, - [MODEL_D2] = &model_xonar_d2, - [MODEL_D2X] = &model_xonar_d2, - [MODEL_HDAV] = &model_xonar_hdav, - [MODEL_ST] = &model_xonar_st, - [MODEL_STX] = &model_xonar_st, - }; - static const char *const names[] = { - [MODEL_D1] = "Xonar D1", - [MODEL_DX] = "Xonar DX", - [MODEL_D2] = "Xonar D2", - [MODEL_D2X] = "Xonar D2X", - [MODEL_HDAV] = "Xonar HDAV1.3", - [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", - [MODEL_ST] = "Xonar Essence ST", - [MODEL_ST_H6] = "Xonar Essence ST+H6", - [MODEL_STX] = "Xonar Essence STX", - }; - unsigned int model = id->driver_data; - - if (model >= ARRAY_SIZE(models) || !models[model]) - return -EINVAL; - chip->model = *models[model]; - - switch (model) { - case MODEL_D2X: - chip->model.init = xonar_d2x_init; - break; - case MODEL_DX: - chip->model.init = xonar_dx_init; - break; - case MODEL_HDAV: - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); - switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { - case GPIO_DB_H6: - model = MODEL_HDAV_H6; - break; - case GPIO_DB_XX: - snd_printk(KERN_ERR "unknown daughterboard\n"); - return -ENODEV; - } - break; - case MODEL_ST: - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); - switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { - case GPIO_DB_H6: - model = MODEL_ST_H6; - break; - } - break; - case MODEL_STX: - chip->model.init = xonar_stx_init; - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); - break; - } - - chip->model.shortname = names[model]; - chip->model.private_data = model; - return 0; + if (get_xonar_pcm179x_model(chip, id) >= 0) + return 0; + if (get_xonar_cs43xx_model(chip, id) >= 0) + return 0; + return -EINVAL; } static int __devinit xonar_probe(struct pci_dev *pci, diff --git a/sound/pci/oxygen/xonar.h b/sound/pci/oxygen/xonar.h new file mode 100644 index 00000000000..89b3ed814d6 --- /dev/null +++ b/sound/pci/oxygen/xonar.h @@ -0,0 +1,50 @@ +#ifndef XONAR_H_INCLUDED +#define XONAR_H_INCLUDED + +#include "oxygen.h" + +struct xonar_generic { + unsigned int anti_pop_delay; + u16 output_enable_bit; + u8 ext_power_reg; + u8 ext_power_int_reg; + u8 ext_power_bit; + u8 has_power; +}; + +struct xonar_hdmi { + u8 params[5]; +}; + +/* generic helper functions */ + +void xonar_enable_output(struct oxygen *chip); +void xonar_disable_output(struct oxygen *chip); +void xonar_init_ext_power(struct oxygen *chip); +void xonar_init_cs53x1(struct oxygen *chip); +void xonar_set_cs53x1_params(struct oxygen *chip, + struct snd_pcm_hw_params *params); +int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value); +int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value); + +/* model-specific card drivers */ + +int get_xonar_pcm179x_model(struct oxygen *chip, + const struct pci_device_id *id); +int get_xonar_cs43xx_model(struct oxygen *chip, + const struct pci_device_id *id); + +/* HDMI helper functions */ + +void xonar_hdmi_init(struct oxygen *chip, struct xonar_hdmi *data); +void xonar_hdmi_cleanup(struct oxygen *chip); +void xonar_hdmi_resume(struct oxygen *chip, struct xonar_hdmi *hdmi); +void xonar_hdmi_pcm_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware); +void xonar_set_hdmi_params(struct oxygen *chip, struct xonar_hdmi *hdmi, + struct snd_pcm_hw_params *params); +void xonar_hdmi_uart_input(struct oxygen *chip); + +#endif diff --git a/sound/pci/oxygen/xonar_cs43xx.c b/sound/pci/oxygen/xonar_cs43xx.c new file mode 100644 index 00000000000..16c226bfcd2 --- /dev/null +++ b/sound/pci/oxygen/xonar_cs43xx.c @@ -0,0 +1,434 @@ +/* + * card driver for models with CS4398/CS4362A DACs (Xonar D1/DX) + * + * Copyright (c) Clemens Ladisch <clemens@ladisch.de> + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see <http://www.gnu.org/licenses/>. + */ + +/* + * Xonar D1/DX + * ----------- + * + * CMI8788: + * + * I²C <-> CS4398 (front) + * <-> CS4362A (surround, center/LFE, back) + * + * GPI 0 <- external power present (DX only) + * + * GPIO 0 -> enable output to speakers + * GPIO 1 -> enable front panel I/O + * GPIO 2 -> M0 of CS5361 + * GPIO 3 -> M1 of CS5361 + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * CS4398: + * + * AD0 <- 1 + * AD1 <- 1 + * + * CS4362A: + * + * AD0 <- 0 + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5361 input + */ + +#include <linux/pci.h> +#include <linux/delay.h> +#include <sound/ac97_codec.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include "xonar.h" +#include "cs4398.h" +#include "cs4362a.h" + +#define GPI_EXT_POWER 0x01 +#define GPIO_D1_OUTPUT_ENABLE 0x0001 +#define GPIO_D1_FRONT_PANEL 0x0002 +#define GPIO_D1_INPUT_ROUTE 0x0100 + +#define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ +#define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ + +struct xonar_cs43xx { + struct xonar_generic generic; + u8 cs4398_regs[8]; + u8 cs4362a_regs[15]; +}; + +static void cs4398_write(struct oxygen *chip, u8 reg, u8 value) +{ + struct xonar_cs43xx *data = chip->model_data; + + oxygen_write_i2c(chip, I2C_DEVICE_CS4398, reg, value); + if (reg < ARRAY_SIZE(data->cs4398_regs)) + data->cs4398_regs[reg] = value; +} + +static void cs4398_write_cached(struct oxygen *chip, u8 reg, u8 value) +{ + struct xonar_cs43xx *data = chip->model_data; + + if (value != data->cs4398_regs[reg]) + cs4398_write(chip, reg, value); +} + +static void cs4362a_write(struct oxygen *chip, u8 reg, u8 value) +{ + struct xonar_cs43xx *data = chip->model_data; + + oxygen_write_i2c(chip, I2C_DEVICE_CS4362A, reg, value); + if (reg < ARRAY_SIZE(data->cs4362a_regs)) + data->cs4362a_regs[reg] = value; +} + +static void cs4362a_write_cached(struct oxygen *chip, u8 reg, u8 value) +{ + struct xonar_cs43xx *data = chip->model_data; + + if (value != data->cs4362a_regs[reg]) + cs4362a_write(chip, reg, value); +} + +static void cs43xx_registers_init(struct oxygen *chip) +{ + struct xonar_cs43xx *data = chip->model_data; + unsigned int i; + + /* set CPEN (control port mode) and power down */ + cs4398_write(chip, 8, CS4398_CPEN | CS4398_PDN); + cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); + /* configure */ + cs4398_write(chip, 2, data->cs4398_regs[2]); + cs4398_write(chip, 3, CS4398_ATAPI_B_R | CS4398_ATAPI_A_L); + cs4398_write(chip, 4, data->cs4398_regs[4]); + cs4398_write(chip, 5, data->cs4398_regs[5]); + cs4398_write(chip, 6, data->cs4398_regs[6]); + cs4398_write(chip, 7, data->cs4398_regs[7]); + cs4362a_write(chip, 0x02, CS4362A_DIF_LJUST); + cs4362a_write(chip, 0x03, CS4362A_MUTEC_6 | CS4362A_AMUTE | + CS4362A_RMP_UP | CS4362A_ZERO_CROSS | CS4362A_SOFT_RAMP); + cs4362a_write(chip, 0x04, data->cs4362a_regs[0x04]); + cs4362a_write(chip, 0x05, 0); + for (i = 6; i <= 14; ++i) + cs4362a_write(chip, i, data->cs4362a_regs[i]); + /* clear power down */ + cs4398_write(chip, 8, CS4398_CPEN); + cs4362a_write(chip, 0x01, CS4362A_CPEN); +} + +static void xonar_d1_init(struct oxygen *chip) +{ + struct xonar_cs43xx *data = chip->model_data; + + data->generic.anti_pop_delay = 800; + data->generic.output_enable_bit = GPIO_D1_OUTPUT_ENABLE; + data->cs4398_regs[2] = + CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; + data->cs4398_regs[4] = CS4398_MUTEP_LOW | + CS4398_MUTE_B | CS4398_MUTE_A | CS4398_PAMUTE; + data->cs4398_regs[5] = 60 * 2; + data->cs4398_regs[6] = 60 * 2; + data->cs4398_regs[7] = CS4398_RMP_DN | CS4398_RMP_UP | + CS4398_ZERO_CROSS | CS4398_SOFT_RAMP; + data->cs4362a_regs[4] = CS4362A_RMP_DN | CS4362A_DEM_NONE; + data->cs4362a_regs[6] = CS4362A_FM_SINGLE | + CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + data->cs4362a_regs[7] = 60 | CS4362A_MUTE; + data->cs4362a_regs[8] = 60 | CS4362A_MUTE; + data->cs4362a_regs[9] = data->cs4362a_regs[6]; + data->cs4362a_regs[10] = 60 | CS4362A_MUTE; + data->cs4362a_regs[11] = 60 | CS4362A_MUTE; + data->cs4362a_regs[12] = data->cs4362a_regs[6]; + data->cs4362a_regs[13] = 60 | CS4362A_MUTE; + data->cs4362a_regs[14] = 60 | CS4362A_MUTE; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + cs43xx_registers_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_D1_FRONT_PANEL | GPIO_D1_INPUT_ROUTE); + + xonar_init_cs53x1(chip); + xonar_enable_output(chip); + + snd_component_add(chip->card, "CS4398"); + snd_component_add(chip->card, "CS4362A"); + snd_component_add(chip->card, "CS5361"); +} + +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_cs43xx *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_d1_init(chip); +} + +static void xonar_d1_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); + cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); + oxygen_clear_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); +} + +static void xonar_d1_suspend(struct oxygen *chip) +{ + xonar_d1_cleanup(chip); +} + +static void xonar_d1_resume(struct oxygen *chip) +{ + oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); + msleep(1); + cs43xx_registers_init(chip); + xonar_enable_output(chip); +} + +static void set_cs43xx_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_cs43xx *data = chip->model_data; + u8 cs4398_fm, cs4362a_fm; + + if (params_rate(params) <= 50000) { + cs4398_fm = CS4398_FM_SINGLE; + cs4362a_fm = CS4362A_FM_SINGLE; + } else if (params_rate(params) <= 100000) { + cs4398_fm = CS4398_FM_DOUBLE; + cs4362a_fm = CS4362A_FM_DOUBLE; + } else { + cs4398_fm = CS4398_FM_QUAD; + cs4362a_fm = CS4362A_FM_QUAD; + } + cs4398_fm |= CS4398_DEM_NONE | CS4398_DIF_LJUST; + cs4398_write_cached(chip, 2, cs4398_fm); + cs4362a_fm |= data->cs4362a_regs[6] & ~CS4362A_FM_MASK; + cs4362a_write_cached(chip, 6, cs4362a_fm); + cs4362a_write_cached(chip, 12, cs4362a_fm); + cs4362a_fm &= CS4362A_FM_MASK; + cs4362a_fm |= data->cs4362a_regs[9] & ~CS4362A_FM_MASK; + cs4362a_write_cached(chip, 9, cs4362a_fm); +} + +static void update_cs4362a_volumes(struct oxygen *chip) +{ + unsigned int i; + u8 mute; + + mute = chip->dac_mute ? CS4362A_MUTE : 0; + for (i = 0; i < 6; ++i) + cs4362a_write_cached(chip, 7 + i + i / 2, + (127 - chip->dac_volume[2 + i]) | mute); +} + +static void update_cs43xx_volume(struct oxygen *chip) +{ + cs4398_write_cached(chip, 5, (127 - chip->dac_volume[0]) * 2); + cs4398_write_cached(chip, 6, (127 - chip->dac_volume[1]) * 2); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_mute(struct oxygen *chip) +{ + u8 reg; + + reg = CS4398_MUTEP_LOW | CS4398_PAMUTE; + if (chip->dac_mute) + reg |= CS4398_MUTE_B | CS4398_MUTE_A; + cs4398_write_cached(chip, 4, reg); + update_cs4362a_volumes(chip); +} + +static void update_cs43xx_center_lfe_mix(struct oxygen *chip, bool mixed) +{ + struct xonar_cs43xx *data = chip->model_data; + u8 reg; + + reg = data->cs4362a_regs[9] & ~CS4362A_ATAPI_MASK; + if (mixed) + reg |= CS4362A_ATAPI_B_LR | CS4362A_ATAPI_A_LR; + else + reg |= CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; + cs4362a_write_cached(chip, 9, reg); +} + +static const struct snd_kcontrol_new front_panel_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Front Panel Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_D1_FRONT_PANEL, +}; + +static int rolloff_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "Fast Roll-off", "Slow Roll-off" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int rolloff_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_cs43xx *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->cs4398_regs[7] & CS4398_FILT_SEL) != 0; + return 0; +} + +static int rolloff_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_cs43xx *data = chip->model_data; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->cs4398_regs[7]; + if (value->value.enumerated.item[0]) + reg |= CS4398_FILT_SEL; + else + reg &= ~CS4398_FILT_SEL; + changed = reg != data->cs4398_regs[7]; + if (changed) { + cs4398_write(chip, 7, reg); + if (reg & CS4398_FILT_SEL) + reg = data->cs4362a_regs[0x04] | CS4362A_FILT_SEL; + else + reg = data->cs4362a_regs[0x04] & ~CS4362A_FILT_SEL; + cs4362a_write(chip, 0x04, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new rolloff_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Filter Playback Enum", + .info = rolloff_info, + .get = rolloff_get, + .put = rolloff_put, +}; + +static void xonar_d1_line_mic_ac97_switch(struct oxygen *chip, + unsigned int reg, unsigned int mute) +{ + if (reg == AC97_LINE) { + spin_lock_irq(&chip->reg_lock); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + mute ? GPIO_D1_INPUT_ROUTE : 0, + GPIO_D1_INPUT_ROUTE); + spin_unlock_irq(&chip->reg_lock); + } +} + +static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); + +static int xonar_d1_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + return 0; +} + +static int xonar_d1_mixer_init(struct oxygen *chip) +{ + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); + if (err < 0) + return err; + return 0; +} + +static const struct oxygen_model model_xonar_d1 = { + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .init = xonar_d1_init, + .control_filter = xonar_d1_control_filter, + .mixer_init = xonar_d1_mixer_init, + .cleanup = xonar_d1_cleanup, + .suspend = xonar_d1_suspend, + .resume = xonar_d1_resume, + .get_i2s_mclk = oxygen_default_i2s_mclk, + .set_dac_params = set_cs43xx_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_cs43xx_volume, + .update_dac_mute = update_cs43xx_mute, + .update_center_lfe_mix = update_cs43xx_center_lfe_mix, + .ac97_switch = xonar_d1_line_mic_ac97_switch, + .dac_tlv = cs4362a_db_scale, + .model_data_size = sizeof(struct xonar_cs43xx), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 8, + .dac_volume_min = 127 - 60, + .dac_volume_max = 127, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +int __devinit get_xonar_cs43xx_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + switch (id->subdevice) { + case 0x834f: + chip->model = model_xonar_d1; + chip->model.shortname = "Xonar D1"; + break; + case 0x8275: + case 0x8327: + chip->model = model_xonar_d1; + chip->model.shortname = "Xonar DX"; + chip->model.init = xonar_dx_init; + break; + default: + return -EINVAL; + } + return 0; +} diff --git a/sound/pci/oxygen/xonar_hdmi.c b/sound/pci/oxygen/xonar_hdmi.c new file mode 100644 index 00000000000..b12db1f1cea --- /dev/null +++ b/sound/pci/oxygen/xonar_hdmi.c @@ -0,0 +1,128 @@ +/* + * helper functions for HDMI models (Xonar HDAV1.3) + * + * Copyright (c) Clemens Ladisch <clemens@ladisch.de> + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/pci.h> +#include <linux/delay.h> +#include <sound/asoundef.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include "xonar.h" + +static void hdmi_write_command(struct oxygen *chip, u8 command, + unsigned int count, const u8 *params) +{ + unsigned int i; + u8 checksum; + + oxygen_write_uart(chip, 0xfb); + oxygen_write_uart(chip, 0xef); + oxygen_write_uart(chip, command); + oxygen_write_uart(chip, count); + for (i = 0; i < count; ++i) + oxygen_write_uart(chip, params[i]); + checksum = 0xfb + 0xef + command + count; + for (i = 0; i < count; ++i) + checksum += params[i]; + oxygen_write_uart(chip, checksum); +} + +static void xonar_hdmi_init_commands(struct oxygen *chip, + struct xonar_hdmi *hdmi) +{ + u8 param; + + oxygen_reset_uart(chip); + param = 0; + hdmi_write_command(chip, 0x61, 1, ¶m); + param = 1; + hdmi_write_command(chip, 0x74, 1, ¶m); + hdmi_write_command(chip, 0x54, 5, hdmi->params); +} + +void xonar_hdmi_init(struct oxygen *chip, struct xonar_hdmi *hdmi) +{ + hdmi->params[1] = IEC958_AES3_CON_FS_48000; + hdmi->params[4] = 1; + xonar_hdmi_init_commands(chip, hdmi); +} + +void xonar_hdmi_cleanup(struct oxygen *chip) +{ + u8 param = 0; + + hdmi_write_command(chip, 0x74, 1, ¶m); +} + +void xonar_hdmi_resume(struct oxygen *chip, struct xonar_hdmi *hdmi) +{ + xonar_hdmi_init_commands(chip, hdmi); +} + +void xonar_hdmi_pcm_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + if (channel == PCM_MULTICH) { + hardware->rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000; + hardware->rate_min = 44100; + } +} + +void xonar_set_hdmi_params(struct oxygen *chip, struct xonar_hdmi *hdmi, + struct snd_pcm_hw_params *params) +{ + hdmi->params[0] = 0; /* 1 = non-audio */ + switch (params_rate(params)) { + case 44100: + hdmi->params[1] = IEC958_AES3_CON_FS_44100; + break; + case 48000: + hdmi->params[1] = IEC958_AES3_CON_FS_48000; + break; + default: /* 96000 */ + hdmi->params[1] = IEC958_AES3_CON_FS_96000; + break; + case 192000: + hdmi->params[1] = IEC958_AES3_CON_FS_192000; + break; + } + hdmi->params[2] = params_channels(params) / 2 - 1; + if (params_format(params) == SNDRV_PCM_FORMAT_S16_LE) + hdmi->params[3] = 0; + else + hdmi->params[3] = 0xc0; + hdmi->params[4] = 1; /* ? */ + hdmi_write_command(chip, 0x54, 5, hdmi->params); +} + +void xonar_hdmi_uart_input(struct oxygen *chip) +{ + if (chip->uart_input_count >= 2 && + chip->uart_input[chip->uart_input_count - 2] == 'O' && + chip->uart_input[chip->uart_input_count - 1] == 'K') { + printk(KERN_DEBUG "message from HDMI chip received:\n"); + print_hex_dump_bytes("", DUMP_PREFIX_OFFSET, + chip->uart_input, chip->uart_input_count); + chip->uart_input_count = 0; + } +} diff --git a/sound/pci/oxygen/xonar_lib.c b/sound/pci/oxygen/xonar_lib.c new file mode 100644 index 00000000000..b3ff7131665 --- /dev/null +++ b/sound/pci/oxygen/xonar_lib.c @@ -0,0 +1,132 @@ +/* + * helper functions for Asus Xonar cards + * + * Copyright (c) Clemens Ladisch <clemens@ladisch.de> + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "xonar.h" + + +#define GPIO_CS53x1_M_MASK 0x000c +#define GPIO_CS53x1_M_SINGLE 0x0000 +#define GPIO_CS53x1_M_DOUBLE 0x0004 +#define GPIO_CS53x1_M_QUAD 0x0008 + + +void xonar_enable_output(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, data->output_enable_bit); + msleep(data->anti_pop_delay); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + +void xonar_disable_output(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); +} + +static void xonar_ext_power_gpio_changed(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + u8 has_power; + + has_power = !!(oxygen_read8(chip, data->ext_power_reg) + & data->ext_power_bit); + if (has_power != data->has_power) { + data->has_power = has_power; + if (has_power) { + snd_printk(KERN_NOTICE "power restored\n"); + } else { + snd_printk(KERN_CRIT + "Hey! Don't unplug the power cable!\n"); + /* TODO: stop PCMs */ + } + } +} + +void xonar_init_ext_power(struct oxygen *chip) +{ + struct xonar_generic *data = chip->model_data; + + oxygen_set_bits8(chip, data->ext_power_int_reg, + data->ext_power_bit); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + chip->model.gpio_changed = xonar_ext_power_gpio_changed; + data->has_power = !!(oxygen_read8(chip, data->ext_power_reg) + & data->ext_power_bit); +} + +void xonar_init_cs53x1(struct oxygen *chip) +{ + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CS53x1_M_MASK); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + GPIO_CS53x1_M_SINGLE, GPIO_CS53x1_M_MASK); +} + +void xonar_set_cs53x1_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + unsigned int value; + + if (params_rate(params) <= 54000) + value = GPIO_CS53x1_M_SINGLE; + else if (params_rate(params) <= 108000) + value = GPIO_CS53x1_M_DOUBLE; + else + value = GPIO_CS53x1_M_QUAD; + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + value, GPIO_CS53x1_M_MASK); +} + +int xonar_gpio_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 bit = ctl->private_value; + + value->value.integer.value[0] = + !!(oxygen_read16(chip, OXYGEN_GPIO_DATA) & bit); + return 0; +} + +int xonar_gpio_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 bit = ctl->private_value; + u16 old_bits, new_bits; + int changed; + + spin_lock_irq(&chip->reg_lock); + old_bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (value->value.integer.value[0]) + new_bits = old_bits | bit; + else + new_bits = old_bits & ~bit; + changed = new_bits != old_bits; + if (changed) + oxygen_write16(chip, OXYGEN_GPIO_DATA, new_bits); + spin_unlock_irq(&chip->reg_lock); + return changed; +} diff --git a/sound/pci/oxygen/xonar_pcm179x.c b/sound/pci/oxygen/xonar_pcm179x.c new file mode 100644 index 00000000000..ba18fb546b4 --- /dev/null +++ b/sound/pci/oxygen/xonar_pcm179x.c @@ -0,0 +1,1115 @@ +/* + * card driver for models with PCM1796 DACs (Xonar D2/D2X/HDAV1.3/ST/STX) + * + * Copyright (c) Clemens Ladisch <clemens@ladisch.de> + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this driver; if not, see <http://www.gnu.org/licenses/>. + */ + +/* + * Xonar D2/D2X + * ------------ + * + * CMI8788: + * + * SPI 0 -> 1st PCM1796 (front) + * SPI 1 -> 2nd PCM1796 (surround) + * SPI 2 -> 3rd PCM1796 (center/LFE) + * SPI 4 -> 4th PCM1796 (back) + * + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 5 <- external power present (D2X only) + * GPIO 7 -> ALT + * GPIO 8 -> enable output to speakers + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + */ + +/* + * Xonar HDAV1.3 (Deluxe) + * ---------------------- + * + * CMI8788: + * + * I²C <-> PCM1796 (front) + * + * GPI 0 <- external power present + * + * GPIO 0 -> enable output to speakers + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * TXD -> HDMI controller + * RXD <- HDMI controller + * + * PCM1796 front: AD1,0 <- 0,0 + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * + * no daughterboard + * ---------------- + * + * GPIO 4 <- 1 + * + * H6 daughterboard + * ---------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 0 + * + * I²C <-> PCM1796 (surround) + * <-> PCM1796 (center/LFE) + * <-> PCM1796 (back) + * + * PCM1796 surround: AD1,0 <- 0,1 + * PCM1796 center/LFE: AD1,0 <- 1,0 + * PCM1796 back: AD1,0 <- 1,1 + * + * unknown daughterboard + * --------------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 1 + * + * I²C <-> CS4362A (surround, center/LFE, back) + * + * CS4362A: AD0 <- 0 + */ + +/* + * Xonar Essence ST (Deluxe)/STX + * ----------------------------- + * + * CMI8788: + * + * I²C <-> PCM1792A + * <-> CS2000 (ST only) + * + * ADC1 MCLK -> REF_CLK of CS2000 (ST only) + * + * GPI 0 <- external power present (STX only) + * + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 7 -> route output to speaker jacks (0) or HP (1) + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * PCM1792A: + * + * AD1,0 <- 0,0 + * SCK <- CLK_OUT of CS2000 (ST only) + * + * CS2000: + * + * AD0 <- 0 + * + * CM9780: + * + * GPO 0 -> route line-in (0) or AC97 output (1) to CS5381 input + * + * H6 daughterboard + * ---------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 0 + */ + +#include <linux/pci.h> +#include <linux/delay.h> +#include <linux/mutex.h> +#include <sound/ac97_codec.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include "xonar.h" +#include "cm9780.h" +#include "pcm1796.h" +#include "cs2000.h" + + +#define GPIO_D2X_EXT_POWER 0x0020 +#define GPIO_D2_ALT 0x0080 +#define GPIO_D2_OUTPUT_ENABLE 0x0100 + +#define GPI_EXT_POWER 0x01 +#define GPIO_INPUT_ROUTE 0x0100 + +#define GPIO_HDAV_OUTPUT_ENABLE 0x0001 + +#define GPIO_DB_MASK 0x0030 +#define GPIO_DB_H6 0x0000 + +#define GPIO_ST_OUTPUT_ENABLE 0x0001 +#define GPIO_ST_HP_REAR 0x0002 +#define GPIO_ST_HP 0x0080 + +#define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ii, /W=0 */ +#define I2C_DEVICE_CS2000 0x9c /* 100111, 0, /W=0 */ + +#define PCM1796_REG_BASE 16 + + +struct xonar_pcm179x { + struct xonar_generic generic; + unsigned int dacs; + u8 pcm1796_regs[4][5]; + unsigned int current_rate; + bool os_128; + bool hp_active; + s8 hp_gain_offset; + bool has_cs2000; + u8 cs2000_fun_cfg_1; +}; + +struct xonar_hdav { + struct xonar_pcm179x pcm179x; + struct xonar_hdmi hdmi; +}; + + +static inline void pcm1796_write_spi(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + /* maps ALSA channel pair number to SPI output */ + static const u8 codec_map[4] = { + 0, 1, 2, 4 + }; + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (codec_map[codec] << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_HI, + (reg << 8) | value); +} + +static inline void pcm1796_write_i2c(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + oxygen_write_i2c(chip, I2C_DEVICE_PCM1796(codec), reg, value); +} + +static void pcm1796_write(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + struct xonar_pcm179x *data = chip->model_data; + + if ((chip->model.function_flags & OXYGEN_FUNCTION_2WIRE_SPI_MASK) == + OXYGEN_FUNCTION_SPI) + pcm1796_write_spi(chip, codec, reg, value); + else + pcm1796_write_i2c(chip, codec, reg, value); + if ((unsigned int)(reg - PCM1796_REG_BASE) + < ARRAY_SIZE(data->pcm1796_regs[codec])) + data->pcm1796_regs[codec][reg - PCM1796_REG_BASE] = value; +} + +static void pcm1796_write_cached(struct oxygen *chip, unsigned int codec, + u8 reg, u8 value) +{ + struct xonar_pcm179x *data = chip->model_data; + + if (value != data->pcm1796_regs[codec][reg - PCM1796_REG_BASE]) + pcm1796_write(chip, codec, reg, value); +} + +static void cs2000_write(struct oxygen *chip, u8 reg, u8 value) +{ + struct xonar_pcm179x *data = chip->model_data; + + oxygen_write_i2c(chip, I2C_DEVICE_CS2000, reg, value); + if (reg == CS2000_FUN_CFG_1) + data->cs2000_fun_cfg_1 = value; +} + +static void cs2000_write_cached(struct oxygen *chip, u8 reg, u8 value) +{ + struct xonar_pcm179x *data = chip->model_data; + + if (reg != CS2000_FUN_CFG_1 || + value != data->cs2000_fun_cfg_1) + cs2000_write(chip, reg, value); +} + +static void pcm1796_registers_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + s8 gain_offset; + + gain_offset = data->hp_active ? data->hp_gain_offset : 0; + for (i = 0; i < data->dacs; ++i) { + /* set ATLD before ATL/ATR */ + pcm1796_write(chip, i, 18, + data->pcm1796_regs[0][18 - PCM1796_REG_BASE]); + pcm1796_write(chip, i, 16, chip->dac_volume[i * 2] + + gain_offset); + pcm1796_write(chip, i, 17, chip->dac_volume[i * 2 + 1] + + gain_offset); + pcm1796_write(chip, i, 19, + data->pcm1796_regs[0][19 - PCM1796_REG_BASE]); + pcm1796_write(chip, i, 20, + data->pcm1796_regs[0][20 - PCM1796_REG_BASE]); + pcm1796_write(chip, i, 21, 0); + } +} + +static void pcm1796_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->pcm1796_regs[0][18 - PCM1796_REG_BASE] = PCM1796_MUTE | + PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + data->pcm1796_regs[0][19 - PCM1796_REG_BASE] = + PCM1796_FLT_SHARP | PCM1796_ATS_1; + data->pcm1796_regs[0][20 - PCM1796_REG_BASE] = PCM1796_OS_64; + pcm1796_registers_init(chip); + data->current_rate = 48000; +} + +static void xonar_d2_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.anti_pop_delay = 300; + data->generic.output_enable_bit = GPIO_D2_OUTPUT_ENABLE; + data->dacs = 4; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2_ALT); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_D2_ALT); + + oxygen_ac97_set_bits(chip, 0, CM9780_JACK, CM9780_FMIC2MIC); + + xonar_init_cs53x1(chip); + xonar_enable_output(chip); + + snd_component_add(chip->card, "PCM1796"); + snd_component_add(chip->card, "CS5381"); +} + +static void xonar_d2x_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.ext_power_reg = OXYGEN_GPIO_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; + data->generic.ext_power_bit = GPIO_D2X_EXT_POWER; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER); + xonar_init_ext_power(chip); + xonar_d2_init(chip); +} + +static void xonar_hdav_init(struct oxygen *chip) +{ + struct xonar_hdav *data = chip->model_data; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + data->pcm179x.generic.anti_pop_delay = 100; + data->pcm179x.generic.output_enable_bit = GPIO_HDAV_OUTPUT_ENABLE; + data->pcm179x.generic.ext_power_reg = OXYGEN_GPI_DATA; + data->pcm179x.generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->pcm179x.generic.ext_power_bit = GPI_EXT_POWER; + data->pcm179x.dacs = chip->model.private_data ? 4 : 1; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_INPUT_ROUTE); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_INPUT_ROUTE); + + xonar_init_cs53x1(chip); + xonar_init_ext_power(chip); + xonar_hdmi_init(chip, &data->hdmi); + xonar_enable_output(chip); + + snd_component_add(chip->card, "PCM1796"); + snd_component_add(chip->card, "CS5381"); +} + +static void xonar_st_init_i2c(struct oxygen *chip) +{ + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); +} + +static void xonar_st_init_common(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->generic.anti_pop_delay = 100; + data->generic.output_enable_bit = GPIO_ST_OUTPUT_ENABLE; + data->dacs = chip->model.private_data ? 4 : 1; + data->hp_gain_offset = 2*-18; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + + xonar_init_cs53x1(chip); + xonar_enable_output(chip); + + snd_component_add(chip->card, "PCM1792A"); + snd_component_add(chip->card, "CS5381"); +} + +static void cs2000_registers_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + cs2000_write(chip, CS2000_GLOBAL_CFG, CS2000_FREEZE); + cs2000_write(chip, CS2000_DEV_CTRL, 0); + cs2000_write(chip, CS2000_DEV_CFG_1, + CS2000_R_MOD_SEL_1 | + (0 << CS2000_R_SEL_SHIFT) | + CS2000_AUX_OUT_SRC_REF_CLK | + CS2000_EN_DEV_CFG_1); + cs2000_write(chip, CS2000_DEV_CFG_2, + (0 << CS2000_LOCK_CLK_SHIFT) | + CS2000_FRAC_N_SRC_STATIC); + cs2000_write(chip, CS2000_RATIO_0 + 0, 0x00); /* 1.0 */ + cs2000_write(chip, CS2000_RATIO_0 + 1, 0x10); + cs2000_write(chip, CS2000_RATIO_0 + 2, 0x00); + cs2000_write(chip, CS2000_RATIO_0 + 3, 0x00); + cs2000_write(chip, CS2000_FUN_CFG_1, data->cs2000_fun_cfg_1); + cs2000_write(chip, CS2000_FUN_CFG_2, 0); + cs2000_write(chip, CS2000_GLOBAL_CFG, CS2000_EN_DEV_CFG_2); +} + +static void xonar_st_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->has_cs2000 = 1; + data->cs2000_fun_cfg_1 = CS2000_REF_CLK_DIV_1; + + oxygen_write16(chip, OXYGEN_I2S_A_FORMAT, + OXYGEN_RATE_48000 | OXYGEN_I2S_FORMAT_I2S | + OXYGEN_I2S_MCLK_128 | OXYGEN_I2S_BITS_16 | + OXYGEN_I2S_MASTER | OXYGEN_I2S_BCLK_64); + + xonar_st_init_i2c(chip); + cs2000_registers_init(chip); + xonar_st_init_common(chip); + + snd_component_add(chip->card, "CS2000"); +} + +static void xonar_stx_init(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + + xonar_st_init_i2c(chip); + data->generic.ext_power_reg = OXYGEN_GPI_DATA; + data->generic.ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->generic.ext_power_bit = GPI_EXT_POWER; + xonar_init_ext_power(chip); + xonar_st_init_common(chip); +} + +static void xonar_d2_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + +static void xonar_hdav_cleanup(struct oxygen *chip) +{ + xonar_hdmi_cleanup(chip); + xonar_disable_output(chip); + msleep(2); +} + +static void xonar_st_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + +static void xonar_d2_suspend(struct oxygen *chip) +{ + xonar_d2_cleanup(chip); +} + +static void xonar_hdav_suspend(struct oxygen *chip) +{ + xonar_hdav_cleanup(chip); +} + +static void xonar_st_suspend(struct oxygen *chip) +{ + xonar_st_cleanup(chip); +} + +static void xonar_d2_resume(struct oxygen *chip) +{ + pcm1796_registers_init(chip); + xonar_enable_output(chip); +} + +static void xonar_hdav_resume(struct oxygen *chip) +{ + struct xonar_hdav *data = chip->model_data; + + pcm1796_registers_init(chip); + xonar_hdmi_resume(chip, &data->hdmi); + xonar_enable_output(chip); +} + +static void xonar_stx_resume(struct oxygen *chip) +{ + pcm1796_registers_init(chip); + xonar_enable_output(chip); +} + +static void xonar_st_resume(struct oxygen *chip) +{ + cs2000_registers_init(chip); + xonar_stx_resume(chip); +} + +static unsigned int mclk_from_rate(struct oxygen *chip, unsigned int rate) +{ + struct xonar_pcm179x *data = chip->model_data; + + if (rate <= 32000) + return OXYGEN_I2S_MCLK_512; + else if (rate <= 48000 && data->os_128) + return OXYGEN_I2S_MCLK_512; + else if (rate <= 96000) + return OXYGEN_I2S_MCLK_256; + else + return OXYGEN_I2S_MCLK_128; +} + +static unsigned int get_pcm1796_i2s_mclk(struct oxygen *chip, + unsigned int channel, + struct snd_pcm_hw_params *params) +{ + if (channel == PCM_MULTICH) + return mclk_from_rate(chip, params_rate(params)); + else + return oxygen_default_i2s_mclk(chip, channel, params); +} + +static void update_pcm1796_oversampling(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + u8 reg; + + if (data->current_rate <= 32000) + reg = PCM1796_OS_128; + else if (data->current_rate <= 48000 && data->os_128) + reg = PCM1796_OS_128; + else if (data->current_rate <= 96000 || data->os_128) + reg = PCM1796_OS_64; + else + reg = PCM1796_OS_32; + for (i = 0; i < data->dacs; ++i) + pcm1796_write_cached(chip, i, 20, reg); +} + +static void set_pcm1796_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_pcm179x *data = chip->model_data; + + data->current_rate = params_rate(params); + update_pcm1796_oversampling(chip); +} + +static void update_pcm1796_volume(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + s8 gain_offset; + + gain_offset = data->hp_active ? data->hp_gain_offset : 0; + for (i = 0; i < data->dacs; ++i) { + pcm1796_write_cached(chip, i, 16, chip->dac_volume[i * 2] + + gain_offset); + pcm1796_write_cached(chip, i, 17, chip->dac_volume[i * 2 + 1] + + gain_offset); + } +} + +static void update_pcm1796_mute(struct oxygen *chip) +{ + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + u8 value; + + value = PCM1796_DMF_DISABLED | PCM1796_FMT_24_LJUST | PCM1796_ATLD; + if (chip->dac_mute) + value |= PCM1796_MUTE; + for (i = 0; i < data->dacs; ++i) + pcm1796_write_cached(chip, i, 18, value); +} + +static void update_cs2000_rate(struct oxygen *chip, unsigned int rate) +{ + struct xonar_pcm179x *data = chip->model_data; + u8 rate_mclk, reg; + + switch (rate) { + /* XXX Why is the I2S A MCLK half the actual I2S MCLK? */ + case 32000: + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + break; + case 44100: + if (data->os_128) + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + else + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_128; + break; + default: /* 48000 */ + if (data->os_128) + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + else + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_128; + break; + case 64000: + rate_mclk = OXYGEN_RATE_32000 | OXYGEN_I2S_MCLK_256; + break; + case 88200: + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + break; + case 96000: + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + break; + case 176400: + rate_mclk = OXYGEN_RATE_44100 | OXYGEN_I2S_MCLK_256; + break; + case 192000: + rate_mclk = OXYGEN_RATE_48000 | OXYGEN_I2S_MCLK_256; + break; + } + oxygen_write16_masked(chip, OXYGEN_I2S_A_FORMAT, rate_mclk, + OXYGEN_I2S_RATE_MASK | OXYGEN_I2S_MCLK_MASK); + if ((rate_mclk & OXYGEN_I2S_MCLK_MASK) <= OXYGEN_I2S_MCLK_128) + reg = CS2000_REF_CLK_DIV_1; + else + reg = CS2000_REF_CLK_DIV_2; + cs2000_write_cached(chip, CS2000_FUN_CFG_1, reg); +} + +static void set_st_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + update_cs2000_rate(chip, params_rate(params)); + set_pcm1796_params(chip, params); +} + +static void set_hdav_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + struct xonar_hdav *data = chip->model_data; + + set_pcm1796_params(chip, params); + xonar_set_hdmi_params(chip, &data->hdmi, params); +} + +static const struct snd_kcontrol_new alt_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Loopback Switch", + .info = snd_ctl_boolean_mono_info, + .get = xonar_gpio_bit_switch_get, + .put = xonar_gpio_bit_switch_put, + .private_value = GPIO_D2_ALT, +}; + +static int rolloff_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "Sharp Roll-off", "Slow Roll-off" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int rolloff_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + value->value.enumerated.item[0] = + (data->pcm1796_regs[0][19 - PCM1796_REG_BASE] & + PCM1796_FLT_MASK) != PCM1796_FLT_SHARP; + return 0; +} + +static int rolloff_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + unsigned int i; + int changed; + u8 reg; + + mutex_lock(&chip->mutex); + reg = data->pcm1796_regs[0][19 - PCM1796_REG_BASE]; + reg &= ~PCM1796_FLT_MASK; + if (!value->value.enumerated.item[0]) + reg |= PCM1796_FLT_SHARP; + else + reg |= PCM1796_FLT_SLOW; + changed = reg != data->pcm1796_regs[0][19 - PCM1796_REG_BASE]; + if (changed) { + for (i = 0; i < data->dacs; ++i) + pcm1796_write(chip, i, 19, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new rolloff_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Filter Playback Enum", + .info = rolloff_info, + .get = rolloff_get, + .put = rolloff_put, +}; + +static int os_128_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { "64x", "128x" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int os_128_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + value->value.enumerated.item[0] = data->os_128; + return 0; +} + +static int os_128_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + int changed; + + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != data->os_128; + if (changed) { + data->os_128 = value->value.enumerated.item[0]; + if (data->has_cs2000) + update_cs2000_rate(chip, data->current_rate); + oxygen_write16_masked(chip, OXYGEN_I2S_MULTICH_FORMAT, + mclk_from_rate(chip, data->current_rate), + OXYGEN_I2S_MCLK_MASK); + update_pcm1796_oversampling(chip); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new os_128_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DAC Oversampling Playback Enum", + .info = os_128_info, + .get = os_128_get, + .put = os_128_put, +}; + +static int st_output_switch_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Speakers", "Headphones", "FP Headphones" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int st_output_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio; + + gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (!(gpio & GPIO_ST_HP)) + value->value.enumerated.item[0] = 0; + else if (gpio & GPIO_ST_HP_REAR) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + + +static int st_output_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + u16 gpio_old, gpio; + + mutex_lock(&chip->mutex); + gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA); + gpio = gpio_old; + switch (value->value.enumerated.item[0]) { + case 0: + gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR); + break; + case 1: + gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR; + break; + case 2: + gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR; + break; + } + oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); + data->hp_active = gpio & GPIO_ST_HP; + update_pcm1796_volume(chip); + mutex_unlock(&chip->mutex); + return gpio != gpio_old; +} + +static int st_hp_volume_offset_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "< 64 ohms", "64-300 ohms", "300-600 ohms" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item > 2) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int st_hp_volume_offset_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + + mutex_lock(&chip->mutex); + if (data->hp_gain_offset < 2*-6) + value->value.enumerated.item[0] = 0; + else if (data->hp_gain_offset < 0) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + mutex_unlock(&chip->mutex); + return 0; +} + + +static int st_hp_volume_offset_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + static const s8 offsets[] = { 2*-18, 2*-6, 0 }; + struct oxygen *chip = ctl->private_data; + struct xonar_pcm179x *data = chip->model_data; + s8 offset; + int changed; + + if (value->value.enumerated.item[0] > 2) + return -EINVAL; + offset = offsets[value->value.enumerated.item[0]]; + mutex_lock(&chip->mutex); + changed = offset != data->hp_gain_offset; + if (changed) { + data->hp_gain_offset = offset; + update_pcm1796_volume(chip); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static const struct snd_kcontrol_new st_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output", + .info = st_output_switch_info, + .get = st_output_switch_get, + .put = st_output_switch_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphones Impedance Playback Enum", + .info = st_hp_volume_offset_info, + .get = st_hp_volume_offset_get, + .put = st_hp_volume_offset_put, + }, +}; + +static void xonar_line_mic_ac97_switch(struct oxygen *chip, + unsigned int reg, unsigned int mute) +{ + if (reg == AC97_LINE) { + spin_lock_irq(&chip->reg_lock); + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + mute ? GPIO_INPUT_ROUTE : 0, + GPIO_INPUT_ROUTE); + spin_unlock_irq(&chip->reg_lock); + } +} + +static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0); + +static int xonar_d2_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + /* CD in is actually connected to the video in pin */ + template->private_value ^= AC97_CD ^ AC97_VIDEO; + return 0; +} + +static int xonar_st_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + return 0; +} + +static int add_pcm1796_controls(struct oxygen *chip) +{ + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&rolloff_control, chip)); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, snd_ctl_new1(&os_128_control, chip)); + if (err < 0) + return err; + return 0; +} + +static int xonar_d2_mixer_init(struct oxygen *chip) +{ + int err; + + err = snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); + if (err < 0) + return err; + err = add_pcm1796_controls(chip); + if (err < 0) + return err; + return 0; +} + +static int xonar_hdav_mixer_init(struct oxygen *chip) +{ + return add_pcm1796_controls(chip); +} + +static int xonar_st_mixer_init(struct oxygen *chip) +{ + unsigned int i; + int err; + + for (i = 0; i < ARRAY_SIZE(st_controls); ++i) { + err = snd_ctl_add(chip->card, + snd_ctl_new1(&st_controls[i], chip)); + if (err < 0) + return err; + } + err = add_pcm1796_controls(chip); + if (err < 0) + return err; + return 0; +} + +static const struct oxygen_model model_xonar_d2 = { + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_d2_init, + .control_filter = xonar_d2_control_filter, + .mixer_init = xonar_d2_mixer_init, + .cleanup = xonar_d2_cleanup, + .suspend = xonar_d2_suspend, + .resume = xonar_d2_resume, + .get_i2s_mclk = get_pcm1796_i2s_mclk, + .set_dac_params = set_pcm1796_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_pcm179x), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF | + MIDI_OUTPUT | + MIDI_INPUT, + .dac_channels = 8, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .misc_flags = OXYGEN_MISC_MIDI, + .function_flags = OXYGEN_FUNCTION_SPI | + OXYGEN_FUNCTION_ENABLE_SPI_4_5, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +static const struct oxygen_model model_xonar_hdav = { + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_hdav_init, + .mixer_init = xonar_hdav_mixer_init, + .cleanup = xonar_hdav_cleanup, + .suspend = xonar_hdav_suspend, + .resume = xonar_hdav_resume, + .pcm_hardware_filter = xonar_hdmi_pcm_hardware_filter, + .get_i2s_mclk = get_pcm1796_i2s_mclk, + .set_dac_params = set_hdav_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .uart_input = xonar_hdmi_uart_input, + .ac97_switch = xonar_line_mic_ac97_switch, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_hdav), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF, + .dac_channels = 8, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .misc_flags = OXYGEN_MISC_MIDI, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +static const struct oxygen_model model_xonar_st = { + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .init = xonar_st_init, + .control_filter = xonar_st_control_filter, + .mixer_init = xonar_st_mixer_init, + .cleanup = xonar_st_cleanup, + .suspend = xonar_st_suspend, + .resume = xonar_st_resume, + .get_i2s_mclk = get_pcm1796_i2s_mclk, + .set_dac_params = set_st_params, + .set_adc_params = xonar_set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .ac97_switch = xonar_line_mic_ac97_switch, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_pcm179x), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 2, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +int __devinit get_xonar_pcm179x_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + switch (id->subdevice) { + case 0x8269: + chip->model = model_xonar_d2; + chip->model.shortname = "Xonar D2"; + break; + case 0x82b7: + chip->model = model_xonar_d2; + chip->model.shortname = "Xonar D2X"; + chip->model.init = xonar_d2x_init; + break; + case 0x8314: + chip->model = model_xonar_hdav; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { + default: + chip->model.shortname = "Xonar HDAV1.3"; + break; + case GPIO_DB_H6: + chip->model.shortname = "Xonar HDAV1.3+H6"; + chip->model.private_data = 1; + break; + } + break; + case 0x835d: + chip->model = model_xonar_st; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & GPIO_DB_MASK) { + default: + chip->model.shortname = "Xonar ST"; + break; + case GPIO_DB_H6: + chip->model.shortname = "Xonar ST+H6"; + chip->model.dac_channels = 8; + chip->model.private_data = 1; + break; + } + break; + case 0x835c: + chip->model = model_xonar_st; + chip->model.shortname = "Xonar STX"; + chip->model.init = xonar_stx_init; + chip->model.resume = xonar_stx_resume; + chip->model.set_dac_params = set_pcm1796_params; + break; + default: + return -EINVAL; + } + return 0; +} diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 2cc0eda4f20..2e156467b81 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -479,7 +479,7 @@ static int snd_pmac_awacs_put_master_amp(struct snd_kcontrol *kcontrol, static struct snd_kcontrol_new snd_pmac_awacs_amp_vol[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Volume", + .name = "Speaker Playback Volume", .info = snd_pmac_awacs_info_volume_amp, .get = snd_pmac_awacs_get_volume_amp, .put = snd_pmac_awacs_put_volume_amp, @@ -525,7 +525,7 @@ static struct snd_kcontrol_new snd_pmac_awacs_amp_hp_sw __devinitdata = { static struct snd_kcontrol_new snd_pmac_awacs_amp_spk_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Switch", + .name = "Speaker Playback Switch", .info = snd_pmac_boolean_stereo_info, .get = snd_pmac_awacs_get_switch_amp, .put = snd_pmac_awacs_put_switch_amp, @@ -696,17 +696,17 @@ static struct snd_kcontrol_new snd_pmac_screamer_mic_boost_imac[] __devinitdata }; static struct snd_kcontrol_new snd_pmac_awacs_speaker_vol[] __devinitdata = { - AWACS_VOLUME("PC Speaker Playback Volume", 4, 6, 1), + AWACS_VOLUME("Speaker Playback Volume", 4, 6, 1), }; static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_SPKMUTE, 1); static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac1 __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 1); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 1); static struct snd_kcontrol_new snd_pmac_awacs_speaker_sw_imac2 __devinitdata = -AWACS_SWITCH("PC Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); +AWACS_SWITCH("Speaker Playback Switch", 1, SHIFT_PAROUT1, 0); /* diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 16ed240e423..0accfe49735 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -505,7 +505,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_imac[] __devinitdata = { MASK_ADDR_BURGUNDY_GAINLINE, 1, 0), BURGUNDY_VOLUME_B("Mic Gain Capture Volume", 0, MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), - BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + BURGUNDY_VOLUME_B("Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), BURGUNDY_VOLUME_B("Line out Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENLINEOUT, 1, 1), @@ -527,7 +527,7 @@ static struct snd_kcontrol_new snd_pmac_burgundy_mixers_pmac[] __devinitdata = { MASK_ADDR_BURGUNDY_VOLMIC, 16), BURGUNDY_VOLUME_B("Line in Gain Capture Volume", 0, MASK_ADDR_BURGUNDY_GAINMIC, 1, 0), - BURGUNDY_VOLUME_B("PC Speaker Playback Volume", 0, + BURGUNDY_VOLUME_B("Speaker Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENMONO, 0, 1), BURGUNDY_VOLUME_B("Line out Playback Volume", 0, MASK_ADDR_BURGUNDY_ATTENSPEAKER, 1, 1), @@ -549,11 +549,11 @@ BURGUNDY_SWITCH_B("Master Playback Switch", 0, BURGUNDY_OUTPUT_INTERN | BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_imac __devinitdata = -BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, +BURGUNDY_SWITCH_B("Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_LEFT, BURGUNDY_OUTPUT_RIGHT, 1); static struct snd_kcontrol_new snd_pmac_burgundy_speaker_sw_pmac __devinitdata = -BURGUNDY_SWITCH_B("PC Speaker Playback Switch", 0, +BURGUNDY_SWITCH_B("Speaker Playback Switch", 0, MASK_ADDR_BURGUNDY_MORE_OUTPUTENABLES, BURGUNDY_OUTPUT_INTERN, 0, 0); static struct snd_kcontrol_new snd_pmac_burgundy_line_sw_imac __devinitdata = diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 08e584d1453..789f44f4ac7 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -905,7 +905,7 @@ static struct snd_kcontrol_new tumbler_hp_sw __devinitdata = { }; static struct snd_kcontrol_new tumbler_speaker_sw __devinitdata = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "PC Speaker Playback Switch", + .name = "Speaker Playback Switch", .info = snd_pmac_boolean_mono_info, .get = tumbler_get_mute_switch, .put = tumbler_put_mute_switch, diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig index aed0f90c391..61139f3c161 100644 --- a/sound/sh/Kconfig +++ b/sound/sh/Kconfig @@ -19,5 +19,13 @@ config SND_AICA help ALSA Sound driver for the SEGA Dreamcast console. +config SND_SH_DAC_AUDIO + tristate "SuperH DAC audio support" + depends on SND + depends on CPU_SH3 && HIGH_RES_TIMERS + select SND_PCM + help + Say Y here to include support for the on-chip DAC. + endif # SND_SUPERH diff --git a/sound/sh/Makefile b/sound/sh/Makefile index 8fdcb6e26f0..7d09b5188cf 100644 --- a/sound/sh/Makefile +++ b/sound/sh/Makefile @@ -3,6 +3,8 @@ # snd-aica-objs := aica.o +snd-sh_dac_audio-objs := sh_dac_audio.o # Toplevel Module Dependency obj-$(CONFIG_SND_AICA) += snd-aica.o +obj-$(CONFIG_SND_SH_DAC_AUDIO) += snd-sh_dac_audio.o diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c new file mode 100644 index 00000000000..76d9ad27d91 --- /dev/null +++ b/sound/sh/sh_dac_audio.c @@ -0,0 +1,453 @@ +/* + * sh_dac_audio.c - SuperH DAC audio driver for ALSA + * + * Copyright (c) 2009 by Rafael Ignacio Zurita <rizurita@yahoo.com> + * + * + * Based on sh_dac_audio.c (Copyright (C) 2004, 2005 by Andriy Skulysh) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/hrtimer.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/sh_dac_audio.h> +#include <asm/clock.h> +#include <asm/hd64461.h> +#include <mach/hp6xx.h> +#include <cpu/dac.h> + +MODULE_AUTHOR("Rafael Ignacio Zurita <rizurita@yahoo.com>"); +MODULE_DESCRIPTION("SuperH DAC audio driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SuperH DAC audio support}}"); + +/* Module Parameters */ +static int index = SNDRV_DEFAULT_IDX1; +static char *id = SNDRV_DEFAULT_STR1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SuperH DAC audio."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SuperH DAC audio."); + +/* main struct */ +struct snd_sh_dac { + struct snd_card *card; + struct snd_pcm_substream *substream; + struct hrtimer hrtimer; + ktime_t wakeups_per_second; + + int rate; + int empty; + char *data_buffer, *buffer_begin, *buffer_end; + int processed; /* bytes proccesed, to compare with period_size */ + int buffer_size; + struct dac_audio_pdata *pdata; +}; + + +static void dac_audio_start_timer(struct snd_sh_dac *chip) +{ + hrtimer_start(&chip->hrtimer, chip->wakeups_per_second, + HRTIMER_MODE_REL); +} + +static void dac_audio_stop_timer(struct snd_sh_dac *chip) +{ + hrtimer_cancel(&chip->hrtimer); +} + +static void dac_audio_reset(struct snd_sh_dac *chip) +{ + dac_audio_stop_timer(chip); + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; +} + +static void dac_audio_set_rate(struct snd_sh_dac *chip) +{ + chip->wakeups_per_second = ktime_set(0, 1000000000 / chip->rate); +} + + +/* PCM INTERFACE */ + +static struct snd_pcm_hardware snd_sh_dac_pcm_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_HALF_DUPLEX), + .formats = SNDRV_PCM_FMTBIT_U8, + .rates = SNDRV_PCM_RATE_8000, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = (48*1024), + .period_bytes_min = 1, + .period_bytes_max = (48*1024), + .periods_min = 1, + .periods_max = 1024, +}; + +static int snd_sh_dac_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = snd_sh_dac_pcm_hw; + + chip->substream = substream; + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; + + chip->pdata->start(chip->pdata); + + return 0; +} + +static int snd_sh_dac_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + + chip->substream = NULL; + + dac_audio_stop_timer(chip); + chip->pdata->stop(chip->pdata); + + return 0; +} + +static int snd_sh_dac_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_sh_dac_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_sh_dac_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = chip->substream->runtime; + + chip->buffer_size = runtime->buffer_size; + memset(chip->data_buffer, 0, chip->pdata->buffer_size); + + return 0; +} + +static int snd_sh_dac_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dac_audio_start_timer(chip); + break; + case SNDRV_PCM_TRIGGER_STOP: + chip->buffer_begin = chip->buffer_end = chip->data_buffer; + chip->processed = 0; + chip->empty = 1; + dac_audio_stop_timer(chip); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int snd_sh_dac_pcm_copy(struct snd_pcm_substream *substream, int channel, + snd_pcm_uframes_t pos, void __user *src, snd_pcm_uframes_t count) +{ + /* channel is not used (interleaved data) */ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + ssize_t b_count = frames_to_bytes(runtime , count); + ssize_t b_pos = frames_to_bytes(runtime , pos); + + if (count < 0) + return -EINVAL; + + if (!count) + return 0; + + memcpy_toio(chip->data_buffer + b_pos, src, b_count); + chip->buffer_end = chip->data_buffer + b_pos + b_count; + + if (chip->empty) { + chip->empty = 0; + dac_audio_start_timer(chip); + } + + return 0; +} + +static int snd_sh_dac_pcm_silence(struct snd_pcm_substream *substream, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + /* channel is not used (interleaved data) */ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + ssize_t b_count = frames_to_bytes(runtime , count); + ssize_t b_pos = frames_to_bytes(runtime , pos); + + if (count < 0) + return -EINVAL; + + if (!count) + return 0; + + memset_io(chip->data_buffer + b_pos, 0, b_count); + chip->buffer_end = chip->data_buffer + b_pos + b_count; + + if (chip->empty) { + chip->empty = 0; + dac_audio_start_timer(chip); + } + + return 0; +} + +static +snd_pcm_uframes_t snd_sh_dac_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_sh_dac *chip = snd_pcm_substream_chip(substream); + int pointer = chip->buffer_begin - chip->data_buffer; + + return pointer; +} + +/* pcm ops */ +static struct snd_pcm_ops snd_sh_dac_pcm_ops = { + .open = snd_sh_dac_pcm_open, + .close = snd_sh_dac_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_sh_dac_pcm_hw_params, + .hw_free = snd_sh_dac_pcm_hw_free, + .prepare = snd_sh_dac_pcm_prepare, + .trigger = snd_sh_dac_pcm_trigger, + .pointer = snd_sh_dac_pcm_pointer, + .copy = snd_sh_dac_pcm_copy, + .silence = snd_sh_dac_pcm_silence, + .mmap = snd_pcm_lib_mmap_iomem, +}; + +static int __devinit snd_sh_dac_pcm(struct snd_sh_dac *chip, int device) +{ + int err; + struct snd_pcm *pcm; + + /* device should be always 0 for us */ + err = snd_pcm_new(chip->card, "SH_DAC PCM", device, 1, 0, &pcm); + if (err < 0) + return err; + + pcm->private_data = chip; + strcpy(pcm->name, "SH_DAC PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sh_dac_pcm_ops); + + /* buffer size=48K */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 48 * 1024, + 48 * 1024); + + return 0; +} +/* END OF PCM INTERFACE */ + + +/* driver .remove -- destructor */ +static int snd_sh_dac_remove(struct platform_device *devptr) +{ + snd_card_free(platform_get_drvdata(devptr)); + platform_set_drvdata(devptr, NULL); + + return 0; +} + +/* free -- it has been defined by create */ +static int snd_sh_dac_free(struct snd_sh_dac *chip) +{ + /* release the data */ + kfree(chip->data_buffer); + kfree(chip); + + return 0; +} + +static int snd_sh_dac_dev_free(struct snd_device *device) +{ + struct snd_sh_dac *chip = device->device_data; + + return snd_sh_dac_free(chip); +} + +static enum hrtimer_restart sh_dac_audio_timer(struct hrtimer *handle) +{ + struct snd_sh_dac *chip = container_of(handle, struct snd_sh_dac, + hrtimer); + struct snd_pcm_runtime *runtime = chip->substream->runtime; + ssize_t b_ps = frames_to_bytes(runtime, runtime->period_size); + + if (!chip->empty) { + sh_dac_output(*chip->buffer_begin, chip->pdata->channel); + chip->buffer_begin++; + + chip->processed++; + if (chip->processed >= b_ps) { + chip->processed -= b_ps; + snd_pcm_period_elapsed(chip->substream); + } + + if (chip->buffer_begin == (chip->data_buffer + + chip->buffer_size - 1)) + chip->buffer_begin = chip->data_buffer; + + if (chip->buffer_begin == chip->buffer_end) + chip->empty = 1; + + } + + if (!chip->empty) + hrtimer_start(&chip->hrtimer, chip->wakeups_per_second, + HRTIMER_MODE_REL); + + return HRTIMER_NORESTART; +} + +/* create -- chip-specific constructor for the cards components */ +static int __devinit snd_sh_dac_create(struct snd_card *card, + struct platform_device *devptr, + struct snd_sh_dac **rchip) +{ + struct snd_sh_dac *chip; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_sh_dac_dev_free, + }; + + *rchip = NULL; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) + return -ENOMEM; + + chip->card = card; + + hrtimer_init(&chip->hrtimer, CLOCK_MONOTONIC, HRTIMER_MODE_REL); + chip->hrtimer.function = sh_dac_audio_timer; + + dac_audio_reset(chip); + chip->rate = 8000; + dac_audio_set_rate(chip); + + chip->pdata = devptr->dev.platform_data; + + chip->data_buffer = kmalloc(chip->pdata->buffer_size, GFP_KERNEL); + if (chip->data_buffer == NULL) { + kfree(chip); + return -ENOMEM; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_sh_dac_free(chip); + return err; + } + + *rchip = chip; + + return 0; +} + +/* driver .probe -- constructor */ +static int __devinit snd_sh_dac_probe(struct platform_device *devptr) +{ + struct snd_sh_dac *chip; + struct snd_card *card; + int err; + + err = snd_card_create(index, id, THIS_MODULE, 0, &card); + if (err < 0) { + snd_printk(KERN_ERR "cannot allocate the card\n"); + return err; + } + + err = snd_sh_dac_create(card, devptr, &chip); + if (err < 0) + goto probe_error; + + err = snd_sh_dac_pcm(chip, 0); + if (err < 0) + goto probe_error; + + strcpy(card->driver, "snd_sh_dac"); + strcpy(card->shortname, "SuperH DAC audio driver"); + printk(KERN_INFO "%s %s", card->longname, card->shortname); + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + snd_printk("ALSA driver for SuperH DAC audio"); + + platform_set_drvdata(devptr, card); + return 0; + +probe_error: + snd_card_free(card); + return err; +} + +/* + * "driver" definition + */ +static struct platform_driver driver = { + .probe = snd_sh_dac_probe, + .remove = snd_sh_dac_remove, + .driver = { + .name = "dac_audio", + }, +}; + +static int __init sh_dac_init(void) +{ + return platform_driver_register(&driver); +} + +static void __exit sh_dac_exit(void) +{ + platform_driver_unregister(&driver); +} + +module_init(sh_dac_init); +module_exit(sh_dac_exit); diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0c5eac01bf2..1470141d416 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ -snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o +snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o obj-$(CONFIG_SND_SOC) += codecs/ diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c index 9eb610c2ba9..9df4c68ef00 100644 --- a/sound/soc/atmel/playpaq_wm8510.c +++ b/sound/soc/atmel/playpaq_wm8510.c @@ -268,7 +268,7 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream, #endif /* CONFIG_SND_AT32_SOC_PLAYPAQ_SLAVE */ - ret = snd_soc_dai_set_pll(codec_dai, 0, + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, clk_get_rate(CODEC_CLK), pll_out); if (ret < 0) { pr_warning("playpaq_wm8510: Failed to set CODEC DAI PLL (%d)\n", diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 885ba012557..e028744c32c 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static int __init at91sam9g20ek_init(void) struct clk *pllb; int ret; - if (!machine_is_at91sam9g20ek()) + if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) return -ENODEV; /* diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 594c6c5b783..19e4d37eba1 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -333,6 +333,30 @@ static int au1xpsc_pcm_new(struct snd_card *card, static int au1xpsc_pcm_probe(struct platform_device *pdev) { + if (!au1xpsc_audio_pcmdma[PCM_TX] || !au1xpsc_audio_pcmdma[PCM_RX]) + return -ENODEV; + + return 0; +} + +static int au1xpsc_pcm_remove(struct platform_device *pdev) +{ + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform au1xpsc_soc_platform = { + .name = "au1xpsc-pcm-dbdma", + .probe = au1xpsc_pcm_probe, + .remove = au1xpsc_pcm_remove, + .pcm_ops = &au1xpsc_pcm_ops, + .pcm_new = au1xpsc_pcm_new, + .pcm_free = au1xpsc_pcm_free_dma_buffers, +}; +EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); + +static int __devinit au1xpsc_pcm_drvprobe(struct platform_device *pdev) +{ struct resource *r; int ret; @@ -365,7 +389,9 @@ static int au1xpsc_pcm_probe(struct platform_device *pdev) } (au1xpsc_audio_pcmdma[PCM_RX])->ddma_id = r->start; - return 0; + ret = snd_soc_register_platform(&au1xpsc_soc_platform); + if (!ret) + return ret; out2: kfree(au1xpsc_audio_pcmdma[PCM_RX]); @@ -376,10 +402,12 @@ out1: return ret; } -static int au1xpsc_pcm_remove(struct platform_device *pdev) +static int __devexit au1xpsc_pcm_drvremove(struct platform_device *pdev) { int i; + snd_soc_unregister_platform(&au1xpsc_soc_platform); + for (i = 0; i < 2; i++) { if (au1xpsc_audio_pcmdma[i]) { au1x_pcm_dbdma_free(au1xpsc_audio_pcmdma[i]); @@ -391,32 +419,81 @@ static int au1xpsc_pcm_remove(struct platform_device *pdev) return 0; } -/* au1xpsc audio platform */ -struct snd_soc_platform au1xpsc_soc_platform = { - .name = "au1xpsc-pcm-dbdma", - .probe = au1xpsc_pcm_probe, - .remove = au1xpsc_pcm_remove, - .pcm_ops = &au1xpsc_pcm_ops, - .pcm_new = au1xpsc_pcm_new, - .pcm_free = au1xpsc_pcm_free_dma_buffers, +static struct platform_driver au1xpsc_pcm_driver = { + .driver = { + .name = "au1xpsc-pcm", + .owner = THIS_MODULE, + }, + .probe = au1xpsc_pcm_drvprobe, + .remove = __devexit_p(au1xpsc_pcm_drvremove), }; -EXPORT_SYMBOL_GPL(au1xpsc_soc_platform); -static int __init au1xpsc_audio_dbdma_init(void) +static int __init au1xpsc_audio_dbdma_load(void) { au1xpsc_audio_pcmdma[PCM_TX] = NULL; au1xpsc_audio_pcmdma[PCM_RX] = NULL; - return snd_soc_register_platform(&au1xpsc_soc_platform); + return platform_driver_register(&au1xpsc_pcm_driver); } -static void __exit au1xpsc_audio_dbdma_exit(void) +static void __exit au1xpsc_audio_dbdma_unload(void) { - snd_soc_unregister_platform(&au1xpsc_soc_platform); + platform_driver_unregister(&au1xpsc_pcm_driver); } -module_init(au1xpsc_audio_dbdma_init); -module_exit(au1xpsc_audio_dbdma_exit); +module_init(au1xpsc_audio_dbdma_load); +module_exit(au1xpsc_audio_dbdma_unload); + + +struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev) +{ + struct resource *res, *r; + struct platform_device *pd; + int id[2]; + int ret; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!r) + return NULL; + id[0] = r->start; + + r = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!r) + return NULL; + id[1] = r->start; + + res = kzalloc(sizeof(struct resource) * 2, GFP_KERNEL); + if (!res) + return NULL; + + res[0].start = res[0].end = id[0]; + res[1].start = res[1].end = id[1]; + res[0].flags = res[1].flags = IORESOURCE_DMA; + + pd = platform_device_alloc("au1xpsc-pcm", -1); + if (!pd) + goto out; + + pd->resource = res; + pd->num_resources = 2; + + ret = platform_device_add(pd); + if (!ret) + return pd; + + platform_device_put(pd); +out: + kfree(res); + return NULL; +} +EXPORT_SYMBOL_GPL(au1xpsc_pcm_add); + +void au1xpsc_pcm_destroy(struct platform_device *dmapd) +{ + if (dmapd) + platform_device_unregister(dmapd); +} +EXPORT_SYMBOL_GPL(au1xpsc_pcm_destroy); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC Audio DMA driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index a521aa90dde..340311d7fed 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -61,7 +61,8 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, retry, tmo; + unsigned short retry, tmo; + unsigned long data; au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); @@ -74,20 +75,26 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, AC97_CDC(pscdata)); au_sync(); - tmo = 2000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) - && --tmo) - udelay(2); + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); - data = au_readl(AC97_CDC(pscdata)) & 0xffff; + data = au_readl(AC97_CDC(pscdata)); au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); mutex_unlock(&pscdata->lock); + + if (reg != ((data >> 16) & 0x7f)) + tmo = 1; /* wrong register, try again */ + } while (--retry && !tmo); - return retry ? data : 0xffff; + return retry ? data & 0xffff : 0xffff; } /* AC97 controller writes to codec register */ @@ -109,10 +116,12 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, AC97_CDC(pscdata)); au_sync(); - tmo = 2000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) - && --tmo) - udelay(2); + tmo = 20; + do { + udelay(21); + if (au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD) + break; + } while (--tmo); au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); @@ -195,7 +204,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; unsigned long r, ro, stat; - int chans, stype = SUBSTREAM_TYPE(substream); + int chans, t, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); @@ -237,8 +246,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, au_sync(); /* ...wait for it... */ - while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) - asm volatile ("nop"); + t = 100; + while ((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't disable!\n"); /* ...write config... */ au_writel(r, AC97_CFG(pscdata)); @@ -249,8 +262,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, au_sync(); /* ...and wait for ready bit */ - while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) - asm volatile ("nop"); + t = 100; + while ((!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && --t) + msleep(1); + + if (!t) + printk(KERN_ERR "PSC-AC97: can't enable!\n"); mutex_unlock(&pscdata->lock); @@ -300,19 +317,55 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, static int au1xpsc_ac97_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { + return au1xpsc_ac97_workdata ? 0 : -ENODEV; +} + +static void au1xpsc_ac97_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ +} + +static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { + .trigger = au1xpsc_ac97_trigger, + .hw_params = au1xpsc_ac97_hw_params, +}; + +struct snd_soc_dai au1xpsc_ac97_dai = { + .name = "au1xpsc_ac97", + .ac97_control = 1, + .probe = au1xpsc_ac97_probe, + .remove = au1xpsc_ac97_remove, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &au1xpsc_ac97_dai_ops, +}; +EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); + +static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev) +{ int ret; struct resource *r; unsigned long sel; + struct au1xpsc_audio_data *wd; if (au1xpsc_ac97_workdata) return -EBUSY; - au1xpsc_ac97_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_ac97_workdata) + wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!wd) return -ENOMEM; - mutex_init(&au1xpsc_ac97_workdata->lock); + mutex_init(&wd->lock); r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { @@ -321,81 +374,95 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, } ret = -EBUSY; - au1xpsc_ac97_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, + wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, "au1xpsc_ac97"); - if (!au1xpsc_ac97_workdata->ioarea) + if (!wd->ioarea) goto out0; - au1xpsc_ac97_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_ac97_workdata->mmio) + wd->mmio = ioremap(r->start, 0xffff); + if (!wd->mmio) goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; - /* preserve PSC clock source set up by platform (dev.platform_data - * is already occupied by soc layer) - */ - sel = au_readl(PSC_SEL(au1xpsc_ac97_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + /* preserve PSC clock source set up by platform */ + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(PSC_SEL_PS_AC97MODE | sel, PSC_SEL(wd)); au_sync(); - /* next up: cold reset. Dont check for PSC-ready now since - * there may not be any codec clock yet. - */ - return 0; + ret = snd_soc_register_dai(&au1xpsc_ac97_dai); + if (ret) + goto out1; + wd->dmapd = au1xpsc_pcm_add(pdev); + if (wd->dmapd) { + platform_set_drvdata(pdev, wd); + au1xpsc_ac97_workdata = wd; /* MDEV */ + return 0; + } + + snd_soc_unregister_dai(&au1xpsc_ac97_dai); out1: - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); + release_resource(wd->ioarea); + kfree(wd->ioarea); out0: - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; + kfree(wd); return ret; } -static void au1xpsc_ac97_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_ac97_drvremove(struct platform_device *pdev) { + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + if (wd->dmapd) + au1xpsc_pcm_destroy(wd->dmapd); + + snd_soc_unregister_dai(&au1xpsc_ac97_dai); + /* disable PSC completely */ - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_ac97_workdata->mmio); - release_resource(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata->ioarea); - kfree(au1xpsc_ac97_workdata); - au1xpsc_ac97_workdata = NULL; + iounmap(wd->mmio); + release_resource(wd->ioarea); + kfree(wd->ioarea); + kfree(wd); + + au1xpsc_ac97_workdata = NULL; /* MDEV */ + + return 0; } -static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai) +#ifdef CONFIG_PM +static int au1xpsc_ac97_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting registers and disable PSC */ - au1xpsc_ac97_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_ac97_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, AC97_CFG(au1xpsc_ac97_workdata)); + au_writel(0, AC97_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_ac97_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) +static int au1xpsc_ac97_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* restore PSC clock config */ - au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE, - PSC_SEL(au1xpsc_ac97_workdata)); + au_writel(wd->pm[0] | PSC_SEL_PS_AC97MODE, PSC_SEL(wd)); au_sync(); /* after this point the ac97 core will cold-reset the codec. @@ -405,48 +472,44 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { - .trigger = au1xpsc_ac97_trigger, - .hw_params = au1xpsc_ac97_hw_params, +static struct dev_pm_ops au1xpscac97_pmops = { + .suspend = au1xpsc_ac97_drvsuspend, + .resume = au1xpsc_ac97_drvresume, }; -struct snd_soc_dai au1xpsc_ac97_dai = { - .name = "au1xpsc_ac97", - .ac97_control = 1, - .probe = au1xpsc_ac97_probe, - .remove = au1xpsc_ac97_remove, - .suspend = au1xpsc_ac97_suspend, - .resume = au1xpsc_ac97_resume, - .playback = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, - }, - .capture = { - .rates = AC97_RATES, - .formats = AC97_FMTS, - .channels_min = 2, - .channels_max = 2, +#define AU1XPSCAC97_PMOPS &au1xpscac97_pmops + +#else + +#define AU1XPSCAC97_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_ac97_driver = { + .driver = { + .name = "au1xpsc_ac97", + .owner = THIS_MODULE, + .pm = AU1XPSCAC97_PMOPS, }, - .ops = &au1xpsc_ac97_dai_ops, + .probe = au1xpsc_ac97_drvprobe, + .remove = __devexit_p(au1xpsc_ac97_drvremove), }; -EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai); -static int __init au1xpsc_ac97_init(void) +static int __init au1xpsc_ac97_load(void) { au1xpsc_ac97_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_ac97_dai); + return platform_driver_register(&au1xpsc_ac97_driver); } -static void __exit au1xpsc_ac97_exit(void) +static void __exit au1xpsc_ac97_unload(void) { - snd_soc_unregister_dai(&au1xpsc_ac97_dai); + platform_driver_unregister(&au1xpsc_ac97_driver); } -module_init(au1xpsc_ac97_init); -module_exit(au1xpsc_ac97_exit); +module_init(au1xpsc_ac97_load); +module_exit(au1xpsc_ac97_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <manuel.lauss@gmail.com>"); +MODULE_AUTHOR("Manuel Lauss"); + diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index bb589327ee3..0cf2ca61c77 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -265,16 +265,52 @@ static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static int au1xpsc_i2s_probe(struct platform_device *pdev, struct snd_soc_dai *dai) { + return au1xpsc_i2s_workdata ? 0 : -ENODEV; +} + +static void au1xpsc_i2s_remove(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ +} + +static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { + .trigger = au1xpsc_i2s_trigger, + .hw_params = au1xpsc_i2s_hw_params, + .set_fmt = au1xpsc_i2s_set_fmt, +}; + +struct snd_soc_dai au1xpsc_i2s_dai = { + .name = "au1xpsc_i2s", + .probe = au1xpsc_i2s_probe, + .remove = au1xpsc_i2s_remove, + .playback = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .capture = { + .rates = AU1XPSC_I2S_RATES, + .formats = AU1XPSC_I2S_FMTS, + .channels_min = 2, + .channels_max = 8, /* 2 without external help */ + }, + .ops = &au1xpsc_i2s_dai_ops, +}; +EXPORT_SYMBOL(au1xpsc_i2s_dai); + +static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev) +{ struct resource *r; unsigned long sel; int ret; + struct au1xpsc_audio_data *wd; if (au1xpsc_i2s_workdata) return -EBUSY; - au1xpsc_i2s_workdata = - kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); - if (!au1xpsc_i2s_workdata) + wd = kzalloc(sizeof(struct au1xpsc_audio_data), GFP_KERNEL); + if (!wd) return -ENOMEM; r = platform_get_resource(pdev, IORESOURCE_MEM, 0); @@ -284,131 +320,146 @@ static int au1xpsc_i2s_probe(struct platform_device *pdev, } ret = -EBUSY; - au1xpsc_i2s_workdata->ioarea = - request_mem_region(r->start, r->end - r->start + 1, + wd->ioarea = request_mem_region(r->start, r->end - r->start + 1, "au1xpsc_i2s"); - if (!au1xpsc_i2s_workdata->ioarea) + if (!wd->ioarea) goto out0; - au1xpsc_i2s_workdata->mmio = ioremap(r->start, 0xffff); - if (!au1xpsc_i2s_workdata->mmio) + wd->mmio = ioremap(r->start, 0xffff); + if (!wd->mmio) goto out1; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) */ - sel = au_readl(PSC_SEL(au1xpsc_i2s_workdata)) & PSC_SEL_CLK_MASK; - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + sel = au_readl(PSC_SEL(wd)) & PSC_SEL_CLK_MASK; + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(au1xpsc_i2s_workdata)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(PSC_SEL_PS_I2SMODE | sel, PSC_SEL(wd)); + au_writel(0, I2S_CFG(wd)); au_sync(); /* preconfigure: set max rx/tx fifo depths */ - au1xpsc_i2s_workdata->cfg |= - PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; + wd->cfg |= PSC_I2SCFG_RT_FIFO8 | PSC_I2SCFG_TT_FIFO8; /* don't wait for I2S core to become ready now; clocks may not * be running yet; depending on clock input for PSC a wait might * time out. */ - return 0; + ret = snd_soc_register_dai(&au1xpsc_i2s_dai); + if (ret) + goto out1; + /* finally add the DMA device for this PSC */ + wd->dmapd = au1xpsc_pcm_add(pdev); + if (wd->dmapd) { + platform_set_drvdata(pdev, wd); + au1xpsc_i2s_workdata = wd; + return 0; + } + + snd_soc_unregister_dai(&au1xpsc_i2s_dai); out1: - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); + release_resource(wd->ioarea); + kfree(wd->ioarea); out0: - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; + kfree(wd); return ret; } -static void au1xpsc_i2s_remove(struct platform_device *pdev, - struct snd_soc_dai *dai) +static int __devexit au1xpsc_i2s_drvremove(struct platform_device *pdev) { - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + struct au1xpsc_audio_data *wd = platform_get_drvdata(pdev); + + if (wd->dmapd) + au1xpsc_pcm_destroy(wd->dmapd); + + snd_soc_unregister_dai(&au1xpsc_i2s_dai); + + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - iounmap(au1xpsc_i2s_workdata->mmio); - release_resource(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata->ioarea); - kfree(au1xpsc_i2s_workdata); - au1xpsc_i2s_workdata = NULL; + iounmap(wd->mmio); + release_resource(wd->ioarea); + kfree(wd->ioarea); + kfree(wd); + + au1xpsc_i2s_workdata = NULL; /* MDEV */ + + return 0; } -static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai) +#ifdef CONFIG_PM +static int au1xpsc_i2s_drvsuspend(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* save interesting register and disable PSC */ - au1xpsc_i2s_workdata->pm[0] = - au_readl(PSC_SEL(au1xpsc_i2s_workdata)); + wd->pm[0] = au_readl(PSC_SEL(wd)); - au_writel(0, I2S_CFG(au1xpsc_i2s_workdata)); + au_writel(0, I2S_CFG(wd)); au_sync(); - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); return 0; } -static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai) +static int au1xpsc_i2s_drvresume(struct device *dev) { + struct au1xpsc_audio_data *wd = dev_get_drvdata(dev); + /* select I2S mode and PSC clock */ - au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata)); + au_writel(PSC_CTRL_DISABLE, PSC_CTRL(wd)); au_sync(); - au_writel(0, PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(0, PSC_SEL(wd)); au_sync(); - au_writel(au1xpsc_i2s_workdata->pm[0], - PSC_SEL(au1xpsc_i2s_workdata)); + au_writel(wd->pm[0], PSC_SEL(wd)); au_sync(); return 0; } -static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = { - .trigger = au1xpsc_i2s_trigger, - .hw_params = au1xpsc_i2s_hw_params, - .set_fmt = au1xpsc_i2s_set_fmt, +static struct dev_pm_ops au1xpsci2s_pmops = { + .suspend = au1xpsc_i2s_drvsuspend, + .resume = au1xpsc_i2s_drvresume, }; -struct snd_soc_dai au1xpsc_i2s_dai = { - .name = "au1xpsc_i2s", - .probe = au1xpsc_i2s_probe, - .remove = au1xpsc_i2s_remove, - .suspend = au1xpsc_i2s_suspend, - .resume = au1xpsc_i2s_resume, - .playback = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ - }, - .capture = { - .rates = AU1XPSC_I2S_RATES, - .formats = AU1XPSC_I2S_FMTS, - .channels_min = 2, - .channels_max = 8, /* 2 without external help */ +#define AU1XPSCI2S_PMOPS &au1xpsci2s_pmops + +#else + +#define AU1XPSCI2S_PMOPS NULL + +#endif + +static struct platform_driver au1xpsc_i2s_driver = { + .driver = { + .name = "au1xpsc_i2s", + .owner = THIS_MODULE, + .pm = AU1XPSCI2S_PMOPS, }, - .ops = &au1xpsc_i2s_dai_ops, + .probe = au1xpsc_i2s_drvprobe, + .remove = __devexit_p(au1xpsc_i2s_drvremove), }; -EXPORT_SYMBOL(au1xpsc_i2s_dai); -static int __init au1xpsc_i2s_init(void) +static int __init au1xpsc_i2s_load(void) { au1xpsc_i2s_workdata = NULL; - return snd_soc_register_dai(&au1xpsc_i2s_dai); + return platform_driver_register(&au1xpsc_i2s_driver); } -static void __exit au1xpsc_i2s_exit(void) +static void __exit au1xpsc_i2s_unload(void) { - snd_soc_unregister_dai(&au1xpsc_i2s_dai); + platform_driver_unregister(&au1xpsc_i2s_driver); } -module_init(au1xpsc_i2s_init); -module_exit(au1xpsc_i2s_exit); +module_init(au1xpsc_i2s_load); +module_exit(au1xpsc_i2s_unload); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC I2S ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 3f474e8ed4f..32d3807d3f5 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -2,7 +2,7 @@ * Au12x0/Au1550 PSC ALSA ASoC audio support. * * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano@roarinelk.homelinux.net> + * Manuel Lauss <manuel.lauss@gmail.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -21,6 +21,10 @@ extern struct snd_soc_dai au1xpsc_i2s_dai; extern struct snd_soc_platform au1xpsc_soc_platform; extern struct snd_ac97_bus_ops soc_ac97_ops; +/* DBDMA helpers */ +extern struct platform_device *au1xpsc_pcm_add(struct platform_device *pdev); +extern void au1xpsc_pcm_destroy(struct platform_device *dmapd); + struct au1xpsc_audio_data { void __iomem *mmio; @@ -30,6 +34,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; struct mutex lock; + struct platform_device *dmapd; }; #define PCM_TX 0 diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e304b0..0f45a3f56be 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e91810..2ef1e5013b8 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), + channel_map, ARRAY_SIZE(channel_map), channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 084b68884ad..3e6ada0dd1c 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -49,7 +49,6 @@ struct bf5xx_i2s_port { u16 rcr1; u16 tcr2; u16 rcr2; - int counter; int configured; }; @@ -133,16 +132,6 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - pr_debug("%s enter\n", __func__); - - /*this counter is used for counting how many pcm streams are opened*/ - bf5xx_i2s.counter++; - return 0; -} - static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -201,9 +190,8 @@ static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); - bf5xx_i2s.counter--; /* No active stream, SPORT is allowed to be configured again. */ - if (!bf5xx_i2s.counter) + if (!dai->active) bf5xx_i2s.configured = 0; } @@ -284,7 +272,6 @@ static int bf5xx_i2s_resume(struct snd_soc_dai *dai) SNDRV_PCM_FMTBIT_S32_LE) static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e823bd1..a8c73cbbd68 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -43,7 +43,7 @@ #include "bf5xx-tdm.h" #include "bf5xx-sport.h" -#define PCM_BUFFER_MAX 0x10000 +#define PCM_BUFFER_MAX 0x8000 #define FRAGMENT_SIZE_MIN (4*1024) #define FRAGMENTS_MIN 2 #define FRAGMENTS_MAX 32 @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index ff546e91a22..4b360124083 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,40 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + unsigned int slot; + unsigned int tx_mapped = 0, rx_mapped = 0; + + if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || + (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) { + slot = tx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(tx_mapped & (1 << slot)))) { + bf5xx_tdm.tx_map[i] = slot; + tx_mapped |= 1 << slot; + } else + return -EINVAL; + } + for (i = 0; i < rx_num; i++) { + slot = rx_slot[i]; + if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && + (!(rx_mapped & (1 << slot)))) { + bf5xx_tdm.rx_map[i] = slot; + rx_mapped |= 1 << slot; + } else + return -EINVAL; + } + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +261,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +327,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d90cd..04189a18c1b 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define BFIN_TDM_DAI_MAX_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[BFIN_TDM_DAI_MAX_SLOTS]; + unsigned int rx_map[BFIN_TDM_DAI_MAX_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93af3b..52b005f8fed 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -15,10 +15,12 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD1938 if SPI_MASTER select SND_SOC_AD1980 if SND_SOC_AC97_BUS + select SND_SOC_ADS117X select SND_SOC_AD73311 if I2C select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -28,6 +30,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C @@ -36,6 +40,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8523 if I2C select SND_SOC_WM8580 if I2C + select SND_SOC_WM8711 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8727 select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI @@ -86,6 +92,9 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate + +config SND_SOC_ADS117X + tristate config SND_SOC_AK4104 tristate @@ -96,6 +105,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_AK4671 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate @@ -136,7 +148,11 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC3X tristate +config SND_SOC_TLV320DAC33 + tristate + config SND_SOC_TWL4030 + select TWL4030_CODEC tristate config SND_SOC_UDA134X @@ -160,6 +176,12 @@ config SND_SOC_WM8523 config SND_SOC_WM8580 tristate +config SND_SOC_WM8711 + tristate + +config SND_SOC_WM8727 + tristate + config SND_SOC_WM8728 tristate @@ -220,3 +242,6 @@ config SND_SOC_WM9713 # Amp config SND_SOC_MAX9877 tristate + +config SND_SOC_TPA6130A2 + tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28486b..dbaecb133ac 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,9 +3,11 @@ snd-soc-ad1836-objs := ad1836.o snd-soc-ad1938-objs := ad1938.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -16,6 +18,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o @@ -24,6 +27,8 @@ snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8523-objs := wm8523.o snd-soc-wm8580-objs := wm8580.o +snd-soc-wm8711-objs := wm8711.o +snd-soc-wm8727-objs := wm8727.o snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o @@ -47,15 +52,18 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o +snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o @@ -66,6 +74,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o @@ -74,6 +83,8 @@ obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8523) += snd-soc-wm8523.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o +obj-$(CONFIG_SND_SOC_WM8711) += snd-soc-wm8711.o +obj-$(CONFIG_SND_SOC_WM8727) += snd-soc-wm8727.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o @@ -97,3 +108,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o +obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 932299bb5d1..69bd0acc81c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -117,9 +117,6 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto bus_err; return 0; bus_err: diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index c48485f2c55..2c18e3d1b71 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -385,19 +385,7 @@ static int ad1836_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets, ARRAY_SIZE(ad1836_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 34b30efc3cb..5d489186c05 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -592,21 +592,9 @@ static int ad1938_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, ad1938_dapm_widgets, ARRAY_SIZE(ad1938_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index d7440a982d2..39c0f7584e6 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -257,11 +257,6 @@ static int ad1980_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, ad1980_snd_ac97_controls, ARRAY_SIZE(ad1980_snd_ac97_controls)); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ad1980: failed to register card\n"); - goto reset_err; - } return 0; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index e61dac5e7b8..d2fcc601722 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -64,16 +64,8 @@ static int ad73311_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ad73311: failed to register card\n"); - goto register_err; - } - return ret; -register_err: - snd_soc_free_pcms(socdev); pcm_err: kfree(socdev->card->codec); socdev->card->codec = NULL; diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c new file mode 100644 index 00000000000..cc96411ca3e --- /dev/null +++ b/sound/soc/codecs/ads117x.c @@ -0,0 +1,123 @@ +/* + * ads117x.c -- Driver for ads1174/8 ADC chips + * + * Copyright 2009 ShotSpotter Inc. + * Author: Graeme Gregory <gg@slimlogic.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "ads117x.h" + +#define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) + +#define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + +struct snd_soc_dai ads117x_dai = { +/* ADC */ + .name = "ADS117X ADC", + .id = 1, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 32, + .rates = ADS117X_RATES, + .formats = ADS117X_FORMATS,}, +}; +EXPORT_SYMBOL_GPL(ads117x_dai); + +static int ads117x_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->card->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->name = "ADS117X"; + codec->owner = THIS_MODULE; + codec->dai = &ads117x_dai; + codec->num_dai = 1; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "ads117x: failed to create pcms\n"); + kfree(codec); + return ret; + } + + return 0; +} + +static int ads117x_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_free_pcms(socdev); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ads117x = { + .probe = ads117x_probe, + .remove = ads117x_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ads117x); + +static __devinit int ads117x_platform_probe(struct platform_device *pdev) +{ + ads117x_dai.dev = &pdev->dev; + return snd_soc_register_dai(&ads117x_dai); +} + +static int __devexit ads117x_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&ads117x_dai); + return 0; +} + +static struct platform_driver ads117x_codec_driver = { + .driver = { + .name = "ads117x", + .owner = THIS_MODULE, + }, + + .probe = ads117x_platform_probe, + .remove = __devexit_p(ads117x_platform_remove), +}; + +static int __init ads117x_init(void) +{ + return platform_driver_register(&ads117x_codec_driver); +} +module_init(ads117x_init); + +static void __exit ads117x_exit(void) +{ + platform_driver_unregister(&ads117x_codec_driver); +} +module_exit(ads117x_exit); + +MODULE_DESCRIPTION("ASoC ads117x driver"); +MODULE_AUTHOR("Graeme Gregory"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ads117x.h b/sound/soc/codecs/ads117x.h new file mode 100644 index 00000000000..dbcf50ec9bd --- /dev/null +++ b/sound/soc/codecs/ads117x.h @@ -0,0 +1,13 @@ +/* + * ads117x.h -- Driver for ads1174/8 ADC chips + * + * Copyright 2009 ShotSpotter Inc. + * Author: Graeme Gregory <gg@slimlogic.co.uk> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +extern struct snd_soc_dai ads117x_dai; +extern struct snd_soc_codec_device soc_codec_dev_ads117x; diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 4d47bc4f742..3a14c6fc4f5 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -313,14 +313,6 @@ static int ak4104_probe(struct platform_device *pdev) return ret; } - /* Register the socdev */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - snd_soc_free_pcms(socdev); - return ret; - } - return 0; } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 0abec0d29a9..ff966567e2b 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -294,7 +294,6 @@ static int ak4535_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -485,17 +484,9 @@ static int ak4535_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, ak4535_snd_controls, ARRAY_SIZE(ak4535_snd_controls)); ak4535_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ak4535: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index e057c7b578d..b69861d5216 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -442,18 +442,9 @@ static int ak4642_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "ak4642: failed to register card\n"); - goto card_err; - } - dev_info(&pdev->dev, "AK4642 Audio Codec %s", AK4642_VERSION); return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c new file mode 100644 index 00000000000..82fca284d00 --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,815 @@ +/* + * ak4671.c -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ak4671.h" + +static struct snd_soc_codec *ak4671_codec; + +/* codec private data */ +struct ak4671_priv { + struct snd_soc_codec codec; + u8 reg_cache[AK4671_CACHEREGNUM]; +}; + +/* ak4671 register cache & default register settings */ +static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { + 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + 0x02, /* AK4671_FORMAT_SELECT (0x03) */ + 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ + 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ + 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ + 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ + 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + 0x00, /* this register not used */ + 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ + 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ + 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ + 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ + 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ + 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ + 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ + 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ + 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ + 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ + 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ + 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ + 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ + 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ + 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ + 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ + 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ + 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ + 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +}; + +/* + * LOUT1/ROUT1 output volume control: + * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB) + */ +static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1); + +/* + * LOUT2/ROUT2 output volume control: + * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB) + */ +static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1); + +/* + * LOUT3/ROUT3 output volume control: + * from -6 to 3 dB in 3 dB steps + */ +static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0); + +/* + * Mic amp gain control: + * from -15 to 30 dB in 3 dB steps + * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not + * available + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new ak4671_snd_controls[] = { + /* Common playback gain controls */ + SOC_SINGLE_TLV("Line Output1 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv), + SOC_SINGLE_TLV("Headphone Output2 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv), + SOC_SINGLE_TLV("Line Output3 Playback Volume", + AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv), + + /* Common capture gain controls */ + SOC_DOUBLE_TLV("Mic Amp Capture Volume", + AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv), +}; + +/* event handlers */ +static int ak4671_out2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + } + + return 0; +} + +/* Output Mixers */ +static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +/* Input MUXs */ +static const char *ak4671_lin_mux_texts[] = + {"LIN1", "LIN2", "LIN3", "LIN4"}; +static const struct soc_enum ak4671_lin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, + ARRAY_SIZE(ak4671_lin_mux_texts), + ak4671_lin_mux_texts); +static const struct snd_kcontrol_new ak4671_lin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); + +static const char *ak4671_rin_mux_texts[] = + {"RIN1", "RIN2", "RIN3", "RIN4"}; +static const struct soc_enum ak4671_rin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, + ARRAY_SIZE(ak4671_rin_mux_texts), + ak4671_rin_mux_texts); +static const struct snd_kcontrol_new ak4671_rin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); + +static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("LIN4"), + SND_SOC_DAPM_INPUT("RIN4"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 6, 0), + SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 7, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 4, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 5, 0), + + /* PGA */ + SND_SOC_DAPM_PGA("LOUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("ROUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0, + &ak4671_lout1_mixer_controls[0], + ARRAY_SIZE(ak4671_lout1_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0, + &ak4671_rout1_mixer_controls[0], + ARRAY_SIZE(ak4671_rout1_mixer_controls)), + SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 0, 0, &ak4671_lout2_mixer_controls[0], + ARRAY_SIZE(ak4671_lout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 1, 0, &ak4671_rout2_mixer_controls[0], + ARRAY_SIZE(ak4671_rout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0, + &ak4671_lout3_mixer_controls[0], + ARRAY_SIZE(ak4671_lout3_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0, + &ak4671_rout3_mixer_controls[0], + ARRAY_SIZE(ak4671_rout3_mixer_controls)), + + /* Input MUXs */ + SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0, + &ak4671_lin_mux_control), + SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0, + &ak4671_rin_mux_control), + + /* Mic Power */ + SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0), + + /* Supply */ + SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DAC Left", "NULL", "PMPLL"}, + {"DAC Right", "NULL", "PMPLL"}, + {"ADC Left", "NULL", "PMPLL"}, + {"ADC Right", "NULL", "PMPLL"}, + + /* Outputs */ + {"LOUT1", "NULL", "LOUT1 Mixer"}, + {"ROUT1", "NULL", "ROUT1 Mixer"}, + {"LOUT2", "NULL", "LOUT2 Mix Amp"}, + {"ROUT2", "NULL", "ROUT2 Mix Amp"}, + {"LOUT3", "NULL", "LOUT3 Mixer"}, + {"ROUT3", "NULL", "ROUT3 Mixer"}, + + {"LOUT1 Mixer", "DACL", "DAC Left"}, + {"ROUT1 Mixer", "DACR", "DAC Right"}, + {"LOUT2 Mixer", "DACHL", "DAC Left"}, + {"ROUT2 Mixer", "DACHR", "DAC Right"}, + {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT3 Mixer", "DACSL", "DAC Left"}, + {"ROUT3 Mixer", "DACSR", "DAC Right"}, + + /* Inputs */ + {"LIN MUX", "LIN1", "LIN1"}, + {"LIN MUX", "LIN2", "LIN2"}, + {"LIN MUX", "LIN3", "LIN3"}, + {"LIN MUX", "LIN4", "LIN4"}, + + {"RIN MUX", "RIN1", "RIN1"}, + {"RIN MUX", "RIN2", "RIN2"}, + {"RIN MUX", "RIN3", "RIN3"}, + {"RIN MUX", "RIN4", "RIN4"}, + + {"LIN1", NULL, "Mic Bias"}, + {"RIN1", NULL, "Mic Bias"}, + {"LIN2", NULL, "Mic Bias"}, + {"RIN2", NULL, "Mic Bias"}, + + {"ADC Left", "NULL", "LIN MUX"}, + {"ADC Right", "NULL", "RIN MUX"}, + + /* Analog Loops */ + {"LIN1 Mixing Circuit", "NULL", "LIN1"}, + {"RIN1 Mixing Circuit", "NULL", "RIN1"}, + {"LIN2 Mixing Circuit", "NULL", "LIN2"}, + {"RIN2 Mixing Circuit", "NULL", "RIN2"}, + {"LIN3 Mixing Circuit", "NULL", "LIN3"}, + {"RIN3 Mixing Circuit", "NULL", "RIN3"}, + {"LIN4 Mixing Circuit", "NULL", "LIN4"}, + {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + + {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, +}; + +static int ak4671_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +static int ak4671_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 fs; + + fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + fs &= ~AK4671_FS; + + switch (params_rate(params)) { + case 8000: + fs |= AK4671_FS_8KHZ; + break; + case 12000: + fs |= AK4671_FS_12KHZ; + break; + case 16000: + fs |= AK4671_FS_16KHZ; + break; + case 24000: + fs |= AK4671_FS_24KHZ; + break; + case 11025: + fs |= AK4671_FS_11_025KHZ; + break; + case 22050: + fs |= AK4671_FS_22_05KHZ; + break; + case 32000: + fs |= AK4671_FS_32KHZ; + break; + case 44100: + fs |= AK4671_FS_44_1KHZ; + break; + case 48000: + fs |= AK4671_FS_48KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs); + + return 0; +} + +static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u8 pll; + + pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + pll &= ~AK4671_PLL; + + switch (freq) { + case 11289600: + pll |= AK4671_PLL_11_2896MHZ; + break; + case 12000000: + pll |= AK4671_PLL_12MHZ; + break; + case 12288000: + pll |= AK4671_PLL_12_288MHZ; + break; + case 13000000: + pll |= AK4671_PLL_13MHZ; + break; + case 13500000: + pll |= AK4671_PLL_13_5MHZ; + break; + case 19200000: + pll |= AK4671_PLL_19_2MHZ; + break; + case 24000000: + pll |= AK4671_PLL_24MHZ; + break; + case 26000000: + pll |= AK4671_PLL_26MHZ; + break; + case 27000000: + pll |= AK4671_PLL_27MHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll); + + return 0; +} + +static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = snd_soc_read(codec, AK4671_FORMAT_SELECT); + format &= ~AK4671_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK4671_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK4671_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_A: + format |= AK4671_DIF_DSP_MODE; + format |= AK4671_BCKP; + format |= AK4671_MSBS; + break; + default: + return -EINVAL; + } + + /* set mode and format */ + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode); + snd_soc_write(codec, AK4671_FORMAT_SELECT, format); + + return 0; +} + +static int ak4671_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops ak4671_dai_ops = { + .hw_params = ak4671_hw_params, + .set_sysclk = ak4671_set_dai_sysclk, + .set_fmt = ak4671_set_dai_fmt, +}; + +struct snd_soc_dai ak4671_dai = { + .name = "AK4671", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .ops = &ak4671_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4671_dai); + +static int ak4671_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ak4671_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4671_codec; + codec = ak4671_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ak4671_snd_controls, + ARRAY_SIZE(ak4671_snd_controls)); + ak4671_add_widgets(codec); + + ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; + +pcm_err: + return ret; +} + +static int ak4671_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4671; + codec->name = "AK4671"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ak4671_set_bias_level; + codec->dai = &ak4671_dai; + codec->num_dai = 1; + codec->reg_cache_size = AK4671_CACHEREGNUM; + codec->reg_cache = &ak4671->reg_cache; + + memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ak4671_dai.dev = codec->dev; + ak4671_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(ak4671); + return ret; +} + +static void ak4671_unregister(struct ak4671_priv *ak4671) +{ + ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ak4671_dai); + snd_soc_unregister_codec(&ak4671->codec); + kfree(ak4671); + ak4671_codec = NULL; +} + +static int __devinit ak4671_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct ak4671_priv *ak4671; + struct snd_soc_codec *codec; + + ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + if (ak4671 == NULL) + return -ENOMEM; + + codec = &ak4671->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(client, ak4671); + codec->control_data = client; + + codec->dev = &client->dev; + + return ak4671_register(ak4671, SND_SOC_I2C); +} + +static __devexit int ak4671_i2c_remove(struct i2c_client *client) +{ + struct ak4671_priv *ak4671 = i2c_get_clientdata(client); + + ak4671_unregister(ak4671); + + return 0; +} + +static const struct i2c_device_id ak4671_i2c_id[] = { + { "ak4671", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); + +static struct i2c_driver ak4671_i2c_driver = { + .driver = { + .name = "ak4671", + .owner = THIS_MODULE, + }, + .probe = ak4671_i2c_probe, + .remove = __devexit_p(ak4671_i2c_remove), + .id_table = ak4671_i2c_id, +}; + +static int __init ak4671_modinit(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_modinit); + +static void __exit ak4671_exit(void) +{ + i2c_del_driver(&ak4671_i2c_driver); +} +module_exit(ak4671_exit); + +MODULE_DESCRIPTION("ASoC AK4671 codec driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h new file mode 100644 index 00000000000..e2fad964e88 --- /dev/null +++ b/sound/soc/codecs/ak4671.h @@ -0,0 +1,156 @@ +/* + * ak4671.h -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _AK4671_H +#define _AK4671_H + +#define AK4671_AD_DA_POWER_MANAGEMENT 0x00 +#define AK4671_PLL_MODE_SELECT0 0x01 +#define AK4671_PLL_MODE_SELECT1 0x02 +#define AK4671_FORMAT_SELECT 0x03 +#define AK4671_MIC_SIGNAL_SELECT 0x04 +#define AK4671_MIC_AMP_GAIN 0x05 +#define AK4671_MIXING_POWER_MANAGEMENT0 0x06 +#define AK4671_MIXING_POWER_MANAGEMENT1 0x07 +#define AK4671_OUTPUT_VOLUME_CONTROL 0x08 +#define AK4671_LOUT1_SIGNAL_SELECT 0x09 +#define AK4671_ROUT1_SIGNAL_SELECT 0x0a +#define AK4671_LOUT2_SIGNAL_SELECT 0x0b +#define AK4671_ROUT2_SIGNAL_SELECT 0x0c +#define AK4671_LOUT3_SIGNAL_SELECT 0x0d +#define AK4671_ROUT3_SIGNAL_SELECT 0x0e +#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f +#define AK4671_LOUT2_POWER_MANAGERMENT 0x10 +#define AK4671_LOUT3_POWER_MANAGERMENT 0x11 +#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12 +#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13 +#define AK4671_ALC_REFERENCE_SELECT 0x14 +#define AK4671_DIGITAL_MIXING_CONTROL 0x15 +#define AK4671_ALC_TIMER_SELECT 0x16 +#define AK4671_ALC_MODE_CONTROL 0x17 +#define AK4671_MODE_CONTROL1 0x18 +#define AK4671_MODE_CONTROL2 0x19 +#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a +#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b +#define AK4671_SIDETONE_A_CONTROL 0x1c +#define AK4671_DIGITAL_FILTER_SELECT 0x1d +#define AK4671_FIL3_COEFFICIENT0 0x1e +#define AK4671_FIL3_COEFFICIENT1 0x1f +#define AK4671_FIL3_COEFFICIENT2 0x20 +#define AK4671_FIL3_COEFFICIENT3 0x21 +#define AK4671_EQ_COEFFICIENT0 0x22 +#define AK4671_EQ_COEFFICIENT1 0x23 +#define AK4671_EQ_COEFFICIENT2 0x24 +#define AK4671_EQ_COEFFICIENT3 0x25 +#define AK4671_EQ_COEFFICIENT4 0x26 +#define AK4671_EQ_COEFFICIENT5 0x27 +#define AK4671_FIL1_COEFFICIENT0 0x28 +#define AK4671_FIL1_COEFFICIENT1 0x29 +#define AK4671_FIL1_COEFFICIENT2 0x2a +#define AK4671_FIL1_COEFFICIENT3 0x2b +#define AK4671_FIL2_COEFFICIENT0 0x2c +#define AK4671_FIL2_COEFFICIENT1 0x2d +#define AK4671_FIL2_COEFFICIENT2 0x2e +#define AK4671_FIL2_COEFFICIENT3 0x2f +#define AK4671_DIGITAL_FILTER_SELECT2 0x30 +#define AK4671_E1_COEFFICIENT0 0x32 +#define AK4671_E1_COEFFICIENT1 0x33 +#define AK4671_E1_COEFFICIENT2 0x34 +#define AK4671_E1_COEFFICIENT3 0x35 +#define AK4671_E1_COEFFICIENT4 0x36 +#define AK4671_E1_COEFFICIENT5 0x37 +#define AK4671_E2_COEFFICIENT0 0x38 +#define AK4671_E2_COEFFICIENT1 0x39 +#define AK4671_E2_COEFFICIENT2 0x3a +#define AK4671_E2_COEFFICIENT3 0x3b +#define AK4671_E2_COEFFICIENT4 0x3c +#define AK4671_E2_COEFFICIENT5 0x3d +#define AK4671_E3_COEFFICIENT0 0x3e +#define AK4671_E3_COEFFICIENT1 0x3f +#define AK4671_E3_COEFFICIENT2 0x40 +#define AK4671_E3_COEFFICIENT3 0x41 +#define AK4671_E3_COEFFICIENT4 0x42 +#define AK4671_E3_COEFFICIENT5 0x43 +#define AK4671_E4_COEFFICIENT0 0x44 +#define AK4671_E4_COEFFICIENT1 0x45 +#define AK4671_E4_COEFFICIENT2 0x46 +#define AK4671_E4_COEFFICIENT3 0x47 +#define AK4671_E4_COEFFICIENT4 0x48 +#define AK4671_E4_COEFFICIENT5 0x49 +#define AK4671_E5_COEFFICIENT0 0x4a +#define AK4671_E5_COEFFICIENT1 0x4b +#define AK4671_E5_COEFFICIENT2 0x4c +#define AK4671_E5_COEFFICIENT3 0x4d +#define AK4671_E5_COEFFICIENT4 0x4e +#define AK4671_E5_COEFFICIENT5 0x4f +#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50 +#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51 +#define AK4671_EQ_CONTRO_10KHZ 0x52 +#define AK4671_PCM_IF_CONTROL0 0x53 +#define AK4671_PCM_IF_CONTROL1 0x54 +#define AK4671_PCM_IF_CONTROL2 0x55 +#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56 +#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57 +#define AK4671_SIDETONE_VOLUME_CONTROL 0x58 +#define AK4671_DIGITAL_MIXING_CONTROL2 0x59 +#define AK4671_SAR_ADC_CONTROL 0x5a + +#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) + +/* Bitfield Definitions */ + +/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ +#define AK4671_PMVCM 0x01 + +/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */ +#define AK4671_PLL 0x0f +#define AK4671_PLL_11_2896MHZ (4 << 0) +#define AK4671_PLL_12_288MHZ (5 << 0) +#define AK4671_PLL_12MHZ (6 << 0) +#define AK4671_PLL_24MHZ (7 << 0) +#define AK4671_PLL_19_2MHZ (8 << 0) +#define AK4671_PLL_13_5MHZ (12 << 0) +#define AK4671_PLL_27MHZ (13 << 0) +#define AK4671_PLL_13MHZ (14 << 0) +#define AK4671_PLL_26MHZ (15 << 0) +#define AK4671_FS 0xf0 +#define AK4671_FS_8KHZ (0 << 4) +#define AK4671_FS_12KHZ (1 << 4) +#define AK4671_FS_16KHZ (2 << 4) +#define AK4671_FS_24KHZ (3 << 4) +#define AK4671_FS_11_025KHZ (5 << 4) +#define AK4671_FS_22_05KHZ (7 << 4) +#define AK4671_FS_32KHZ (10 << 4) +#define AK4671_FS_48KHZ (11 << 4) +#define AK4671_FS_44_1KHZ (15 << 4) + +/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */ +#define AK4671_PMPLL 0x01 +#define AK4671_M_S 0x02 + +/* AK4671_FORMAT_SELECT (0x03) Fields */ +#define AK4671_DIF 0x03 +#define AK4671_DIF_DSP_MODE (0 << 0) +#define AK4671_DIF_MSB_MODE (2 << 0) +#define AK4671_DIF_I2S_MODE (3 << 0) +#define AK4671_BCKP 0x04 +#define AK4671_MSBS 0x08 +#define AK4671_SDOD 0x10 + +/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */ +#define AK4671_MUTEN 0x04 + +extern struct snd_soc_dai ak4671_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4671; + +#endif diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index ca1e24a8f12..ffe122d1cd7 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -520,6 +520,7 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0), SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0), + SOC_SINGLE("De-emphasis filter", CS4270_TRANS, 0, 1, 0), SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1), SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0), SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1), @@ -598,13 +599,6 @@ static int cs4270_probe(struct platform_device *pdev) goto error_free_pcms; } - /* And finally, register the socdev */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - goto error_free_pcms; - } - return 0; error_free_pcms: @@ -802,22 +796,6 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); * and all registers are written back to the hardware when resuming. */ -static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) -{ - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; - - return snd_soc_suspend_device(codec->dev); -} - -static int cs4270_i2c_resume(struct i2c_client *client) -{ - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; - - return snd_soc_resume_device(codec->dev); -} - static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { struct snd_soc_codec *codec = cs4270_codec; @@ -853,8 +831,6 @@ static int cs4270_soc_resume(struct platform_device *pdev) return snd_soc_write(codec, CS4270_PWRCTL, reg); } #else -#define cs4270_i2c_suspend NULL -#define cs4270_i2c_resume NULL #define cs4270_soc_suspend NULL #define cs4270_soc_resume NULL #endif /* CONFIG_PM */ @@ -873,8 +849,6 @@ static struct i2c_driver cs4270_i2c_driver = { .id_table = cs4270_id, .probe = cs4270_i2c_probe, .remove = cs4270_i2c_remove, - .suspend = cs4270_i2c_suspend, - .resume = cs4270_i2c_resume, }; /* diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index 38eac9c866e..e000cdfec1e 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -93,7 +93,6 @@ static int cx20442_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, cx20442_audio_map, ARRAY_SIZE(cx20442_audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -355,17 +354,6 @@ static int cx20442_codec_probe(struct platform_device *pdev) cx20442_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 5cda9e6b5a7..2afcd0a8669 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -90,13 +90,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev) goto pcm_err; } - /* Register Card. */ - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "pcm3008: failed to register card\n"); - goto card_err; - } - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF @@ -136,8 +129,6 @@ static int pcm3008_soc_probe(struct platform_device *pdev) gpio_err: pcm3008_gpio_free(setup); -card_err: - snd_soc_free_pcms(socdev); pcm_err: kfree(socdev->card->codec); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c550750c79c..d2ff1cde688 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -210,7 +210,6 @@ static int ssm2602_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_conn, ARRAY_SIZE(audio_conn)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -613,17 +612,9 @@ static int ssm2602_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, ssm2602_snd_controls, ARRAY_SIZE(ssm2602_snd_controls)); ssm2602_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - pr_err("ssm2602: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index befc6488c39..bbc72c2ddfc 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -418,9 +418,6 @@ static int stac9766_codec_probe(struct platform_device *pdev) snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto reset_err; return 0; reset_err: diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 90a0264f753..a9dc5fb5477 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -85,7 +85,7 @@ static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg, * of data into val */ - if ((reg < 0 || reg > 9) && (reg != 15)) { + if (reg > 9 && reg != 15) { printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); return -1; } @@ -395,7 +395,6 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -706,17 +705,9 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, tlv320aic23_snd_controls, ARRAY_SIZE(tlv320aic23_snd_controls)); tlv320aic23_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "tlv320aic23: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 3387d9e736e..357b609196e 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -356,18 +356,7 @@ static int aic26_probe(struct platform_device *pdev) ARRAY_SIZE(aic26_snd_controls)); WARN_ON(err < 0); - /* CODEC is setup, we can register the card now */ - dev_dbg(&pdev->dev, "Registering card\n"); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "aic26: failed to register card\n"); - goto card_err; - } return 0; - - card_err: - snd_soc_free_pcms(socdev); - return ret; } static int aic26_remove(struct platform_device *pdev) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 3395cf945d5..2b4dc2b0b01 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -753,7 +753,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -1405,18 +1404,8 @@ static int aic3x_probe(struct platform_device *pdev) aic3x_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "aic3x: failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c new file mode 100644 index 00000000000..9c8903dbe64 --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.c @@ -0,0 +1,1229 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include <sound/tlv320dac33-plat.h> +#include "tlv320dac33.h" + +#define DAC33_BUFFER_SIZE_BYTES 24576 /* bytes, 12288 16 bit words, + * 6144 stereo */ +#define DAC33_BUFFER_SIZE_SAMPLES 6144 + +#define NSAMPLE_MAX 5700 + +#define LATENCY_TIME_MS 20 + +static struct snd_soc_codec *tlv320dac33_codec; + +enum dac33_state { + DAC33_IDLE = 0, + DAC33_PREFILL, + DAC33_PLAYBACK, + DAC33_FLUSH, +}; + +struct tlv320dac33_priv { + struct mutex mutex; + struct workqueue_struct *dac33_wq; + struct work_struct work; + struct snd_soc_codec codec; + int power_gpio; + int chip_power; + int irq; + unsigned int refclk; + + unsigned int alarm_threshold; /* set to be half of LATENCY_TIME_MS */ + unsigned int nsample_min; /* nsample should not be lower than + * this */ + unsigned int nsample_max; /* nsample should not be higher than + * this */ + unsigned int nsample_switch; /* Use FIFO or bypass FIFO switch */ + unsigned int nsample; /* burst read amount from host */ + + enum dac33_state state; +}; + +static const u8 dac33_reg[DAC33_CACHEREGNUM] = { +0x00, 0x00, 0x00, 0x00, /* 0x00 - 0x03 */ +0x00, 0x00, 0x00, 0x00, /* 0x04 - 0x07 */ +0x00, 0x00, 0x00, 0x00, /* 0x08 - 0x0b */ +0x00, 0x00, 0x00, 0x00, /* 0x0c - 0x0f */ +0x00, 0x00, 0x00, 0x00, /* 0x10 - 0x13 */ +0x00, 0x00, 0x00, 0x00, /* 0x14 - 0x17 */ +0x00, 0x00, 0x00, 0x00, /* 0x18 - 0x1b */ +0x00, 0x00, 0x00, 0x00, /* 0x1c - 0x1f */ +0x00, 0x00, 0x00, 0x00, /* 0x20 - 0x23 */ +0x00, 0x00, 0x00, 0x00, /* 0x24 - 0x27 */ +0x00, 0x00, 0x00, 0x00, /* 0x28 - 0x2b */ +0x00, 0x00, 0x00, 0x80, /* 0x2c - 0x2f */ +0x80, 0x00, 0x00, 0x00, /* 0x30 - 0x33 */ +0x00, 0x00, 0x00, 0x00, /* 0x34 - 0x37 */ +0x00, 0x00, /* 0x38 - 0x39 */ +/* Registers 0x3a - 0x3f are reserved */ + 0x00, 0x00, /* 0x3a - 0x3b */ +0x00, 0x00, 0x00, 0x00, /* 0x3c - 0x3f */ + +0x00, 0x00, 0x00, 0x00, /* 0x40 - 0x43 */ +0x00, 0x80, /* 0x44 - 0x45 */ +/* Registers 0x46 - 0x47 are reserved */ + 0x80, 0x80, /* 0x46 - 0x47 */ + +0x80, 0x00, 0x00, /* 0x48 - 0x4a */ +/* Registers 0x4b - 0x7c are reserved */ + 0x00, /* 0x4b */ +0x00, 0x00, 0x00, 0x00, /* 0x4c - 0x4f */ +0x00, 0x00, 0x00, 0x00, /* 0x50 - 0x53 */ +0x00, 0x00, 0x00, 0x00, /* 0x54 - 0x57 */ +0x00, 0x00, 0x00, 0x00, /* 0x58 - 0x5b */ +0x00, 0x00, 0x00, 0x00, /* 0x5c - 0x5f */ +0x00, 0x00, 0x00, 0x00, /* 0x60 - 0x63 */ +0x00, 0x00, 0x00, 0x00, /* 0x64 - 0x67 */ +0x00, 0x00, 0x00, 0x00, /* 0x68 - 0x6b */ +0x00, 0x00, 0x00, 0x00, /* 0x6c - 0x6f */ +0x00, 0x00, 0x00, 0x00, /* 0x70 - 0x73 */ +0x00, 0x00, 0x00, 0x00, /* 0x74 - 0x77 */ +0x00, 0x00, 0x00, 0x00, /* 0x78 - 0x7b */ +0x00, /* 0x7c */ + + 0xda, 0x33, 0x03, /* 0x7d - 0x7f */ +}; + +/* Register read and write */ +static inline unsigned int dac33_read_reg_cache(struct snd_soc_codec *codec, + unsigned reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return 0; + + return cache[reg]; +} + +static inline void dac33_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + if (reg >= DAC33_CACHEREGNUM) + return; + + cache[reg] = value; +} + +static int dac33_read(struct snd_soc_codec *codec, unsigned int reg, + u8 *value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int val; + + *value = reg & 0xff; + + /* If powered off, return the cached value */ + if (dac33->chip_power) { + val = i2c_smbus_read_byte_data(codec->control_data, value[0]); + if (val < 0) { + dev_err(codec->dev, "Read failed (%d)\n", val); + value[0] = dac33_read_reg_cache(codec, reg); + } else { + value[0] = val; + dac33_write_reg_cache(codec, reg, val); + } + } else { + value[0] = dac33_read_reg_cache(codec, reg); + } + + return 0; +} + +static int dac33_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[2]; + int ret = 0; + + /* + * data is + * D15..D8 dac33 register offset + * D7...D0 register data + */ + data[0] = reg & 0xff; + data[1] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + if (dac33->chip_power) { + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +static int dac33_write_locked(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret; + + mutex_lock(&dac33->mutex); + ret = dac33_write(codec, reg, value); + mutex_unlock(&dac33->mutex); + + return ret; +} + +#define DAC33_I2C_ADDR_AUTOINC 0x80 +static int dac33_write16(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 data[3]; + int ret = 0; + + /* + * data is + * D23..D16 dac33 register offset + * D15..D8 register data MSB + * D7...D0 register data LSB + */ + data[0] = reg & 0xff; + data[1] = (value >> 8) & 0xff; + data[2] = value & 0xff; + + dac33_write_reg_cache(codec, data[0], data[1]); + dac33_write_reg_cache(codec, data[0] + 1, data[2]); + + if (dac33->chip_power) { + /* We need to set autoincrement mode for 16 bit writes */ + data[0] |= DAC33_I2C_ADDR_AUTOINC; + ret = codec->hw_write(codec->control_data, data, 3); + if (ret != 3) + dev_err(codec->dev, "Write failed (%d)\n", ret); + else + ret = 0; + } + + return ret; +} + +static void dac33_restore_regs(struct snd_soc_codec *codec) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 *cache = codec->reg_cache; + u8 data[2]; + int i, ret; + + if (!dac33->chip_power) + return; + + for (i = DAC33_PWR_CTRL; i <= DAC33_INTP_CTRL_B; i++) { + data[0] = i; + data[1] = cache[i]; + /* Skip the read only registers */ + if ((i >= DAC33_INT_OSC_STATUS && + i <= DAC33_INT_OSC_FREQ_RAT_READ_B) || + (i >= DAC33_FIFO_WPTR_MSB && i <= DAC33_FIFO_IRQ_FLAG) || + i == DAC33_DAC_STATUS_FLAGS || + i == DAC33_SRC_EST_REF_CLK_RATIO_A || + i == DAC33_SRC_EST_REF_CLK_RATIO_B) + continue; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LDAC_PWR_CTRL; i <= DAC33_LINEL_TO_LLO_VOL; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } + for (i = DAC33_LINER_TO_RLO_VOL; i <= DAC33_OSC_TRIM; i++) { + data[0] = i; + data[1] = cache[i]; + ret = codec->hw_write(codec->control_data, data, 2); + if (ret != 2) + dev_err(codec->dev, "Write failed (%d)\n", ret); + } +} + +static inline void dac33_soft_power(struct snd_soc_codec *codec, int power) +{ + u8 reg; + + reg = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + if (power) + reg |= DAC33_PDNALLB; + else + reg &= ~DAC33_PDNALLB; + dac33_write(codec, DAC33_PWR_CTRL, reg); +} + +static void dac33_hard_power(struct snd_soc_codec *codec, int power) +{ + struct tlv320dac33_priv *dac33 = codec->private_data; + + mutex_lock(&dac33->mutex); + if (power) { + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 1); + dac33->chip_power = 1; + /* Restore registers */ + dac33_restore_regs(codec); + } + dac33_soft_power(codec, 1); + } else { + dac33_soft_power(codec, 0); + if (dac33->power_gpio >= 0) { + gpio_set_value(dac33->power_gpio, 0); + dac33->chip_power = 0; + } + } + mutex_unlock(&dac33->mutex); + +} + +static int dac33_get_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample; + + return 0; +} + +static int dac33_set_nsample(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample == ucontrol->value.integer.value[0]) + return 0; + + if (ucontrol->value.integer.value[0] < dac33->nsample_min || + ucontrol->value.integer.value[0] > dac33->nsample_max) + ret = -EINVAL; + else + dac33->nsample = ucontrol->value.integer.value[0]; + + return ret; +} + +static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + + ucontrol->value.integer.value[0] = dac33->nsample_switch; + + return 0; +} + +static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + if (dac33->nsample_switch == ucontrol->value.integer.value[0]) + return 0; + /* Do not allow changes while stream is running*/ + if (codec->active) + return -EPERM; + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 1) + ret = -EINVAL; + else + dac33->nsample_switch = ucontrol->value.integer.value[0]; + + return ret; +} + +/* + * DACL/R digital volume control: + * from 0 dB to -63.5 in 0.5 dB steps + * Need to be inverted later on: + * 0x00 == 0 dB + * 0x7f == -63.5 dB + */ +static DECLARE_TLV_DB_SCALE(dac_digivol_tlv, -6350, 50, 0); + +static const struct snd_kcontrol_new dac33_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC Digital Playback Volume", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, + 0, 0x7f, 1, dac_digivol_tlv), + SOC_DOUBLE_R("DAC Digital Playback Switch", + DAC33_LDAC_DIG_VOL_CTRL, DAC33_RDAC_DIG_VOL_CTRL, 7, 1, 1), + SOC_DOUBLE_R("Line to Line Out Volume", + DAC33_LINEL_TO_LLO_VOL, DAC33_LINER_TO_RLO_VOL, 0, 127, 1), +}; + +static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = { + SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0, + dac33_get_nsample, dac33_set_nsample), + SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0, + dac33_get_nsample_switch, dac33_set_nsample_switch), +}; + +/* Analog bypass */ +static const struct snd_kcontrol_new dac33_dapm_abypassl_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINEL_TO_LLO_VOL, 7, 1, 1); + +static const struct snd_kcontrol_new dac33_dapm_abypassr_control = + SOC_DAPM_SINGLE("Switch", DAC33_LINER_TO_RLO_VOL, 7, 1, 1); + +static const struct snd_soc_dapm_widget dac33_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("LEFT_LO"), + SND_SOC_DAPM_OUTPUT("RIGHT_LO"), + + SND_SOC_DAPM_INPUT("LINEL"), + SND_SOC_DAPM_INPUT("LINER"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", DAC33_LDAC_PWR_CTRL, 2, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", DAC33_RDAC_PWR_CTRL, 2, 0), + + /* Analog bypass */ + SND_SOC_DAPM_SWITCH("Analog Left Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassl_control), + SND_SOC_DAPM_SWITCH("Analog Right Bypass", SND_SOC_NOPM, 0, 0, + &dac33_dapm_abypassr_control), + + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Left Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_mixer, "Output Right Amp Power", + DAC33_OUT_AMP_PWR_CTRL, 4, 3, 3, 0), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Analog bypass */ + {"Analog Left Bypass", "Switch", "LINEL"}, + {"Analog Right Bypass", "Switch", "LINER"}, + + {"Output Left Amp Power", NULL, "DACL"}, + {"Output Right Amp Power", NULL, "DACR"}, + + {"Output Left Amp Power", NULL, "Analog Left Bypass"}, + {"Output Right Amp Power", NULL, "Analog Right Bypass"}, + + /* output */ + {"LEFT_LO", NULL, "Output Left Amp Power"}, + {"RIGHT_LO", NULL, "Output Right Amp Power"}, +}; + +static int dac33_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dac33_dapm_widgets, + ARRAY_SIZE(dac33_dapm_widgets)); + + /* set up audio path interconnects */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +static int dac33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + dac33_soft_power(codec, 1); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) + dac33_hard_power(codec, 1); + dac33_soft_power(codec, 0); + break; + case SND_SOC_BIAS_OFF: + dac33_hard_power(codec, 0); + break; + } + codec->bias_level = level; + + return 0; +} + +static void dac33_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + u8 reg; + + dac33 = container_of(work, struct tlv320dac33_priv, work); + codec = &dac33->codec; + + mutex_lock(&dac33->mutex); + switch (dac33->state) { + case DAC33_PREFILL: + dac33->state = DAC33_PLAYBACK; + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + dac33_write16(codec, DAC33_PREFILL_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + break; + case DAC33_PLAYBACK: + dac33_write16(codec, DAC33_NSAMPLE_MSB, + DAC33_THRREG(dac33->nsample)); + break; + case DAC33_IDLE: + break; + case DAC33_FLUSH: + dac33->state = DAC33_IDLE; + /* Mask all interrupts from dac33 */ + dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0); + + /* flush fifo */ + reg = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + reg |= DAC33_FIFOFLUSH; + dac33_write(codec, DAC33_FIFO_CTRL_A, reg); + break; + } + mutex_unlock(&dac33->mutex); +} + +static irqreturn_t dac33_interrupt_handler(int irq, void *dev) +{ + struct snd_soc_codec *codec = dev; + struct tlv320dac33_priv *dac33 = codec->private_data; + + queue_work(dac33->dac33_wq, &dac33->work); + + return IRQ_HANDLED; +} + +static void dac33_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int pwr_ctrl; + + /* Stop pending workqueue */ + if (dac33->nsample_switch) + cancel_work_sync(&dac33->work); + + mutex_lock(&dac33->mutex); + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl &= ~(DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB); + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + mutex_unlock(&dac33->mutex); +} + +static void dac33_oscwait(struct snd_soc_codec *codec) +{ + int timeout = 20; + u8 reg; + + do { + msleep(1); + dac33_read(codec, DAC33_INT_OSC_STATUS, ®); + } while (((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) && timeout--); + if ((reg & 0x03) != DAC33_OSCSTATUS_NORMAL) + dev_err(codec->dev, + "internal oscillator calibration failed\n"); +} + +static int dac33_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* Check parameters for validity */ + switch (params_rate(params)) { + case 44100: + case 48000: + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + params_rate(params)); + return -EINVAL; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + params_format(params)); + return -EINVAL; + } + + return 0; +} + +#define CALC_OSCSET(rate, refclk) ( \ + ((((rate * 10000) / refclk) * 4096) + 5000) / 10000) +#define CALC_RATIOSET(rate, refclk) ( \ + ((((refclk * 100000) / rate) * 16384) + 50000) / 100000) + +/* + * tlv320dac33 is strict on the sequence of the register writes, if the register + * writes happens in different order, than dac33 might end up in unknown state. + * Use the known, working sequence of register writes to initialize the dac33. + */ +static int dac33_prepare_chip(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int oscset, ratioset, pwr_ctrl, reg_tmp; + u8 aictrl_a, fifoctrl_a; + + switch (substream->runtime->rate) { + case 44100: + case 48000: + oscset = CALC_OSCSET(substream->runtime->rate, dac33->refclk); + ratioset = CALC_RATIOSET(substream->runtime->rate, + dac33->refclk); + break; + default: + dev_err(codec->dev, "unsupported rate %d\n", + substream->runtime->rate); + return -EINVAL; + } + + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK); + fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A); + fifoctrl_a &= ~DAC33_WIDTH; + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + aictrl_a |= (DAC33_NCYCL_16 | DAC33_WLEN_16); + fifoctrl_a |= DAC33_WIDTH; + break; + default: + dev_err(codec->dev, "unsupported format %d\n", + substream->runtime->format); + return -EINVAL; + } + + mutex_lock(&dac33->mutex); + dac33_soft_power(codec, 1); + + reg_tmp = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + dac33_write(codec, DAC33_INT_OSC_CTRL, reg_tmp); + + /* Write registers 0x08 and 0x09 (MSB, LSB) */ + dac33_write16(codec, DAC33_INT_OSC_FREQ_RAT_A, oscset); + + /* calib time: 128 is a nice number ;) */ + dac33_write(codec, DAC33_CALIB_TIME, 128); + + /* adjustment treshold & step */ + dac33_write(codec, DAC33_INT_OSC_CTRL_B, DAC33_ADJTHRSHLD(2) | + DAC33_ADJSTEP(1)); + + /* div=4 / gain=1 / div */ + dac33_write(codec, DAC33_INT_OSC_CTRL_C, DAC33_REFDIV(4)); + + pwr_ctrl = dac33_read_reg_cache(codec, DAC33_PWR_CTRL); + pwr_ctrl |= DAC33_OSCPDNB | DAC33_DACRPDNB | DAC33_DACLPDNB; + dac33_write(codec, DAC33_PWR_CTRL, pwr_ctrl); + + dac33_oscwait(codec); + + if (dac33->nsample_switch) { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ + + /* Write registers 0x34 and 0x35 (MSB, LSB) */ + dac33_write16(codec, DAC33_SRC_REF_CLK_RATIO_A, ratioset); + + /* Set interrupts to high active */ + dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH); + + dac33_write(codec, DAC33_FIFO_IRQ_MODE_B, + DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL)); + dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT); + } else { + /* 50-51 : ASRC Control registers */ + dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP); + dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */ + } + + if (dac33->nsample_switch) + fifoctrl_a &= ~DAC33_FBYPAS; + else + fifoctrl_a |= DAC33_FBYPAS; + dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a); + + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + if (dac33->nsample_switch) + reg_tmp &= ~DAC33_BCLKON; + else + reg_tmp |= DAC33_BCLKON; + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp); + + if (dac33->nsample_switch) { + /* 20: BCLK divide ratio */ + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3); + + dac33_write16(codec, DAC33_ATHR_MSB, + DAC33_THRREG(dac33->alarm_threshold)); + } else { + dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32); + } + + mutex_unlock(&dac33->mutex); + + return 0; +} + +static void dac33_calculate_times(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + unsigned int nsample_limit; + + /* Number of samples (16bit, stereo) in one period */ + dac33->nsample_min = snd_pcm_lib_period_bytes(substream) / 4; + + /* Number of samples (16bit, stereo) in ALSA buffer */ + dac33->nsample_max = snd_pcm_lib_buffer_bytes(substream) / 4; + /* Subtract one period from the total */ + dac33->nsample_max -= dac33->nsample_min; + + /* Number of samples for LATENCY_TIME_MS / 2 */ + dac33->alarm_threshold = substream->runtime->rate / + (1000 / (LATENCY_TIME_MS / 2)); + + /* Find and fix up the lowest nsmaple limit */ + nsample_limit = substream->runtime->rate / (1000 / LATENCY_TIME_MS); + + if (dac33->nsample_min < nsample_limit) + dac33->nsample_min = nsample_limit; + + if (dac33->nsample < dac33->nsample_min) + dac33->nsample = dac33->nsample_min; + + /* + * Find and fix up the highest nsmaple limit + * In order to not overflow the DAC33 buffer substract the + * alarm_threshold value from the size of the DAC33 buffer + */ + nsample_limit = DAC33_BUFFER_SIZE_SAMPLES - dac33->alarm_threshold; + + if (dac33->nsample_max > nsample_limit) + dac33->nsample_max = nsample_limit; + + if (dac33->nsample > dac33->nsample_max) + dac33->nsample = dac33->nsample_max; +} + +static int dac33_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + dac33_calculate_times(substream); + dac33_prepare_chip(substream); + + return 0; +} + +static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (dac33->nsample_switch) { + dac33->state = DAC33_PREFILL; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (dac33->nsample_switch) { + dac33->state = DAC33_FLUSH; + queue_work(dac33->dac33_wq, &dac33->work); + } + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int dac33_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tlv320dac33_priv *dac33 = codec->private_data; + u8 ioc_reg, asrcb_reg; + + ioc_reg = dac33_read_reg_cache(codec, DAC33_INT_OSC_CTRL); + asrcb_reg = dac33_read_reg_cache(codec, DAC33_ASRC_CTRL_B); + switch (clk_id) { + case TLV320DAC33_MCLK: + ioc_reg |= DAC33_REFSEL; + asrcb_reg |= DAC33_SRCREFSEL; + break; + case TLV320DAC33_SLEEPCLK: + ioc_reg &= ~DAC33_REFSEL; + asrcb_reg &= ~DAC33_SRCREFSEL; + break; + default: + dev_err(codec->dev, "Invalid clock ID (%d)\n", clk_id); + break; + } + dac33->refclk = freq; + + dac33_write_reg_cache(codec, DAC33_INT_OSC_CTRL, ioc_reg); + dac33_write_reg_cache(codec, DAC33_ASRC_CTRL_B, asrcb_reg); + + return 0; +} + +static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 aictrl_a, aictrl_b; + + aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A); + aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B); + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + /* Codec Master */ + aictrl_a |= (DAC33_MSBCLK | DAC33_MSWCLK); + break; + case SND_SOC_DAIFMT_CBS_CFS: + /* Codec Slave */ + aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK); + break; + default: + return -EINVAL; + } + + aictrl_a &= ~DAC33_AFMT_MASK; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aictrl_a |= DAC33_AFMT_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; + aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */ + break; + case SND_SOC_DAIFMT_DSP_B: + aictrl_a |= DAC33_AFMT_DSP; + aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */ + break; + case SND_SOC_DAIFMT_RIGHT_J: + aictrl_a |= DAC33_AFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + aictrl_a |= DAC33_AFMT_LEFT_J; + break; + default: + dev_err(codec->dev, "Unsupported format (%u)\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a); + dac33_write_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b); + + return 0; +} + +static void dac33_init_chip(struct snd_soc_codec *codec) +{ + /* 44-46: DAC Control Registers */ + /* A : DAC sample rate Fsref/1.5 */ + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1)); + /* B : DAC src=normal, not muted */ + dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | + DAC33_DACSRCL_LEFT); + /* C : (defaults) */ + dac33_write(codec, DAC33_DAC_CTRL_C, 0x00); + + /* 64-65 : L&R DAC power control + Line In -> OUT 1V/V Gain, DAC -> OUT 4V/V Gain*/ + dac33_write(codec, DAC33_LDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + dac33_write(codec, DAC33_RDAC_PWR_CTRL, DAC33_LROUT_GAIN(2)); + + /* 73 : volume soft stepping control, + clock source = internal osc (?) */ + dac33_write(codec, DAC33_ANA_VOL_SOFT_STEP_CTRL, DAC33_VOLCLKEN); + + /* 66 : LOP/LOM Modes */ + dac33_write(codec, DAC33_OUT_AMP_CM_CTRL, 0xff); + + /* 68 : LOM inverted from LOP */ + dac33_write(codec, DAC33_OUT_AMP_CTRL, (3<<2)); + + dac33_write(codec, DAC33_PWR_CTRL, DAC33_PDNALLB); +} + +static int dac33_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct tlv320dac33_priv *dac33; + int ret = 0; + + BUG_ON(!tlv320dac33_codec); + + codec = tlv320dac33_codec; + socdev->card->codec = codec; + dac33 = codec->private_data; + + /* Power up the codec */ + dac33_hard_power(codec, 1); + /* Set default configuration */ + dac33_init_chip(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms\n"); + goto pcm_err; + } + + snd_soc_add_controls(codec, dac33_snd_controls, + ARRAY_SIZE(dac33_snd_controls)); + /* Only add the nSample controls, if we have valid IRQ number */ + if (dac33->irq >= 0) + snd_soc_add_controls(codec, dac33_nsample_snd_controls, + ARRAY_SIZE(dac33_nsample_snd_controls)); + + dac33_add_widgets(codec); + + /* power on device */ + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; + +pcm_err: + dac33_hard_power(codec, 0); + return ret; +} + +static int dac33_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +static int dac33_soc_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int dac33_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + dac33_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_tlv320dac33 = { + .probe = dac33_soc_probe, + .remove = dac33_soc_remove, + .suspend = dac33_soc_suspend, + .resume = dac33_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320dac33); + +#define DAC33_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) +#define DAC33_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops dac33_dai_ops = { + .shutdown = dac33_shutdown, + .hw_params = dac33_hw_params, + .prepare = dac33_pcm_prepare, + .trigger = dac33_pcm_trigger, + .set_sysclk = dac33_set_dai_sysclk, + .set_fmt = dac33_set_dai_fmt, +}; + +struct snd_soc_dai dac33_dai = { + .name = "tlv320dac33", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = DAC33_RATES, + .formats = DAC33_FORMATS,}, + .ops = &dac33_dai_ops, +}; +EXPORT_SYMBOL_GPL(dac33_dai); + +static int dac33_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_priv *dac33; + struct snd_soc_codec *codec; + int ret = 0; + + if (client->dev.platform_data == NULL) { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } + pdata = client->dev.platform_data; + + dac33 = kzalloc(sizeof(struct tlv320dac33_priv), GFP_KERNEL); + if (dac33 == NULL) + return -ENOMEM; + + codec = &dac33->codec; + codec->private_data = dac33; + codec->control_data = client; + + mutex_init(&codec->mutex); + mutex_init(&dac33->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "tlv320dac33"; + codec->owner = THIS_MODULE; + codec->read = dac33_read_reg_cache; + codec->write = dac33_write_locked; + codec->hw_write = (hw_write_t) i2c_master_send; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = dac33_set_bias_level; + codec->dai = &dac33_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(dac33_reg); + codec->reg_cache = kmemdup(dac33_reg, ARRAY_SIZE(dac33_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_reg; + } + + i2c_set_clientdata(client, dac33); + + dac33->power_gpio = pdata->power_gpio; + dac33->irq = client->irq; + dac33->nsample = NSAMPLE_MAX; + /* Disable FIFO use by default */ + dac33->nsample_switch = 0; + + tlv320dac33_codec = codec; + + codec->dev = &client->dev; + dac33_dai.dev = codec->dev; + + /* Check if the reset GPIO number is valid and request it */ + if (dac33->power_gpio >= 0) { + ret = gpio_request(dac33->power_gpio, "tlv320dac33 reset"); + if (ret < 0) { + dev_err(codec->dev, + "Failed to request reset GPIO (%d)\n", + dac33->power_gpio); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(codec); + goto error_gpio; + } + gpio_direction_output(dac33->power_gpio, 0); + } else { + dac33->chip_power = 1; + } + + /* Check if the IRQ number is valid and request it */ + if (dac33->irq >= 0) { + ret = request_irq(dac33->irq, dac33_interrupt_handler, + IRQF_TRIGGER_RISING | IRQF_DISABLED, + codec->name, codec); + if (ret < 0) { + dev_err(codec->dev, "Could not request IRQ%d (%d)\n", + dac33->irq, ret); + dac33->irq = -1; + } + if (dac33->irq != -1) { + /* Setup work queue */ + dac33->dac33_wq = + create_singlethread_workqueue("tlv320dac33"); + if (dac33->dac33_wq == NULL) { + free_irq(dac33->irq, &dac33->codec); + ret = -ENOMEM; + goto error_wq; + } + + INIT_WORK(&dac33->work, dac33_work); + } + } + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dai(&dac33_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } + + /* Shut down the codec for now */ + dac33_hard_power(codec, 0); + + return ret; + +error_codec: + if (dac33->irq >= 0) { + free_irq(dac33->irq, &dac33->codec); + destroy_workqueue(dac33->dac33_wq); + } +error_wq: + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); +error_gpio: + kfree(codec->reg_cache); +error_reg: + tlv320dac33_codec = NULL; + kfree(dac33); + + return ret; +} + +static int dac33_i2c_remove(struct i2c_client *client) +{ + struct tlv320dac33_priv *dac33; + + dac33 = i2c_get_clientdata(client); + dac33_hard_power(&dac33->codec, 0); + + if (dac33->power_gpio >= 0) + gpio_free(dac33->power_gpio); + if (dac33->irq >= 0) + free_irq(dac33->irq, &dac33->codec); + + destroy_workqueue(dac33->dac33_wq); + snd_soc_unregister_dai(&dac33_dai); + snd_soc_unregister_codec(&dac33->codec); + kfree(dac33->codec.reg_cache); + kfree(dac33); + tlv320dac33_codec = NULL; + + return 0; +} + +static const struct i2c_device_id tlv320dac33_i2c_id[] = { + { + .name = "tlv320dac33", + .driver_data = 0, + }, + { }, +}; + +static struct i2c_driver tlv320dac33_i2c_driver = { + .driver = { + .name = "tlv320dac33", + .owner = THIS_MODULE, + }, + .probe = dac33_i2c_probe, + .remove = __devexit_p(dac33_i2c_remove), + .id_table = tlv320dac33_i2c_id, +}; + +static int __init dac33_module_init(void) +{ + int r; + r = i2c_add_driver(&tlv320dac33_i2c_driver); + if (r < 0) { + printk(KERN_ERR "DAC33: driver registration failed\n"); + return r; + } + return 0; +} +module_init(dac33_module_init); + +static void __exit dac33_module_exit(void) +{ + i2c_del_driver(&tlv320dac33_i2c_driver); +} +module_exit(dac33_module_exit); + + +MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver"); +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@nokia.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320dac33.h b/sound/soc/codecs/tlv320dac33.h new file mode 100644 index 00000000000..eb8ae07f0bd --- /dev/null +++ b/sound/soc/codecs/tlv320dac33.h @@ -0,0 +1,267 @@ +/* + * ALSA SoC Texas Instruments TLV320DAC33 codec driver + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * Copyright: (C) 2009 Nokia Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TLV320DAC33_H +#define __TLV320DAC33_H + +#define DAC33_PAGE_SELECT 0x00 +#define DAC33_PWR_CTRL 0x01 +#define DAC33_PLL_CTRL_A 0x02 +#define DAC33_PLL_CTRL_B 0x03 +#define DAC33_PLL_CTRL_C 0x04 +#define DAC33_PLL_CTRL_D 0x05 +#define DAC33_PLL_CTRL_E 0x06 +#define DAC33_INT_OSC_CTRL 0x07 +#define DAC33_INT_OSC_FREQ_RAT_A 0x08 +#define DAC33_INT_OSC_FREQ_RAT_B 0x09 +#define DAC33_INT_OSC_DAC_RATIO_SET 0x0A +#define DAC33_CALIB_TIME 0x0B +#define DAC33_INT_OSC_CTRL_B 0x0C +#define DAC33_INT_OSC_CTRL_C 0x0D +#define DAC33_INT_OSC_STATUS 0x0E +#define DAC33_INT_OSC_DAC_RATIO_READ 0x0F +#define DAC33_INT_OSC_FREQ_RAT_READ_A 0x10 +#define DAC33_INT_OSC_FREQ_RAT_READ_B 0x11 +#define DAC33_SER_AUDIOIF_CTRL_A 0x12 +#define DAC33_SER_AUDIOIF_CTRL_B 0x13 +#define DAC33_SER_AUDIOIF_CTRL_C 0x14 +#define DAC33_FIFO_CTRL_A 0x15 +#define DAC33_UTHR_MSB 0x16 +#define DAC33_UTHR_LSB 0x17 +#define DAC33_ATHR_MSB 0x18 +#define DAC33_ATHR_LSB 0x19 +#define DAC33_LTHR_MSB 0x1A +#define DAC33_LTHR_LSB 0x1B +#define DAC33_PREFILL_MSB 0x1C +#define DAC33_PREFILL_LSB 0x1D +#define DAC33_NSAMPLE_MSB 0x1E +#define DAC33_NSAMPLE_LSB 0x1F +#define DAC33_FIFO_WPTR_MSB 0x20 +#define DAC33_FIFO_WPTR_LSB 0x21 +#define DAC33_FIFO_RPTR_MSB 0x22 +#define DAC33_FIFO_RPTR_LSB 0x23 +#define DAC33_FIFO_DEPTH_MSB 0x24 +#define DAC33_FIFO_DEPTH_LSB 0x25 +#define DAC33_SAMPLES_REMAINING_MSB 0x26 +#define DAC33_SAMPLES_REMAINING_LSB 0x27 +#define DAC33_FIFO_IRQ_FLAG 0x28 +#define DAC33_FIFO_IRQ_MASK 0x29 +#define DAC33_FIFO_IRQ_MODE_A 0x2A +#define DAC33_FIFO_IRQ_MODE_B 0x2B +#define DAC33_DAC_CTRL_A 0x2C +#define DAC33_DAC_CTRL_B 0x2D +#define DAC33_DAC_CTRL_C 0x2E +#define DAC33_LDAC_DIG_VOL_CTRL 0x2F +#define DAC33_RDAC_DIG_VOL_CTRL 0x30 +#define DAC33_DAC_STATUS_FLAGS 0x31 +#define DAC33_ASRC_CTRL_A 0x32 +#define DAC33_ASRC_CTRL_B 0x33 +#define DAC33_SRC_REF_CLK_RATIO_A 0x34 +#define DAC33_SRC_REF_CLK_RATIO_B 0x35 +#define DAC33_SRC_EST_REF_CLK_RATIO_A 0x36 +#define DAC33_SRC_EST_REF_CLK_RATIO_B 0x37 +#define DAC33_INTP_CTRL_A 0x38 +#define DAC33_INTP_CTRL_B 0x39 +/* Registers 0x3A - 0x3F Reserved */ +#define DAC33_LDAC_PWR_CTRL 0x40 +#define DAC33_RDAC_PWR_CTRL 0x41 +#define DAC33_OUT_AMP_CM_CTRL 0x42 +#define DAC33_OUT_AMP_PWR_CTRL 0x43 +#define DAC33_OUT_AMP_CTRL 0x44 +#define DAC33_LINEL_TO_LLO_VOL 0x45 +/* Registers 0x45 - 0x47 Reserved */ +#define DAC33_LINER_TO_RLO_VOL 0x48 +#define DAC33_ANA_VOL_SOFT_STEP_CTRL 0x49 +#define DAC33_OSC_TRIM 0x4A +/* Registers 0x4B - 0x7C Reserved */ +#define DAC33_DEVICE_ID_MSB 0x7D +#define DAC33_DEVICE_ID_LSB 0x7E +#define DAC33_DEVICE_REV_ID 0x7F + +#define DAC33_CACHEREGNUM 128 + +/* Bit definitions */ + +/* DAC33_PWR_CTRL (0x01) */ +#define DAC33_DACRPDNB (0x01 << 0) +#define DAC33_DACLPDNB (0x01 << 1) +#define DAC33_OSCPDNB (0x01 << 2) +#define DAC33_PLLPDNB (0x01 << 3) +#define DAC33_PDNALLB (0x01 << 4) +#define DAC33_SOFT_RESET (0x01 << 7) + +/* DAC33_INT_OSC_CTRL (0x07) */ +#define DAC33_REFSEL (0x01 << 1) + +/* DAC33_INT_OSC_CTRL_B (0x0C) */ +#define DAC33_ADJSTEP(x) (x << 0) +#define DAC33_ADJTHRSHLD(x) (x << 4) + +/* DAC33_INT_OSC_CTRL_C (0x0D) */ +#define DAC33_REFDIV(x) (x << 4) + +/* DAC33_INT_OSC_STATUS (0x0E) */ +#define DAC33_OSCSTATUS_IDLE_CALIB (0x00) +#define DAC33_OSCSTATUS_NORMAL (0x01) +#define DAC33_OSCSTATUS_ADJUSTMENT (0x03) +#define DAC33_OSCSTATUS_NOT_USED (0x02) + +/* DAC33_SER_AUDIOIF_CTRL_A (0x12) */ +#define DAC33_MSWCLK (0x01 << 0) +#define DAC33_MSBCLK (0x01 << 1) +#define DAC33_AFMT_MASK (0x03 << 2) +#define DAC33_AFMT_I2S (0x00 << 2) +#define DAC33_AFMT_DSP (0x01 << 2) +#define DAC33_AFMT_RIGHT_J (0x02 << 2) +#define DAC33_AFMT_LEFT_J (0x03 << 2) +#define DAC33_WLEN_MASK (0x03 << 4) +#define DAC33_WLEN_16 (0x00 << 4) +#define DAC33_WLEN_20 (0x01 << 4) +#define DAC33_WLEN_24 (0x02 << 4) +#define DAC33_WLEN_32 (0x03 << 4) +#define DAC33_NCYCL_MASK (0x03 << 6) +#define DAC33_NCYCL_16 (0x00 << 6) +#define DAC33_NCYCL_20 (0x01 << 6) +#define DAC33_NCYCL_24 (0x02 << 6) +#define DAC33_NCYCL_32 (0x03 << 6) + +/* DAC33_SER_AUDIOIF_CTRL_B (0x13) */ +#define DAC33_DATA_DELAY_MASK (0x03 << 2) +#define DAC33_DATA_DELAY(x) (x << 2) +#define DAC33_BCLKON (0x01 << 5) + +/* DAC33_FIFO_CTRL_A (0x15) */ +#define DAC33_WIDTH (0x01 << 0) +#define DAC33_FBYPAS (0x01 << 1) +#define DAC33_FAUTO (0x01 << 2) +#define DAC33_FIFOFLUSH (0x01 << 3) + +/* + * UTHR, ATHR, LTHR, PREFILL, NSAMPLE (0x16 - 0x1F) + * 13-bit values +*/ +#define DAC33_THRREG(x) (((x) & 0x1FFF) << 3) + +/* DAC33_FIFO_IRQ_MASK (0x29) */ +#define DAC33_MNS (0x01 << 0) +#define DAC33_MPS (0x01 << 1) +#define DAC33_MAT (0x01 << 2) +#define DAC33_MLT (0x01 << 3) +#define DAC33_MUT (0x01 << 4) +#define DAC33_MUF (0x01 << 5) +#define DAC33_MOF (0x01 << 6) + +#define DAC33_FIFO_IRQ_MODE_MASK (0x03) +#define DAC33_FIFO_IRQ_MODE_RISING (0x00) +#define DAC33_FIFO_IRQ_MODE_FALLING (0x01) +#define DAC33_FIFO_IRQ_MODE_LEVEL (0x02) +#define DAC33_FIFO_IRQ_MODE_EDGE (0x03) + +/* DAC33_FIFO_IRQ_MODE_A (0x2A) */ +#define DAC33_UTM(x) (x << 0) +#define DAC33_UFM(x) (x << 2) +#define DAC33_OFM(x) (x << 4) + +/* DAC33_FIFO_IRQ_MODE_B (0x2B) */ +#define DAC33_NSM(x) (x << 0) +#define DAC33_PSM(x) (x << 2) +#define DAC33_ATM(x) (x << 4) +#define DAC33_LTM(x) (x << 6) + +/* DAC33_DAC_CTRL_A (0x2C) */ +#define DAC33_DACRATE(x) (x << 0) +#define DAC33_DACDUAL (0x01 << 4) +#define DAC33_DACLKSEL_MASK (0x03 << 5) +#define DAC33_DACLKSEL_INTSOC (0x00 << 5) +#define DAC33_DACLKSEL_PLL (0x01 << 5) +#define DAC33_DACLKSEL_MCLK (0x02 << 5) +#define DAC33_DACLKSEL_BCLK (0x03 << 5) + +/* DAC33_DAC_CTRL_B (0x2D) */ +#define DAC33_DACSRCR_MASK (0x03 << 0) +#define DAC33_DACSRCR_MUTE (0x00 << 0) +#define DAC33_DACSRCR_RIGHT (0x01 << 0) +#define DAC33_DACSRCR_LEFT (0x02 << 0) +#define DAC33_DACSRCR_MONOMIX (0x03 << 0) +#define DAC33_DACSRCL_MASK (0x03 << 2) +#define DAC33_DACSRCL_MUTE (0x00 << 2) +#define DAC33_DACSRCL_LEFT (0x01 << 2) +#define DAC33_DACSRCL_RIGHT (0x02 << 2) +#define DAC33_DACSRCL_MONOMIX (0x03 << 2) +#define DAC33_DVOLSTEP_MASK (0x03 << 4) +#define DAC33_DVOLSTEP_SS_PERFS (0x00 << 4) +#define DAC33_DVOLSTEP_SS_PER2FS (0x01 << 4) +#define DAC33_DVOLSTEP_SS_DISABLED (0x02 << 4) +#define DAC33_DVOLCTRL_MASK (0x03 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT1 (0x00 << 6) +#define DAC33_DVOLCTRL_LR_RIGHT_CONTROL (0x01 << 6) +#define DAC33_DVOLCTRL_LR_LEFT_CONTROL (0x02 << 6) +#define DAC33_DVOLCTRL_LR_INDEPENDENT2 (0x03 << 6) + +/* DAC33_DAC_CTRL_C (0x2E) */ +#define DAC33_DEEMENR (0x01 << 0) +#define DAC33_EFFENR (0x01 << 1) +#define DAC33_DEEMENL (0x01 << 2) +#define DAC33_EFFENL (0x01 << 3) +#define DAC33_EN3D (0x01 << 4) +#define DAC33_RESYNMUTE (0x01 << 5) +#define DAC33_RESYNEN (0x01 << 6) + +/* DAC33_ASRC_CTRL_A (0x32) */ +#define DAC33_SRCBYP (0x01 << 0) +#define DAC33_SRCLKSEL_MASK (0x03 << 1) +#define DAC33_SRCLKSEL_INTSOC (0x00 << 1) +#define DAC33_SRCLKSEL_PLL (0x01 << 1) +#define DAC33_SRCLKSEL_MCLK (0x02 << 1) +#define DAC33_SRCLKSEL_BCLK (0x03 << 1) +#define DAC33_SRCLKDIV(x) (x << 3) + +/* DAC33_ASRC_CTRL_B (0x33) */ +#define DAC33_SRCSETUP(x) (x << 0) +#define DAC33_SRCREFSEL (0x01 << 4) +#define DAC33_SRCREFDIV(x) (x << 5) + +/* DAC33_INTP_CTRL_A (0x38) */ +#define DAC33_INTPSEL (0x01 << 0) +#define DAC33_INTPM_MASK (0x03 << 1) +#define DAC33_INTPM_ALOW_OPENDRAIN (0x00 << 1) +#define DAC33_INTPM_ALOW (0x01 << 1) +#define DAC33_INTPM_AHIGH (0x02 << 1) + +/* DAC33_LDAC_PWR_CTRL (0x40) */ +/* DAC33_RDAC_PWR_CTRL (0x41) */ +#define DAC33_DACLRNUM (0x01 << 2) +#define DAC33_LROUT_GAIN(x) (x << 0) + +/* DAC33_ANA_VOL_SOFT_STEP_CTRL (0x49) */ +#define DAC33_VOLCLKSEL (0x01 << 0) +#define DAC33_VOLCLKEN (0x01 << 1) +#define DAC33_VOLBYPASS (0x01 << 2) + +#define TLV320DAC33_MCLK 0 +#define TLV320DAC33_SLEEPCLK 1 + +extern struct snd_soc_dai dac33_dai; +extern struct snd_soc_codec_device soc_codec_dev_tlv320dac33; + +#endif /* __TLV320DAC33_H */ diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c new file mode 100644 index 00000000000..6b650c1aa3d --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.c @@ -0,0 +1,463 @@ +/* + * ALSA SoC Texas Instruments TPA6130A2 headset stereo amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <sound/tpa6130a2-plat.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tpa6130a2.h" + +static struct i2c_client *tpa6130a2_client; + +/* This struct is used to save the context */ +struct tpa6130a2_data { + struct mutex mutex; + unsigned char regs[TPA6130A2_CACHEREGNUM]; + int power_gpio; + unsigned char power_state; +}; + +static int tpa6130a2_i2c_read(int reg) +{ + struct tpa6130a2_data *data; + int val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + /* If powered off, return the cached value */ + if (data->power_state) { + val = i2c_smbus_read_byte_data(tpa6130a2_client, reg); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Read failed\n"); + else + data->regs[reg] = val; + } else { + val = data->regs[reg]; + } + + return val; +} + +static int tpa6130a2_i2c_write(int reg, u8 value) +{ + struct tpa6130a2_data *data; + int val = 0; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (data->power_state) { + val = i2c_smbus_write_byte_data(tpa6130a2_client, reg, value); + if (val < 0) + dev_err(&tpa6130a2_client->dev, "Write failed\n"); + } + + /* Either powered on or off, we save the context */ + data->regs[reg] = value; + + return val; +} + +static u8 tpa6130a2_read(int reg) +{ + struct tpa6130a2_data *data; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + return data->regs[reg]; +} + +static void tpa6130a2_initialize(void) +{ + struct tpa6130a2_data *data; + int i; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + for (i = 1; i < TPA6130A2_REG_VERSION; i++) + tpa6130a2_i2c_write(i, data->regs[i]); +} + +static void tpa6130a2_power(int power) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + if (power) { + /* Power on */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 1); + data->power_state = 1; + tpa6130a2_initialize(); + } + /* Clear SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } else { + /* set SWS */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= TPA6130A2_SWS; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + /* Power off */ + if (data->power_gpio >= 0) { + gpio_set_value(data->power_gpio, 0); + data->power_state = 0; + } + } + mutex_unlock(&data->mutex); +} + +static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + mutex_lock(&data->mutex); + + ucontrol->value.integer.value[0] = + (tpa6130a2_read(reg) >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + mask - ucontrol->value.integer.value[0]; + + mutex_unlock(&data->mutex); + return 0; +} + +static int tpa6130a2_set_reg(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct tpa6130a2_data *data; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int mask = mc->max; + unsigned int invert = mc->invert; + unsigned int val = (ucontrol->value.integer.value[0] & mask); + unsigned int val_reg; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (invert) + val = mask - val; + + mutex_lock(&data->mutex); + + val_reg = tpa6130a2_read(reg); + if (((val_reg >> shift) & mask) == val) { + mutex_unlock(&data->mutex); + return 0; + } + + val_reg &= ~(mask << shift); + val_reg |= val << shift; + tpa6130a2_i2c_write(reg, val_reg); + + mutex_unlock(&data->mutex); + + return 1; +} + +/* + * TPA6130 volume. From -59.5 to 4 dB with increasing step size when going + * down in gain. + */ +static const unsigned int tpa6130_tlv[] = { + TLV_DB_RANGE_HEAD(10), + 0, 1, TLV_DB_SCALE_ITEM(-5950, 600, 0), + 2, 3, TLV_DB_SCALE_ITEM(-5000, 250, 0), + 4, 5, TLV_DB_SCALE_ITEM(-4550, 160, 0), + 6, 7, TLV_DB_SCALE_ITEM(-4140, 190, 0), + 8, 9, TLV_DB_SCALE_ITEM(-3650, 120, 0), + 10, 11, TLV_DB_SCALE_ITEM(-3330, 160, 0), + 12, 13, TLV_DB_SCALE_ITEM(-3040, 180, 0), + 14, 20, TLV_DB_SCALE_ITEM(-2710, 110, 0), + 21, 37, TLV_DB_SCALE_ITEM(-1960, 74, 0), + 38, 63, TLV_DB_SCALE_ITEM(-720, 45, 0), +}; + +static const struct snd_kcontrol_new tpa6130a2_controls[] = { + SOC_SINGLE_EXT_TLV("TPA6130A2 Headphone Playback Volume", + TPA6130A2_REG_VOL_MUTE, 0, 0x3f, 0, + tpa6130a2_get_reg, tpa6130a2_set_reg, + tpa6130_tlv), +}; + +/* + * Enable or disable channel (left or right) + * The bit number for mute and amplifier are the same per channel: + * bit 6: Right channel + * bit 7: Left channel + * in both registers. + */ +static void tpa6130a2_channel_enable(u8 channel, int enable) +{ + struct tpa6130a2_data *data; + u8 val; + + BUG_ON(tpa6130a2_client == NULL); + data = i2c_get_clientdata(tpa6130a2_client); + + if (enable) { + /* Enable channel */ + /* Enable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + + /* Unmute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + } else { + /* Disable channel */ + /* Mute channel */ + val = tpa6130a2_read(TPA6130A2_REG_VOL_MUTE); + val |= channel; + tpa6130a2_i2c_write(TPA6130A2_REG_VOL_MUTE, val); + + /* Disable amplifier */ + val = tpa6130a2_read(TPA6130A2_REG_CONTROL); + val &= ~channel; + tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val); + } +} + +static int tpa6130a2_left_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_L, 0); + break; + } + return 0; +} + +static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_channel_enable(TPA6130A2_HP_EN_R, 0); + break; + } + return 0; +} + +static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + tpa6130a2_power(1); + break; + case SND_SOC_DAPM_POST_PMD: + tpa6130a2_power(0); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { + SND_SOC_DAPM_PGA_E("TPA6130A2 Left", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_left_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("TPA6130A2 Right", SND_SOC_NOPM, + 0, 0, NULL, 0, tpa6130a2_right_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("TPA6130A2 Enable", SND_SOC_NOPM, + 0, 0, tpa6130a2_supply_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + /* Outputs */ + SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL), + SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Left"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Right"}, + + {"TPA6130A2 Headphone Left", NULL, "TPA6130A2 Enable"}, + {"TPA6130A2 Headphone Right", NULL, "TPA6130A2 Enable"}, +}; + +int tpa6130a2_add_controls(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, tpa6130a2_dapm_widgets, + ARRAY_SIZE(tpa6130a2_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + return snd_soc_add_controls(codec, tpa6130a2_controls, + ARRAY_SIZE(tpa6130a2_controls)); + +} +EXPORT_SYMBOL_GPL(tpa6130a2_add_controls); + +static int tpa6130a2_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tpa6130a2_data *data; + struct tpa6130a2_platform_data *pdata; + int ret; + + dev = &client->dev; + + if (client->dev.platform_data == NULL) { + dev_err(dev, "Platform data not set\n"); + dump_stack(); + return -ENODEV; + } + + data = kzalloc(sizeof(*data), GFP_KERNEL); + if (data == NULL) { + dev_err(dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + tpa6130a2_client = client; + + i2c_set_clientdata(tpa6130a2_client, data); + + pdata = client->dev.platform_data; + data->power_gpio = pdata->power_gpio; + + mutex_init(&data->mutex); + + /* Set default register values */ + data->regs[TPA6130A2_REG_CONTROL] = TPA6130A2_SWS; + data->regs[TPA6130A2_REG_VOL_MUTE] = TPA6130A2_MUTE_R | + TPA6130A2_MUTE_L; + + if (data->power_gpio >= 0) { + ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + if (ret < 0) { + dev_err(dev, "Failed to request power GPIO (%d)\n", + data->power_gpio); + goto fail; + } + gpio_direction_output(data->power_gpio, 0); + } else { + data->power_state = 1; + tpa6130a2_initialize(); + } + + tpa6130a2_power(1); + + /* Read version */ + ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) & + TPA6130A2_VERSION_MASK; + if ((ret != 1) && (ret != 2)) + dev_warn(dev, "UNTESTED version detected (%d)\n", ret); + + /* Disable the chip */ + tpa6130a2_power(0); + + return 0; +fail: + kfree(data); + i2c_set_clientdata(tpa6130a2_client, NULL); + tpa6130a2_client = NULL; + + return ret; +} + +static int tpa6130a2_remove(struct i2c_client *client) +{ + struct tpa6130a2_data *data = i2c_get_clientdata(client); + + tpa6130a2_power(0); + + if (data->power_gpio >= 0) + gpio_free(data->power_gpio); + kfree(data); + tpa6130a2_client = NULL; + + return 0; +} + +static const struct i2c_device_id tpa6130a2_id[] = { + { "tpa6130a2", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); + +static struct i2c_driver tpa6130a2_i2c_driver = { + .driver = { + .name = "tpa6130a2", + .owner = THIS_MODULE, + }, + .probe = tpa6130a2_probe, + .remove = __devexit_p(tpa6130a2_remove), + .id_table = tpa6130a2_id, +}; + +static int __init tpa6130a2_init(void) +{ + return i2c_add_driver(&tpa6130a2_i2c_driver); +} + +static void __exit tpa6130a2_exit(void) +{ + i2c_del_driver(&tpa6130a2_i2c_driver); +} + +MODULE_AUTHOR("Peter Ujfalusi"); +MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver"); +MODULE_LICENSE("GPL"); + +module_init(tpa6130a2_init); +module_exit(tpa6130a2_exit); diff --git a/sound/soc/codecs/tpa6130a2.h b/sound/soc/codecs/tpa6130a2.h new file mode 100644 index 00000000000..57e867fd86d --- /dev/null +++ b/sound/soc/codecs/tpa6130a2.h @@ -0,0 +1,61 @@ +/* + * ALSA SoC TPA6130A2 amplifier driver + * + * Copyright (C) Nokia Corporation + * + * Author: Peter Ujfalusi <peter.ujfalusi@nokia.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TPA6130A2_H__ +#define __TPA6130A2_H__ + +/* Register addresses */ +#define TPA6130A2_REG_CONTROL 0x01 +#define TPA6130A2_REG_VOL_MUTE 0x02 +#define TPA6130A2_REG_OUT_IMPEDANCE 0x03 +#define TPA6130A2_REG_VERSION 0x04 + +#define TPA6130A2_CACHEREGNUM (TPA6130A2_REG_VERSION + 1) + +/* Register bits */ +/* TPA6130A2_REG_CONTROL (0x01) */ +#define TPA6130A2_SWS (0x01 << 0) +#define TPA6130A2_TERMAL (0x01 << 1) +#define TPA6130A2_MODE(x) (x << 4) +#define TPA6130A2_MODE_STEREO (0x00) +#define TPA6130A2_MODE_DUAL_MONO (0x01) +#define TPA6130A2_MODE_BRIDGE (0x02) +#define TPA6130A2_MODE_MASK (0x03) +#define TPA6130A2_HP_EN_R (0x01 << 6) +#define TPA6130A2_HP_EN_L (0x01 << 7) + +/* TPA6130A2_REG_VOL_MUTE (0x02) */ +#define TPA6130A2_VOLUME(x) ((x & 0x3f) << 0) +#define TPA6130A2_MUTE_R (0x01 << 6) +#define TPA6130A2_MUTE_L (0x01 << 7) + +/* TPA6130A2_REG_OUT_IMPEDANCE (0x03) */ +#define TPA6130A2_HIZ_R (0x01 << 0) +#define TPA6130A2_HIZ_L (0x01 << 1) + +/* TPA6130A2_REG_VERSION (0x04) */ +#define TPA6130A2_VERSION_MASK (0x0f) + +extern int tpa6130a2_add_controls(struct snd_soc_codec *codec); + +#endif /* __TPA6130A2_H__ */ diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4df7c6c61c7..5f1681f6ca7 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -120,9 +120,10 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { - unsigned int bypass_state; + struct snd_soc_codec codec; + unsigned int codec_powered; - unsigned int codec_muted; + unsigned int apll_enabled; struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; @@ -183,19 +184,20 @@ static int twl4030_write(struct snd_soc_codec *codec, static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = codec->private_data; - u8 mode; + int mode; if (enable == twl4030->codec_powered) return; - mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); if (enable) - mode |= TWL4030_CODECPDZ; + mode = twl4030_codec_enable_resource(TWL4030_CODEC_RES_POWER); else - mode &= ~TWL4030_CODECPDZ; + mode = twl4030_codec_disable_resource(TWL4030_CODEC_RES_POWER); - twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); - twl4030->codec_powered = enable; + if (mode >= 0) { + twl4030_write_reg_cache(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030->codec_powered = enable; + } /* REVISIT: this delay is present in TI sample drivers */ /* but there seems to be no TRM requirement for it */ @@ -212,31 +214,30 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* set all audio section registers to reasonable defaults */ for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) - twl4030_write(codec, i, cache[i]); + if (i != TWL4030_REG_APLL_CTL) + twl4030_write(codec, i, cache[i]); } -static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute) +static void twl4030_apll_enable(struct snd_soc_codec *codec, int enable) { struct twl4030_priv *twl4030 = codec->private_data; - u8 reg_val; + int status; - if (mute == twl4030->codec_muted) + if (enable == twl4030->apll_enabled) return; - if (mute) { - /* Disable PLL */ - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - reg_val &= ~TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); - } else { + if (enable) /* Enable PLL */ - reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL); - reg_val |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val); - } + status = twl4030_codec_enable_resource(TWL4030_CODEC_RES_APLL); + else + /* Disable PLL */ + status = twl4030_codec_disable_resource(TWL4030_CODEC_RES_APLL); + + if (status >= 0) + twl4030_write_reg_cache(codec, TWL4030_REG_APLL_CTL, status); - twl4030->codec_muted = mute; + twl4030->apll_enabled = enable; } static void twl4030_power_up(struct snd_soc_codec *codec) @@ -613,6 +614,27 @@ static int handsfreerpga_event(struct snd_soc_dapm_widget *w, return 0; } +static int vibramux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + twl4030_write(w->codec, TWL4030_REG_VIBRA_SET, 0xff); + return 0; +} + +static int apll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + twl4030_apll_enable(w->codec, 1); + break; + case SND_SOC_DAPM_POST_PMD: + twl4030_apll_enable(w->codec, 0); + break; + } + return 0; +} + static void headset_ramp(struct snd_soc_codec *codec, int ramp) { struct snd_soc_device *socdev = codec->socdev; @@ -724,67 +746,6 @@ static int headsetrpga_event(struct snd_soc_dapm_widget *w, return 0; } -static int bypass_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct soc_mixer_control *m = - (struct soc_mixer_control *)w->kcontrols->private_value; - struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg, misc; - - reg = twl4030_read_reg_cache(w->codec, m->reg); - - /* - * bypass_state[0:3] - analog HiFi bypass - * bypass_state[4] - analog voice bypass - * bypass_state[5] - digital voice bypass - * bypass_state[6:7] - digital HiFi bypass - */ - if (m->reg == TWL4030_REG_VSTPGA) { - /* Voice digital bypass */ - if (reg) - twl4030->bypass_state |= (1 << 5); - else - twl4030->bypass_state &= ~(1 << 5); - } else if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) { - /* Analog bypass */ - if (reg & (1 << m->shift)) - twl4030->bypass_state |= - (1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); - else - twl4030->bypass_state &= - ~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL)); - } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) { - /* Analog voice bypass */ - if (reg & (1 << m->shift)) - twl4030->bypass_state |= (1 << 4); - else - twl4030->bypass_state &= ~(1 << 4); - } else { - /* Digital bypass */ - if (reg & (0x7 << m->shift)) - twl4030->bypass_state |= (1 << (m->shift ? 7 : 6)); - else - twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6)); - } - - /* Enable master analog loopback mode if any analog switch is enabled*/ - misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); - if (twl4030->bypass_state & 0x1F) - misc |= TWL4030_FMLOOP_EN; - else - misc &= ~TWL4030_FMLOOP_EN; - twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); - - if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { - if (twl4030->bypass_state) - twl4030_codec_mute(w->codec, 0); - else - twl4030_codec_mute(w->codec, 1); - } - return 0; -} - /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -1192,32 +1153,28 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), /* Analog bypasses */ - SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassr1_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassl1_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassr2_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassl2_control, - bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_abypassv_control, - bypass_event, SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr1_control), + SND_SOC_DAPM_SWITCH("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl1_control), + SND_SOC_DAPM_SWITCH("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassr2_control), + SND_SOC_DAPM_SWITCH("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassl2_control), + SND_SOC_DAPM_SWITCH("Voice Analog Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_abypassv_control), + + /* Master analog loopback switch */ + SND_SOC_DAPM_SUPPLY("FM Loop Enable", TWL4030_REG_MISC_SET_1, 5, 0, + NULL, 0), /* Digital bypasses */ - SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassl_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassr_control, bypass_event, - SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_dbypassv_control, bypass_event, - SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH("Left Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassl_control), + SND_SOC_DAPM_SWITCH("Right Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassr_control), + SND_SOC_DAPM_SWITCH("Voice Digital Loopback", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_dbypassv_control), /* Digital mixers, power control for the physical DACs */ SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer", @@ -1243,6 +1200,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer", TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event, + SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD), + /* Output MIXER controls */ /* Earpiece */ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, @@ -1308,8 +1268,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { 0, 0, NULL, 0, handsfreerpga_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Vibra */ - SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, - &twl4030_dapm_vibra_control), + SND_SOC_DAPM_MUX_E("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0, + &twl4030_dapm_vibra_control, vibramux_event, + SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0, &twl4030_dapm_vibrapath_control), @@ -1369,6 +1330,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Digital R2 Playback Mixer", NULL, "DAC Right2"}, {"Digital Voice Playback Mixer", NULL, "DAC Voice"}, + /* Supply for the digital part (APLL) */ + {"Digital R1 Playback Mixer", NULL, "APLL Enable"}, + {"Digital L1 Playback Mixer", NULL, "APLL Enable"}, + {"Digital R2 Playback Mixer", NULL, "APLL Enable"}, + {"Digital L2 Playback Mixer", NULL, "APLL Enable"}, + {"Digital Voice Playback Mixer", NULL, "APLL Enable"}, + {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"}, {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"}, {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"}, @@ -1482,6 +1450,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"ADC Virtual Left2", NULL, "TX2 Capture Route"}, {"ADC Virtual Right2", NULL, "TX2 Capture Route"}, + {"ADC Virtual Left1", NULL, "APLL Enable"}, + {"ADC Virtual Right1", NULL, "APLL Enable"}, + {"ADC Virtual Left2", NULL, "APLL Enable"}, + {"ADC Virtual Right2", NULL, "APLL Enable"}, + /* Analog bypass routes */ {"Right1 Analog Loopback", "Switch", "Analog Right"}, {"Left1 Analog Loopback", "Switch", "Analog Left"}, @@ -1489,6 +1462,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left2 Analog Loopback", "Switch", "Analog Left"}, {"Voice Analog Loopback", "Switch", "Analog Left"}, + /* Supply for the Analog loopbacks */ + {"Right1 Analog Loopback", NULL, "FM Loop Enable"}, + {"Left1 Analog Loopback", NULL, "FM Loop Enable"}, + {"Right2 Analog Loopback", NULL, "FM Loop Enable"}, + {"Left2 Analog Loopback", NULL, "FM Loop Enable"}, + {"Voice Analog Loopback", NULL, "FM Loop Enable"}, + {"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"}, {"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"}, {"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"}, @@ -1513,32 +1493,20 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } static int twl4030_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - struct twl4030_priv *twl4030 = codec->private_data; - switch (level) { case SND_SOC_BIAS_ON: - twl4030_codec_mute(codec, 0); break; case SND_SOC_BIAS_PREPARE: - twl4030_power_up(codec); - if (twl4030->bypass_state) - twl4030_codec_mute(codec, 0); - else - twl4030_codec_mute(codec, 1); break; case SND_SOC_BIAS_STANDBY: - twl4030_power_up(codec); - if (twl4030->bypass_state) - twl4030_codec_mute(codec, 0); - else - twl4030_codec_mute(codec, 1); + if (codec->bias_level == SND_SOC_BIAS_OFF) + twl4030_power_up(codec); break; case SND_SOC_BIAS_OFF: twl4030_power_down(codec); @@ -1785,29 +1753,23 @@ static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_codec *codec = codec_dai->codec; struct twl4030_priv *twl4030 = codec->private_data; - u8 infreq; switch (freq) { case 19200000: - infreq = TWL4030_APLL_INFREQ_19200KHZ; - twl4030->sysclk = 19200; - break; case 26000000: - infreq = TWL4030_APLL_INFREQ_26000KHZ; - twl4030->sysclk = 26000; - break; case 38400000: - infreq = TWL4030_APLL_INFREQ_38400KHZ; - twl4030->sysclk = 38400; break; default: - printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", - freq); + dev_err(codec->dev, "Unsupported APLL mclk: %u\n", freq); return -EINVAL; } - infreq |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + if ((freq / 1000) != twl4030->sysclk) { + dev_err(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + freq, twl4030->sysclk * 1000); + return -EINVAL; + } return 0; } @@ -1905,18 +1867,16 @@ static int twl4030_voice_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->card->codec; - u8 infreq; + struct twl4030_priv *twl4030 = codec->private_data; u8 mode; /* If the system master clock is not 26MHz, the voice PCM interface is * not avilable. */ - infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) - & TWL4030_APLL_INFREQ; - - if (infreq != TWL4030_APLL_INFREQ_26000KHZ) { - printk(KERN_ERR "TWL4030 voice startup: " - "MCLK is not 26MHz, call set_sysclk() on init\n"); + if (twl4030->sysclk != 26000) { + dev_err(codec->dev, "The board is configured for %u Hz, while" + "the Voice interface needs 26MHz APLL mclk\n", + twl4030->sysclk * 1000); return -EINVAL; } @@ -1989,21 +1949,19 @@ static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - u8 infreq; + struct twl4030_priv *twl4030 = codec->private_data; - switch (freq) { - case 26000000: - infreq = TWL4030_APLL_INFREQ_26000KHZ; - break; - default: - printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", - freq); + if (freq != 26000000) { + dev_err(codec->dev, "Unsupported APLL mclk: %u, the Voice" + "interface needs 26MHz APLL mclk\n", freq); + return -EINVAL; + } + if ((freq / 1000) != twl4030->sysclk) { + dev_err(codec->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + freq, twl4030->sysclk * 1000); return -EINVAL; } - - infreq |= TWL4030_APLL_EN; - twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); - return 0; } @@ -2121,7 +2079,7 @@ struct snd_soc_dai twl4030_dai[] = { }; EXPORT_SYMBOL_GPL(twl4030_dai); -static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) +static int twl4030_soc_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -2131,7 +2089,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) return 0; } -static int twl4030_resume(struct platform_device *pdev) +static int twl4030_soc_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; @@ -2141,147 +2099,181 @@ static int twl4030_resume(struct platform_device *pdev) return 0; } -/* - * initialize the driver - * register the mixer and dsp interfaces with the kernel - */ +static struct snd_soc_codec *twl4030_codec; -static int twl4030_init(struct snd_soc_device *socdev) +static int twl4030_soc_probe(struct platform_device *pdev) { - struct snd_soc_codec *codec = socdev->card->codec; + struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct twl4030_setup_data *setup = socdev->codec_data; - struct twl4030_priv *twl4030 = codec->private_data; - int ret = 0; + struct snd_soc_codec *codec; + struct twl4030_priv *twl4030; + int ret; - printk(KERN_INFO "TWL4030 Audio Codec init \n"); + BUG_ON(!twl4030_codec); - codec->name = "twl4030"; - codec->owner = THIS_MODULE; - codec->read = twl4030_read_reg_cache; - codec->write = twl4030_write; - codec->set_bias_level = twl4030_set_bias_level; - codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), - codec->reg_cache_size = sizeof(twl4030_reg); - codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), - GFP_KERNEL); - if (codec->reg_cache == NULL) - return -ENOMEM; + codec = twl4030_codec; + twl4030 = codec->private_data; + socdev->card->codec = codec; /* Configuration for headset ramp delay from setup data */ if (setup) { unsigned char hs_pop; - if (setup->sysclk) - twl4030->sysclk = setup->sysclk; - else - twl4030->sysclk = 26000; + if (setup->sysclk != twl4030->sysclk) + dev_warn(&pdev->dev, + "Mismatch in APLL mclk: %u (configured: %u)\n", + setup->sysclk, twl4030->sysclk); hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); hs_pop &= ~TWL4030_RAMP_DELAY; hs_pop |= (setup->ramp_delay_value << 2); twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop); - } else { - twl4030->sysclk = 26000; } /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { - printk(KERN_ERR "twl4030: failed to create pcms\n"); - goto pcm_err; + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; } - twl4030_init_chip(codec); - - /* power on device */ - twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_controls(codec, twl4030_snd_controls, ARRAY_SIZE(twl4030_snd_controls)); twl4030_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "twl4030: failed to register card\n"); - goto card_err; - } + return 0; +} - return ret; +static int twl4030_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; -card_err: + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); - return ret; -} + kfree(codec->private_data); + kfree(codec); -static struct snd_soc_device *twl4030_socdev; + return 0; +} -static int twl4030_probe(struct platform_device *pdev) +static int __devinit twl4030_codec_probe(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct twl4030_codec_audio_data *pdata = pdev->dev.platform_data; struct snd_soc_codec *codec; struct twl4030_priv *twl4030; + int ret; - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; + if (!pdata) { + dev_err(&pdev->dev, "platform_data is missing\n"); + return -EINVAL; + } twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL); if (twl4030 == NULL) { - kfree(codec); + dev_err(&pdev->dev, "Can not allocate memroy\n"); return -ENOMEM; } + codec = &twl4030->codec; codec->private_data = twl4030; - socdev->card->codec = codec; + codec->dev = &pdev->dev; + twl4030_dai[0].dev = &pdev->dev; + twl4030_dai[1].dev = &pdev->dev; + mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); - twl4030_socdev = socdev; - twl4030_init(socdev); + codec->name = "twl4030"; + codec->owner = THIS_MODULE; + codec->read = twl4030_read_reg_cache; + codec->write = twl4030_write; + codec->set_bias_level = twl4030_set_bias_level; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->reg_cache_size = sizeof(twl4030_reg); + codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto error_cache; + } + + platform_set_drvdata(pdev, twl4030); + twl4030_codec = codec; + + /* Set the defaults, and power up the codec */ + twl4030->sysclk = twl4030_codec_get_mclk() / 1000; + twl4030_init_chip(codec); + codec->bias_level = SND_SOC_BIAS_OFF; + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto error_codec; + } + + ret = snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + snd_soc_unregister_codec(codec); + goto error_codec; + } return 0; + +error_codec: + twl4030_power_down(codec); + kfree(codec->reg_cache); +error_cache: + kfree(twl4030); + return ret; } -static int twl4030_remove(struct platform_device *pdev) +static int __devexit twl4030_codec_remove(struct platform_device *pdev) { - struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->card->codec; + struct twl4030_priv *twl4030 = platform_get_drvdata(pdev); - printk(KERN_INFO "TWL4030 Audio Codec remove\n"); - twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - kfree(codec->private_data); - kfree(codec); + kfree(twl4030); + twl4030_codec = NULL; return 0; } -struct snd_soc_codec_device soc_codec_dev_twl4030 = { - .probe = twl4030_probe, - .remove = twl4030_remove, - .suspend = twl4030_suspend, - .resume = twl4030_resume, +MODULE_ALIAS("platform:twl4030_codec_audio"); + +static struct platform_driver twl4030_codec_driver = { + .probe = twl4030_codec_probe, + .remove = __devexit_p(twl4030_codec_remove), + .driver = { + .name = "twl4030_codec_audio", + .owner = THIS_MODULE, + }, }; -EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + return platform_driver_register(&twl4030_codec_driver); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); + platform_driver_unregister(&twl4030_codec_driver); } module_exit(twl4030_exit); +struct snd_soc_codec_device soc_codec_dev_twl4030 = { + .probe = twl4030_soc_probe, + .remove = twl4030_soc_remove, + .suspend = twl4030_soc_suspend, + .resume = twl4030_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); + MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 2b4bfa23f98..dd6396ec9c7 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -22,245 +22,13 @@ #ifndef __TWL4030_AUDIO_H__ #define __TWL4030_AUDIO_H__ -#define TWL4030_REG_CODEC_MODE 0x1 -#define TWL4030_REG_OPTION 0x2 -#define TWL4030_REG_UNKNOWN 0x3 -#define TWL4030_REG_MICBIAS_CTL 0x4 -#define TWL4030_REG_ANAMICL 0x5 -#define TWL4030_REG_ANAMICR 0x6 -#define TWL4030_REG_AVADC_CTL 0x7 -#define TWL4030_REG_ADCMICSEL 0x8 -#define TWL4030_REG_DIGMIXING 0x9 -#define TWL4030_REG_ATXL1PGA 0xA -#define TWL4030_REG_ATXR1PGA 0xB -#define TWL4030_REG_AVTXL2PGA 0xC -#define TWL4030_REG_AVTXR2PGA 0xD -#define TWL4030_REG_AUDIO_IF 0xE -#define TWL4030_REG_VOICE_IF 0xF -#define TWL4030_REG_ARXR1PGA 0x10 -#define TWL4030_REG_ARXL1PGA 0x11 -#define TWL4030_REG_ARXR2PGA 0x12 -#define TWL4030_REG_ARXL2PGA 0x13 -#define TWL4030_REG_VRXPGA 0x14 -#define TWL4030_REG_VSTPGA 0x15 -#define TWL4030_REG_VRX2ARXPGA 0x16 -#define TWL4030_REG_AVDAC_CTL 0x17 -#define TWL4030_REG_ARX2VTXPGA 0x18 -#define TWL4030_REG_ARXL1_APGA_CTL 0x19 -#define TWL4030_REG_ARXR1_APGA_CTL 0x1A -#define TWL4030_REG_ARXL2_APGA_CTL 0x1B -#define TWL4030_REG_ARXR2_APGA_CTL 0x1C -#define TWL4030_REG_ATX2ARXPGA 0x1D -#define TWL4030_REG_BT_IF 0x1E -#define TWL4030_REG_BTPGA 0x1F -#define TWL4030_REG_BTSTPGA 0x20 -#define TWL4030_REG_EAR_CTL 0x21 -#define TWL4030_REG_HS_SEL 0x22 -#define TWL4030_REG_HS_GAIN_SET 0x23 -#define TWL4030_REG_HS_POPN_SET 0x24 -#define TWL4030_REG_PREDL_CTL 0x25 -#define TWL4030_REG_PREDR_CTL 0x26 -#define TWL4030_REG_PRECKL_CTL 0x27 -#define TWL4030_REG_PRECKR_CTL 0x28 -#define TWL4030_REG_HFL_CTL 0x29 -#define TWL4030_REG_HFR_CTL 0x2A -#define TWL4030_REG_ALC_CTL 0x2B -#define TWL4030_REG_ALC_SET1 0x2C -#define TWL4030_REG_ALC_SET2 0x2D -#define TWL4030_REG_BOOST_CTL 0x2E -#define TWL4030_REG_SOFTVOL_CTL 0x2F -#define TWL4030_REG_DTMF_FREQSEL 0x30 -#define TWL4030_REG_DTMF_TONEXT1H 0x31 -#define TWL4030_REG_DTMF_TONEXT1L 0x32 -#define TWL4030_REG_DTMF_TONEXT2H 0x33 -#define TWL4030_REG_DTMF_TONEXT2L 0x34 -#define TWL4030_REG_DTMF_TONOFF 0x35 -#define TWL4030_REG_DTMF_WANONOFF 0x36 -#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 -#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 -#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 -#define TWL4030_REG_APLL_CTL 0x3A -#define TWL4030_REG_DTMF_CTL 0x3B -#define TWL4030_REG_DTMF_PGA_CTL2 0x3C -#define TWL4030_REG_DTMF_PGA_CTL1 0x3D -#define TWL4030_REG_MISC_SET_1 0x3E -#define TWL4030_REG_PCMBTMUX 0x3F -#define TWL4030_REG_RX_PATH_SEL 0x43 -#define TWL4030_REG_VDL_APGA_CTL 0x44 -#define TWL4030_REG_VIBRA_CTL 0x45 -#define TWL4030_REG_VIBRA_SET 0x46 -#define TWL4030_REG_VIBRA_PWM_SET 0x47 -#define TWL4030_REG_ANAMIC_GAIN 0x48 -#define TWL4030_REG_MISC_SET_2 0x49 -#define TWL4030_REG_SW_SHADOW 0x4A +/* Register descriptions are here */ +#include <linux/mfd/twl4030-codec.h> +/* Sgadow register used by the audio driver */ +#define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -/* Bitfield Definitions */ - -/* TWL4030_CODEC_MODE (0x01) Fields */ - -#define TWL4030_APLL_RATE 0xF0 -#define TWL4030_APLL_RATE_8000 0x00 -#define TWL4030_APLL_RATE_11025 0x10 -#define TWL4030_APLL_RATE_12000 0x20 -#define TWL4030_APLL_RATE_16000 0x40 -#define TWL4030_APLL_RATE_22050 0x50 -#define TWL4030_APLL_RATE_24000 0x60 -#define TWL4030_APLL_RATE_32000 0x80 -#define TWL4030_APLL_RATE_44100 0x90 -#define TWL4030_APLL_RATE_48000 0xA0 -#define TWL4030_APLL_RATE_96000 0xE0 -#define TWL4030_SEL_16K 0x08 -#define TWL4030_CODECPDZ 0x02 -#define TWL4030_OPT_MODE 0x01 -#define TWL4030_OPTION_1 (1 << 0) -#define TWL4030_OPTION_2 (0 << 0) - -/* TWL4030_OPTION (0x02) Fields */ - -#define TWL4030_ATXL1_EN (1 << 0) -#define TWL4030_ATXR1_EN (1 << 1) -#define TWL4030_ATXL2_VTXL_EN (1 << 2) -#define TWL4030_ATXR2_VTXR_EN (1 << 3) -#define TWL4030_ARXL1_VRX_EN (1 << 4) -#define TWL4030_ARXR1_EN (1 << 5) -#define TWL4030_ARXL2_EN (1 << 6) -#define TWL4030_ARXR2_EN (1 << 7) - -/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ - -#define TWL4030_MICBIAS2_CTL 0x40 -#define TWL4030_MICBIAS1_CTL 0x20 -#define TWL4030_HSMICBIAS_EN 0x04 -#define TWL4030_MICBIAS2_EN 0x02 -#define TWL4030_MICBIAS1_EN 0x01 - -/* ANAMICL (0x05) Fields */ - -#define TWL4030_CNCL_OFFSET_START 0x80 -#define TWL4030_OFFSET_CNCL_SEL 0x60 -#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 -#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 -#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 -#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 -#define TWL4030_MICAMPL_EN 0x10 -#define TWL4030_CKMIC_EN 0x08 -#define TWL4030_AUXL_EN 0x04 -#define TWL4030_HSMIC_EN 0x02 -#define TWL4030_MAINMIC_EN 0x01 - -/* ANAMICR (0x06) Fields */ - -#define TWL4030_MICAMPR_EN 0x10 -#define TWL4030_AUXR_EN 0x04 -#define TWL4030_SUBMIC_EN 0x01 - -/* AVADC_CTL (0x07) Fields */ - -#define TWL4030_ADCL_EN 0x08 -#define TWL4030_AVADC_CLK_PRIORITY 0x04 -#define TWL4030_ADCR_EN 0x02 - -/* TWL4030_REG_ADCMICSEL (0x08) Fields */ - -#define TWL4030_DIGMIC1_EN 0x08 -#define TWL4030_TX2IN_SEL 0x04 -#define TWL4030_DIGMIC0_EN 0x02 -#define TWL4030_TX1IN_SEL 0x01 - -/* AUDIO_IF (0x0E) Fields */ - -#define TWL4030_AIF_SLAVE_EN 0x80 -#define TWL4030_DATA_WIDTH 0x60 -#define TWL4030_DATA_WIDTH_16S_16W 0x00 -#define TWL4030_DATA_WIDTH_32S_16W 0x40 -#define TWL4030_DATA_WIDTH_32S_24W 0x60 -#define TWL4030_AIF_FORMAT 0x18 -#define TWL4030_AIF_FORMAT_CODEC 0x00 -#define TWL4030_AIF_FORMAT_LEFT 0x08 -#define TWL4030_AIF_FORMAT_RIGHT 0x10 -#define TWL4030_AIF_FORMAT_TDM 0x18 -#define TWL4030_AIF_TRI_EN 0x04 -#define TWL4030_CLK256FS_EN 0x02 -#define TWL4030_AIF_EN 0x01 - -/* VOICE_IF (0x0F) Fields */ - -#define TWL4030_VIF_SLAVE_EN 0x80 -#define TWL4030_VIF_DIN_EN 0x40 -#define TWL4030_VIF_DOUT_EN 0x20 -#define TWL4030_VIF_SWAP 0x10 -#define TWL4030_VIF_FORMAT 0x08 -#define TWL4030_VIF_TRI_EN 0x04 -#define TWL4030_VIF_SUB_EN 0x02 -#define TWL4030_VIF_EN 0x01 - -/* EAR_CTL (0x21) */ -#define TWL4030_EAR_GAIN 0x30 - -/* HS_GAIN_SET (0x23) Fields */ - -#define TWL4030_HSR_GAIN 0x0C -#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 -#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 -#define TWL4030_HSR_GAIN_0DB 0x08 -#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C -#define TWL4030_HSL_GAIN 0x03 -#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 -#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 -#define TWL4030_HSL_GAIN_0DB 0x02 -#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 - -/* HS_POPN_SET (0x24) Fields */ - -#define TWL4030_VMID_EN 0x40 -#define TWL4030_EXTMUTE 0x20 -#define TWL4030_RAMP_DELAY 0x1C -#define TWL4030_RAMP_DELAY_20MS 0x00 -#define TWL4030_RAMP_DELAY_40MS 0x04 -#define TWL4030_RAMP_DELAY_81MS 0x08 -#define TWL4030_RAMP_DELAY_161MS 0x0C -#define TWL4030_RAMP_DELAY_323MS 0x10 -#define TWL4030_RAMP_DELAY_645MS 0x14 -#define TWL4030_RAMP_DELAY_1291MS 0x18 -#define TWL4030_RAMP_DELAY_2581MS 0x1C -#define TWL4030_RAMP_EN 0x02 - -/* PREDL_CTL (0x25) */ -#define TWL4030_PREDL_GAIN 0x30 - -/* PREDR_CTL (0x26) */ -#define TWL4030_PREDR_GAIN 0x30 - -/* PRECKL_CTL (0x27) */ -#define TWL4030_PRECKL_GAIN 0x30 - -/* PRECKR_CTL (0x28) */ -#define TWL4030_PRECKR_GAIN 0x30 - -/* HFL_CTL (0x29, 0x2A) Fields */ -#define TWL4030_HF_CTL_HB_EN 0x04 -#define TWL4030_HF_CTL_LOOP_EN 0x08 -#define TWL4030_HF_CTL_RAMP_EN 0x10 -#define TWL4030_HF_CTL_REF_EN 0x20 - -/* APLL_CTL (0x3A) Fields */ - -#define TWL4030_APLL_EN 0x10 -#define TWL4030_APLL_INFREQ 0x0F -#define TWL4030_APLL_INFREQ_19200KHZ 0x05 -#define TWL4030_APLL_INFREQ_26000KHZ 0x06 -#define TWL4030_APLL_INFREQ_38400KHZ 0x0F - -/* REG_MISC_SET_1 (0x3E) Fields */ - -#define TWL4030_CLK64_EN 0x80 -#define TWL4030_SCRAMBLE_EN 0x40 -#define TWL4030_FMLOOP_EN 0x20 -#define TWL4030_SMOOTH_ANAVOL_EN 0x02 -#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 - /* TWL4030_REG_SW_SHADOW (0x4A) Fields */ #define TWL4030_HFL_EN 0x01 #define TWL4030_HFR_EN 0x02 @@ -279,3 +47,5 @@ struct twl4030_setup_data { }; #endif /* End of __TWL4030_AUDIO_H__ */ + + diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c33b92edbde..aa40d985138 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -562,17 +562,8 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "UDA134X: failed to register card\n"); - goto card_err; - } - return 0; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); reg_err: diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 92ec0344215..a2763c2e734 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -378,7 +378,6 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -713,17 +712,9 @@ static int uda1380_probe(struct platform_device *pdev) snd_soc_add_controls(codec, uda1380_snd_controls, ARRAY_SIZE(uda1380_snd_controls)); uda1380_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 593d5b9c9f0..f82125d9e85 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -800,7 +800,7 @@ static int wm8350_add_widgets(struct snd_soc_codec *codec) return ret; } - return snd_soc_dapm_new_widgets(codec); + return 0; } static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -1101,7 +1101,7 @@ static inline int fll_factors(struct _fll_div *fll_div, unsigned int input, } static int wm8350_set_fll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, + int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; @@ -1501,18 +1501,7 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - return 0; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - return ret; } static int wm8350_remove(struct platform_device *pdev) @@ -1680,21 +1669,6 @@ static int __devexit wm8350_codec_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM -static int wm8350_codec_suspend(struct platform_device *pdev, pm_message_t m) -{ - return snd_soc_suspend_device(&pdev->dev); -} - -static int wm8350_codec_resume(struct platform_device *pdev) -{ - return snd_soc_resume_device(&pdev->dev); -} -#else -#define wm8350_codec_suspend NULL -#define wm8350_codec_resume NULL -#endif - static struct platform_driver wm8350_codec_driver = { .driver = { .name = "wm8350-codec", @@ -1702,8 +1676,6 @@ static struct platform_driver wm8350_codec_driver = { }, .probe = wm8350_codec_probe, .remove = __devexit_p(wm8350_codec_remove), - .suspend = wm8350_codec_suspend, - .resume = wm8350_codec_resume, }; static __init int wm8350_init(void) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index b9ef4d91522..b432f4d4a32 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -915,7 +915,6 @@ static int wm8400_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -1011,7 +1010,8 @@ static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors, } static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, - unsigned int freq_in, unsigned int freq_out) + int source, unsigned int freq_in, + unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; struct wm8400_priv *wm8400 = codec->private_data; @@ -1399,17 +1399,6 @@ static int wm8400_probe(struct platform_device *pdev) wm8400_add_controls(codec); wm8400_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "failed to register card\n"); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -1558,21 +1547,6 @@ static int __exit wm8400_codec_remove(struct platform_device *dev) return 0; } -#ifdef CONFIG_PM -static int wm8400_pdev_suspend(struct platform_device *pdev, pm_message_t msg) -{ - return snd_soc_suspend_device(&pdev->dev); -} - -static int wm8400_pdev_resume(struct platform_device *pdev) -{ - return snd_soc_resume_device(&pdev->dev); -} -#else -#define wm8400_pdev_suspend NULL -#define wm8400_pdev_resume NULL -#endif - static struct platform_driver wm8400_codec_driver = { .driver = { .name = "wm8400-codec", @@ -1580,8 +1554,6 @@ static struct platform_driver wm8400_codec_driver = { }, .probe = wm8400_codec_probe, .remove = __exit_p(wm8400_codec_remove), - .suspend = wm8400_pdev_suspend, - .resume = wm8400_pdev_resume, }; static int __init wm8400_codec_init(void) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 060d5d06ba9..265e68c75df 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -219,7 +219,6 @@ static int wm8510_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -271,8 +270,8 @@ static void pll_factors(unsigned int target, unsigned int source) pll_div.k = K; } -static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -604,16 +603,9 @@ static int wm8510_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8510_snd_controls, ARRAY_SIZE(wm8510_snd_controls)); wm8510_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8510: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 25870a4652f..d3a61d7ea0c 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -117,7 +117,6 @@ static int wm8523_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -448,17 +447,9 @@ static int wm8523_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8523_snd_controls, ARRAY_SIZE(wm8523_snd_controls)); wm8523_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -638,21 +629,6 @@ static __devexit int wm8523_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8523_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8523_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8523_i2c_suspend NULL -#define wm8523_i2c_resume NULL -#endif - static const struct i2c_device_id wm8523_i2c_id[] = { { "wm8523", 0 }, { } @@ -666,8 +642,6 @@ static struct i2c_driver wm8523_i2c_driver = { }, .probe = wm8523_i2c_probe, .remove = __devexit_p(wm8523_i2c_remove), - .suspend = wm8523_i2c_suspend, - .resume = wm8523_i2c_resume, .id_table = wm8523_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c7815..d077df6f5e7 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -315,7 +315,6 @@ static int wm8580_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -407,8 +406,8 @@ static int pll_factors(struct _pll_div *pll_div, unsigned int target, return 0; } -static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { int offset; struct snd_soc_codec *codec = codec_dai->codec; @@ -800,17 +799,9 @@ static int wm8580_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8580_snd_controls, ARRAY_SIZE(wm8580_snd_controls)); wm8580_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -961,21 +952,6 @@ static int wm8580_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8580_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8580_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8580_i2c_suspend NULL -#define wm8580_i2c_resume NULL -#endif - static const struct i2c_device_id wm8580_i2c_id[] = { { "wm8580", 0 }, { } @@ -989,8 +965,6 @@ static struct i2c_driver wm8580_i2c_driver = { }, .probe = wm8580_i2c_probe, .remove = wm8580_i2c_remove, - .suspend = wm8580_i2c_suspend, - .resume = wm8580_i2c_resume, .id_table = wm8580_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c new file mode 100644 index 00000000000..24a35603bcf --- /dev/null +++ b/sound/soc/codecs/wm8711.c @@ -0,0 +1,633 @@ +/* + * wm8711.c -- WM8711 ALSA SoC Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur <linux@wolfsonmicro.com> + * + * Based on wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "wm8711.h" + +static struct snd_soc_codec *wm8711_codec; + +/* codec private data */ +struct wm8711_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8711_CACHEREGNUM]; + unsigned int sysclk; +}; + +/* + * wm8711 register cache + * We can't read the WM8711 register space when we are + * using 2 wire for device control, so we cache them instead. + * There is no point in caching the reset register + */ +static const u16 wm8711_reg[WM8711_CACHEREGNUM] = { + 0x0079, 0x0079, 0x000a, 0x0008, + 0x009f, 0x000a, 0x0000, 0x0000 +}; + +#define wm8711_reset(c) snd_soc_write(c, WM8711_RESET, 0) + +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); + +static const struct snd_kcontrol_new wm8711_snd_controls[] = { + +SOC_DOUBLE_R_TLV("Master Playback Volume", WM8711_LOUT1V, WM8711_ROUT1V, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Master Playback ZC Switch", WM8711_LOUT1V, WM8711_ROUT1V, + 7, 1, 0), + +}; + +/* Output Mixer */ +static const struct snd_kcontrol_new wm8711_output_mixer_controls[] = { +SOC_DAPM_SINGLE("Line Bypass Switch", WM8711_APANA, 3, 1, 0), +SOC_DAPM_SINGLE("HiFi Playback Switch", WM8711_APANA, 4, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8711_dapm_widgets[] = { +SND_SOC_DAPM_MIXER("Output Mixer", WM8711_PWR, 4, 1, + &wm8711_output_mixer_controls[0], + ARRAY_SIZE(wm8711_output_mixer_controls)), +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8711_PWR, 3, 1), +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("LHPOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +SND_SOC_DAPM_OUTPUT("RHPOUT"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* output mixer */ + {"Output Mixer", "Line Bypass Switch", "Line Input"}, + {"Output Mixer", "HiFi Playback Switch", "DAC"}, + + /* outputs */ + {"RHPOUT", NULL, "Output Mixer"}, + {"ROUT", NULL, "Output Mixer"}, + {"LHPOUT", NULL, "Output Mixer"}, + {"LOUT", NULL, "Output Mixer"}, +}; + +static int wm8711_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8711_dapm_widgets, + ARRAY_SIZE(wm8711_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + return 0; +} + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:4; + u8 bosr:1; + u8 usb:1; +}; + +/* codec mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0, 0x0}, + {18432000, 48000, 384, 0x0, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x0, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0x6, 0x0, 0x0}, + {18432000, 32000, 576, 0x6, 0x1, 0x0}, + {12000000, 32000, 375, 0x6, 0x0, 0x1}, + + /* 8k */ + {12288000, 8000, 1536, 0x3, 0x0, 0x0}, + {18432000, 8000, 2304, 0x3, 0x1, 0x0}, + {11289600, 8000, 1408, 0xb, 0x0, 0x0}, + {16934400, 8000, 2112, 0xb, 0x1, 0x0}, + {12000000, 8000, 1500, 0x3, 0x0, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0x7, 0x0, 0x0}, + {18432000, 96000, 192, 0x7, 0x1, 0x0}, + {12000000, 96000, 125, 0x7, 0x0, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x8, 0x0, 0x0}, + {16934400, 44100, 384, 0x8, 0x1, 0x0}, + {12000000, 44100, 272, 0x8, 0x1, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0xf, 0x0, 0x0}, + {16934400, 88200, 192, 0xf, 0x1, 0x0}, + {12000000, 88200, 136, 0xf, 0x1, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + return 0; +} + +static int wm8711_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8711_priv *wm8711 = codec->private_data; + u16 iface = snd_soc_read(codec, WM8711_IFACE) & 0xfffc; + int i = get_coeff(wm8711->sysclk, params_rate(params)); + u16 srate = (coeff_div[i].sr << 2) | + (coeff_div[i].bosr << 1) | coeff_div[i].usb; + + snd_soc_write(codec, WM8711_SRATE, srate); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + } + + snd_soc_write(codec, WM8711_IFACE, iface); + return 0; +} + +static int wm8711_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* set active */ + snd_soc_write(codec, WM8711_ACTIVE, 0x0001); + + return 0; +} + +static void wm8711_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + /* deactivate */ + if (!codec->active) { + udelay(50); + snd_soc_write(codec, WM8711_ACTIVE, 0x0); + } +} + +static int wm8711_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = snd_soc_read(codec, WM8711_APDIGI) & 0xfff7; + + if (mute) + snd_soc_write(codec, WM8711_APDIGI, mute_reg | 0x8); + else + snd_soc_write(codec, WM8711_APDIGI, mute_reg); + + return 0; +} + +static int wm8711_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8711_priv *wm8711 = codec->private_data; + + switch (freq) { + case 11289600: + case 12000000: + case 12288000: + case 16934400: + case 18432000: + wm8711->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + snd_soc_write(codec, WM8711_IFACE, iface); + return 0; +} + + +static int wm8711_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg = snd_soc_read(codec, WM8711_PWR) & 0xff7f; + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_write(codec, WM8711_PWR, reg); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, WM8711_PWR, reg | 0x0040); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, WM8711_ACTIVE, 0x0); + snd_soc_write(codec, WM8711_PWR, 0xffff); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8711_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8711_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8711_ops = { + .prepare = wm8711_pcm_prepare, + .hw_params = wm8711_hw_params, + .shutdown = wm8711_shutdown, + .digital_mute = wm8711_mute, + .set_sysclk = wm8711_set_dai_sysclk, + .set_fmt = wm8711_set_dai_fmt, +}; + +struct snd_soc_dai wm8711_dai = { + .name = "WM8711", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8711_RATES, + .formats = WM8711_FORMATS, + }, + .ops = &wm8711_ops, +}; +EXPORT_SYMBOL_GPL(wm8711_dai); + +static int wm8711_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + snd_soc_write(codec, WM8711_ACTIVE, 0x0); + wm8711_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8711_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8711_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static int wm8711_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8711_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8711_codec; + codec = wm8711_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8711_snd_controls, + ARRAY_SIZE(wm8711_snd_controls)); + wm8711_add_widgets(codec); + + return ret; + +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8711_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8711 = { + .probe = wm8711_probe, + .remove = wm8711_remove, + .suspend = wm8711_suspend, + .resume = wm8711_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8711); + +static int wm8711_register(struct wm8711_priv *wm8711, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &wm8711->codec; + u16 reg; + + if (wm8711_codec) { + dev_err(codec->dev, "Another WM8711 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8711; + codec->name = "WM8711"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8711_set_bias_level; + codec->dai = &wm8711_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8711_CACHEREGNUM; + codec->reg_cache = &wm8711->reg_cache; + + memcpy(codec->reg_cache, wm8711_reg, sizeof(wm8711_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ret = wm8711_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + goto err; + } + + wm8711_dai.dev = codec->dev; + + wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits */ + reg = snd_soc_read(codec, WM8711_LOUT1V); + snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100); + reg = snd_soc_read(codec, WM8711_ROUT1V); + snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100); + + wm8711_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8711_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8711); + return ret; +} + +static void wm8711_unregister(struct wm8711_priv *wm8711) +{ + wm8711_set_bias_level(&wm8711->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8711_dai); + snd_soc_unregister_codec(&wm8711->codec); + kfree(wm8711); + wm8711_codec = NULL; +} + +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8711_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8711_priv *wm8711; + + wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + if (wm8711 == NULL) + return -ENOMEM; + + codec = &wm8711->codec; + codec->control_data = spi; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, wm8711); + + return wm8711_register(wm8711, SND_SOC_SPI); +} + +static int __devexit wm8711_spi_remove(struct spi_device *spi) +{ + struct wm8711_priv *wm8711 = dev_get_drvdata(&spi->dev); + + wm8711_unregister(wm8711); + + return 0; +} + +static struct spi_driver wm8711_spi_driver = { + .driver = { + .name = "wm8711", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8711_spi_probe, + .remove = __devexit_p(wm8711_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8711_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8711_priv *wm8711; + struct snd_soc_codec *codec; + + wm8711 = kzalloc(sizeof(struct wm8711_priv), GFP_KERNEL); + if (wm8711 == NULL) + return -ENOMEM; + + codec = &wm8711->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8711); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8711_register(wm8711, SND_SOC_I2C); +} + +static __devexit int wm8711_i2c_remove(struct i2c_client *client) +{ + struct wm8711_priv *wm8711 = i2c_get_clientdata(client); + wm8711_unregister(wm8711); + return 0; +} + +static const struct i2c_device_id wm8711_i2c_id[] = { + { "wm8711", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8711_i2c_id); + +static struct i2c_driver wm8711_i2c_driver = { + .driver = { + .name = "WM8711 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8711_i2c_probe, + .remove = __devexit_p(wm8711_i2c_remove), + .id_table = wm8711_i2c_id, +}; +#endif + +static int __init wm8711_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8711_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8711 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8711_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8711 SPI driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8711_modinit); + +static void __exit wm8711_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8711_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8711_spi_driver); +#endif +} +module_exit(wm8711_exit); + +MODULE_DESCRIPTION("ASoC WM8711 driver"); +MODULE_AUTHOR("Mike Arthur"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8711.h b/sound/soc/codecs/wm8711.h new file mode 100644 index 00000000000..381e84a4381 --- /dev/null +++ b/sound/soc/codecs/wm8711.h @@ -0,0 +1,42 @@ +/* + * wm8711.h -- WM8711 Soc Audio driver + * + * Copyright 2006 Wolfson Microelectronics + * + * Author: Mike Arthur <linux@wolfsonmicro.com> + * + * Based on wm8731.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8711_H +#define _WM8711_H + +/* WM8711 register space */ + +#define WM8711_LOUT1V 0x02 +#define WM8711_ROUT1V 0x03 +#define WM8711_APANA 0x04 +#define WM8711_APDIGI 0x05 +#define WM8711_PWR 0x06 +#define WM8711_IFACE 0x07 +#define WM8711_SRATE 0x08 +#define WM8711_ACTIVE 0x09 +#define WM8711_RESET 0x0f + +#define WM8711_CACHEREGNUM 8 + +#define WM8711_SYSCLK 0 +#define WM8711_DAI 0 + +struct wm8711_setup_data { + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8711_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8711; + +#endif diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c new file mode 100644 index 00000000000..d8ffbd641d7 --- /dev/null +++ b/sound/soc/codecs/wm8727.c @@ -0,0 +1,135 @@ +/* + * wm8727.c + * + * Created on: 15-Oct-2009 + * Author: neil.jones@imgtec.com + * + * Copyright (C) 2009 Imagination Technologies Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "wm8727.h" +/* + * Note this is a simple chip with no configuration interface, sample rate is + * determined automatically by examining the Master clock and Bit clock ratios + */ +#define WM8727_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000) + + +struct snd_soc_dai wm8727_dai = { + .name = "WM8727", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8727_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, + }, +}; +EXPORT_SYMBOL_GPL(wm8727_dai); + +static int wm8727_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + mutex_init(&codec->mutex); + codec->name = "WM8727"; + codec->owner = THIS_MODULE; + codec->dai = &wm8727_dai; + codec->num_dai = 1; + socdev->card->codec = codec; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8727: failed to create pcms\n"); + goto pcm_err; + } + + return ret; + +pcm_err: + kfree(socdev->card->codec); + socdev->card->codec = NULL; + return ret; +} + +static int wm8727_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + if (codec == NULL) + return 0; + snd_soc_free_pcms(socdev); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8727 = { + .probe = wm8727_soc_probe, + .remove = wm8727_soc_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8727); + + +static __devinit int wm8727_platform_probe(struct platform_device *pdev) +{ + wm8727_dai.dev = &pdev->dev; + return snd_soc_register_dai(&wm8727_dai); +} + +static int __devexit wm8727_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_dai(&wm8727_dai); + return 0; +} + +static struct platform_driver wm8727_codec_driver = { + .driver = { + .name = "wm8727-codec", + .owner = THIS_MODULE, + }, + + .probe = wm8727_platform_probe, + .remove = __devexit_p(wm8727_platform_remove), +}; + +static int __init wm8727_init(void) +{ + return platform_driver_register(&wm8727_codec_driver); +} +module_init(wm8727_init); + +static void __exit wm8727_exit(void) +{ + platform_driver_unregister(&wm8727_codec_driver); +} +module_exit(wm8727_exit); + +MODULE_DESCRIPTION("ASoC wm8727 driver"); +MODULE_AUTHOR("Neil Jones"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8727.h b/sound/soc/codecs/wm8727.h new file mode 100644 index 00000000000..ee19aa71bcd --- /dev/null +++ b/sound/soc/codecs/wm8727.h @@ -0,0 +1,21 @@ +/* + * wm8727.h + * + * Created on: 15-Oct-2009 + * Author: neil.jones@imgtec.com + * + * Copyright (C) 2009 Imagination Technologies Ltd. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef WM8727_H_ +#define WM8727_H_ + +extern struct snd_soc_dai wm8727_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8727; + +#endif /* WM8727_H_ */ diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 16e969a762c..3fb653ba363 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -74,8 +74,6 @@ static int wm8728_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -287,17 +285,9 @@ static int wm8728_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8728_snd_controls, ARRAY_SIZE(wm8728_snd_controls)); wm8728_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8728: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index d3fd4f28d96..3a497810f93 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -19,6 +19,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regulator/consumer.h> #include <linux/spi/spi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -33,9 +34,18 @@ static struct snd_soc_codec *wm8731_codec; struct snd_soc_codec_device soc_codec_dev_wm8731; +#define WM8731_NUM_SUPPLIES 4 +static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { + "AVDD", + "HPVDD", + "DCVDD", + "DBVDD", +}; + /* codec private data */ struct wm8731_priv { struct snd_soc_codec codec; + struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; u16 reg_cache[WM8731_CACHEREGNUM]; unsigned int sysclk; }; @@ -149,7 +159,6 @@ static int wm8731_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -422,9 +431,12 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + struct wm8731_priv *wm8731 = codec->private_data; snd_soc_write(codec, WM8731_ACTIVE, 0x0); wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); return 0; } @@ -432,10 +444,16 @@ static int wm8731_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; - int i; + struct wm8731_priv *wm8731 = codec->private_data; + int i, ret; u8 data[2]; u16 *cache = codec->reg_cache; + ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) + return ret; + /* Sync reg_cache with the hardware */ for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) { data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); @@ -444,6 +462,7 @@ static int wm8731_resume(struct platform_device *pdev) } wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8731_set_bias_level(codec, codec->suspend_bias_level); + return 0; } #else @@ -475,17 +494,9 @@ static int wm8731_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8731_snd_controls, ARRAY_SIZE(wm8731_snd_controls)); wm8731_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -512,7 +523,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); static int wm8731_register(struct wm8731_priv *wm8731, enum snd_soc_control_type control) { - int ret; + int ret, i; struct snd_soc_codec *codec = &wm8731->codec; if (wm8731_codec) { @@ -543,10 +554,27 @@ static int wm8731_register(struct wm8731_priv *wm8731, goto err; } + for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) + wm8731->supplies[i].supply = wm8731_supply_names[i]; + + ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to request supplies: %d\n", ret); + goto err; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8731->supplies), + wm8731->supplies); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + goto err_regulator_get; + } + ret = wm8731_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); - goto err; + goto err_regulator_enable; } wm8731_dai.dev = codec->dev; @@ -567,7 +595,7 @@ static int wm8731_register(struct wm8731_priv *wm8731, ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); - goto err; + goto err_regulator_enable; } ret = snd_soc_register_dai(&wm8731_dai); @@ -581,6 +609,10 @@ static int wm8731_register(struct wm8731_priv *wm8731, err_codec: snd_soc_unregister_codec(codec); +err_regulator_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); +err_regulator_get: + regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); err: kfree(wm8731); return ret; @@ -591,6 +623,8 @@ static void wm8731_unregister(struct wm8731_priv *wm8731) wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF); snd_soc_unregister_dai(&wm8731_dai); snd_soc_unregister_codec(&wm8731->codec); + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + regulator_bulk_free(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); kfree(wm8731); wm8731_codec = NULL; } @@ -623,21 +657,6 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8731_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8731_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8731_spi_suspend NULL -#define wm8731_spi_resume NULL -#endif - static struct spi_driver wm8731_spi_driver = { .driver = { .name = "wm8731", @@ -645,8 +664,6 @@ static struct spi_driver wm8731_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8731_spi_probe, - .suspend = wm8731_spi_suspend, - .resume = wm8731_spi_resume, .remove = __devexit_p(wm8731_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -679,21 +696,6 @@ static __devexit int wm8731_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8731_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8731_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8731_i2c_suspend NULL -#define wm8731_i2c_resume NULL -#endif - static const struct i2c_device_id wm8731_i2c_id[] = { { "wm8731", 0 }, { } @@ -707,8 +709,6 @@ static struct i2c_driver wm8731_i2c_driver = { }, .probe = wm8731_i2c_probe, .remove = __devexit_p(wm8731_i2c_remove), - .suspend = wm8731_i2c_suspend, - .resume = wm8731_i2c_resume, .id_table = wm8731_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 4ba1e7e93fb..475c67ac781 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -403,7 +403,6 @@ static int wm8750_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -772,16 +771,8 @@ static int wm8750_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8750_snd_controls, ARRAY_SIZE(wm8750_snd_controls)); wm8750_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to register card\n"); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 5ad677ce80d..d6850dacda2 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -673,7 +673,6 @@ static int wm8753_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -724,8 +723,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { u16 reg, enable; int offset; @@ -1583,18 +1582,9 @@ static int wm8753_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8753_snd_controls, ARRAY_SIZE(wm8753_snd_controls)); wm8753_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8753: failed to register card\n"); - goto card_err; - } return 0; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); - pcm_err: return ret; } @@ -1767,21 +1757,6 @@ static int wm8753_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8753_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8753_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8753_i2c_suspend NULL -#define wm8753_i2c_resume NULL -#endif - static const struct i2c_device_id wm8753_i2c_id[] = { { "wm8753", 0 }, { } @@ -1795,8 +1770,6 @@ static struct i2c_driver wm8753_i2c_driver = { }, .probe = wm8753_i2c_probe, .remove = wm8753_i2c_remove, - .suspend = wm8753_i2c_suspend, - .resume = wm8753_i2c_resume, .id_table = wm8753_i2c_id, }; #endif @@ -1852,22 +1825,6 @@ static int __devexit wm8753_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8753_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8753_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} - -#else -#define wm8753_spi_suspend NULL -#define wm8753_spi_resume NULL -#endif - static struct spi_driver wm8753_spi_driver = { .driver = { .name = "wm8753", @@ -1876,8 +1833,6 @@ static struct spi_driver wm8753_spi_driver = { }, .probe = wm8753_spi_probe, .remove = __devexit_p(wm8753_spi_remove), - .suspend = wm8753_spi_suspend, - .resume = wm8753_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index a9829aa26e5..ab2c0da1809 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -447,17 +447,8 @@ static int wm8776_probe(struct platform_device *pdev) ARRAY_SIZE(wm8776_dapm_widgets)); snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -616,21 +607,6 @@ static int __devexit wm8776_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8776_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8776_spi_suspend NULL -#define wm8776_spi_resume NULL -#endif - static struct spi_driver wm8776_spi_driver = { .driver = { .name = "wm8776", @@ -638,8 +614,6 @@ static struct spi_driver wm8776_spi_driver = { .owner = THIS_MODULE, }, .probe = wm8776_spi_probe, - .suspend = wm8776_spi_suspend, - .resume = wm8776_spi_resume, .remove = __devexit_p(wm8776_spi_remove), }; #endif /* CONFIG_SPI_MASTER */ @@ -673,21 +647,6 @@ static __devexit int wm8776_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) -{ - return snd_soc_suspend_device(&i2c->dev); -} - -static int wm8776_i2c_resume(struct i2c_client *i2c) -{ - return snd_soc_resume_device(&i2c->dev); -} -#else -#define wm8776_i2c_suspend NULL -#define wm8776_i2c_resume NULL -#endif - static const struct i2c_device_id wm8776_i2c_id[] = { { "wm8776", 0 }, { } @@ -701,8 +660,6 @@ static struct i2c_driver wm8776_i2c_driver = { }, .probe = wm8776_i2c_probe, .remove = __devexit_p(wm8776_i2c_remove), - .suspend = wm8776_i2c_suspend, - .resume = wm8776_i2c_resume, .id_table = wm8776_i2c_id, }; #endif diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 5e9c855c003..c9438dd62df 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -618,8 +618,6 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -814,8 +812,8 @@ reenable: return 0; } -static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8900_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { return wm8900_set_fll(codec_dai->codec, pll_id, freq_in, freq_out); } @@ -1312,21 +1310,6 @@ static __devexit int wm8900_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8900_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8900_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8900_i2c_suspend NULL -#define wm8900_i2c_resume NULL -#endif - static const struct i2c_device_id wm8900_i2c_id[] = { { "wm8900", 0 }, { } @@ -1340,8 +1323,6 @@ static struct i2c_driver wm8900_i2c_driver = { }, .probe = wm8900_i2c_probe, .remove = __devexit_p(wm8900_i2c_remove), - .suspend = wm8900_i2c_suspend, - .resume = wm8900_i2c_resume, .id_table = wm8900_i2c_id, }; @@ -1370,17 +1351,6 @@ static int wm8900_probe(struct platform_device *pdev) ARRAY_SIZE(wm8900_snd_controls)); wm8900_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "Failed to register card\n"); - goto card_err; - } - - return ret; - -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index fe1307b500c..b8cae175864 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -919,8 +919,6 @@ static int wm8903_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -1655,21 +1653,6 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8903_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8903_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8903_i2c_suspend NULL -#define wm8903_i2c_resume NULL -#endif - /* i2c codec control layer */ static const struct i2c_device_id wm8903_i2c_id[] = { { "wm8903", 0 }, @@ -1684,8 +1667,6 @@ static struct i2c_driver wm8903_i2c_driver = { }, .probe = wm8903_i2c_probe, .remove = __devexit_p(wm8903_i2c_remove), - .suspend = wm8903_i2c_suspend, - .resume = wm8903_i2c_resume, .id_table = wm8903_i2c_id, }; @@ -1712,17 +1693,8 @@ static int wm8903_probe(struct platform_device *pdev) ARRAY_SIZE(wm8903_snd_controls)); wm8903_add_widgets(socdev->card->codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&pdev->dev, "wm8903: failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: return ret; } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 1ef2454c520..3d850b97037 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -298,7 +298,6 @@ static int wm8940_add_widgets(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); if (ret) goto error_ret; - ret = snd_soc_dapm_new_widgets(codec); error_ret: return ret; @@ -536,8 +535,8 @@ static void pll_factors(unsigned int target, unsigned int source) } /* Untested at the moment */ -static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -731,12 +730,6 @@ static int wm8940_probe(struct platform_device *pdev) if (ret) goto error_free_pcms; - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto error_free_pcms; - } - return ret; error_free_pcms: @@ -877,21 +870,6 @@ static int __devexit wm8940_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8940_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8940_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8940_i2c_suspend NULL -#define wm8940_i2c_resume NULL -#endif - static const struct i2c_device_id wm8940_i2c_id[] = { { "wm8940", 0 }, { } @@ -905,8 +883,6 @@ static struct i2c_driver wm8940_i2c_driver = { }, .probe = wm8940_i2c_probe, .remove = __devexit_p(wm8940_i2c_remove), - .suspend = wm8940_i2c_suspend, - .resume = wm8940_i2c_resume, .id_table = wm8940_i2c_id, }; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f59703be61c..d07bcc1e1c6 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -307,7 +307,6 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -540,8 +539,8 @@ static int pll_factors(unsigned int source, unsigned int target, return 0; } -static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; u16 reg; @@ -713,17 +712,9 @@ static int wm8960_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8960_snd_controls, ARRAY_SIZE(wm8960_snd_controls)); wm8960_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -883,21 +874,6 @@ static __devexit int wm8960_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8960_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8960_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8960_i2c_suspend NULL -#define wm8960_i2c_resume NULL -#endif - static const struct i2c_device_id wm8960_i2c_id[] = { { "wm8960", 0 }, { } @@ -911,8 +887,6 @@ static struct i2c_driver wm8960_i2c_driver = { }, .probe = wm8960_i2c_probe, .remove = __devexit_p(wm8960_i2c_remove), - .suspend = wm8960_i2c_suspend, - .resume = wm8960_i2c_resume, .id_table = wm8960_i2c_id, }; diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 50303208589..a8007d58813 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -986,19 +986,9 @@ static int wm8961_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets, ARRAY_SIZE(wm8961_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -1206,21 +1196,6 @@ static __devexit int wm8961_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8961_i2c_suspend(struct i2c_client *client, pm_message_t state) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8961_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8961_i2c_suspend NULL -#define wm8961_i2c_resume NULL -#endif - static const struct i2c_device_id wm8961_i2c_id[] = { { "wm8961", 0 }, { } @@ -1234,8 +1209,6 @@ static struct i2c_driver wm8961_i2c_driver = { }, .probe = wm8961_i2c_probe, .remove = __devexit_p(wm8961_i2c_remove), - .suspend = wm8961_i2c_suspend, - .resume = wm8961_i2c_resume, .id_table = wm8961_i2c_id, }; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index d66efb0546e..d9540d55fc8 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -338,8 +338,6 @@ static int wm8971_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - return 0; } @@ -703,16 +701,9 @@ static int wm8971_init(struct snd_soc_device *socdev, snd_soc_add_controls(codec, wm8971_snd_controls, ARRAY_SIZE(wm8971_snd_controls)); wm8971_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8971: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 98d663afc97..81c57b5c591 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -276,41 +276,42 @@ static int wm8974_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } struct pll_ { - unsigned int pre_div:4; /* prescale - 1 */ + unsigned int pre_div:1; unsigned int n:4; unsigned int k; }; -static struct pll_ pll_div; - /* The size in bits of the pll divide multiplied by 10 * to allow rounding later */ #define FIXED_PLL_SIZE ((1 << 24) * 10) -static void pll_factors(unsigned int target, unsigned int source) +static void pll_factors(struct pll_ *pll_div, + unsigned int target, unsigned int source) { unsigned long long Kpart; unsigned int K, Ndiv, Nmod; + /* There is a fixed divide by 4 in the output path */ + target *= 4; + Ndiv = target / source; if (Ndiv < 6) { - source >>= 1; - pll_div.pre_div = 1; + source /= 2; + pll_div->pre_div = 1; Ndiv = target / source; } else - pll_div.pre_div = 0; + pll_div->pre_div = 0; if ((Ndiv < 6) || (Ndiv > 12)) printk(KERN_WARNING "WM8974 N value %u outwith recommended range!\n", Ndiv); - pll_div.n = Ndiv; + pll_div->n = Ndiv; Nmod = target % source; Kpart = FIXED_PLL_SIZE * (long long)Nmod; @@ -325,13 +326,14 @@ static void pll_factors(unsigned int target, unsigned int source) /* Move down to proper range now rounding is done */ K /= 10; - pll_div.k = K; + pll_div->k = K; } -static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; + struct pll_ pll_div; u16 reg; if (freq_in == 0 || freq_out == 0) { @@ -345,7 +347,7 @@ static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } - pll_factors(freq_out*4, freq_in); + pll_factors(&pll_div, freq_out, freq_in); snd_soc_write(codec, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n); snd_soc_write(codec, WM8974_PLLK1, pll_div.k >> 18); @@ -638,17 +640,9 @@ static int wm8974_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm8974_snd_controls, ARRAY_SIZE(wm8974_snd_controls)); wm8974_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 3f530f8a972..2862e4dced2 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -790,19 +790,9 @@ static int wm8988_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets, ARRAY_SIZE(wm8988_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -944,21 +934,6 @@ static int wm8988_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm8988_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm8988_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm8988_i2c_suspend NULL -#define wm8988_i2c_resume NULL -#endif - static const struct i2c_device_id wm8988_i2c_id[] = { { "wm8988", 0 }, { } @@ -972,8 +947,6 @@ static struct i2c_driver wm8988_i2c_driver = { }, .probe = wm8988_i2c_probe, .remove = wm8988_i2c_remove, - .suspend = wm8988_i2c_suspend, - .resume = wm8988_i2c_resume, .id_table = wm8988_i2c_id, }; #endif @@ -1006,21 +979,6 @@ static int __devexit wm8988_spi_remove(struct spi_device *spi) return 0; } -#ifdef CONFIG_PM -static int wm8988_spi_suspend(struct spi_device *spi, pm_message_t msg) -{ - return snd_soc_suspend_device(&spi->dev); -} - -static int wm8988_spi_resume(struct spi_device *spi) -{ - return snd_soc_resume_device(&spi->dev); -} -#else -#define wm8988_spi_suspend NULL -#define wm8988_spi_resume NULL -#endif - static struct spi_driver wm8988_spi_driver = { .driver = { .name = "wm8988", @@ -1029,8 +987,6 @@ static struct spi_driver wm8988_spi_driver = { }, .probe = wm8988_spi_probe, .remove = __devexit_p(wm8988_spi_remove), - .suspend = wm8988_spi_suspend, - .resume = wm8988_spi_resume, }; #endif diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 2d702db4131..341481e0e83 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,6 @@ static int wm8990_add_widgets(struct snd_soc_codec *codec) /* set up the WM8990 audio map */ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -972,8 +971,8 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target, pll_div->k = K; } -static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { u16 reg; struct snd_soc_codec *codec = codec_dai->codec; @@ -1409,16 +1408,9 @@ static int wm8990_init(struct snd_soc_device *socdev) snd_soc_add_controls(codec, wm8990_snd_controls, ARRAY_SIZE(wm8990_snd_controls)); wm8990_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm8990: failed to register card\n"); - goto card_err; - } + return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: kfree(codec->reg_cache); return ret; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index d9987999e92..5e32f2ed5fc 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -422,7 +422,7 @@ static int fll_factors(struct _fll_div *fll_div, unsigned int Fref, return 0; } -static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, +static int wm8993_set_fll(struct snd_soc_dai *dai, int fll_id, int source, unsigned int Fref, unsigned int Fout) { struct snd_soc_codec *codec = dai->codec; @@ -1464,19 +1464,8 @@ static int wm8993_probe(struct platform_device *pdev) wm_hubs_add_analogue_routes(codec, wm8993->pdata.lineout1_diff, wm8993->pdata.lineout2_diff); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card\n"); - goto card_err; - } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); err: return ret; } @@ -1572,33 +1561,15 @@ static int wm8993_i2c_probe(struct i2c_client *i2c, /* Use automatic clock configuration */ snd_soc_update_bits(codec, WM8993_CLOCKING_4, WM8993_SR_MODE, 0); - if (!wm8993->pdata.lineout1_diff) - snd_soc_update_bits(codec, WM8993_LINE_MIXER1, - WM8993_LINEOUT1_MODE, - WM8993_LINEOUT1_MODE); - if (!wm8993->pdata.lineout2_diff) - snd_soc_update_bits(codec, WM8993_LINE_MIXER2, - WM8993_LINEOUT2_MODE, - WM8993_LINEOUT2_MODE); - - if (wm8993->pdata.lineout1fb) - snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, - WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); - - if (wm8993->pdata.lineout2fb) - snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, - WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); - - /* Apply the microphone bias/detection configuration - the - * platform data is directly applicable to the register. */ - snd_soc_update_bits(codec, WM8993_MICBIAS, - WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | - WM8993_MICB1_LVL | WM8993_MICB2_LVL, - wm8993->pdata.jd_scthr << WM8993_JD_SCTHR_SHIFT | - wm8993->pdata.jd_thr << WM8993_JD_THR_SHIFT | - wm8993->pdata.micbias1_lvl | - wm8993->pdata.micbias1_lvl << 1); - + wm_hubs_handle_analogue_pdata(codec, wm8993->pdata.lineout1_diff, + wm8993->pdata.lineout2_diff, + wm8993->pdata.lineout1fb, + wm8993->pdata.lineout2fb, + wm8993->pdata.jd_scthr, + wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_lvl, + wm8993->pdata.micbias2_lvl); + ret = wm8993_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 686e5aa9720..c468497314b 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1262,19 +1262,9 @@ static int wm9081_probe(struct platform_device *pdev) snd_soc_dapm_new_controls(codec, wm9081_dapm_widgets, ARRAY_SIZE(wm9081_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); - snd_soc_dapm_new_widgets(codec); - - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(codec->dev, "failed to register card: %d\n", ret); - goto card_err; - } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); pcm_err: return ret; } @@ -1452,21 +1442,6 @@ static __devexit int wm9081_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM -static int wm9081_i2c_suspend(struct i2c_client *client, pm_message_t msg) -{ - return snd_soc_suspend_device(&client->dev); -} - -static int wm9081_i2c_resume(struct i2c_client *client) -{ - return snd_soc_resume_device(&client->dev); -} -#else -#define wm9081_i2c_suspend NULL -#define wm9081_i2c_resume NULL -#endif - static const struct i2c_device_id wm9081_i2c_id[] = { { "wm9081", 0 }, { } @@ -1480,8 +1455,6 @@ static struct i2c_driver wm9081_i2c_driver = { }, .probe = wm9081_i2c_probe, .remove = __devexit_p(wm9081_i2c_remove), - .suspend = wm9081_i2c_suspend, - .resume = wm9081_i2c_resume, .id_table = wm9081_i2c_id, }; diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index e7d2840d9e5..ec54c6da985 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -205,7 +205,6 @@ static int wm9705_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets, ARRAY_SIZE(wm9705_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -403,12 +402,6 @@ static int wm9705_soc_probe(struct platform_device *pdev) ARRAY_SIZE(wm9705_snd_ac97_controls)); wm9705_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm9705: failed to register card\n"); - goto reset_err; - } - return 0; reset_err: diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fd4e88f50c..0ac1215dcd9 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -436,7 +436,6 @@ static int wm9712_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -695,17 +694,11 @@ static int wm9712_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm9712_snd_ac97_controls, ARRAY_SIZE(wm9712_snd_ac97_controls)); wm9712_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - printk(KERN_ERR "wm9712: failed to register card\n"); - goto reset_err; - } return 0; reset_err: snd_soc_free_pcms(socdev); - pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index abed37acf78..c58aab375ed 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -165,9 +165,9 @@ SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), -SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), -SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), -SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), +SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), +SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), +SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), @@ -266,7 +266,7 @@ static int mixer_event(struct snd_soc_dapm_widget *w, /* Left Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Beep Playback Switch", HPL_MIXER, 5, 1, 0), SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), @@ -276,7 +276,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), /* Right Headphone Mixers */ static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Beep Playback Switch", HPR_MIXER, 5, 1, 0), SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), @@ -294,7 +294,7 @@ SOC_DAPM_ENUM("Route", wm9713_enum[0]); /* Speaker Mixer */ static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), +SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 11, 1, 1), SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), @@ -304,7 +304,7 @@ SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), /* Mono Mixer */ static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { -SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), +SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 7, 1, 1), SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), @@ -463,7 +463,7 @@ SND_SOC_DAPM_VMID("VMID"), static const struct snd_soc_dapm_route audio_map[] = { /* left HP mixer */ - {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Left HP Mixer", "Beep Playback Switch", "PCBEEP"}, {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, @@ -472,7 +472,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left HP Mixer", NULL, "Capture Headphone Mux"}, /* right HP mixer */ - {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Right HP Mixer", "Beep Playback Switch", "PCBEEP"}, {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, @@ -491,7 +491,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Capture Mixer", NULL, "Right Capture Source"}, /* speaker mixer */ - {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Speaker Mixer", "Beep Playback Switch", "PCBEEP"}, {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, @@ -499,7 +499,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, /* mono mixer */ - {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Mono Mixer", "Beep Playback Switch", "PCBEEP"}, {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, @@ -625,7 +625,6 @@ static int wm9713_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); - snd_soc_dapm_new_widgets(codec); return 0; } @@ -800,8 +799,8 @@ static int wm9713_set_pll(struct snd_soc_codec *codec, return 0; } -static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct snd_soc_codec *codec = codec_dai->codec; return wm9713_set_pll(codec, pll_id, freq_in, freq_out); @@ -1247,14 +1246,11 @@ static int wm9713_soc_probe(struct platform_device *pdev) snd_soc_add_controls(codec, wm9713_snd_ac97_controls, ARRAY_SIZE(wm9713_snd_ac97_controls)); wm9713_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) - goto reset_err; + return 0; reset_err: snd_soc_free_pcms(socdev); - pcm_err: snd_soc_free_ac97_codec(codec); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e542027eea8..d73c30536a2 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -438,11 +438,11 @@ static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1LN"), SND_SOC_DAPM_INPUT("IN1LP"), SND_SOC_DAPM_INPUT("IN2LN"), -SND_SOC_DAPM_INPUT("IN2LP/VXRN"), +SND_SOC_DAPM_INPUT("IN2LP:VXRN"), SND_SOC_DAPM_INPUT("IN1RN"), SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), -SND_SOC_DAPM_INPUT("IN2RP/VXRP"), +SND_SOC_DAPM_INPUT("IN2RP:VXRP"), SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0), SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0), @@ -537,14 +537,14 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "IN1R PGA", "IN1RP Switch", "IN1RP" }, { "IN1R PGA", "IN1RN Switch", "IN1RN" }, - { "IN2L PGA", "IN2LP Switch", "IN2LP/VXRN" }, + { "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" }, { "IN2L PGA", "IN2LN Switch", "IN2LN" }, - { "IN2R PGA", "IN2RP Switch", "IN2RP/VXRP" }, + { "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" }, { "IN2R PGA", "IN2RN Switch", "IN2RN" }, - { "Direct Voice", NULL, "IN2LP/VXRN" }, - { "Direct Voice", NULL, "IN2RP/VXRP" }, + { "Direct Voice", NULL, "IN2LP:VXRN" }, + { "Direct Voice", NULL, "IN2RP:VXRP" }, { "MIXINL", "IN1L Switch", "IN1L PGA" }, { "MIXINL", "IN2L Switch", "IN2L PGA" }, @@ -565,7 +565,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Left Output Mixer", "Right Input Switch", "MIXINR" }, { "Left Output Mixer", "IN2RN Switch", "IN2RN" }, { "Left Output Mixer", "IN2LN Switch", "IN2LN" }, - { "Left Output Mixer", "IN2LP Switch", "IN2LP/VXRN" }, + { "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" }, { "Left Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Left Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -573,7 +573,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "Right Output Mixer", "Right Input Switch", "MIXINR" }, { "Right Output Mixer", "IN2LN Switch", "IN2LN" }, { "Right Output Mixer", "IN2RN Switch", "IN2RN" }, - { "Right Output Mixer", "IN2RP Switch", "IN2RP/VXRP" }, + { "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" }, { "Right Output Mixer", "IN1L Switch", "IN1L PGA" }, { "Right Output Mixer", "IN1R Switch", "IN1R PGA" }, @@ -738,6 +738,41 @@ int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); +int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, + int lineout1_diff, int lineout2_diff, + int lineout1fb, int lineout2fb, + int jd_scthr, int jd_thr, int micbias1_lvl, + int micbias2_lvl) +{ + if (!lineout1_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER1, + WM8993_LINEOUT1_MODE, + WM8993_LINEOUT1_MODE); + if (!lineout2_diff) + snd_soc_update_bits(codec, WM8993_LINE_MIXER2, + WM8993_LINEOUT2_MODE, + WM8993_LINEOUT2_MODE); + + if (lineout1fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB); + + if (lineout2fb) + snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL, + WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + + snd_soc_update_bits(codec, WM8993_MICBIAS, + WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | + WM8993_MICB1_LVL | WM8993_MICB2_LVL, + jd_scthr << WM8993_JD_SCTHR_SHIFT | + jd_thr << WM8993_JD_THR_SHIFT | + micbias1_lvl | + micbias2_lvl << WM8993_MICB2_LVL_SHIFT); + + return 0; +} +EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata); + MODULE_DESCRIPTION("Shared support for Wolfson hubs products"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index ec09cb6a293..36d3fba1de8 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -20,5 +20,10 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int); +extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, + int lineout1_diff, int lineout2_diff, + int lineout1fb, int lineout2fb, + int jd_scthr, int jd_thr, + int micbias1_lvl, int micbias2_lvl); #endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 4dfd4ad9d90..047ee39418c 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -13,9 +13,9 @@ config SND_DAVINCI_SOC_MCASP tristate config SND_DAVINCI_SOC_EVM - tristate "SoC Audio support for DaVinci DM6446 or DM355 EVM" + tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC - depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM + depends on MACH_DAVINCI_EVM || MACH_DAVINCI_DM355_EVM || MACH_DAVINCI_DM365_EVM select SND_DAVINCI_SOC_I2S select SND_SOC_TLV320AIC3X help diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 67414f65940..7ccbe6684fc 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -45,7 +45,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, unsigned sysclk; /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm()) + if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || + machine_is_davinci_dm365_evm()) sysclk = 27000000; /* ASP0 in DM6446 EVM is clocked by U55, as configured by @@ -176,7 +177,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .ops = &evm_ops, }; -/* davinci-evm audio machine driver */ +/* davinci dm6446, dm355 or dm365 evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { .name = "DaVinci EVM", .platform = &davinci_soc_platform, @@ -243,7 +244,7 @@ static int __init evm_init(void) int index; int ret; - if (machine_is_davinci_evm()) { + if (machine_is_davinci_evm() || machine_is_davinci_dm365_evm()) { evm_snd_dev_data = &evm_snd_devdata; index = 0; } else if (machine_is_davinci_dm355_evm()) { diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 4ae70704802..6362ca05506 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -97,12 +97,24 @@ enum { DAVINCI_MCBSP_WORD_32, }; +static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = 1, + [SNDRV_PCM_FORMAT_S16_LE] = 2, + [SNDRV_PCM_FORMAT_S32_LE] = 4, +}; + +static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = DAVINCI_MCBSP_WORD_8, + [SNDRV_PCM_FORMAT_S16_LE] = DAVINCI_MCBSP_WORD_16, + [SNDRV_PCM_FORMAT_S32_LE] = DAVINCI_MCBSP_WORD_32, +}; + +static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = { + [SNDRV_PCM_FORMAT_S8] = SNDRV_PCM_FORMAT_S16_LE, + [SNDRV_PCM_FORMAT_S16_LE] = SNDRV_PCM_FORMAT_S32_LE, +}; + struct davinci_mcbsp_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; #define MOD_DSP_A 0 @@ -110,6 +122,27 @@ struct davinci_mcbsp_dev { int mode; u32 pcr; struct clk *clk; + /* + * Combining both channels into 1 element will at least double the + * amount of time between servicing the dma channel, increase + * effiency, and reduce the chance of overrun/underrun. But, + * it will result in the left & right channels being swapped. + * + * If relabeling the left and right channels is not possible, + * you may want to let the codec know to swap them back. + * + * It may allow x10 the amount of time to service dma requests, + * if the codec is master and is using an unnecessarily fast bit clock + * (ie. tlvaic23b), independent of the sample rate. So, having an + * entire frame at once means it can be serviced at the sample rate + * instead of the bit clock rate. + * + * In the now unlikely case that an underrun still + * occurs, both the left and right samples will be repeated + * so that no pops are heard, and the left and right channels + * won't end up being swapped because of the underrun. + */ + unsigned enable_channel_combine:1; }; static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, @@ -349,6 +382,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, int mcbsp_word_length; unsigned int rcr, xcr, srgr; u32 spcr; + snd_pcm_format_t fmt; + unsigned element_cnt = 1; /* general line settings */ spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); @@ -378,27 +413,24 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1); } /* Determine xfer data type */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - dma_params->data_type = 1; - mcbsp_word_length = DAVINCI_MCBSP_WORD_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - dma_params->data_type = 2; - mcbsp_word_length = DAVINCI_MCBSP_WORD_16; - break; - case SNDRV_PCM_FORMAT_S32_LE: - dma_params->data_type = 4; - mcbsp_word_length = DAVINCI_MCBSP_WORD_32; - break; - default: + fmt = params_format(params); + if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) { printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n"); return -EINVAL; } - dma_params->acnt = dma_params->data_type; - rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); - xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); + if (params_channels(params) == 2) { + element_cnt = 2; + if (double_fmt[fmt] && dev->enable_channel_combine) { + element_cnt = 1; + fmt = double_fmt[fmt]; + } + } + dma_params->acnt = dma_params->data_type = data_type[fmt]; + dma_params->fifo_level = 0; + mcbsp_word_length = asp_word_length[fmt]; + rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1); + xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1); rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length); @@ -513,7 +545,13 @@ static int davinci_i2s_probe(struct platform_device *pdev) ret = -ENOMEM; goto err_release_region; } - + if (pdata) { + dev->enable_channel_combine = pdata->enable_channel_combine; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].sram_size = + pdata->sram_size_playback; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].sram_size = + pdata->sram_size_capture; + } dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) { ret = -ENODEV; @@ -547,6 +585,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; + davinci_i2s_dai.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 5d1f98a4c97..0a302e1080d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -714,16 +714,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = &dev->dma_params[substream->stream]; int word_length; - u8 numevt; + u8 fifo_level; davinci_hw_common_param(dev, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - numevt = dev->txnumevt; + fifo_level = dev->txnumevt; else - numevt = dev->rxnumevt; - - if (!numevt) - numevt = 1; + fifo_level = dev->rxnumevt; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -751,12 +748,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (dev->version == MCASP_VERSION_2) { - dma_params->data_type *= numevt; - dma_params->acnt = 4 * numevt; - } else + if (dev->version == MCASP_VERSION_2 && !fifo_level) + dma_params->acnt = 4; + else dma_params->acnt = dma_params->data_type; + dma_params->fifo_level = fifo_level; davinci_config_channel_size(dev, word_length); return 0; @@ -907,6 +904,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 9d179cc88f7..582c9249ef0 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -39,11 +39,6 @@ enum { }; struct davinci_audio_dev { - /* - * dma_params must be first because rtd->dai->cpu_dai->private_data - * is cast to a pointer of an array of struct davinci_pcm_dma_params in - * davinci_pcm_open. - */ struct davinci_pcm_dma_params dma_params[2]; void __iomem *base; int sample_rate; diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c73a915f233..ad4d7f47a86 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -3,6 +3,7 @@ * * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> + * added SRAM ping/pong (C) 2008 Troy Kisky <troy.kisky@boundarydevices.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as @@ -23,10 +24,29 @@ #include <asm/dma.h> #include <mach/edma.h> +#include <mach/sram.h> #include "davinci-pcm.h" -static struct snd_pcm_hardware davinci_pcm_hardware = { +#ifdef DEBUG +static void print_buf_info(int slot, char *name) +{ + struct edmacc_param p; + if (slot < 0) + return; + edma_read_slot(slot, &p); + printk(KERN_DEBUG "%s: 0x%x, opt=%x, src=%x, a_b_cnt=%x dst=%x\n", + name, slot, p.opt, p.src, p.a_b_cnt, p.dst); + printk(KERN_DEBUG " src_dst_bidx=%x link_bcntrld=%x src_dst_cidx=%x ccnt=%x\n", + p.src_dst_bidx, p.link_bcntrld, p.src_dst_cidx, p.ccnt); +} +#else +static void print_buf_info(int slot, char *name) +{ +} +#endif + +static struct snd_pcm_hardware pcm_hardware_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE), @@ -48,102 +68,432 @@ static struct snd_pcm_hardware davinci_pcm_hardware = { .fifo_size = 0, }; +static struct snd_pcm_hardware pcm_hardware_capture = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +/* + * How ping/pong works.... + * + * Playback: + * ram_params - copys 2*ping_size from start of SDRAM to iram, + * links to ram_link2 + * ram_link2 - copys rest of SDRAM to iram in ping_size units, + * links to ram_link + * ram_link - copys entire SDRAM to iram in ping_size uints, + * links to self + * + * asp_params - same as asp_link[0] + * asp_link[0] - copys from lower half of iram to asp port + * links to asp_link[1], triggers iram copy event on completion + * asp_link[1] - copys from upper half of iram to asp port + * links to asp_link[0], triggers iram copy event on completion + * triggers interrupt only needed to let upper SOC levels update position + * in stream on completion + * + * When playback is started: + * ram_params started + * asp_params started + * + * Capture: + * ram_params - same as ram_link, + * links to ram_link + * ram_link - same as playback + * links to self + * + * asp_params - same as playback + * asp_link[0] - same as playback + * asp_link[1] - same as playback + * + * When capture is started: + * asp_params started + */ struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ - int master_lch; /* Master DMA channel */ - int slave_lch; /* linked parameter RAM reload slot */ + int asp_channel; /* Master DMA channel */ + int asp_link[2]; /* asp parameter link channel, ping/pong */ struct davinci_pcm_dma_params *params; /* DMA params */ + int ram_channel; + int ram_link; + int ram_link2; + struct edmacc_param asp_params; + struct edmacc_param ram_params; }; +/* + * Not used with ping/pong + */ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int lch = prtd->slave_lch; + int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; + unsigned short src_cidx, dst_cidx; unsigned int data_type; unsigned short acnt; unsigned int count; + unsigned int fifo_level; period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; + fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; + if (fifo_level) + count /= fifo_level; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { src = dma_pos; dst = prtd->params->dma_addr; src_bidx = data_type; dst_bidx = 0; + src_cidx = data_type * fifo_level; + dst_cidx = 0; } else { src = prtd->params->dma_addr; dst = dma_pos; src_bidx = 0; dst_bidx = data_type; + src_cidx = 0; + dst_cidx = data_type * fifo_level; } acnt = prtd->params->acnt; - edma_set_src(lch, src, INCR, W8BIT); - edma_set_dest(lch, dst, INCR, W8BIT); - edma_set_src_index(lch, src_bidx, 0); - edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + edma_set_src(link, src, INCR, W8BIT); + edma_set_dest(link, dst, INCR, W8BIT); + + edma_set_src_index(link, src_bidx, src_cidx); + edma_set_dest_index(link, dst_bidx, dst_cidx); + + if (!fifo_level) + edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); + else + edma_set_transfer_params(link, acnt, fifo_level, count, + fifo_level, ABSYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) prtd->period = 0; } -static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; - pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status); + print_buf_info(prtd->ram_channel, "i ram_channel"); + pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; if (snd_pcm_running(substream)) { + if (prtd->ram_channel < 0) { + /* No ping/pong must fix up link dma data*/ + spin_lock(&prtd->lock); + davinci_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } snd_pcm_period_elapsed(substream); + } +} + +static int allocate_sram(struct snd_pcm_substream *substream, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + struct snd_dma_buffer *iram_dma = NULL; + dma_addr_t iram_phys = 0; + void *iram_virt = NULL; + + if (buf->private_data || !size) + return 0; + + ppcm->period_bytes_max = size; + iram_virt = sram_alloc(size, &iram_phys); + if (!iram_virt) + goto exit1; + iram_dma = kzalloc(sizeof(*iram_dma), GFP_KERNEL); + if (!iram_dma) + goto exit2; + iram_dma->area = iram_virt; + iram_dma->addr = iram_phys; + memset(iram_dma->area, 0, size); + iram_dma->bytes = size; + buf->private_data = iram_dma; + return 0; +exit2: + if (iram_virt) + sram_free(iram_virt, size); +exit1: + return -ENOMEM; +} + +/* + * Only used with ping/pong. + * This is called after runtime->dma_addr, period_bytes and data_type are valid + */ +static int ping_pong_dma_setup(struct snd_pcm_substream *substream) +{ + unsigned short ram_src_cidx, ram_dst_cidx; + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + struct snd_dma_buffer *iram_dma = + (struct snd_dma_buffer *)substream->dma_buffer.private_data; + struct davinci_pcm_dma_params *params = prtd->params; + unsigned int data_type = params->data_type; + unsigned int acnt = params->acnt; + /* divide by 2 for ping/pong */ + unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; + int link = prtd->asp_link[1]; + unsigned int fifo_level = prtd->params->fifo_level; + unsigned int count; + if ((data_type == 0) || (data_type > 4)) { + printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type); + return -EINVAL; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_addr_t asp_src_pong = iram_dma->addr + ping_size; + ram_src_cidx = ping_size; + ram_dst_cidx = -ping_size; + edma_set_src(link, asp_src_pong, INCR, W8BIT); + + link = prtd->asp_link[0]; + edma_set_src_index(link, data_type, data_type * fifo_level); + link = prtd->asp_link[1]; + edma_set_src_index(link, data_type, data_type * fifo_level); + + link = prtd->ram_link; + edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + } else { + dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; + ram_src_cidx = -ping_size; + ram_dst_cidx = ping_size; + edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + + link = prtd->asp_link[0]; + edma_set_dest_index(link, data_type, data_type * fifo_level); + link = prtd->asp_link[1]; + edma_set_dest_index(link, data_type, data_type * fifo_level); + + link = prtd->ram_link; + edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + } - spin_lock(&prtd->lock); - davinci_pcm_enqueue_dma(substream); - spin_unlock(&prtd->lock); + if (!fifo_level) { + count = ping_size / data_type; + edma_set_transfer_params(prtd->asp_link[0], acnt, count, + 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, count, + 1, 0, ASYNC); + } else { + count = ping_size / (data_type * fifo_level); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[1], acnt, fifo_level, + count, fifo_level, ABSYNC); } + + link = prtd->ram_link; + edma_set_src_index(link, ping_size, ram_src_cidx); + edma_set_dest_index(link, ping_size, ram_dst_cidx); + edma_set_transfer_params(link, ping_size, 2, + runtime->periods, 2, ASYNC); + + /* init master params */ + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_read_slot(prtd->ram_link, &prtd->ram_params); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + struct edmacc_param p_ram; + /* Copy entire iram buffer before playback started */ + prtd->ram_params.a_b_cnt = (1 << 16) | (ping_size << 1); + /* 0 dst_bidx */ + prtd->ram_params.src_dst_bidx = (ping_size << 1); + /* 0 dst_cidx */ + prtd->ram_params.src_dst_cidx = (ping_size << 1); + prtd->ram_params.ccnt = 1; + + /* Skip 1st period */ + edma_read_slot(prtd->ram_link, &p_ram); + p_ram.src += (ping_size << 1); + p_ram.ccnt -= 1; + edma_write_slot(prtd->ram_link2, &p_ram); + /* + * When 1st started, ram -> iram dma channel will fill the + * entire iram. Then, whenever a ping/pong asp buffer finishes, + * 1/2 iram will be filled. + */ + prtd->ram_params.link_bcntrld = + EDMA_CHAN_SLOT(prtd->ram_link2) << 5; + } + return 0; +} + +/* 1 asp tx or rx channel using 2 parameter channels + * 1 ram to/from iram channel using 1 parameter channel + * + * Playback + * ram copy channel kicks off first, + * 1st ram copy of entire iram buffer completion kicks off asp channel + * asp tcc always kicks off ram copy of 1/2 iram buffer + * + * Record + * asp channel starts, tcc kicks off ram copy + */ +static int request_ping_pong(struct snd_pcm_substream *substream, + struct davinci_runtime_data *prtd, + struct snd_dma_buffer *iram_dma) +{ + dma_addr_t asp_src_ping; + dma_addr_t asp_dst_ping; + int link; + struct davinci_pcm_dma_params *params = prtd->params; + + /* Request ram master channel */ + link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + davinci_pcm_dma_irq, substream, + EVENTQ_1); + if (link < 0) + goto exit1; + + /* Request ram link channel */ + link = prtd->ram_link = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit2; + + link = prtd->asp_link[1] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit3; + + prtd->ram_link2 = -1; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + link = prtd->ram_link2 = edma_alloc_slot( + EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit4; + } + /* circle ping-pong buffers */ + edma_link(prtd->asp_link[0], prtd->asp_link[1]); + edma_link(prtd->asp_link[1], prtd->asp_link[0]); + /* circle ram buffers */ + edma_link(prtd->ram_link, prtd->ram_link); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + asp_src_ping = iram_dma->addr; + asp_dst_ping = params->dma_addr; /* fifo */ + } else { + asp_src_ping = params->dma_addr; /* fifo */ + asp_dst_ping = iram_dma->addr; + } + /* ping */ + link = prtd->asp_link[0]; + edma_set_src(link, asp_src_ping, INCR, W16BIT); + edma_set_dest(link, asp_dst_ping, INCR, W16BIT); + edma_set_src_index(link, 0, 0); + edma_set_dest_index(link, 0, 0); + + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); + prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); + edma_write_slot(link, &prtd->asp_params); + + /* pong */ + link = prtd->asp_link[1]; + edma_set_src(link, asp_src_ping, INCR, W16BIT); + edma_set_dest(link, asp_dst_ping, INCR, W16BIT); + edma_set_src_index(link, 0, 0); + edma_set_dest_index(link, 0, 0); + + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); + /* interrupt after every pong completion */ + prtd->asp_params.opt |= TCINTEN | TCCHEN | + EDMA_TCC(EDMA_CHAN_SLOT(prtd->ram_channel)); + edma_write_slot(link, &prtd->asp_params); + + /* ram */ + link = prtd->ram_link; + edma_set_src(link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," + "for asp:%u %u %u\n", __func__, + prtd->ram_channel, prtd->ram_link, prtd->ram_link2, + prtd->asp_channel, prtd->asp_link[0], + prtd->asp_link[1]); + return 0; +exit4: + edma_free_channel(prtd->asp_link[1]); + prtd->asp_link[1] = -1; +exit3: + edma_free_channel(prtd->ram_link); + prtd->ram_link = -1; +exit2: + edma_free_channel(prtd->ram_channel); + prtd->ram_channel = -1; +exit1: + return link; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) { + struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct edmacc_param p_ram; - int ret; + struct davinci_pcm_dma_params *params = prtd->params; + int link; - /* Request master DMA channel */ - ret = edma_alloc_channel(prtd->params->channel, - davinci_pcm_dma_irq, substream, - EVENTQ_0); - if (ret < 0) - return ret; - prtd->master_lch = ret; + if (!params) + return -ENODEV; - /* Request parameter RAM reload slot */ - ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY); - if (ret < 0) { - edma_free_channel(prtd->master_lch); - return ret; + /* Request asp master DMA channel */ + link = prtd->asp_channel = edma_alloc_channel(params->channel, + davinci_pcm_dma_irq, substream, EVENTQ_0); + if (link < 0) + goto exit1; + + /* Request asp link channels */ + link = prtd->asp_link[0] = edma_alloc_slot( + EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); + if (link < 0) + goto exit2; + + iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; + if (iram_dma) { + if (request_ping_pong(substream, prtd, iram_dma) == 0) + return 0; + printk(KERN_WARNING "%s: dma channel allocation failed," + "not using sram\n", __func__); } - prtd->slave_lch = ret; /* Issue transfer completion IRQ when the channel completes a * transfer, then always reload from the same slot (by a kind @@ -154,12 +504,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(prtd->slave_lch, &p_ram); - p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch)); - p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5; - edma_write_slot(prtd->slave_lch, &p_ram); - + edma_read_slot(link, &prtd->asp_params); + prtd->asp_params.opt |= TCINTEN | + EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; + edma_write_slot(link, &prtd->asp_params); return 0; +exit2: + edma_free_channel(prtd->asp_channel); + prtd->asp_channel = -1; +exit1: + return link; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -173,12 +528,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_start(prtd->master_lch); + edma_resume(prtd->asp_channel); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_stop(prtd->master_lch); + edma_pause(prtd->asp_channel); break; default: ret = -EINVAL; @@ -193,15 +548,37 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) static int davinci_pcm_prepare(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; - struct edmacc_param temp; + if (prtd->ram_channel >= 0) { + int ret = ping_pong_dma_setup(substream); + if (ret < 0) + return ret; + + edma_write_slot(prtd->ram_channel, &prtd->ram_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); + + print_buf_info(prtd->ram_channel, "ram_channel"); + print_buf_info(prtd->ram_link, "ram_link"); + print_buf_info(prtd->ram_link2, "ram_link2"); + print_buf_info(prtd->asp_channel, "asp_channel"); + print_buf_info(prtd->asp_link[0], "asp_link[0]"); + print_buf_info(prtd->asp_link[1], "asp_link[1]"); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* copy 1st iram buffer */ + edma_start(prtd->ram_channel); + } + edma_start(prtd->asp_channel); + return 0; + } prtd->period = 0; davinci_pcm_enqueue_dma(substream); /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->slave_lch, &temp); - edma_write_slot(prtd->master_lch, &temp); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); + edma_write_slot(prtd->asp_channel, &prtd->asp_params); davinci_pcm_enqueue_dma(substream); + edma_start(prtd->asp_channel); return 0; } @@ -212,20 +589,53 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; unsigned int offset; - dma_addr_t count; - dma_addr_t src, dst; + int asp_count; + dma_addr_t asp_src, asp_dst; spin_lock(&prtd->lock); - - edma_get_position(prtd->master_lch, &src, &dst); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - count = src - runtime->dma_addr; - else - count = dst - runtime->dma_addr; - + if (prtd->ram_channel >= 0) { + int ram_count; + int mod_ram; + dma_addr_t ram_src, ram_dst; + unsigned int period_size = snd_pcm_lib_period_bytes(substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* reading ram before asp should be safe + * as long as the asp transfers less than a ping size + * of bytes between the 2 reads + */ + edma_get_position(prtd->ram_channel, + &ram_src, &ram_dst); + edma_get_position(prtd->asp_channel, + &asp_src, &asp_dst); + asp_count = asp_src - prtd->asp_params.src; + ram_count = ram_src - prtd->ram_params.src; + mod_ram = ram_count % period_size; + mod_ram -= asp_count; + if (mod_ram < 0) + mod_ram += period_size; + else if (mod_ram == 0) { + if (snd_pcm_running(substream)) + mod_ram += period_size; + } + ram_count -= mod_ram; + if (ram_count < 0) + ram_count += period_size * runtime->periods; + } else { + edma_get_position(prtd->ram_channel, + &ram_src, &ram_dst); + ram_count = ram_dst - prtd->ram_params.dst; + } + asp_count = ram_count; + } else { + edma_get_position(prtd->asp_channel, &asp_src, &asp_dst); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + asp_count = asp_src - runtime->dma_addr; + else + asp_count = asp_dst - runtime->dma_addr; + } spin_unlock(&prtd->lock); - offset = bytes_to_frames(runtime, count); + offset = bytes_to_frames(runtime, asp_count); if (offset >= runtime->buffer_size) offset = 0; @@ -236,14 +646,19 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd; + struct snd_pcm_hardware *ppcm; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->private_data; - struct davinci_pcm_dma_params *params = &pa[substream->stream]; - if (!params) + struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *params; + if (!pa) return -ENODEV; + params = &pa[substream->stream]; - snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + ppcm = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + &pcm_hardware_playback : &pcm_hardware_capture; + allocate_sram(substream, params->sram_size, ppcm); + snd_soc_set_runtime_hwparams(substream, ppcm); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -256,6 +671,11 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) spin_lock_init(&prtd->lock); prtd->params = params; + prtd->asp_channel = -1; + prtd->asp_link[0] = prtd->asp_link[1] = -1; + prtd->ram_channel = -1; + prtd->ram_link = -1; + prtd->ram_link2 = -1; runtime->private_data = prtd; @@ -273,10 +693,29 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - edma_unlink(prtd->slave_lch); - - edma_free_slot(prtd->slave_lch); - edma_free_channel(prtd->master_lch); + if (prtd->ram_channel >= 0) + edma_stop(prtd->ram_channel); + if (prtd->asp_channel >= 0) + edma_stop(prtd->asp_channel); + if (prtd->asp_link[0] >= 0) + edma_unlink(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_unlink(prtd->asp_link[1]); + if (prtd->ram_link >= 0) + edma_unlink(prtd->ram_link); + + if (prtd->asp_link[0] >= 0) + edma_free_slot(prtd->asp_link[0]); + if (prtd->asp_link[1] >= 0) + edma_free_slot(prtd->asp_link[1]); + if (prtd->asp_channel >= 0) + edma_free_channel(prtd->asp_channel); + if (prtd->ram_link >= 0) + edma_free_slot(prtd->ram_link); + if (prtd->ram_link2 >= 0) + edma_free_slot(prtd->ram_link2); + if (prtd->ram_channel >= 0) + edma_free_channel(prtd->ram_channel); kfree(prtd); @@ -318,11 +757,11 @@ static struct snd_pcm_ops davinci_pcm_ops = { .mmap = davinci_pcm_mmap, }; -static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, + size_t size) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = davinci_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -347,6 +786,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) int stream; for (stream = 0; stream < 2; stream++) { + struct snd_dma_buffer *iram_dma; substream = pcm->streams[stream].substream; if (!substream) continue; @@ -358,6 +798,11 @@ static void davinci_pcm_free(struct snd_pcm *pcm) dma_free_writecombine(pcm->card->dev, buf->bytes, buf->area, buf->addr); buf->area = NULL; + iram_dma = (struct snd_dma_buffer *)buf->private_data; + if (iram_dma) { + sram_free(iram_dma->area, iram_dma->bytes); + kfree(iram_dma); + } } } @@ -375,14 +820,16 @@ static int davinci_pcm_new(struct snd_card *card, if (dai->playback.channels_min) { ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); + SNDRV_PCM_STREAM_PLAYBACK, + pcm_hardware_playback.buffer_bytes_max); if (ret) return ret; } if (dai->capture.channels_min) { ret = davinci_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); + SNDRV_PCM_STREAM_CAPTURE, + pcm_hardware_capture.buffer_bytes_max); if (ret) return ret; } diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index 8746606efc8..0764944cf10 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -20,9 +20,11 @@ struct davinci_pcm_dma_params { int channel; /* sync dma channel ID */ unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned sram_size; enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ unsigned char convert_mono_stereo; + unsigned int fifo_level; }; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 6096d22283e..30ed568afb2 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -58,47 +58,15 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) /* Prepare and enqueue the next buffer descriptor */ bd = bcom_prepare_next_buffer(s->bcom_task); bd->status = s->period_bytes; - bd->data[0] = s->period_next_pt; + bd->data[0] = s->runtime->dma_addr + (s->period_next * s->period_bytes); bcom_submit_next_buffer(s->bcom_task, NULL); /* Update for next period */ - s->period_next_pt += s->period_bytes; - if (s->period_next_pt >= s->period_end) - s->period_next_pt = s->period_start; -} - -static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) -{ - if (s->appl_ptr > s->runtime->control->appl_ptr) { - /* - * In this case s->runtime->control->appl_ptr has wrapped around. - * Play the data to the end of the boundary, then wrap our own - * appl_ptr back around. - */ - while (s->appl_ptr < s->runtime->boundary) { - if (bcom_queue_full(s->bcom_task)) - return; - - s->appl_ptr += s->period_size; - - psc_dma_bcom_enqueue_next_buffer(s); - } - s->appl_ptr -= s->runtime->boundary; - } - - while (s->appl_ptr < s->runtime->control->appl_ptr) { - - if (bcom_queue_full(s->bcom_task)) - return; - - s->appl_ptr += s->period_size; - - psc_dma_bcom_enqueue_next_buffer(s); - } + s->period_next = (s->period_next + 1) % s->runtime->periods; } /* Bestcomm DMA irq handler */ -static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) +static irqreturn_t psc_dma_bcom_irq(int irq, void *_psc_dma_stream) { struct psc_dma_stream *s = _psc_dma_stream; @@ -108,34 +76,8 @@ static irqreturn_t psc_dma_bcom_irq_tx(int irq, void *_psc_dma_stream) while (bcom_buffer_done(s->bcom_task)) { bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - s->period_current_pt += s->period_bytes; - if (s->period_current_pt >= s->period_end) - s->period_current_pt = s->period_start; - } - psc_dma_bcom_enqueue_tx(s); - spin_unlock(&s->psc_dma->lock); - - /* If the stream is active, then also inform the PCM middle layer - * of the period finished event. */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - return IRQ_HANDLED; -} - -static irqreturn_t psc_dma_bcom_irq_rx(int irq, void *_psc_dma_stream) -{ - struct psc_dma_stream *s = _psc_dma_stream; - - spin_lock(&s->psc_dma->lock); - /* For each finished period, dequeue the completed period buffer - * and enqueue a new one in it's place. */ - while (bcom_buffer_done(s->bcom_task)) { - bcom_retrieve_buffer(s->bcom_task, NULL, NULL); - - s->period_current_pt += s->period_bytes; - if (s->period_current_pt >= s->period_end) - s->period_current_pt = s->period_start; + s->period_current = (s->period_current+1) % s->runtime->periods; + s->period_count++; psc_dma_bcom_enqueue_next_buffer(s); } @@ -166,54 +108,38 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - struct psc_dma_stream *s; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs; u16 imr; unsigned long flags; int i; - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - s = &psc_dma->capture; - else - s = &psc_dma->playback; - - dev_dbg(psc_dma->dev, "psc_dma_trigger(substream=%p, cmd=%i)" - " stream_id=%i\n", - substream, cmd, substream->pstr->stream); - switch (cmd) { case SNDRV_PCM_TRIGGER_START: + dev_dbg(psc_dma->dev, "START: stream=%i fbits=%u ps=%u #p=%u\n", + substream->pstr->stream, runtime->frame_bits, + (int)runtime->period_size, runtime->periods); s->period_bytes = frames_to_bytes(runtime, runtime->period_size); - s->period_start = virt_to_phys(runtime->dma_area); - s->period_end = s->period_start + - (s->period_bytes * runtime->periods); - s->period_next_pt = s->period_start; - s->period_current_pt = s->period_start; - s->period_size = runtime->period_size; + s->period_next = 0; + s->period_current = 0; s->active = 1; - - /* track appl_ptr so that we have a better chance of detecting - * end of stream and not over running it. - */ + s->period_count = 0; s->runtime = runtime; - s->appl_ptr = s->runtime->control->appl_ptr - - (runtime->period_size * runtime->periods); /* Fill up the bestcomm bd queue and enable DMA. * This will begin filling the PSC's fifo. */ spin_lock_irqsave(&psc_dma->lock, flags); - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) bcom_gen_bd_rx_reset(s->bcom_task); - for (i = 0; i < runtime->periods; i++) - if (!bcom_queue_full(s->bcom_task)) - psc_dma_bcom_enqueue_next_buffer(s); - } else { + else bcom_gen_bd_tx_reset(s->bcom_task); - psc_dma_bcom_enqueue_tx(s); - } + + for (i = 0; i < runtime->periods; i++) + if (!bcom_queue_full(s->bcom_task)) + psc_dma_bcom_enqueue_next_buffer(s); bcom_enable(s->bcom_task); spin_unlock_irqrestore(&psc_dma->lock, flags); @@ -223,6 +149,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_STOP: + dev_dbg(psc_dma->dev, "STOP: stream=%i periods_count=%i\n", + substream->pstr->stream, s->period_count); s->active = 0; spin_lock_irqsave(&psc_dma->lock, flags); @@ -236,7 +164,8 @@ static int psc_dma_trigger(struct snd_pcm_substream *substream, int cmd) break; default: - dev_dbg(psc_dma->dev, "invalid command\n"); + dev_dbg(psc_dma->dev, "unhandled trigger: stream=%i cmd=%i\n", + substream->pstr->stream, cmd); return -EINVAL; } @@ -343,7 +272,7 @@ psc_dma_pointer(struct snd_pcm_substream *substream) else s = &psc_dma->playback; - count = s->period_current_pt - s->period_start; + count = s->period_current * s->period_bytes; return bytes_to_frames(substream->runtime, count); } @@ -532,11 +461,9 @@ int mpc5200_audio_dma_create(struct of_device *op) rc = request_irq(psc_dma->irq, &psc_dma_status_irq, IRQF_SHARED, "psc-dma-status", psc_dma); - rc |= request_irq(psc_dma->capture.irq, - &psc_dma_bcom_irq_rx, IRQF_SHARED, + rc |= request_irq(psc_dma->capture.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-capture", &psc_dma->capture); - rc |= request_irq(psc_dma->playback.irq, - &psc_dma_bcom_irq_tx, IRQF_SHARED, + rc |= request_irq(psc_dma->playback.irq, &psc_dma_bcom_irq, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { ret = -ENODEV; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 8d396bb9d9f..22208b373fb 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -13,26 +13,25 @@ * @psc_dma: pointer back to parent psc_dma data structure * @bcom_task: bestcomm task structure * @irq: irq number for bestcomm task - * @period_start: physical address of start of DMA region * @period_end: physical address of end of DMA region * @period_next_pt: physical address of next DMA buffer to enqueue * @period_bytes: size of DMA period in bytes + * @ac97_slot_bits: Enable bits for turning on the correct AC97 slot */ struct psc_dma_stream { struct snd_pcm_runtime *runtime; - snd_pcm_uframes_t appl_ptr; - int active; struct psc_dma *psc_dma; struct bcom_task *bcom_task; int irq; struct snd_pcm_substream *stream; - dma_addr_t period_start; - dma_addr_t period_end; - dma_addr_t period_next_pt; - dma_addr_t period_current_pt; + int period_next; + int period_current; int period_bytes; - int period_size; + int period_count; + + /* AC97 state */ + u32 ac97_slot_bits; }; /** @@ -73,6 +72,15 @@ struct psc_dma { } stats; }; +/* Utility for retrieving psc_dma_stream structure from a substream */ +inline struct psc_dma_stream * +to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma) +{ + if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) + return &psc_dma->capture; + return &psc_dma->playback; +} + int mpc5200_audio_dma_create(struct of_device *op); int mpc5200_audio_dma_destroy(struct of_device *op); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index c4ae3e096bb..3dbc7f7cd7b 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -130,6 +130,7 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct psc_dma *psc_dma = cpu_dai->private_data; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i" " periods=%i buffer_size=%i buffer_bytes=%i channels=%i" @@ -140,20 +141,10 @@ static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream, params_channels(params), params_rate(params), params_format(params)); - - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (params_channels(params) == 1) - psc_dma->slots |= 0x00000100; - else - psc_dma->slots |= 0x00000300; - } else { - if (params_channels(params) == 1) - psc_dma->slots |= 0x01000000; - else - psc_dma->slots |= 0x03000000; - } - out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); - + /* Determine the set of enable bits to turn on */ + s->ac97_slot_bits = (params_channels(params) == 1) ? 0x100 : 0x300; + if (substream->pstr->stream != SNDRV_PCM_STREAM_CAPTURE) + s->ac97_slot_bits <<= 16; return 0; } @@ -163,6 +154,8 @@ static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream, { struct psc_dma *psc_dma = cpu_dai->private_data; + dev_dbg(psc_dma->dev, "%s(substream=%p)\n", __func__, substream); + if (params_channels(params) == 1) out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000); else @@ -176,14 +169,24 @@ static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data; + struct psc_dma_stream *s = to_psc_dma_stream(substream, psc_dma); switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + dev_dbg(psc_dma->dev, "AC97 START: stream=%i\n", + substream->pstr->stream); + + /* Set the slot enable bits */ + psc_dma->slots |= s->ac97_slot_bits; + out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); + break; + case SNDRV_PCM_TRIGGER_STOP: - if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) - psc_dma->slots &= 0xFFFF0000; - else - psc_dma->slots &= 0x0000FFFF; + dev_dbg(psc_dma->dev, "AC97 STOP: stream=%i\n", + substream->pstr->stream); + /* Clear the slot enable bits */ + psc_dma->slots &= ~(s->ac97_slot_bits); out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots); break; } diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c index e4dcb539108..0267d2d9168 100644 --- a/sound/soc/imx/mx27vis_wm8974.c +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -157,7 +157,7 @@ static int mx27vis_hifi_hw_params(struct snd_pcm_substream *substream, /* codec PLL input is 25 MHz */ - ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, + ret = codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, 25000000, pll_out); if (ret < 0) { printk(KERN_ERR "Error when setting PLL input\n"); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 653a362425d..61952aa6cd5 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -43,12 +43,13 @@ config SND_OMAP_SOC_OSK5912 Say Y if you want to add support for SoC audio on osk5912. config SND_OMAP_SOC_OVERO - tristate "SoC Audio support for Gumstix Overo" - depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO + tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35" + depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35) select SND_OMAP_SOC_MCBSP select SND_SOC_TWL4030 help - Say Y if you want to add support for SoC audio on the Gumstix Overo. + Say Y if you want to add support for SoC audio on the + Gumstix Overo or CompuLab CM-T35 config SND_OMAP_SOC_OMAP2EVM tristate "SoC Audio support for OMAP2EVM board" @@ -66,6 +67,15 @@ config SND_OMAP_SOC_OMAP3EVM help Say Y if you want to add support for SoC audio on the omap3evm board. +config SND_OMAP_SOC_AM3517EVM + tristate "SoC Audio support for OMAP3517 / AM3517 EVM" + depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C + select SND_OMAP_SOC_MCBSP + select SND_SOC_TLV320AIC23 + help + Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 + EVM. + config SND_OMAP_SOC_SDP3430 tristate "SoC Audio support for Texas Instruments SDP3430" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP @@ -99,3 +109,10 @@ config SND_OMAP_SOC_ZOOM2 help Say Y if you want to add support for Soc audio on Zoom2 board. +config SND_OMAP_SOC_IGEP0020 + tristate "SoC Audio support for IGEP v2" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020 + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for Soc audio on IGEP v2 board. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 02d69471dcb..d49458a29bb 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -12,10 +12,12 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o +snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o +snd-soc-igep0020-objs := igep0020.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o @@ -23,7 +25,9 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o +obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c new file mode 100644 index 00000000000..135901b2ea1 --- /dev/null +++ b/sound/soc/omap/am3517evm.c @@ -0,0 +1,202 @@ +/* + * am3517evm.c -- ALSA SoC support for OMAP3517 / AM3517 EVM + * + * Author: Anuj Aggarwal <anuj.aggarwal@ti.com> + * + * Based on sound/soc/omap/beagle.c by Steve Sakoman + * + * Copyright (C) 2009 Texas Instruments Incorporated + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation version 2. + * + * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind, + * whether express or implied; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" + +#include "../codecs/tlv320aic23.h" + +#define CODEC_CLOCK 12000000 + +static int am3517evm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n"); + return ret; + } + + snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops am3517evm_ops = { + .hw_params = am3517evm_hw_params, +}; + +/* am3517evm machine dapm widgets */ +static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { + SND_SOC_DAPM_HP("Line Out", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_MIC("Mic In", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LOUT"}, + {"Line Out", NULL, "ROUT"}, + + {"LLINEIN", NULL, "Line In"}, + {"RLINEIN", NULL, "Line In"}, + + {"MICIN", NULL, "Mic In"}, +}; + +static int am3517evm_aic23_init(struct snd_soc_codec *codec) +{ + /* Add am3517-evm specific widgets */ + snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, + ARRAY_SIZE(tlv320aic23_dapm_widgets)); + + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Mic In"); + + snd_soc_dapm_sync(codec); + + return 0; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link am3517evm_dai = { + .name = "TLV320AIC23", + .stream_name = "AIC23", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &tlv320aic23_dai, + .init = am3517evm_aic23_init, + .ops = &am3517evm_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_am3517evm = { + .name = "am3517evm", + .platform = &omap_soc_platform, + .dai_link = &am3517evm_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device am3517evm_snd_devdata = { + .card = &snd_soc_am3517evm, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static struct platform_device *am3517evm_snd_device; + +static int __init am3517evm_soc_init(void) +{ + int ret; + + if (!machine_is_omap3517evm()) { + pr_err("Not OMAP3517 / AM3517 EVM!\n"); + return -ENODEV; + } + pr_info("OMAP3517 / AM3517 EVM SoC init\n"); + + am3517evm_snd_device = platform_device_alloc("soc-audio", -1); + if (!am3517evm_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(am3517evm_snd_device, &am3517evm_snd_devdata); + am3517evm_snd_devdata.dev = &am3517evm_snd_device->dev; + *(unsigned int *)am3517evm_dai.cpu_dai->private_data = 0; /* McBSP1 */ + + ret = platform_device_add(am3517evm_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(am3517evm_snd_device); + + return ret; +} + +static void __exit am3517evm_soc_exit(void) +{ + platform_device_unregister(am3517evm_snd_device); +} + +module_init(am3517evm_soc_init); +module_exit(am3517evm_soc_exit); + +MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 5a5166ac727..ae0fc9b135d 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -40,7 +40,7 @@ /* Board specific DAPM widgets */ - const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { +static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = { /* Handset */ SND_SOC_DAPM_MIC("Mouthpiece", NULL), SND_SOC_DAPM_HP("Earpiece", NULL), @@ -81,7 +81,7 @@ static const char *ams_delta_audio_mode[] = (1 << AMS_DELTA_SPEAKER)) #define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC)) -unsigned short ams_delta_audio_mode_pins[] = { +static const unsigned short ams_delta_audio_mode_pins[] = { AMS_DELTA_MIXED, AMS_DELTA_HANDSET, AMS_DELTA_HANDSFREE, diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c new file mode 100644 index 00000000000..3583c429f9b --- /dev/null +++ b/sound/soc/omap/igep0020.c @@ -0,0 +1,148 @@ +/* + * igep0020.c -- SoC audio for IGEP v2 + * + * Based on sound/soc/omap/overo.c by Steve Sakoman + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/hardware.h> +#include <mach/gpio.h> +#include <plat/mcbsp.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +static int igep2_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + printk(KERN_ERR "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } + + return 0; +} + +static struct snd_soc_ops igep2_ops = { + .hw_params = igep2_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link igep2_dai = { + .name = "TWL4030", + .stream_name = "TWL4030", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], + .ops = &igep2_ops, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_igep2 = { + .name = "igep2", + .platform = &omap_soc_platform, + .dai_link = &igep2_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device igep2_snd_devdata = { + .card = &snd_soc_card_igep2, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *igep2_snd_device; + +static int __init igep2_soc_init(void) +{ + int ret; + + if (!machine_is_igep0020()) { + pr_debug("Not IGEP v2!\n"); + return -ENODEV; + } + printk(KERN_INFO "IGEP v2 SoC init\n"); + + igep2_snd_device = platform_device_alloc("soc-audio", -1); + if (!igep2_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(igep2_snd_device, &igep2_snd_devdata); + igep2_snd_devdata.dev = &igep2_snd_device->dev; + *(unsigned int *)igep2_dai.cpu_dai->private_data = 1; /* McBSP2 */ + + ret = platform_device_add(igep2_snd_device); + if (ret) + goto err1; + + return 0; + +err1: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(igep2_snd_device); + + return ret; +} +module_init(igep2_soc_init); + +static void __exit igep2_soc_exit(void) +{ + platform_device_unregister(igep2_snd_device); +} +module_exit(igep2_soc_exit); + +MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo@iseebcn.com>"); +MODULE_DESCRIPTION("ALSA SoC IGEP v2"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3341f49402c..45be94201c8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -49,6 +49,8 @@ struct omap_mcbsp_data { */ int active; int configured; + unsigned int in_freq; + int clk_div; }; #define to_mcbsp(priv) container_of((priv), struct omap_mcbsp_data, bus_id) @@ -257,7 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT; unsigned long port; - unsigned int format; + unsigned int format, div, framesize, master; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; @@ -294,28 +296,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; wpf = channels = params_channels(params); - switch (channels) { - case 2: - if (format == SND_SOC_DAIFMT_I2S) { - /* Use dual-phase frames */ - regs->rcr2 |= RPHASE; - regs->xcr2 |= XPHASE; - /* Set 1 word per (McBSP) frame for phase1 and phase2 */ - wpf--; - regs->rcr2 |= RFRLEN2(wpf - 1); - regs->xcr2 |= XFRLEN2(wpf - 1); - } - case 1: - case 4: - /* Set word per (McBSP) frame for phase1 */ - regs->rcr1 |= RFRLEN1(wpf - 1); - regs->xcr1 |= XFRLEN1(wpf - 1); - break; - default: - /* Unsupported number of channels */ - return -EINVAL; + if (channels == 2 && format == SND_SOC_DAIFMT_I2S) { + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + /* Set 1 word per (McBSP) frame for phase1 and phase2 */ + wpf--; + regs->rcr2 |= RFRLEN2(wpf - 1); + regs->xcr2 |= XFRLEN2(wpf - 1); } + regs->rcr1 |= RFRLEN1(wpf - 1); + regs->xcr1 |= XFRLEN1(wpf - 1); + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ @@ -330,15 +323,30 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* In McBSP master modes, FRAME (i.e. sample rate) is generated + * by _counting_ BCLKs. Calculate frame size in BCLKs */ + master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK; + if (master == SND_SOC_DAIFMT_CBS_CFS) { + div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1; + framesize = (mcbsp_data->in_freq / div) / params_rate(params); + + if (framesize < wlen * channels) { + printk(KERN_ERR "%s: not enough bandwidth for desired rate and " + "channels\n", __func__); + return -EINVAL; + } + } else + framesize = wlen * channels; + /* Set FS period and length in terms of bit clock periods */ switch (format) { case SND_SOC_DAIFMT_I2S: - regs->srgr2 |= FPER(wlen * channels - 1); - regs->srgr1 |= FWID(wlen - 1); + regs->srgr2 |= FPER(framesize - 1); + regs->srgr1 |= FWID((framesize >> 1) - 1); break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - regs->srgr2 |= FPER(wlen * channels - 1); + regs->srgr2 |= FPER(framesize - 1); regs->srgr1 |= FWID(0); break; } @@ -454,6 +462,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai, if (div_id != OMAP_MCBSP_CLKGDV) return -ENODEV; + mcbsp_data->clk_div = div; regs->srgr1 |= CLKGDV(div - 1); return 0; @@ -554,6 +563,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + mcbsp_data->in_freq = freq; + switch (clk_id) { case OMAP_MCBSP_SYSCLK_CLK: regs->srgr2 |= CLKSM; @@ -598,13 +609,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .id = (link_id), \ .playback = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ .capture = { \ .channels_min = 1, \ - .channels_max = 4, \ + .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ .formats = SNDRV_PCM_FMTBIT_S16_LE, \ }, \ diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 13aa380de16..f484dcd6340 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = { .num_links = 1, }; +/* twl4030 setup */ +static struct twl4030_setup_data twl4030_setup = { + .ramp_delay_value = 4, + .sysclk = 26000, +}; + /* Audio subsystem */ static struct snd_soc_device omap3evm_snd_devdata = { .card = &snd_soc_omap3evm, .codec_dev = &soc_codec_dev_twl4030, + .codec_data = &twl4030_setup, }; static struct platform_device *omap3evm_snd_device; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 0cd06f5dd35..71b2c161158 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -40,9 +40,12 @@ #define PREFIX "ASoC omap3pandora: " -static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, - struct snd_soc_dai *cpu_dai, unsigned int fmt) +static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, unsigned int fmt) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; /* Set codec DAI configuration */ @@ -68,8 +71,9 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, } /* Set McBSP clock to external */ - ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, - SND_SOC_CLOCK_IN); + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, + 256 * params_rate(params), + SND_SOC_CLOCK_IN); if (ret < 0) { pr_err(PREFIX "can't set cpu system clock\n"); return ret; @@ -87,11 +91,7 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBS_CFS); @@ -100,11 +100,7 @@ static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - - return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + return omap3pandora_cmn_hw_params(substream, params, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index ec4f8fd8b3a..97a4d6308bd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -107,8 +107,8 @@ static int __init overo_soc_init(void) { int ret; - if (!machine_is_overo()) { - pr_debug("Not Overo!\n"); + if (!(machine_is_overo() || machine_is_cm_t35())) { + pr_debug("Incomatible machine!\n"); return -ENODEV; } printk(KERN_INFO "overo SoC init\n"); diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index dcb3181bb34..376e14a9c27 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -90,7 +90,8 @@ config SND_PXA2XX_SOC_E800 config SND_PXA2XX_SOC_EM_X270 tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300" - depends on SND_PXA2XX_SOC && MACH_EM_X270 + depends on SND_PXA2XX_SOC && (MACH_EM_X270 || MACH_EXEDA || \ + MACH_CM_X300) select SND_PXA2XX_SOC_AC97 select SND_SOC_WM9712 help @@ -117,6 +118,15 @@ config SND_SOC_ZYLONITE Say Y if you want to add support for SoC audio on the Marvell Zylonite reference platform. +config SND_SOC_RAUMFELD + tristate "SoC Audio support Raumfeld audio adapter" + depends on SND_PXA2XX_SOC && (MACH_RAUMFELD_SPEAKER || MACH_RAUMFELD_CONNECTOR) + select SND_PXA_SOC_SSP + select SND_SOC_CS4270 + select SND_SOC_AK4104 + help + Say Y if you want to add support for SoC audio on Raumfeld devices + config SND_PXA2XX_SOC_MAGICIAN tristate "SoC Audio support for HTC Magician" depends on SND_PXA2XX_SOC && MACH_MAGICIAN diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 6e096b48033..f3e08fd40ca 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -23,6 +23,7 @@ snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-imote2-objs := imote2.o +snd-soc-raumfeld-objs := raumfeld.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -37,3 +38,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o +obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 9f7c61e23da..4c8d99a8d38 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -213,7 +213,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, return ret; /* set SSP audio pll clock */ - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, acps); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d11a6d7e384..3bd7712f029 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -305,8 +305,8 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, /* * Configure the PLL frequency pxa27x and (afaik - pxa320 only) */ -static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; @@ -760,13 +760,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -780,13 +780,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -801,13 +801,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, @@ -822,13 +822,13 @@ struct snd_soc_dai pxa_ssp_dai[] = { .resume = pxa_ssp_resume, .playback = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, .capture = { .channels_min = 1, - .channels_max = 2, + .channels_max = 8, .rates = PXA_SSP_RATES, .formats = PXA_SSP_FORMATS, }, diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c new file mode 100644 index 00000000000..acfce1c0f1c --- /dev/null +++ b/sound/soc/pxa/raumfeld.c @@ -0,0 +1,335 @@ +/* + * raumfeld_audio.c -- SoC audio for Raumfeld audio devices + * + * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de> + * + * based on code from: + * + * Wolfson Microelectronics PLC. + * Openedhand Ltd. + * Liam Girdwood <lrg@slimlogic.co.uk> + * Richard Purdie <richard@openedhand.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "../codecs/cs4270.h" +#include "../codecs/ak4104.h" +#include "pxa2xx-pcm.h" +#include "pxa-ssp.h" + +#define GPIO_SPDIF_RESET (38) +#define GPIO_MCLK_RESET (111) +#define GPIO_CODEC_RESET (120) + +static struct i2c_client *max9486_client; +static struct i2c_board_info max9486_hwmon_info = { + I2C_BOARD_INFO("max9485", 0x63), +}; + +#define MAX9485_MCLK_FREQ_112896 0x22 +#define MAX9485_MCLK_FREQ_122880 0x23 + +static void set_max9485_clk(char clk) +{ + i2c_master_send(max9486_client, &clk, 1); +} + +static void raumfeld_enable_audio(bool en) +{ + if (en) { + gpio_set_value(GPIO_MCLK_RESET, 1); + + /* wait some time to let the clocks become stable */ + msleep(100); + + gpio_set_value(GPIO_SPDIF_RESET, 1); + gpio_set_value(GPIO_CODEC_RESET, 1); + } else { + gpio_set_value(GPIO_MCLK_RESET, 0); + gpio_set_value(GPIO_SPDIF_RESET, 0); + gpio_set_value(GPIO_CODEC_RESET, 0); + } +} + +/* CS4270 */ +static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + + return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); +} + +static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + unsigned int fmt, clk = 0; + int ret = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, clk, 0); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_cs4270_ops = { + .startup = raumfeld_cs4270_startup, + .hw_params = raumfeld_cs4270_hw_params, +}; + +static int raumfeld_line_suspend(struct platform_device *pdev, pm_message_t state) +{ + raumfeld_enable_audio(false); + return 0; +} + +static int raumfeld_line_resume(struct platform_device *pdev) +{ + raumfeld_enable_audio(true); + return 0; +} + +static struct snd_soc_dai_link raumfeld_line_dai = { + .name = "CS4270", + .stream_name = "CS4270", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP1], + .codec_dai = &cs4270_dai, + .ops = &raumfeld_cs4270_ops, +}; + +static struct snd_soc_card snd_soc_line_raumfeld = { + .name = "Raumfeld analog", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_line_dai, + .suspend_post = raumfeld_line_suspend, + .resume_pre = raumfeld_line_resume, + .num_links = 1, +}; + + +/* AK4104 */ + +static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int fmt, ret = 0, clk = 0; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + clk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; + } + + fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; + + /* setup the CODEC DAI */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* setup the CPU DAI */ + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, clk); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_DIV_SCR, 4); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops raumfeld_ak4104_ops = { + .hw_params = raumfeld_ak4104_hw_params, +}; + +static struct snd_soc_dai_link raumfeld_spdif_dai = { + .name = "ak4104", + .stream_name = "Playback", + .cpu_dai = &pxa_ssp_dai[PXA_DAI_SSP2], + .codec_dai = &ak4104_dai, + .ops = &raumfeld_ak4104_ops, +}; + +static struct snd_soc_card snd_soc_spdif_raumfeld = { + .name = "Raumfeld S/PDIF", + .platform = &pxa2xx_soc_platform, + .dai_link = &raumfeld_spdif_dai, + .num_links = 1 +}; + +/* raumfeld_audio audio subsystem */ +static struct snd_soc_device raumfeld_line_devdata = { + .card = &snd_soc_line_raumfeld, + .codec_dev = &soc_codec_device_cs4270, +}; + +static struct snd_soc_device raumfeld_spdif_devdata = { + .card = &snd_soc_spdif_raumfeld, + .codec_dev = &soc_codec_device_ak4104, +}; + +static struct platform_device *raumfeld_audio_line_device; +static struct platform_device *raumfeld_audio_spdif_device; + +static int __init raumfeld_audio_init(void) +{ + int ret; + + if (!machine_is_raumfeld_speaker() && + !machine_is_raumfeld_connector()) + return 0; + + max9486_client = i2c_new_device(i2c_get_adapter(0), + &max9486_hwmon_info); + + if (!max9486_client) + return -ENOMEM; + + set_max9485_clk(MAX9485_MCLK_FREQ_122880); + + /* LINE */ + raumfeld_audio_line_device = platform_device_alloc("soc-audio", 0); + if (!raumfeld_audio_line_device) + return -ENOMEM; + + platform_set_drvdata(raumfeld_audio_line_device, + &raumfeld_line_devdata); + raumfeld_line_devdata.dev = &raumfeld_audio_line_device->dev; + ret = platform_device_add(raumfeld_audio_line_device); + if (ret) + platform_device_put(raumfeld_audio_line_device); + + /* no S/PDIF on Speakers */ + if (machine_is_raumfeld_speaker()) + return ret; + + /* S/PDIF */ + raumfeld_audio_spdif_device = platform_device_alloc("soc-audio", 1); + if (!raumfeld_audio_spdif_device) { + platform_device_put(raumfeld_audio_line_device); + return -ENOMEM; + } + + platform_set_drvdata(raumfeld_audio_spdif_device, + &raumfeld_spdif_devdata); + raumfeld_spdif_devdata.dev = &raumfeld_audio_spdif_device->dev; + ret = platform_device_add(raumfeld_audio_spdif_device); + if (ret) { + platform_device_put(raumfeld_audio_line_device); + platform_device_put(raumfeld_audio_spdif_device); + } + + raumfeld_enable_audio(true); + + return ret; +} + +static void __exit raumfeld_audio_exit(void) +{ + raumfeld_enable_audio(false); + + platform_device_unregister(raumfeld_audio_line_device); + + if (machine_is_raumfeld_connector()) + platform_device_unregister(raumfeld_audio_spdif_device); + + i2c_unregister_device(max9486_client); + + gpio_free(GPIO_MCLK_RESET); + gpio_free(GPIO_CODEC_RESET); + gpio_free(GPIO_SPDIF_RESET); +} + +module_init(raumfeld_audio_init); +module_exit(raumfeld_audio_exit); + +/* Module information */ +MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); +MODULE_DESCRIPTION("Raumfeld audio SoC"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index 9a386b4c4ed..dd678ae2439 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -74,7 +74,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_codec *codec) { if (clk_pout) - snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0); + snd_soc_dai_set_pll(&codec->dai[0], 0, 0, + clk_get_rate(pout), 0); snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets, ARRAY_SIZE(zylonite_dapm_widgets)); @@ -128,7 +129,7 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, pll_out); + ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, pll_out); if (ret < 0) return ret; diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 923428fc1ad..b489f1ae103 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -24,6 +24,9 @@ config SND_S3C64XX_SOC_I2S select SND_S3C_I2SV2_SOC select S3C64XX_DMA +config SND_S3C_SOC_PCM + tristate + config SND_S3C2443_SOC_AC97 tristate select S3C2410_DMA @@ -56,6 +59,15 @@ config SND_S3C24XX_SOC_JIVE_WM8750 help Sat Y if you want to add support for SoC audio on the Jive. +config SND_S3C64XX_SOC_WM8580 + tristate "SoC I2S Audio support for WM8580 on SMDK64XX" + depends on SND_S3C24XX_SOC && (MACH_SMDK6400 || MACH_SMDK6410) + depends on BROKEN + select SND_SOC_WM8580 + select SND_S3C64XX_SOC_I2S + help + Sat Y if you want to add support for SoC audio on the SMDK64XX. + config SND_S3C24XX_SOC_SMDK2443_WM9710 tristate "SoC AC97 Audio support for SMDK2443 - WM9710" depends on SND_S3C24XX_SOC && MACH_SMDK2443 diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile index 99f5a7dd3fc..b744657733d 100644 --- a/sound/soc/s3c24xx/Makefile +++ b/sound/soc/s3c24xx/Makefile @@ -1,10 +1,11 @@ # S3c24XX Platform Support -snd-soc-s3c24xx-objs := s3c24xx-pcm.o +snd-soc-s3c24xx-objs := s3c-dma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o +snd-soc-s3c-pcm-objs := s3c-pcm.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o @@ -12,6 +13,7 @@ obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o +obj-$(CONFIG_SND_S3C_SOC_PCM) += snd-soc-s3c-pcm.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o @@ -23,6 +25,7 @@ snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o +snd-soc-smdk64xx-wm8580-objs := smdk64xx_wm8580.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,4 +36,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o +obj-$(CONFIG_SND_S3C64XX_SOC_WM8580) += snd-soc-smdk64xx-wm8580.o diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c index 93e6c87b739..59dc2c6b56d 100644 --- a/sound/soc/s3c24xx/jive_wm8750.c +++ b/sound/soc/s3c24xx/jive_wm8750.c @@ -25,7 +25,7 @@ #include <asm/mach-types.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #include "../codecs/wm8750.h" diff --git a/sound/soc/s3c24xx/ln2440sbc_alc650.c b/sound/soc/s3c24xx/ln2440sbc_alc650.c index 12c71482d25..d00d359a03e 100644 --- a/sound/soc/s3c24xx/ln2440sbc_alc650.c +++ b/sound/soc/s3c24xx/ln2440sbc_alc650.c @@ -24,7 +24,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card ln2440sbc; diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c index 0c52e36ddd8..dea83d30a5c 100644 --- a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c @@ -32,7 +32,7 @@ #include <asm/io.h> #include <mach/gta02.h> #include "../codecs/wm8753.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct snd_soc_card neo1973_gta02; @@ -119,7 +119,7 @@ static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream, return ret; /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -133,7 +133,7 @@ static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); } /* @@ -183,7 +183,7 @@ static int neo1973_gta02_voice_hw_params( return ret; /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -197,7 +197,7 @@ static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } static struct snd_soc_ops neo1973_gta02_voice_ops = { diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 906709e6dd5..0cb4f86f6d1 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -29,7 +29,6 @@ #include <mach/regs-clock.h> #include <mach/regs-gpio.h> #include <mach/hardware.h> -#include <plat/audio.h> #include <linux/io.h> #include <mach/spi-gpio.h> @@ -37,7 +36,7 @@ #include "../codecs/wm8753.h" #include "lm4857.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" /* define the scenarios */ @@ -137,7 +136,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, return ret; /* codec PLL input is PCLK/4 */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -153,7 +152,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0); } /* @@ -203,7 +202,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, return ret; /* configue and enable PLL for 12.288MHz output */ - ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -219,7 +218,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); /* disable the PLL */ - return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c-dma.c index 1f35c6fcf5f..7725e26d6c9 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -1,5 +1,5 @@ /* - * s3c24xx-pcm.c -- ALSA Soc Audio Layer + * s3c-dma.c -- ALSA Soc Audio Layer * * (c) 2006 Wolfson Microelectronics PLC. * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com @@ -29,11 +29,10 @@ #include <asm/dma.h> #include <mach/hardware.h> #include <mach/dma.h> -#include <plat/audio.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" -static const struct snd_pcm_hardware s3c24xx_pcm_hardware = { +static const struct snd_pcm_hardware s3c_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -63,15 +62,15 @@ struct s3c24xx_runtime_data { dma_addr_t dma_start; dma_addr_t dma_pos; dma_addr_t dma_end; - struct s3c24xx_pcm_dma_params *params; + struct s3c_dma_params *params; }; -/* s3c24xx_pcm_enqueue +/* s3c_dma_enqueue * * place a dma buffer onto the queue for the dma system * to handle. */ -static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) +static void s3c_dma_enqueue(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; @@ -80,12 +79,13 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); - if (s3c_dma_has_circular()) { + if (s3c_dma_has_circular()) limit = (prtd->dma_end - prtd->dma_start) / prtd->dma_period; - } else + else limit = prtd->dma_limit; - pr_debug("%s: loaded %d, limit %d\n", __func__, prtd->dma_loaded, limit); + pr_debug("%s: loaded %d, limit %d\n", + __func__, prtd->dma_loaded, limit); while (prtd->dma_loaded < limit) { unsigned long len = prtd->dma_period; @@ -133,19 +133,19 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, spin_lock(&prtd->lock); if (prtd->state & ST_RUNNING && !s3c_dma_has_circular()) { prtd->dma_loaded--; - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); } spin_unlock(&prtd->lock); } -static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, +static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); int ret = 0; @@ -198,7 +198,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) +static int s3c_dma_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; @@ -215,7 +215,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) +static int s3c_dma_prepare(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -248,12 +248,12 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) prtd->dma_pos = prtd->dma_start; /* enqueue dma buffers */ - s3c24xx_pcm_enqueue(substream); + s3c_dma_enqueue(substream); return ret; } -static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c_dma_trigger(struct snd_pcm_substream *substream, int cmd) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; @@ -288,7 +288,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } static snd_pcm_uframes_t -s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) +s3c_dma_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -323,7 +323,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream) return bytes_to_frames(substream->runtime, res); } -static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) +static int s3c_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; @@ -331,7 +331,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &s3c_dma_hardware); prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); if (prtd == NULL) @@ -343,7 +343,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) return 0; } -static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) +static int s3c_dma_close(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; @@ -351,14 +351,14 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); if (!prtd) - pr_debug("s3c24xx_pcm_close called with prtd == NULL\n"); + pr_debug("s3c_dma_close called with prtd == NULL\n"); kfree(prtd); return 0; } -static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, +static int s3c_dma_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -371,23 +371,23 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, runtime->dma_bytes); } -static struct snd_pcm_ops s3c24xx_pcm_ops = { - .open = s3c24xx_pcm_open, - .close = s3c24xx_pcm_close, +static struct snd_pcm_ops s3c_dma_ops = { + .open = s3c_dma_open, + .close = s3c_dma_close, .ioctl = snd_pcm_lib_ioctl, - .hw_params = s3c24xx_pcm_hw_params, - .hw_free = s3c24xx_pcm_hw_free, - .prepare = s3c24xx_pcm_prepare, - .trigger = s3c24xx_pcm_trigger, - .pointer = s3c24xx_pcm_pointer, - .mmap = s3c24xx_pcm_mmap, + .hw_params = s3c_dma_hw_params, + .hw_free = s3c_dma_hw_free, + .prepare = s3c_dma_prepare, + .trigger = s3c_dma_trigger, + .pointer = s3c_dma_pointer, + .mmap = s3c_dma_mmap, }; -static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +static int s3c_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; + size_t size = s3c_dma_hardware.buffer_bytes_max; pr_debug("Entered %s\n", __func__); @@ -402,7 +402,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) return 0; } -static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) +static void s3c_dma_free_dma_buffers(struct snd_pcm *pcm) { struct snd_pcm_substream *substream; struct snd_dma_buffer *buf; @@ -425,9 +425,9 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32); +static u64 s3c_dma_mask = DMA_BIT_MASK(32); -static int s3c24xx_pcm_new(struct snd_card *card, +static int s3c_dma_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; @@ -435,19 +435,19 @@ static int s3c24xx_pcm_new(struct snd_card *card, pr_debug("Entered %s\n", __func__); if (!card->dev->dma_mask) - card->dev->dma_mask = &s3c24xx_pcm_dmamask; + card->dev->dma_mask = &s3c_dma_mask; if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = 0xffffffff; if (dai->playback.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } if (dai->capture.channels_min) { - ret = s3c24xx_pcm_preallocate_dma_buffer(pcm, + ret = s3c_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) goto out; @@ -458,9 +458,9 @@ static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_platform s3c24xx_soc_platform = { .name = "s3c24xx-audio", - .pcm_ops = &s3c24xx_pcm_ops, - .pcm_new = s3c24xx_pcm_new, - .pcm_free = s3c24xx_pcm_free_dma_buffers, + .pcm_ops = &s3c_dma_ops, + .pcm_new = s3c_dma_new, + .pcm_free = s3c_dma_free_dma_buffers, }; EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); @@ -477,5 +477,5 @@ static void __exit s3c24xx_soc_platform_exit(void) module_exit(s3c24xx_soc_platform_exit); MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>"); -MODULE_DESCRIPTION("Samsung S3C24XX PCM DMA module"); +MODULE_DESCRIPTION("Samsung S3C Audio DMA module"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.h b/sound/soc/s3c24xx/s3c-dma.h index 0088c79822e..69bb6bf6fc1 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.h +++ b/sound/soc/s3c24xx/s3c-dma.h @@ -1,5 +1,5 @@ /* - * s3c24xx-pcm.h -- + * s3c-dma.h -- * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the @@ -9,13 +9,13 @@ * ALSA PCM interface for the Samsung S3C24xx CPU */ -#ifndef _S3C24XX_PCM_H -#define _S3C24XX_PCM_H +#ifndef _S3C_AUDIO_H +#define _S3C_AUDIO_H #define ST_RUNNING (1<<0) #define ST_OPENED (1<<1) -struct s3c24xx_pcm_dma_params { +struct s3c_dma_params { struct s3c2410_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 9bc4aa35caa..e994d8374fe 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -32,11 +32,10 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/dma.h> #include "s3c-i2s-v2.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #undef S3C_IIS_V2_SUPPORTED @@ -312,12 +311,15 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_MSB; break; case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_LSB; break; case SND_SOC_DAIFMT_I2S: + iismod &= ~S3C2412_IISMOD_LR_RLOW; iismod |= S3C2412_IISMOD_SDF_IIS; break; default: @@ -392,7 +394,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); @@ -467,6 +469,31 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, switch (div_id) { case S3C_I2SV2_DIV_BCLK: + if (div > 3) { + /* convert value to bit field */ + + switch (div) { + case 16: + div = S3C2412_IISMOD_BCLK_16FS; + break; + + case 32: + div = S3C2412_IISMOD_BCLK_32FS; + break; + + case 24: + div = S3C2412_IISMOD_BCLK_24FS; + break; + + case 48: + div = S3C2412_IISMOD_BCLK_48FS; + break; + + default: + return -EINVAL; + } + } + reg = readl(i2s->regs + S3C2412_IISMOD); reg &= ~S3C2412_IISMOD_BCLK_MASK; writel(reg | div, i2s->regs + S3C2412_IISMOD); @@ -626,7 +653,7 @@ int s3c_i2sv2_probe(struct platform_device *pdev, } i2s->iis_pclk = clk_get(dev, "iis"); - if (i2s->iis_pclk == NULL) { + if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); iounmap(i2s->regs); return -ENOENT; diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h index f66854a77fb..ecf8eaaed1d 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.h +++ b/sound/soc/s3c24xx/s3c-i2s-v2.h @@ -49,8 +49,8 @@ struct s3c_i2sv2_info { unsigned char master; - struct s3c24xx_pcm_dma_params *dma_playback; - struct s3c24xx_pcm_dma_params *dma_capture; + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; u32 suspend_iismod; u32 suspend_iiscon; diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c new file mode 100644 index 00000000000..9e61a7c2d9a --- /dev/null +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -0,0 +1,552 @@ +/* sound/soc/s3c24xx/s3c-pcm.c + * + * ALSA SoC Audio Layer - S3C PCM-Controller driver + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar@samsung.com> + * based upon I2S drivers by Ben Dooks. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include <plat/audio.h> +#include <plat/dma.h> + +#include "s3c-dma.h" +#include "s3c-pcm.h" + +static struct s3c2410_dma_client s3c_pcm_dma_client_out = { + .name = "PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c_pcm_dma_client_in = { + .name = "PCM Stereo in" +}; + +static struct s3c_dma_params s3c_pcm_stereo_out[] = { + [0] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_out, + .dma_size = 4, + }, +}; + +static struct s3c_dma_params s3c_pcm_stereo_in[] = { + [0] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, + [1] = { + .client = &s3c_pcm_dma_client_in, + .dma_size = 4, + }, +}; + +static struct s3c_pcm_info s3c_pcm[2]; + +static inline struct s3c_pcm_info *to_info(struct snd_soc_dai *cpu_dai) +{ + return cpu_dai->private_data; +} + +static void s3c_pcm_snd_txctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + clkctl = readl(regs + S3C_PCM_CLKCTL); + ctl = readl(regs + S3C_PCM_CTL); + ctl &= ~(S3C_PCM_CTL_TXDIPSTICK_MASK + << S3C_PCM_CTL_TXDIPSTICK_SHIFT); + + if (on) { + ctl |= S3C_PCM_CTL_TXDMA_EN; + ctl |= S3C_PCM_CTL_TXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + ctl |= (0x20<<S3C_PCM_CTL_TXDIPSTICK_SHIFT); + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_TXDMA_EN; + ctl &= ~S3C_PCM_CTL_TXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_RXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static void s3c_pcm_snd_rxctrl(struct s3c_pcm_info *pcm, int on) +{ + void __iomem *regs = pcm->regs; + u32 ctl, clkctl; + + ctl = readl(regs + S3C_PCM_CTL); + clkctl = readl(regs + S3C_PCM_CLKCTL); + + if (on) { + ctl |= S3C_PCM_CTL_RXDMA_EN; + ctl |= S3C_PCM_CTL_RXFIFO_EN; + ctl |= S3C_PCM_CTL_ENABLE; + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } else { + ctl &= ~S3C_PCM_CTL_RXDMA_EN; + ctl &= ~S3C_PCM_CTL_RXFIFO_EN; + + if (!(ctl & S3C_PCM_CTL_TXFIFO_EN)) { + ctl &= ~S3C_PCM_CTL_ENABLE; + if (!pcm->idleclk) + clkctl |= S3C_PCM_CLKCTL_SERCLK_EN; + } + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + writel(ctl, regs + S3C_PCM_CTL); +} + +static int s3c_pcm_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_pcm_info *pcm = to_info(rtd->dai->cpu_dai); + unsigned long flags; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 1); + else + s3c_pcm_snd_txctrl(pcm, 1); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&pcm->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + s3c_pcm_snd_rxctrl(pcm, 0); + else + s3c_pcm_snd_txctrl(pcm, 0); + + spin_unlock_irqrestore(&pcm->lock, flags); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *socdai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai_link *dai = rtd->dai; + struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + void __iomem *regs = pcm->regs; + struct clk *clk; + int sclk_div, sync_div; + unsigned long flags; + u32 clkctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dai->cpu_dai->dma_data = pcm->dma_playback; + else + dai->cpu_dai->dma_data = pcm->dma_capture; + + /* Strictly check for sample size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + default: + return -EINVAL; + } + + spin_lock_irqsave(&pcm->lock, flags); + + /* Get hold of the PCMSOURCE_CLK */ + clkctl = readl(regs + S3C_PCM_CLKCTL); + if (clkctl & S3C_PCM_CLKCTL_SERCLKSEL_PCLK) + clk = pcm->pclk; + else + clk = pcm->cclk; + + /* Set the SCLK divider */ + sclk_div = clk_get_rate(clk) / pcm->sclk_per_fs / + params_rate(params) / 2 - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SCLKDIV_MASK + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + clkctl |= ((sclk_div & S3C_PCM_CLKCTL_SCLKDIV_MASK) + << S3C_PCM_CLKCTL_SCLKDIV_SHIFT); + + /* Set the SYNC divider */ + sync_div = pcm->sclk_per_fs - 1; + + clkctl &= ~(S3C_PCM_CLKCTL_SYNCDIV_MASK + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + clkctl |= ((sync_div & S3C_PCM_CLKCTL_SYNCDIV_MASK) + << S3C_PCM_CLKCTL_SYNCDIV_SHIFT); + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + spin_unlock_irqrestore(&pcm->lock, flags); + + dev_dbg(pcm->dev, "PCMSOURCE_CLK-%lu SCLK=%ufs \ + SCLK_DIV=%d SYNC_DIV=%d\n", + clk_get_rate(clk), pcm->sclk_per_fs, + sclk_div, sync_div); + + return 0; +} + +static int s3c_pcm_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + unsigned long flags; + int ret = 0; + u32 ctl; + + dev_dbg(pcm->dev, "Entered %s\n", __func__); + + spin_lock_irqsave(&pcm->lock, flags); + + ctl = readl(regs + S3C_PCM_CTL); + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do, NB_NF by default */ + break; + default: + dev_err(pcm->dev, "Unsupported clock inversion!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + /* Nothing to do, Master by default */ + break; + default: + dev_err(pcm->dev, "Unsupported master/slave format!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + pcm->idleclk = 1; + break; + case SND_SOC_DAIFMT_GATED: + pcm->idleclk = 0; + break; + default: + dev_err(pcm->dev, "Invalid Clock gating request!\n"); + ret = -EINVAL; + goto exit; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + ctl |= S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl |= S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + case SND_SOC_DAIFMT_DSP_B: + ctl &= ~S3C_PCM_CTL_TXMSB_AFTER_FSYNC; + ctl &= ~S3C_PCM_CTL_RXMSB_AFTER_FSYNC; + break; + default: + dev_err(pcm->dev, "Unsupported data format!\n"); + ret = -EINVAL; + goto exit; + } + + writel(ctl, regs + S3C_PCM_CTL); + +exit: + spin_unlock_irqrestore(&pcm->lock, flags); + + return ret; +} + +static int s3c_pcm_set_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + + switch (div_id) { + case S3C_PCM_SCLK_PER_FS: + pcm->sclk_per_fs = div; + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int s3c_pcm_set_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct s3c_pcm_info *pcm = to_info(cpu_dai); + void __iomem *regs = pcm->regs; + u32 clkctl = readl(regs + S3C_PCM_CLKCTL); + + switch (clk_id) { + case S3C_PCM_CLKSRC_PCLK: + clkctl |= S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + break; + + case S3C_PCM_CLKSRC_MUX: + clkctl &= ~S3C_PCM_CLKCTL_SERCLKSEL_PCLK; + + if (clk_get_rate(pcm->cclk) != freq) + clk_set_rate(pcm->cclk, freq); + + break; + + default: + return -EINVAL; + } + + writel(clkctl, regs + S3C_PCM_CLKCTL); + + return 0; +} + +static struct snd_soc_dai_ops s3c_pcm_dai_ops = { + .set_sysclk = s3c_pcm_set_sysclk, + .set_clkdiv = s3c_pcm_set_clkdiv, + .trigger = s3c_pcm_trigger, + .hw_params = s3c_pcm_hw_params, + .set_fmt = s3c_pcm_set_fmt, +}; + +#define S3C_PCM_RATES SNDRV_PCM_RATE_8000_96000 + +#define S3C_PCM_DECLARE(n) \ +{ \ + .name = "samsung-pcm", \ + .id = (n), \ + .symmetric_rates = 1, \ + .ops = &s3c_pcm_dai_ops, \ + .playback = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ + .capture = { \ + .channels_min = 2, \ + .channels_max = 2, \ + .rates = S3C_PCM_RATES, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + }, \ +} + +struct snd_soc_dai s3c_pcm_dai[] = { + S3C_PCM_DECLARE(0), + S3C_PCM_DECLARE(1), +}; +EXPORT_SYMBOL_GPL(s3c_pcm_dai); + +static __devinit int s3c_pcm_dev_probe(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm; + struct snd_soc_dai *dai; + struct resource *mem_res, *dmatx_res, *dmarx_res; + struct s3c_audio_pdata *pcm_pdata; + int ret; + + /* Check for valid device index */ + if ((pdev->id < 0) || pdev->id >= ARRAY_SIZE(s3c_pcm)) { + dev_err(&pdev->dev, "id %d out of range\n", pdev->id); + return -EINVAL; + } + + pcm_pdata = pdev->dev.platform_data; + + /* Check for availability of necessary resource */ + dmatx_res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmatx_res) { + dev_err(&pdev->dev, "Unable to get PCM-TX dma resource\n"); + return -ENXIO; + } + + dmarx_res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!dmarx_res) { + dev_err(&pdev->dev, "Unable to get PCM-RX dma resource\n"); + return -ENXIO; + } + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "Unable to get register resource\n"); + return -ENXIO; + } + + if (pcm_pdata && pcm_pdata->cfg_gpio && pcm_pdata->cfg_gpio(pdev)) { + dev_err(&pdev->dev, "Unable to configure gpio\n"); + return -EINVAL; + } + + pcm = &s3c_pcm[pdev->id]; + pcm->dev = &pdev->dev; + + spin_lock_init(&pcm->lock); + + dai = &s3c_pcm_dai[pdev->id]; + dai->dev = &pdev->dev; + + /* Default is 128fs */ + pcm->sclk_per_fs = 128; + + pcm->cclk = clk_get(&pdev->dev, "audio-bus"); + if (IS_ERR(pcm->cclk)) { + dev_err(&pdev->dev, "failed to get audio-bus\n"); + ret = PTR_ERR(pcm->cclk); + goto err1; + } + clk_enable(pcm->cclk); + + /* record our pcm structure for later use in the callbacks */ + dai->private_data = pcm; + + if (!request_mem_region(mem_res->start, + resource_size(mem_res), "samsung-pcm")) { + dev_err(&pdev->dev, "Unable to request register region\n"); + ret = -EBUSY; + goto err2; + } + + pcm->regs = ioremap(mem_res->start, 0x100); + if (pcm->regs == NULL) { + dev_err(&pdev->dev, "cannot ioremap registers\n"); + ret = -ENXIO; + goto err3; + } + + pcm->pclk = clk_get(&pdev->dev, "pcm"); + if (IS_ERR(pcm->pclk)) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + ret = -ENOENT; + goto err4; + } + clk_enable(pcm->pclk); + + ret = snd_soc_register_dai(dai); + if (ret != 0) { + dev_err(&pdev->dev, "failed to get pcm_clock\n"); + goto err5; + } + + s3c_pcm_stereo_in[pdev->id].dma_addr = mem_res->start + + S3C_PCM_RXFIFO; + s3c_pcm_stereo_out[pdev->id].dma_addr = mem_res->start + + S3C_PCM_TXFIFO; + + s3c_pcm_stereo_in[pdev->id].channel = dmarx_res->start; + s3c_pcm_stereo_out[pdev->id].channel = dmatx_res->start; + + pcm->dma_capture = &s3c_pcm_stereo_in[pdev->id]; + pcm->dma_playback = &s3c_pcm_stereo_out[pdev->id]; + + return 0; + +err5: + clk_disable(pcm->pclk); + clk_put(pcm->pclk); +err4: + iounmap(pcm->regs); +err3: + release_mem_region(mem_res->start, resource_size(mem_res)); +err2: + clk_disable(pcm->cclk); + clk_put(pcm->cclk); +err1: + return ret; +} + +static __devexit int s3c_pcm_dev_remove(struct platform_device *pdev) +{ + struct s3c_pcm_info *pcm = &s3c_pcm[pdev->id]; + struct resource *mem_res; + + iounmap(pcm->regs); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem_res->start, resource_size(mem_res)); + + clk_disable(pcm->cclk); + clk_disable(pcm->pclk); + clk_put(pcm->pclk); + clk_put(pcm->cclk); + + return 0; +} + +static struct platform_driver s3c_pcm_driver = { + .probe = s3c_pcm_dev_probe, + .remove = s3c_pcm_dev_remove, + .driver = { + .name = "samsung-pcm", + .owner = THIS_MODULE, + }, +}; + +static int __init s3c_pcm_init(void) +{ + return platform_driver_register(&s3c_pcm_driver); +} +module_init(s3c_pcm_init); + +static void __exit s3c_pcm_exit(void) +{ + platform_driver_unregister(&s3c_pcm_driver); +} +module_exit(s3c_pcm_exit); + +/* Module information */ +MODULE_AUTHOR("Jaswinder Singh, <jassi.brar@samsung.com>"); +MODULE_DESCRIPTION("S3C PCM Controller Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s3c24xx/s3c-pcm.h b/sound/soc/s3c24xx/s3c-pcm.h new file mode 100644 index 00000000000..69ff9971692 --- /dev/null +++ b/sound/soc/s3c24xx/s3c-pcm.h @@ -0,0 +1,123 @@ +/* sound/soc/s3c24xx/s3c-pcm.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __S3C_PCM_H +#define __S3C_PCM_H __FILE__ + +/*Register Offsets */ +#define S3C_PCM_CTL (0x00) +#define S3C_PCM_CLKCTL (0x04) +#define S3C_PCM_TXFIFO (0x08) +#define S3C_PCM_RXFIFO (0x0C) +#define S3C_PCM_IRQCTL (0x10) +#define S3C_PCM_IRQSTAT (0x14) +#define S3C_PCM_FIFOSTAT (0x18) +#define S3C_PCM_CLRINT (0x20) + +/* PCM_CTL Bit-Fields */ +#define S3C_PCM_CTL_TXDIPSTICK_MASK (0x3f) +#define S3C_PCM_CTL_TXDIPSTICK_SHIFT (13) +#define S3C_PCM_CTL_RXDIPSTICK_MSK (0x3f<<7) +#define S3C_PCM_CTL_TXDMA_EN (0x1<<6) +#define S3C_PCM_CTL_RXDMA_EN (0x1<<5) +#define S3C_PCM_CTL_TXMSB_AFTER_FSYNC (0x1<<4) +#define S3C_PCM_CTL_RXMSB_AFTER_FSYNC (0x1<<3) +#define S3C_PCM_CTL_TXFIFO_EN (0x1<<2) +#define S3C_PCM_CTL_RXFIFO_EN (0x1<<1) +#define S3C_PCM_CTL_ENABLE (0x1<<0) + +/* PCM_CLKCTL Bit-Fields */ +#define S3C_PCM_CLKCTL_SERCLK_EN (0x1<<19) +#define S3C_PCM_CLKCTL_SERCLKSEL_PCLK (0x1<<18) +#define S3C_PCM_CLKCTL_SCLKDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SYNCDIV_MASK (0x1ff) +#define S3C_PCM_CLKCTL_SCLKDIV_SHIFT (9) +#define S3C_PCM_CLKCTL_SYNCDIV_SHIFT (0) + +/* PCM_TXFIFO Bit-Fields */ +#define S3C_PCM_TXFIFO_DVALID (0x1<<16) +#define S3C_PCM_TXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_RXFIFO Bit-Fields */ +#define S3C_PCM_RXFIFO_DVALID (0x1<<16) +#define S3C_PCM_RXFIFO_DATA_MSK (0xffff<<0) + +/* PCM_IRQCTL Bit-Fields */ +#define S3C_PCM_IRQCTL_IRQEN (0x1<<14) +#define S3C_PCM_IRQCTL_WRDEN (0x1<<12) +#define S3C_PCM_IRQCTL_TXEMPTYEN (0x1<<11) +#define S3C_PCM_IRQCTL_TXALMSTEMPTYEN (0x1<<10) +#define S3C_PCM_IRQCTL_TXFULLEN (0x1<<9) +#define S3C_PCM_IRQCTL_TXALMSTFULLEN (0x1<<8) +#define S3C_PCM_IRQCTL_TXSTARVEN (0x1<<7) +#define S3C_PCM_IRQCTL_TXERROVRFLEN (0x1<<6) +#define S3C_PCM_IRQCTL_RXEMPTEN (0x1<<5) +#define S3C_PCM_IRQCTL_RXALMSTEMPTEN (0x1<<4) +#define S3C_PCM_IRQCTL_RXFULLEN (0x1<<3) +#define S3C_PCM_IRQCTL_RXALMSTFULLEN (0x1<<2) +#define S3C_PCM_IRQCTL_RXSTARVEN (0x1<<1) +#define S3C_PCM_IRQCTL_RXERROVRFLEN (0x1<<0) + +/* PCM_IRQSTAT Bit-Fields */ +#define S3C_PCM_IRQSTAT_IRQPND (0x1<<13) +#define S3C_PCM_IRQSTAT_WRD_XFER (0x1<<12) +#define S3C_PCM_IRQSTAT_TXEMPTY (0x1<<11) +#define S3C_PCM_IRQSTAT_TXALMSTEMPTY (0x1<<10) +#define S3C_PCM_IRQSTAT_TXFULL (0x1<<9) +#define S3C_PCM_IRQSTAT_TXALMSTFULL (0x1<<8) +#define S3C_PCM_IRQSTAT_TXSTARV (0x1<<7) +#define S3C_PCM_IRQSTAT_TXERROVRFL (0x1<<6) +#define S3C_PCM_IRQSTAT_RXEMPT (0x1<<5) +#define S3C_PCM_IRQSTAT_RXALMSTEMPT (0x1<<4) +#define S3C_PCM_IRQSTAT_RXFULL (0x1<<3) +#define S3C_PCM_IRQSTAT_RXALMSTFULL (0x1<<2) +#define S3C_PCM_IRQSTAT_RXSTARV (0x1<<1) +#define S3C_PCM_IRQSTAT_RXERROVRFL (0x1<<0) + +/* PCM_FIFOSTAT Bit-Fields */ +#define S3C_PCM_FIFOSTAT_TXCNT_MSK (0x3f<<14) +#define S3C_PCM_FIFOSTAT_TXFIFOEMPTY (0x1<<13) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTEMPTY (0x1<<12) +#define S3C_PCM_FIFOSTAT_TXFIFOFULL (0x1<<11) +#define S3C_PCM_FIFOSTAT_TXFIFOALMSTFULL (0x1<<10) +#define S3C_PCM_FIFOSTAT_RXCNT_MSK (0x3f<<4) +#define S3C_PCM_FIFOSTAT_RXFIFOEMPTY (0x1<<3) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTEMPTY (0x1<<2) +#define S3C_PCM_FIFOSTAT_RXFIFOFULL (0x1<<1) +#define S3C_PCM_FIFOSTAT_RXFIFOALMSTFULL (0x1<<0) + +#define S3C_PCM_CLKSRC_PCLK 0 +#define S3C_PCM_CLKSRC_MUX 1 + +#define S3C_PCM_SCLK_PER_FS 0 + +/** + * struct s3c_pcm_info - S3C PCM Controller information + * @dev: The parent device passed to use from the probe. + * @regs: The pointer to the device register block. + * @dma_playback: DMA information for playback channel. + * @dma_capture: DMA information for capture channel. + */ +struct s3c_pcm_info { + spinlock_t lock; + struct device *dev; + void __iomem *regs; + + unsigned int sclk_per_fs; + + /* Whether to keep PCMSCLK enabled even when idle(no active xfer) */ + unsigned int idleclk; + + struct clk *pclk; + struct clk *cclk; + + struct s3c_dma_params *dma_playback; + struct s3c_dma_params *dma_capture; +}; + +#endif /* __S3C_PCM_H */ diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index a587ec40b44..359e59346ba 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -34,11 +34,10 @@ #include <plat/regs-s3c2412-iis.h> -#include <plat/audio.h> #include <mach/regs-gpio.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c2412-i2s.h" #define S3C2412_I2S_DEBUG 0 @@ -51,14 +50,14 @@ static struct s3c2410_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { .client = &s3c2412_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { .client = &s3c2412_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index fc1beb0930b..0191e3acb0b 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -32,11 +32,10 @@ #include <plat/regs-ac97.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> #include <asm/dma.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" struct s3c24xx_ac97_info { @@ -189,21 +188,21 @@ static struct s3c2410_dma_client s3c2443_dma_client_micin = { .name = "AC97 Mic Mono in" }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_out = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_out = { .client = &s3c2443_dma_client_out, .channel = DMACH_PCM_OUT, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_pcm_stereo_in = { +static struct s3c_dma_params s3c2443_ac97_pcm_stereo_in = { .client = &s3c2443_dma_client_in, .channel = DMACH_PCM_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_PCM_DATA, .dma_size = 4, }; -static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { +static struct s3c_dma_params s3c2443_ac97_mic_mono_in = { .client = &s3c2443_dma_client_micin, .channel = DMACH_MIC_IN, .dma_addr = S3C2440_PA_AC97 + S3C_AC97_MIC_DATA, @@ -291,7 +290,7 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); @@ -340,7 +339,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 40e2c4790f0..0bc5950b9f0 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -32,13 +32,13 @@ #include <mach/hardware.h> #include <mach/regs-gpio.h> #include <mach/regs-clock.h> -#include <plat/audio.h> + #include <asm/dma.h> #include <mach/dma.h> #include <plat/regs-iis.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" static struct s3c2410_dma_client s3c24xx_dma_client_out = { @@ -49,14 +49,14 @@ static struct s3c2410_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { .client = &s3c24xx_dma_client_out, .channel = DMACH_I2S_OUT, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, .dma_size = 2, }; -static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = { +static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { .client = &s3c24xx_dma_client_in, .channel = DMACH_I2S_IN, .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, @@ -258,12 +258,12 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c24xx_pcm_dma_params *) + ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; default: @@ -280,7 +280,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c24xx_pcm_dma_params *) + int channel = ((struct s3c_dma_params *) rtd->dai->cpu_dai->dma_data)->channel; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 1966e0d5652..507b2ed5d58 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -21,7 +21,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c index 8346bd96eaf..bdf8951af8e 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c index 25797e09617..185c0acb5ce 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c @@ -18,7 +18,7 @@ #include <plat/audio-simtec.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "s3c24xx_simtec.h" diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index c215d32d632..052d59659c2 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -24,7 +24,7 @@ #include <plat/regs-iis.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-i2s.h" #include "../codecs/uda134x.h" diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c index 105a77eeded..cc7edb5f792 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.c +++ b/sound/soc/s3c24xx/s3c64xx-i2s.c @@ -31,12 +31,11 @@ #include <plat/gpio-bank-d.h> #include <plat/gpio-bank-e.h> #include <plat/gpio-cfg.h> -#include <plat/audio.h> #include <mach/map.h> #include <mach/dma.h> -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c64xx-i2s.h" static struct s3c2410_dma_client s3c64xx_dma_client_out = { @@ -47,7 +46,7 @@ static struct s3c2410_dma_client s3c64xx_dma_client_in = { .name = "I2S PCM Stereo in" }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { [0] = { .channel = DMACH_I2S0_OUT, .client = &s3c64xx_dma_client_out, @@ -62,7 +61,7 @@ static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = { }, }; -static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { +static struct s3c_dma_params s3c64xx_i2s_pcm_stereo_in[2] = { [0] = { .channel = DMACH_I2S0_IN, .client = &s3c64xx_dma_client_in, @@ -99,6 +98,19 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, iismod |= S3C64XX_IISMOD_IMS_SYSMUX; break; + case S3C64XX_CLKSRC_CDCLK: + switch (dir) { + case SND_SOC_CLOCK_IN: + iismod |= S3C64XX_IISMOD_CDCLKCON; + break; + case SND_SOC_CLOCK_OUT: + iismod &= ~S3C64XX_IISMOD_CDCLKCON; + break; + default: + return -EINVAL; + } + break; + default: return -EINVAL; } @@ -111,8 +123,12 @@ static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, struct clk *s3c64xx_i2s_get_clock(struct snd_soc_dai *dai) { struct s3c_i2sv2_info *i2s = to_info(dai); + u32 iismod = readl(i2s->regs + S3C2412_IISMOD); - return i2s->iis_cclk; + if (iismod & S3C64XX_IISMOD_IMS_SYSMUX) + return i2s->iis_cclk; + else + return i2s->iis_pclk; } EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clock); diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h index 02148cee261..abe7253b55f 100644 --- a/sound/soc/s3c24xx/s3c64xx-i2s.h +++ b/sound/soc/s3c24xx/s3c64xx-i2s.h @@ -25,6 +25,7 @@ struct clk; #define S3C64XX_CLKSRC_PCLK (0) #define S3C64XX_CLKSRC_MUX (1) +#define S3C64XX_CLKSRC_CDCLK (2) extern struct snd_soc_dai s3c64xx_i2s_dai[]; diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index a2a4f5323c1..12b783b12fc 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -20,7 +20,7 @@ #include <sound/soc-dapm.h> #include "../codecs/ac97.h" -#include "s3c24xx-pcm.h" +#include "s3c-dma.h" #include "s3c24xx-ac97.h" static struct snd_soc_card smdk2443; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c new file mode 100644 index 00000000000..efe4901213a --- /dev/null +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -0,0 +1,268 @@ +/* + * smdk64xx_wm8580.c + * + * Copyright (c) 2009 Samsung Electronics Co. Ltd + * Author: Jaswinder Singh <jassi.brar@samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/platform_device.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/wm8580.h" +#include "s3c-dma.h" +#include "s3c64xx-i2s.h" + +#define S3C64XX_I2S_V4 2 + +/* SMDK64XX has a 12MHZ crystal attached to WM8580 */ +#define SMDK64XX_WM8580_FREQ 12000000 + +static int smdk64xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int pll_out; + int bfs, rfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U8: + case SNDRV_PCM_FORMAT_S8: + bfs = 16; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + /* The Fvco for WM8580 PLLs must fall within [90,100]MHz. + * This criterion can't be met if we request PLL output + * as {8000x256, 64000x256, 11025x256}Hz. + * As a wayout, we rather change rfs to a minimum value that + * results in (params_rate(params) * rfs), and itself, acceptable + * to both - the CODEC and the CPU. + */ + switch (params_rate(params)) { + case 16000: + case 22050: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 256; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + rfs = 512; + break; + default: + return -EINVAL; + } + pll_out = params_rate(params) * rfs; + + /* Set the Codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* Set the AP DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_CDCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* We use PCLK for basic ops in SoC-Slave mode */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C64XX_CLKSRC_PCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set WM8580 to drive MCLK from its PLLA */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_MCLK, + WM8580_CLKSRC_PLLA); + if (ret < 0) + return ret; + + /* Explicitly set WM8580-DAC to source from MCLK */ + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8580_DAC_CLKSEL, + WM8580_CLKSRC_MCLK); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8580_PLLA, 0, + SMDK64XX_WM8580_FREQ, pll_out); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_I2SV2_DIV_RCLK, rfs); + if (ret < 0) + return ret; + + return 0; +} + +/* + * SMDK64XX WM8580 DAI operations. + */ +static struct snd_soc_ops smdk64xx_ops = { + .hw_params = smdk64xx_hw_params, +}; + +/* SMDK64xx Playback widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_pbk[] = { + SND_SOC_DAPM_HP("Front-L/R", NULL), + SND_SOC_DAPM_HP("Center/Sub", NULL), + SND_SOC_DAPM_HP("Rear-L/R", NULL), +}; + +/* SMDK64xx Capture widgets */ +static const struct snd_soc_dapm_widget wm8580_dapm_widgets_cpt[] = { + SND_SOC_DAPM_MIC("MicIn", NULL), + SND_SOC_DAPM_LINE("LineIn", NULL), +}; + +/* SMDK-PAIFTX connections */ +static const struct snd_soc_dapm_route audio_map_tx[] = { + /* MicIn feeds AINL */ + {"AINL", NULL, "MicIn"}, + + /* LineIn feeds AINL/R */ + {"AINL", NULL, "LineIn"}, + {"AINR", NULL, "LineIn"}, +}; + +/* SMDK-PAIFRX connections */ +static const struct snd_soc_dapm_route audio_map_rx[] = { + /* Front Left/Right are fed VOUT1L/R */ + {"Front-L/R", NULL, "VOUT1L"}, + {"Front-L/R", NULL, "VOUT1R"}, + + /* Center/Sub are fed VOUT2L/R */ + {"Center/Sub", NULL, "VOUT2L"}, + {"Center/Sub", NULL, "VOUT2R"}, + + /* Rear Left/Right are fed VOUT3L/R */ + {"Rear-L/R", NULL, "VOUT3L"}, + {"Rear-L/R", NULL, "VOUT3R"}, +}; + +static int smdk64xx_wm8580_init_paiftx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Capture widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_cpt, + ARRAY_SIZE(wm8580_dapm_widgets_cpt)); + + /* Set up PAIFTX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_tx, ARRAY_SIZE(audio_map_tx)); + + /* Enabling the microphone requires the fitting of a 0R + * resistor to connect the line from the microphone jack. + */ + snd_soc_dapm_disable_pin(codec, "MicIn"); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static int smdk64xx_wm8580_init_paifrx(struct snd_soc_codec *codec) +{ + /* Add smdk64xx specific Playback widgets */ + snd_soc_dapm_new_controls(codec, wm8580_dapm_widgets_pbk, + ARRAY_SIZE(wm8580_dapm_widgets_pbk)); + + /* Set up PAIFRX audio path */ + snd_soc_dapm_add_routes(codec, audio_map_rx, ARRAY_SIZE(audio_map_rx)); + + /* signal a DAPM event */ + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link smdk64xx_dai[] = { +{ /* Primary Playback i/f */ + .name = "WM8580 PAIF RX", + .stream_name = "Playback", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFRX], + .init = smdk64xx_wm8580_init_paifrx, + .ops = &smdk64xx_ops, +}, +{ /* Primary Capture i/f */ + .name = "WM8580 PAIF TX", + .stream_name = "Capture", + .cpu_dai = &s3c64xx_i2s_dai[S3C64XX_I2S_V4], + .codec_dai = &wm8580_dai[WM8580_DAI_PAIFTX], + .init = smdk64xx_wm8580_init_paiftx, + .ops = &smdk64xx_ops, +}, +}; + +static struct snd_soc_card smdk64xx = { + .name = "smdk64xx", + .platform = &s3c24xx_soc_platform, + .dai_link = smdk64xx_dai, + .num_links = ARRAY_SIZE(smdk64xx_dai), +}; + +static struct snd_soc_device smdk64xx_snd_devdata = { + .card = &smdk64xx, + .codec_dev = &soc_codec_dev_wm8580, +}; + +static struct platform_device *smdk64xx_snd_device; + +static int __init smdk64xx_audio_init(void) +{ + int ret; + + smdk64xx_snd_device = platform_device_alloc("soc-audio", -1); + if (!smdk64xx_snd_device) + return -ENOMEM; + + platform_set_drvdata(smdk64xx_snd_device, &smdk64xx_snd_devdata); + smdk64xx_snd_devdata.dev = &smdk64xx_snd_device->dev; + ret = platform_device_add(smdk64xx_snd_device); + + if (ret) + platform_device_put(smdk64xx_snd_device); + + return ret; +} +module_init(smdk64xx_audio_init); + +MODULE_AUTHOR("Jaswinder Singh, jassi.brar@samsung.com"); +MODULE_DESCRIPTION("ALSA SoC SMDK64XX WM8580"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 83b8028e209..0eb1722f658 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -423,7 +423,7 @@ static void s6000_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; +static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); static int s6000_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -435,7 +435,7 @@ static int s6000_pcm_new(struct snd_card *card, if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (params->dma_in) { s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 9154b4363db..9e697658655 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -23,7 +23,6 @@ config SND_SOC_SH4_SSI config SND_SOC_SH4_FSI tristate "SH4 FSI support" depends on CPU_SUBTYPE_SH7724 - select SH_DMA help This option enables FSI sound support diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 44123248b63..9c49c11c43c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -17,7 +17,7 @@ #include <linux/platform_device.h> #include <linux/delay.h> #include <linux/list.h> -#include <linux/clk.h> +#include <linux/pm_runtime.h> #include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -26,8 +26,6 @@ #include <sound/pcm_params.h> #include <sound/sh_fsi.h> #include <asm/atomic.h> -#include <asm/dma.h> -#include <asm/dma-sh.h> #define DO_FMT 0x0000 #define DOFF_CTL 0x0004 @@ -97,7 +95,6 @@ struct fsi_priv { int fifo_max; int chan; - int dma_chan; int byte_offset; int period_len; @@ -108,7 +105,6 @@ struct fsi_priv { struct fsi_master { void __iomem *base; int irq; - struct clk *clk; struct fsi_priv fsia; struct fsi_priv fsib; struct sh_fsi_platform_info *info; @@ -308,62 +304,6 @@ static int fsi_get_fifo_residue(struct fsi_priv *fsi, int is_play) return residue; } -static int fsi_get_residue(struct fsi_priv *fsi, int is_play) -{ - int residue; - int width; - struct snd_pcm_runtime *runtime; - - runtime = fsi->substream->runtime; - - /* get 1 channel data width */ - width = frames_to_bytes(runtime, 1) / fsi->chan; - - if (2 == width) - residue = fsi_get_fifo_residue(fsi, is_play); - else - residue = get_dma_residue(fsi->dma_chan); - - return residue; -} - -/************************************************************************ - - - basic dma function - - -************************************************************************/ -#define PORTA_DMA 0 -#define PORTB_DMA 1 - -static int fsi_get_dma_chan(void) -{ - if (0 != request_dma(PORTA_DMA, "fsia")) - return -EIO; - - if (0 != request_dma(PORTB_DMA, "fsib")) { - free_dma(PORTA_DMA); - return -EIO; - } - - master->fsia.dma_chan = PORTA_DMA; - master->fsib.dma_chan = PORTB_DMA; - - return 0; -} - -static void fsi_free_dma_chan(void) -{ - dma_wait_for_completion(PORTA_DMA); - dma_wait_for_completion(PORTB_DMA); - free_dma(PORTA_DMA); - free_dma(PORTB_DMA); - - master->fsia.dma_chan = -1; - master->fsib.dma_chan = -1; -} - /************************************************************************ @@ -435,44 +375,6 @@ static void fsi_soft_all_reset(void) mdelay(10); } -static void fsi_16data_push(struct fsi_priv *fsi, - struct snd_pcm_runtime *runtime, - int send) -{ - u16 *dma_start; - u32 snd; - int i; - - /* get dma start position for FSI */ - dma_start = (u16 *)runtime->dma_area; - dma_start += fsi->byte_offset / 2; - - /* - * soft dma - * FSI can not use DMA when 16bpp - */ - for (i = 0; i < send; i++) { - snd = (u32)dma_start[i]; - fsi_reg_write(fsi, DODT, snd << 8); - } -} - -static void fsi_32data_push(struct fsi_priv *fsi, - struct snd_pcm_runtime *runtime, - int send) -{ - u32 *dma_start; - - /* get dma start position for FSI */ - dma_start = (u32 *)runtime->dma_area; - dma_start += fsi->byte_offset / 4; - - dma_wait_for_completion(fsi->dma_chan); - dma_configure_channel(fsi->dma_chan, (SM_INC|0x400|TS_32|TM_BUR)); - dma_write(fsi->dma_chan, (u32)dma_start, - (u32)(fsi->base + DODT), send * 4); -} - /* playback interrupt */ static int fsi_data_push(struct fsi_priv *fsi) { @@ -481,6 +383,8 @@ static int fsi_data_push(struct fsi_priv *fsi) int send; int fifo_free; int width; + u8 *start; + int i; if (!fsi || !fsi->substream || @@ -515,12 +419,22 @@ static int fsi_data_push(struct fsi_priv *fsi) if (fifo_free < send) send = fifo_free; - if (2 == width) - fsi_16data_push(fsi, runtime, send); - else if (4 == width) - fsi_32data_push(fsi, runtime, send); - else + start = runtime->dma_area; + start += fsi->byte_offset; + + switch (width) { + case 2: + for (i = 0; i < send; i++) + fsi_reg_write(fsi, DODT, + ((u32)*((u16 *)start + i) << 8)); + break; + case 4: + for (i = 0; i < send; i++) + fsi_reg_write(fsi, DODT, *((u32 *)start + i)); + break; + default: return -EINVAL; + } fsi->byte_offset += send * width; @@ -532,6 +446,75 @@ static int fsi_data_push(struct fsi_priv *fsi) return 0; } +static int fsi_data_pop(struct fsi_priv *fsi) +{ + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream = NULL; + int free; + int fifo_fill; + int width; + u8 *start; + int i; + + if (!fsi || + !fsi->substream || + !fsi->substream->runtime) + return -EINVAL; + + runtime = fsi->substream->runtime; + + /* FSI FIFO has limit. + * So, this driver can not send periods data at a time + */ + if (fsi->byte_offset >= + fsi->period_len * (fsi->periods + 1)) { + + substream = fsi->substream; + fsi->periods = (fsi->periods + 1) % runtime->periods; + + if (0 == fsi->periods) + fsi->byte_offset = 0; + } + + /* get 1 channel data width */ + width = frames_to_bytes(runtime, 1) / fsi->chan; + + /* get free space for alsa */ + free = (fsi->buffer_len - fsi->byte_offset) / width; + + /* get recv size */ + fifo_fill = fsi_get_fifo_residue(fsi, 0); + + if (free < fifo_fill) + fifo_fill = free; + + start = runtime->dma_area; + start += fsi->byte_offset; + + switch (width) { + case 2: + for (i = 0; i < fifo_fill; i++) + *((u16 *)start + i) = + (u16)(fsi_reg_read(fsi, DIDT) >> 8); + break; + case 4: + for (i = 0; i < fifo_fill; i++) + *((u32 *)start + i) = fsi_reg_read(fsi, DIDT); + break; + default: + return -EINVAL; + } + + fsi->byte_offset += fifo_fill * width; + + fsi_irq_enable(fsi, 0); + + if (substream) + snd_pcm_period_elapsed(substream); + + return 0; +} + static irqreturn_t fsi_interrupt(int irq, void *data) { u32 status = fsi_master_read(SOFT_RST) & ~0x00000010; @@ -545,6 +528,10 @@ static irqreturn_t fsi_interrupt(int irq, void *data) fsi_data_push(&master->fsia); if (int_st & INT_B_OUT) fsi_data_push(&master->fsib); + if (int_st & INT_A_IN) + fsi_data_pop(&master->fsia); + if (int_st & INT_B_IN) + fsi_data_pop(&master->fsib); fsi_master_write(INT_ST, 0x0000000); @@ -571,7 +558,7 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, int is_master; int ret = 0; - clk_enable(master->clk); + pm_runtime_get_sync(dai->dev); /* CKG1 */ data = is_play ? (1 << 0) : (1 << 4); @@ -664,8 +651,6 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream, } fsi_reg_write(fsi, reg, data); - dev_dbg(dai->dev, "use %s format (%d channel) use %d DMAC\n", - msg, fsi->chan, fsi->dma_chan); /* * clear clk reset if master mode @@ -688,7 +673,7 @@ static void fsi_dai_shutdown(struct snd_pcm_substream *substream, fsi_irq_disable(fsi, is_play); fsi_clk_ctrl(fsi, 0); - clk_disable(master->clk); + pm_runtime_put_sync(dai->dev); } static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, @@ -699,16 +684,12 @@ static int fsi_dai_trigger(struct snd_pcm_substream *substream, int cmd, int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; int ret = 0; - /* capture not supported */ - if (!is_play) - return -ENODEV; - switch (cmd) { case SNDRV_PCM_TRIGGER_START: fsi_stream_push(fsi, substream, frames_to_bytes(runtime, runtime->buffer_size), frames_to_bytes(runtime, runtime->period_size)); - ret = fsi_data_push(fsi); + ret = is_play ? fsi_data_push(fsi) : fsi_data_pop(fsi); break; case SNDRV_PCM_TRIGGER_STOP: fsi_irq_disable(fsi, is_play); @@ -780,10 +761,9 @@ static snd_pcm_uframes_t fsi_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsi_priv *fsi = fsi_get(substream); - int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; long location; - location = (fsi->byte_offset - 1) - fsi_get_residue(fsi, is_play); + location = (fsi->byte_offset - 1); if (location < 0) location = 0; @@ -845,7 +825,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, { @@ -857,7 +842,12 @@ struct snd_soc_dai fsi_soc_dai[] = { .channels_min = 1, .channels_max = 8, }, - /* capture not supported */ + .capture = { + .rates = FSI_RATES, + .formats = FSI_FMTS, + .channels_min = 1, + .channels_max = 8, + }, .ops = &fsi_dai_ops, }, }; @@ -881,7 +871,6 @@ EXPORT_SYMBOL_GPL(fsi_soc_platform); static int fsi_probe(struct platform_device *pdev) { struct resource *res; - char clk_name[8]; unsigned int irq; int ret; @@ -912,23 +901,8 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.base = master->base; master->fsib.base = master->base + 0x40; - master->fsia.dma_chan = -1; - master->fsib.dma_chan = -1; - - ret = fsi_get_dma_chan(); - if (ret < 0) { - dev_err(&pdev->dev, "cannot get dma api\n"); - goto exit_iounmap; - } - - /* FSI is based on SPU mstp */ - snprintf(clk_name, sizeof(clk_name), "spu%d", pdev->id); - master->clk = clk_get(NULL, clk_name); - if (IS_ERR(master->clk)) { - dev_err(&pdev->dev, "cannot get %s mstp\n", clk_name); - ret = -EIO; - goto exit_free_dma; - } + pm_runtime_enable(&pdev->dev); + pm_runtime_resume(&pdev->dev); fsi_soc_dai[0].dev = &pdev->dev; fsi_soc_dai[1].dev = &pdev->dev; @@ -938,7 +912,7 @@ static int fsi_probe(struct platform_device *pdev) ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, "fsi", master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); - goto exit_free_dma; + goto exit_iounmap; } ret = snd_soc_register_platform(&fsi_soc_platform); @@ -951,10 +925,9 @@ static int fsi_probe(struct platform_device *pdev) exit_free_irq: free_irq(irq, master); -exit_free_dma: - fsi_free_dma_chan(); exit_iounmap: iounmap(master->base); + pm_runtime_disable(&pdev->dev); exit_kfree: kfree(master); master = NULL; @@ -967,9 +940,7 @@ static int fsi_remove(struct platform_device *pdev) snd_soc_unregister_dais(fsi_soc_dai, ARRAY_SIZE(fsi_soc_dai)); snd_soc_unregister_platform(&fsi_soc_platform); - clk_put(master->clk); - - fsi_free_dma_chan(); + pm_runtime_disable(&pdev->dev); free_irq(master->irq, master); @@ -979,9 +950,27 @@ static int fsi_remove(struct platform_device *pdev) return 0; } +static int fsi_runtime_nop(struct device *dev) +{ + /* Runtime PM callback shared between ->runtime_suspend() + * and ->runtime_resume(). Simply returns success. + * + * This driver re-initializes all registers after + * pm_runtime_get_sync() anyway so there is no need + * to save and restore registers here. + */ + return 0; +} + +static struct dev_pm_ops fsi_pm_ops = { + .runtime_suspend = fsi_runtime_nop, + .runtime_resume = fsi_runtime_nop, +}; + static struct platform_driver fsi_driver = { .driver = { .name = "sh_fsi", + .pm = &fsi_pm_ops, }, .probe = fsi_probe, .remove = fsi_remove, diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c8ceddc2a26..d2505e8b06c 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -77,6 +77,35 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data, #define snd_soc_7_9_spi_write NULL #endif +static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -150,9 +179,20 @@ static struct { unsigned int (*read)(struct snd_soc_codec *, unsigned int); unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { - { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read }, - { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read, - snd_soc_8_16_read_i2c }, + { + .addr_bits = 7, .data_bits = 9, + .write = snd_soc_7_9_write, .read = snd_soc_7_9_read, + .spi_write = snd_soc_7_9_spi_write + }, + { + .addr_bits = 8, .data_bits = 8, + .write = snd_soc_8_8_write, .read = snd_soc_8_8_read, + }, + { + .addr_bits = 8, .data_bits = 16, + .write = snd_soc_8_16_write, .read = snd_soc_8_16_read, + .i2c_read = snd_soc_8_16_read_i2c, + }, }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0a1b2f64bbe..ef8f28284cb 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -37,7 +37,6 @@ #include <sound/initval.h> static DEFINE_MUTEX(pcm_mutex); -static DEFINE_MUTEX(io_mutex); static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq); #ifdef CONFIG_DEBUG_FS @@ -81,6 +80,173 @@ static int run_delayed_work(struct delayed_work *dwork) return ret; } +/* codec register dump */ +static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) +{ + int i, step = 1, count = 0; + + if (!codec->reg_cache_size) + return 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + count += sprintf(buf, "%s registers\n", codec->name); + for (i = 0; i < codec->reg_cache_size; i += step) { + if (codec->readable_register && !codec->readable_register(i)) + continue; + + count += sprintf(buf + count, "%2x: ", i); + if (count >= PAGE_SIZE - 1) + break; + + if (codec->display_register) + count += codec->display_register(codec, buf + count, + PAGE_SIZE - count, i); + else + count += snprintf(buf + count, PAGE_SIZE - count, + "%4x", codec->read(codec, i)); + + if (count >= PAGE_SIZE - 1) + break; + + count += snprintf(buf + count, PAGE_SIZE - count, "\n"); + if (count >= PAGE_SIZE - 1) + break; + } + + /* Truncate count; min() would cause a warning */ + if (count >= PAGE_SIZE) + count = PAGE_SIZE - 1; + + return count; +} +static ssize_t codec_reg_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct snd_soc_device *devdata = dev_get_drvdata(dev); + return soc_codec_reg_show(devdata->card->codec, buf); +} + +static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); + +#ifdef CONFIG_DEBUG_FS +static int codec_reg_open_file(struct inode *inode, struct file *file) +{ + file->private_data = inode->i_private; + return 0; +} + +static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, + size_t count, loff_t *ppos) +{ + ssize_t ret; + struct snd_soc_codec *codec = file->private_data; + char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = soc_codec_reg_show(codec, buf); + if (ret >= 0) + ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); + kfree(buf); + return ret; +} + +static ssize_t codec_reg_write_file(struct file *file, + const char __user *user_buf, size_t count, loff_t *ppos) +{ + char buf[32]; + int buf_size; + char *start = buf; + unsigned long reg, value; + int step = 1; + struct snd_soc_codec *codec = file->private_data; + + buf_size = min(count, (sizeof(buf)-1)); + if (copy_from_user(buf, user_buf, buf_size)) + return -EFAULT; + buf[buf_size] = 0; + + if (codec->reg_cache_step) + step = codec->reg_cache_step; + + while (*start == ' ') + start++; + reg = simple_strtoul(start, &start, 16); + if ((reg >= codec->reg_cache_size) || (reg % step)) + return -EINVAL; + while (*start == ' ') + start++; + if (strict_strtoul(start, 16, &value)) + return -EINVAL; + codec->write(codec, reg, value); + return buf_size; +} + +static const struct file_operations codec_reg_fops = { + .open = codec_reg_open_file, + .read = codec_reg_read_file, + .write = codec_reg_write_file, +}; + +static void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ + char codec_root[128]; + + if (codec->dev) + snprintf(codec_root, sizeof(codec_root), + "%s.%s", codec->name, dev_name(codec->dev)); + else + snprintf(codec_root, sizeof(codec_root), + "%s", codec->name); + + codec->debugfs_codec_root = debugfs_create_dir(codec_root, + debugfs_root); + if (!codec->debugfs_codec_root) { + printk(KERN_WARNING + "ASoC: Failed to create codec debugfs directory\n"); + return; + } + + codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, + codec->debugfs_codec_root, + codec, &codec_reg_fops); + if (!codec->debugfs_reg) + printk(KERN_WARNING + "ASoC: Failed to create codec register debugfs file\n"); + + codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, + codec->debugfs_codec_root, + &codec->pop_time); + if (!codec->debugfs_pop_time) + printk(KERN_WARNING + "Failed to create pop time debugfs file\n"); + + codec->debugfs_dapm = debugfs_create_dir("dapm", + codec->debugfs_codec_root); + if (!codec->debugfs_dapm) + printk(KERN_WARNING + "Failed to create DAPM debugfs directory\n"); + + snd_soc_dapm_debugfs_init(codec); +} + +static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ + debugfs_remove_recursive(codec->debugfs_codec_root); +} + +#else + +static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) +{ +} + +static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) +{ +} +#endif + #ifdef CONFIG_SND_SOC_AC97_BUS /* unregister ac97 codec */ static int soc_ac97_dev_unregister(struct snd_soc_codec *codec) @@ -790,45 +956,6 @@ static int soc_resume(struct device *dev) return 0; } - -/** - * snd_soc_suspend_device: Notify core of device suspend - * - * @dev: Device being suspended. - * - * In order to ensure that the entire audio subsystem is suspended in a - * coordinated fashion ASoC devices should suspend themselves when - * called by ASoC. When the standard kernel suspend process asks the - * device to suspend it should call this function to initiate a suspend - * of the entire ASoC card. - * - * \note Currently this function is stubbed out. - */ -int snd_soc_suspend_device(struct device *dev) -{ - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_suspend_device); - -/** - * snd_soc_resume_device: Notify core of device resume - * - * @dev: Device being resumed. - * - * In order to ensure that the entire audio subsystem is resumed in a - * coordinated fashion ASoC devices should resume themselves when called - * by ASoC. When the standard kernel resume process asks the device - * to resume it should call this function. Once all the components of - * the card have notified that they are ready to be resumed the card - * will be resumed. - * - * \note Currently this function is stubbed out. - */ -int snd_soc_resume_device(struct device *dev) -{ - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_resume_device); #else #define soc_suspend NULL #define soc_resume NULL @@ -843,6 +970,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct platform_device, dev); struct snd_soc_codec_device *codec_dev = card->socdev->codec_dev; + struct snd_soc_codec *codec; struct snd_soc_platform *platform; struct snd_soc_dai *dai; int i, found, ret, ac97; @@ -931,6 +1059,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) if (ret < 0) goto cpu_dai_err; } + codec = card->codec; if (platform->probe) { ret = platform->probe(pdev); @@ -945,10 +1074,69 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) INIT_WORK(&card->deferred_resume_work, soc_resume_deferred); #endif + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].init) { + ret = card->dai_link[i].init(codec); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to init %s\n", + card->dai_link[i].stream_name); + continue; + } + } + if (card->dai_link[i].codec_dai->ac97_control) + ac97 = 1; + } + + snprintf(codec->card->shortname, sizeof(codec->card->shortname), + "%s", card->name); + snprintf(codec->card->longname, sizeof(codec->card->longname), + "%s (%s)", card->name, codec->name); + + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(codec); + + ret = snd_card_register(codec->card); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register soundcard for %s\n", + codec->name); + goto card_err; + } + + mutex_lock(&codec->mutex); +#ifdef CONFIG_SND_SOC_AC97_BUS + /* Only instantiate AC97 if not already done by the adaptor + * for the generic AC97 subsystem. + */ + if (ac97 && strcmp(codec->name, "AC97") != 0) { + ret = soc_ac97_dev_register(codec); + if (ret < 0) { + printk(KERN_ERR "asoc: AC97 device register failed\n"); + snd_card_free(codec->card); + mutex_unlock(&codec->mutex); + goto card_err; + } + } +#endif + + ret = snd_soc_dapm_sys_add(card->socdev->dev); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); + + ret = device_create_file(card->socdev->dev, &dev_attr_codec_reg); + if (ret < 0) + printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); + + soc_init_codec_debugfs(codec); + mutex_unlock(&codec->mutex); + card->instantiated = 1; return; +card_err: + if (platform->remove) + platform->remove(pdev); + platform_err: if (codec_dev->remove) codec_dev->remove(pdev); @@ -1151,157 +1339,6 @@ int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg) } EXPORT_SYMBOL_GPL(snd_soc_codec_volatile_register); -/* codec register dump */ -static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf) -{ - int i, step = 1, count = 0; - - if (!codec->reg_cache_size) - return 0; - - if (codec->reg_cache_step) - step = codec->reg_cache_step; - - count += sprintf(buf, "%s registers\n", codec->name); - for (i = 0; i < codec->reg_cache_size; i += step) { - if (codec->readable_register && !codec->readable_register(i)) - continue; - - count += sprintf(buf + count, "%2x: ", i); - if (count >= PAGE_SIZE - 1) - break; - - if (codec->display_register) - count += codec->display_register(codec, buf + count, - PAGE_SIZE - count, i); - else - count += snprintf(buf + count, PAGE_SIZE - count, - "%4x", codec->read(codec, i)); - - if (count >= PAGE_SIZE - 1) - break; - - count += snprintf(buf + count, PAGE_SIZE - count, "\n"); - if (count >= PAGE_SIZE - 1) - break; - } - - /* Truncate count; min() would cause a warning */ - if (count >= PAGE_SIZE) - count = PAGE_SIZE - 1; - - return count; -} -static ssize_t codec_reg_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct snd_soc_device *devdata = dev_get_drvdata(dev); - return soc_codec_reg_show(devdata->card->codec, buf); -} - -static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL); - -#ifdef CONFIG_DEBUG_FS -static int codec_reg_open_file(struct inode *inode, struct file *file) -{ - file->private_data = inode->i_private; - return 0; -} - -static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - ssize_t ret; - struct snd_soc_codec *codec = file->private_data; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); - if (!buf) - return -ENOMEM; - ret = soc_codec_reg_show(codec, buf); - if (ret >= 0) - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - kfree(buf); - return ret; -} - -static ssize_t codec_reg_write_file(struct file *file, - const char __user *user_buf, size_t count, loff_t *ppos) -{ - char buf[32]; - int buf_size; - char *start = buf; - unsigned long reg, value; - int step = 1; - struct snd_soc_codec *codec = file->private_data; - - buf_size = min(count, (sizeof(buf)-1)); - if (copy_from_user(buf, user_buf, buf_size)) - return -EFAULT; - buf[buf_size] = 0; - - if (codec->reg_cache_step) - step = codec->reg_cache_step; - - while (*start == ' ') - start++; - reg = simple_strtoul(start, &start, 16); - if ((reg >= codec->reg_cache_size) || (reg % step)) - return -EINVAL; - while (*start == ' ') - start++; - if (strict_strtoul(start, 16, &value)) - return -EINVAL; - codec->write(codec, reg, value); - return buf_size; -} - -static const struct file_operations codec_reg_fops = { - .open = codec_reg_open_file, - .read = codec_reg_read_file, - .write = codec_reg_write_file, -}; - -static void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ - codec->debugfs_reg = debugfs_create_file("codec_reg", 0644, - debugfs_root, codec, - &codec_reg_fops); - if (!codec->debugfs_reg) - printk(KERN_WARNING - "ASoC: Failed to create codec register debugfs file\n"); - - codec->debugfs_pop_time = debugfs_create_u32("dapm_pop_time", 0744, - debugfs_root, - &codec->pop_time); - if (!codec->debugfs_pop_time) - printk(KERN_WARNING - "Failed to create pop time debugfs file\n"); - - codec->debugfs_dapm = debugfs_create_dir("dapm", debugfs_root); - if (!codec->debugfs_dapm) - printk(KERN_WARNING - "Failed to create DAPM debugfs directory\n"); - - snd_soc_dapm_debugfs_init(codec); -} - -static void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ - debugfs_remove_recursive(codec->debugfs_dapm); - debugfs_remove(codec->debugfs_pop_time); - debugfs_remove(codec->debugfs_reg); -} - -#else - -static inline void soc_init_codec_debugfs(struct snd_soc_codec *codec) -{ -} - -static inline void soc_cleanup_codec_debugfs(struct snd_soc_codec *codec) -{ -} -#endif - /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec @@ -1369,19 +1406,41 @@ int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, int change; unsigned int old, new; - mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); new = (old & ~mask) | value; change = old != new; if (change) snd_soc_write(codec, reg, new); - mutex_unlock(&io_mutex); return change; } EXPORT_SYMBOL_GPL(snd_soc_update_bits); /** + * snd_soc_update_bits_locked - update codec register bits + * @codec: audio codec + * @reg: codec register + * @mask: register mask + * @value: new value + * + * Writes new register value, and takes the codec mutex. + * + * Returns 1 for change else 0. + */ +static int snd_soc_update_bits_locked(struct snd_soc_codec *codec, + unsigned short reg, unsigned int mask, + unsigned int value) +{ + int change; + + mutex_lock(&codec->mutex); + change = snd_soc_update_bits(codec, reg, mask, value); + mutex_unlock(&codec->mutex); + + return change; +} + +/** * snd_soc_test_bits - test register for change * @codec: audio codec * @reg: codec register @@ -1399,11 +1458,9 @@ int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, int change; unsigned int old, new; - mutex_lock(&io_mutex); old = snd_soc_read(codec, reg); new = (old & ~mask) | value; change = old != new; - mutex_unlock(&io_mutex); return change; } @@ -1450,89 +1507,16 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) mutex_unlock(&codec->mutex); return ret; } - } - - mutex_unlock(&codec->mutex); - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_new_pcms); - -/** - * snd_soc_init_card - register sound card - * @socdev: the SoC audio device - * - * Register a SoC sound card. Also registers an AC97 device if the - * codec is AC97 for ad hoc devices. - * - * Returns 0 for success, else error. - */ -int snd_soc_init_card(struct snd_soc_device *socdev) -{ - struct snd_soc_card *card = socdev->card; - struct snd_soc_codec *codec = card->codec; - int ret = 0, i, ac97 = 0, err = 0; - - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].init) { - err = card->dai_link[i].init(codec); - if (err < 0) { - printk(KERN_ERR "asoc: failed to init %s\n", - card->dai_link[i].stream_name); - continue; - } - } if (card->dai_link[i].codec_dai->ac97_control) { - ac97 = 1; snd_ac97_dev_add_pdata(codec->ac97, card->dai_link[i].cpu_dai->ac97_pdata); } } - snprintf(codec->card->shortname, sizeof(codec->card->shortname), - "%s", card->name); - snprintf(codec->card->longname, sizeof(codec->card->longname), - "%s (%s)", card->name, codec->name); - - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(codec); - - ret = snd_card_register(codec->card); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register soundcard for %s\n", - codec->name); - goto out; - } - - mutex_lock(&codec->mutex); -#ifdef CONFIG_SND_SOC_AC97_BUS - /* Only instantiate AC97 if not already done by the adaptor - * for the generic AC97 subsystem. - */ - if (ac97 && strcmp(codec->name, "AC97") != 0) { - ret = soc_ac97_dev_register(codec); - if (ret < 0) { - printk(KERN_ERR "asoc: AC97 device register failed\n"); - snd_card_free(codec->card); - mutex_unlock(&codec->mutex); - goto out; - } - } -#endif - - err = snd_soc_dapm_sys_add(socdev->dev); - if (err < 0) - printk(KERN_WARNING "asoc: failed to add dapm sysfs entries\n"); - - err = device_create_file(socdev->dev, &dev_attr_codec_reg); - if (err < 0) - printk(KERN_WARNING "asoc: failed to add codec sysfs files\n"); - soc_init_codec_debugfs(codec); mutex_unlock(&codec->mutex); - -out: return ret; } -EXPORT_SYMBOL_GPL(snd_soc_init_card); +EXPORT_SYMBOL_GPL(snd_soc_new_pcms); /** * snd_soc_free_pcms - free sound card and pcms @@ -1734,7 +1718,7 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, mask |= (bitmask - 1) << e->shift_r; } - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg, mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); @@ -1808,7 +1792,7 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, mask |= e->mask << e->shift_r; } - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg, mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); @@ -1969,7 +1953,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val_mask |= mask << rshift; val |= val2 << rshift; } - return snd_soc_update_bits(codec, reg, val_mask, val); + return snd_soc_update_bits_locked(codec, reg, val_mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw); @@ -2075,11 +2059,11 @@ int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol, val = val << shift; val2 = val2 << shift; - err = snd_soc_update_bits(codec, reg, val_mask, val); + err = snd_soc_update_bits_locked(codec, reg, val_mask, val); if (err < 0) return err; - err = snd_soc_update_bits(codec, reg2, val_mask, val2); + err = snd_soc_update_bits_locked(codec, reg2, val_mask, val2); return err; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_2r); @@ -2158,7 +2142,7 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, val = (ucontrol->value.integer.value[0]+min) & 0xff; val |= ((ucontrol->value.integer.value[1]+min) & 0xff) << 8; - return snd_soc_update_bits(codec, reg, 0xffff, val); + return snd_soc_update_bits_locked(codec, reg, 0xffff, val); } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); @@ -2205,16 +2189,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); * snd_soc_dai_set_pll - configure DAI PLL. * @dai: DAI * @pll_id: DAI specific PLL ID + * @source: DAI specific source for the PLL * @freq_in: PLL input clock frequency in Hz * @freq_out: requested PLL output clock frequency in Hz * * Configures and enables PLL to generate output clock based on input clock. */ -int snd_soc_dai_set_pll(struct snd_soc_dai *dai, - int pll_id, unsigned int freq_in, unsigned int freq_out) +int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) { if (dai->ops && dai->ops->set_pll) - return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); + return dai->ops->set_pll(dai, pll_id, source, + freq_in, freq_out); else return -EINVAL; } @@ -2259,6 +2245,30 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** + * snd_soc_dai_set_channel_map - configure DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + if (dai->ops && dai->ops->set_channel_map) + return dai->ops->set_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 66d4c165f99..0d294ef7259 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -719,6 +719,10 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + if (path->connected && + !path->connected(path->source, path->sink)) + continue; + if (path->sink && path->sink->power_check && path->sink->power_check(path->sink)) { power = 1; @@ -1152,6 +1156,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->active ? "active" : "inactive"); list_for_each_entry(p, &w->sources, list_sink) { + if (p->connected && !p->connected(w, p->sink)) + continue; + if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, " in %s %s\n", @@ -1159,6 +1166,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, p->source->name); } list_for_each_entry(p, &w->sinks, list_source) { + if (p->connected && !p->connected(w, p->sink)) + continue; + if (p->connect) ret += snprintf(buf + ret, PAGE_SIZE - ret, " out %s %s\n", @@ -1206,8 +1216,8 @@ void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec) /* test and update the power status of a mux widget */ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, - struct snd_kcontrol *kcontrol, int mask, - int mux, int val, struct soc_enum *e) + struct snd_kcontrol *kcontrol, int change, + int mux, struct soc_enum *e) { struct snd_soc_dapm_path *path; int found = 0; @@ -1216,7 +1226,7 @@ static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget, widget->id != snd_soc_dapm_value_mux) return -ENODEV; - if (!snd_soc_test_bits(widget->codec, e->reg, mask, val)) + if (!change) return 0; /* find dapm widget path assoc with kcontrol */ @@ -1401,10 +1411,13 @@ int snd_soc_dapm_sync(struct snd_soc_codec *codec) EXPORT_SYMBOL_GPL(snd_soc_dapm_sync); static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, - const char *sink, const char *control, const char *source) + const struct snd_soc_dapm_route *route) { struct snd_soc_dapm_path *path; struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w; + const char *sink = route->sink; + const char *control = route->control; + const char *source = route->source; int ret = 0; /* find src and dest widgets */ @@ -1428,6 +1441,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec, path->source = wsource; path->sink = wsink; + path->connected = route->connected; INIT_LIST_HEAD(&path->list); INIT_LIST_HEAD(&path->list_source); INIT_LIST_HEAD(&path->list_sink); @@ -1528,8 +1542,7 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec, int i, ret; for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(codec, route->sink, - route->control, route->source); + ret = snd_soc_dapm_add_route(codec, route); if (ret < 0) { printk(KERN_ERR "Failed to add route %s->%s\n", route->source, @@ -1766,7 +1779,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux; + unsigned int val, mux, change; unsigned int mask, bitmask; int ret = 0; @@ -1786,20 +1799,21 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); - if (widget->event) { - if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret < 0) - goto out; - } - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); - if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_POST_REG); - } else - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + dapm_mux_update_power(widget, kcontrol, change, mux, e); + + if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret < 0) + goto out; + } + + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_POST_REG) + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); out: mutex_unlock(&widget->codec->mutex); @@ -1808,6 +1822,54 @@ out: EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = widget->value; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); + +/** + * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); + struct soc_enum *e = + (struct soc_enum *)kcontrol->private_value; + int change; + int ret = 0; + + if (ucontrol->value.enumerated.item[0] >= e->max) + return -EINVAL; + + mutex_lock(&widget->codec->mutex); + + change = widget->value != ucontrol->value.enumerated.item[0]; + widget->value = ucontrol->value.enumerated.item[0]; + dapm_mux_update_power(widget, kcontrol, change, widget->value, e); + + mutex_unlock(&widget->codec->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); + +/** * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get * callback * @kcontrol: mixer control @@ -1865,7 +1927,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux; + unsigned int val, mux, change; unsigned int mask; int ret = 0; @@ -1883,20 +1945,21 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, mutex_lock(&widget->codec->mutex); widget->value = val; - dapm_mux_update_power(widget, kcontrol, mask, mux, val, e); - if (widget->event) { - if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_PRE_REG); - if (ret < 0) - goto out; - } - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); - if (widget->event_flags & SND_SOC_DAPM_POST_REG) - ret = widget->event(widget, - kcontrol, SND_SOC_DAPM_POST_REG); - } else - ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + change = snd_soc_test_bits(widget->codec, e->reg, mask, val); + dapm_mux_update_power(widget, kcontrol, change, mux, e); + + if (widget->event_flags & SND_SOC_DAPM_PRE_REG) { + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_PRE_REG); + if (ret < 0) + goto out; + } + + ret = snd_soc_update_bits(widget->codec, e->reg, mask, val); + + if (widget->event_flags & SND_SOC_DAPM_POST_REG) + ret = widget->event(widget, + kcontrol, SND_SOC_DAPM_POST_REG); out: mutex_unlock(&widget->codec->mutex); diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 1d455ab7949..3c07a94c2e3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -58,7 +58,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_new); */ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) { - struct snd_soc_codec *codec = jack->card->codec; + struct snd_soc_codec *codec; struct snd_soc_jack_pin *pin; int enable; int oldstatus; @@ -67,6 +67,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) WARN_ON_ONCE(!jack); return; } + codec = jack->card->codec; mutex_lock(&codec->mutex); @@ -162,6 +163,9 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) else report = 0; + if (gpio->jack_status_check) + report = gpio->jack_status_check(); + snd_soc_jack_report(jack, report, gpio->report); } diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c new file mode 100644 index 00000000000..1d07b931f3d --- /dev/null +++ b/sound/soc/soc-utils.c @@ -0,0 +1,74 @@ +/* + * soc-util.c -- ALSA SoC Audio Layer utility functions + * + * Copyright 2009 Wolfson Microelectronics PLC. + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * Liam Girdwood <lrg@slimlogic.co.uk> + * + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots) +{ + return sample_size * channels * tdm_slots; +} +EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size); + +int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params) +{ + int sample_size; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + case SNDRV_PCM_FORMAT_S16_BE: + sample_size = 16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + case SNDRV_PCM_FORMAT_S20_3BE: + sample_size = 20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S24_BE: + sample_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + case SNDRV_PCM_FORMAT_S32_BE: + sample_size = 32; + break; + default: + return -ENOTSUPP; + } + + return snd_soc_calc_frame_size(sample_size, params_channels(params), + 1); +} +EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size); + +int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots) +{ + return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots); +} +EXPORT_SYMBOL_GPL(snd_soc_calc_bclk); + +int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params) +{ + int ret; + + ret = snd_soc_params_to_frame_size(params); + + if (ret > 0) + return ret * params_rate(params); + else + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8db0374e10d..b074a594c59 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2893,7 +2893,9 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { - if (snd_usb_create_midi_interface(chip, iface, NULL) < 0) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j); continue; } @@ -3038,12 +3040,11 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &uaxx_ep }; - if (chip->usb_id == USB_ID(0x0582, 0x002b)) - return snd_usb_create_midi_interface(chip, iface, - &ua700_quirk); - else - return snd_usb_create_midi_interface(chip, iface, - &uaxx_quirk); + const struct snd_usb_audio_quirk *quirk = + chip->usb_id == USB_ID(0x0582, 0x002b) + ? &ua700_quirk : &uaxx_quirk; + return snd_usbmidi_create(chip->card, iface, + &chip->midi_list, quirk); } if (altsd->bNumEndpoints != 1) @@ -3370,6 +3371,13 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } +static int create_any_midi_quirk(struct snd_usb_audio *chip, + struct usb_interface *intf, + const struct snd_usb_audio_quirk *quirk) +{ + return snd_usbmidi_create(chip->card, intf, &chip->midi_list, quirk); +} + /* * audio-interface quirks * @@ -3387,14 +3395,14 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, static const quirk_func_t quirk_funcs[] = { [QUIRK_IGNORE_INTERFACE] = ignore_interface_quirk, [QUIRK_COMPOSITE] = create_composite_quirk, - [QUIRK_MIDI_STANDARD_INTERFACE] = snd_usb_create_midi_interface, - [QUIRK_MIDI_FIXED_ENDPOINT] = snd_usb_create_midi_interface, - [QUIRK_MIDI_YAMAHA] = snd_usb_create_midi_interface, - [QUIRK_MIDI_MIDIMAN] = snd_usb_create_midi_interface, - [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, - [QUIRK_MIDI_FASTLANE] = snd_usb_create_midi_interface, - [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, - [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, + [QUIRK_MIDI_STANDARD_INTERFACE] = create_any_midi_quirk, + [QUIRK_MIDI_FIXED_ENDPOINT] = create_any_midi_quirk, + [QUIRK_MIDI_YAMAHA] = create_any_midi_quirk, + [QUIRK_MIDI_MIDIMAN] = create_any_midi_quirk, + [QUIRK_MIDI_NOVATION] = create_any_midi_quirk, + [QUIRK_MIDI_FASTLANE] = create_any_midi_quirk, + [QUIRK_MIDI_EMAGIC] = create_any_midi_quirk, + [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index e9a3a9dca15..40ba8115fb8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -132,7 +132,6 @@ struct snd_usb_audio { int pcm_devs; struct list_head midi_list; /* list of midi interfaces */ - int next_midi_device; struct list_head mixer_list; /* list of mixer interfaces */ }; @@ -227,8 +226,10 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error); void snd_usb_mixer_disconnect(struct list_head *p); -int snd_usb_create_midi_interface(struct snd_usb_audio *chip, struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk); +int snd_usbmidi_create(struct snd_card *card, + struct usb_interface *iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk *quirk); void snd_usbmidi_input_stop(struct list_head* p); void snd_usbmidi_input_start(struct list_head* p); void snd_usbmidi_disconnect(struct list_head *p); diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 0eff19ceb7e..6e89b8368d9 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1,7 +1,7 @@ /* * usbmidi.c - ALSA USB MIDI driver * - * Copyright (c) 2002-2007 Clemens Ladisch + * Copyright (c) 2002-2009 Clemens Ladisch * All rights reserved. * * Based on the OSS usb-midi driver by NAGANO Daisuke, @@ -47,6 +47,7 @@ #include <linux/usb.h> #include <linux/wait.h> #include <sound/core.h> +#include <sound/control.h> #include <sound/rawmidi.h> #include <sound/asequencer.h> #include "usbaudio.h" @@ -101,7 +102,8 @@ struct usb_protocol_ops { }; struct snd_usb_midi { - struct snd_usb_audio *chip; + struct usb_device *dev; + struct snd_card *card; struct usb_interface *iface; const struct snd_usb_audio_quirk *quirk; struct snd_rawmidi *rmidi; @@ -109,13 +111,19 @@ struct snd_usb_midi { struct list_head list; struct timer_list error_timer; spinlock_t disc_lock; + struct mutex mutex; + u32 usb_id; + int next_midi_device; struct snd_usb_midi_endpoint { struct snd_usb_midi_out_endpoint *out; struct snd_usb_midi_in_endpoint *in; } endpoints[MIDI_MAX_ENDPOINTS]; unsigned long input_triggered; + unsigned int opened; unsigned char disconnected; + + struct snd_kcontrol *roland_load_ctl; }; struct snd_usb_midi_out_endpoint { @@ -255,7 +263,7 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb) } } - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; snd_usbmidi_submit_urb(urb, GFP_ATOMIC); } @@ -296,7 +304,7 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) unsigned long flags; spin_lock_irqsave(&ep->buffer_lock, flags); - if (ep->umidi->chip->shutdown) { + if (ep->umidi->disconnected) { spin_unlock_irqrestore(&ep->buffer_lock, flags); return; } @@ -312,7 +320,7 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint* ep) dump_urb("sending", urb->transfer_buffer, urb->transfer_buffer_length); - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; if (snd_usbmidi_submit_urb(urb, GFP_ATOMIC) < 0) break; ep->active_urbs |= 1 << urb_index; @@ -349,7 +357,7 @@ static void snd_usbmidi_error_timer(unsigned long data) if (in && in->error_resubmit) { in->error_resubmit = 0; for (j = 0; j < INPUT_URBS; ++j) { - in->urbs[j]->dev = umidi->chip->dev; + in->urbs[j]->dev = umidi->dev; snd_usbmidi_submit_urb(in->urbs[j], GFP_ATOMIC); } } @@ -369,7 +377,7 @@ static int send_bulk_static_data(struct snd_usb_midi_out_endpoint* ep, return -ENOMEM; dump_urb("sending", buf, len); if (ep->urbs[0].urb) - err = usb_bulk_msg(ep->umidi->chip->dev, ep->urbs[0].urb->pipe, + err = usb_bulk_msg(ep->umidi->dev, ep->urbs[0].urb->pipe, buf, len, NULL, 250); kfree(buf); return err; @@ -724,8 +732,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, if (!ep->ports[0].active) return; - count = snd_usb_get_speed(ep->umidi->chip->dev) == USB_SPEED_HIGH - ? 1 : 2; + count = snd_usb_get_speed(ep->umidi->dev) == USB_SPEED_HIGH ? 1 : 2; count = snd_rawmidi_transmit(ep->ports[0].substream, urb->transfer_buffer, count); @@ -879,6 +886,50 @@ static struct usb_protocol_ops snd_usbmidi_emagic_ops = { }; +static void update_roland_altsetting(struct snd_usb_midi* umidi) +{ + struct usb_interface *intf; + struct usb_host_interface *hostif; + struct usb_interface_descriptor *intfd; + int is_light_load; + + intf = umidi->iface; + is_light_load = intf->cur_altsetting != intf->altsetting; + if (umidi->roland_load_ctl->private_value == is_light_load) + return; + hostif = &intf->altsetting[umidi->roland_load_ctl->private_value]; + intfd = get_iface_desc(hostif); + snd_usbmidi_input_stop(&umidi->list); + usb_set_interface(umidi->dev, intfd->bInterfaceNumber, + intfd->bAlternateSetting); + snd_usbmidi_input_start(&umidi->list); +} + +static void substream_open(struct snd_rawmidi_substream *substream, int open) +{ + struct snd_usb_midi* umidi = substream->rmidi->private_data; + struct snd_kcontrol *ctl; + + mutex_lock(&umidi->mutex); + if (open) { + if (umidi->opened++ == 0 && umidi->roland_load_ctl) { + ctl = umidi->roland_load_ctl; + ctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(umidi->card, + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + update_roland_altsetting(umidi); + } + } else { + if (--umidi->opened == 0 && umidi->roland_load_ctl) { + ctl = umidi->roland_load_ctl; + ctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(umidi->card, + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + } + } + mutex_unlock(&umidi->mutex); +} + static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) { struct snd_usb_midi* umidi = substream->rmidi->private_data; @@ -898,11 +949,13 @@ static int snd_usbmidi_output_open(struct snd_rawmidi_substream *substream) } substream->runtime->private_data = port; port->state = STATE_UNKNOWN; + substream_open(substream, 1); return 0; } static int snd_usbmidi_output_close(struct snd_rawmidi_substream *substream) { + substream_open(substream, 0); return 0; } @@ -912,7 +965,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream, port->active = up; if (up) { - if (port->ep->umidi->chip->shutdown) { + if (port->ep->umidi->disconnected) { /* gobble up remaining bytes to prevent wait in * snd_rawmidi_drain_output */ while (!snd_rawmidi_transmit_empty(substream)) @@ -954,11 +1007,13 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) static int snd_usbmidi_input_open(struct snd_rawmidi_substream *substream) { + substream_open(substream, 1); return 0; } static int snd_usbmidi_input_close(struct snd_rawmidi_substream *substream) { + substream_open(substream, 0); return 0; } @@ -988,7 +1043,7 @@ static struct snd_rawmidi_ops snd_usbmidi_input_ops = { static void free_urb_and_buffer(struct snd_usb_midi *umidi, struct urb *urb, unsigned int buffer_length) { - usb_buffer_free(umidi->chip->dev, buffer_length, + usb_buffer_free(umidi->dev, buffer_length, urb->transfer_buffer, urb->transfer_dma); usb_free_urb(urb); } @@ -1035,24 +1090,24 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, } } if (ep_info->in_interval) - pipe = usb_rcvintpipe(umidi->chip->dev, ep_info->in_ep); + pipe = usb_rcvintpipe(umidi->dev, ep_info->in_ep); else - pipe = usb_rcvbulkpipe(umidi->chip->dev, ep_info->in_ep); - length = usb_maxpacket(umidi->chip->dev, pipe, 0); + pipe = usb_rcvbulkpipe(umidi->dev, ep_info->in_ep); + length = usb_maxpacket(umidi->dev, pipe, 0); for (i = 0; i < INPUT_URBS; ++i) { - buffer = usb_buffer_alloc(umidi->chip->dev, length, GFP_KERNEL, + buffer = usb_buffer_alloc(umidi->dev, length, GFP_KERNEL, &ep->urbs[i]->transfer_dma); if (!buffer) { snd_usbmidi_in_endpoint_delete(ep); return -ENOMEM; } if (ep_info->in_interval) - usb_fill_int_urb(ep->urbs[i], umidi->chip->dev, + usb_fill_int_urb(ep->urbs[i], umidi->dev, pipe, buffer, length, snd_usbmidi_in_urb_complete, ep, ep_info->in_interval); else - usb_fill_bulk_urb(ep->urbs[i], umidi->chip->dev, + usb_fill_bulk_urb(ep->urbs[i], umidi->dev, pipe, buffer, length, snd_usbmidi_in_urb_complete, ep); ep->urbs[i]->transfer_flags = URB_NO_TRANSFER_DMA_MAP; @@ -1062,15 +1117,6 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, return 0; } -static unsigned int snd_usbmidi_count_bits(unsigned int x) -{ - unsigned int bits; - - for (bits = 0; x; ++bits) - x &= x - 1; - return bits; -} - /* * Frees an output endpoint. * May be called when ep hasn't been initialized completely. @@ -1113,15 +1159,15 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, ep->urbs[i].ep = ep; } if (ep_info->out_interval) - pipe = usb_sndintpipe(umidi->chip->dev, ep_info->out_ep); + pipe = usb_sndintpipe(umidi->dev, ep_info->out_ep); else - pipe = usb_sndbulkpipe(umidi->chip->dev, ep_info->out_ep); - if (umidi->chip->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ + pipe = usb_sndbulkpipe(umidi->dev, ep_info->out_ep); + if (umidi->usb_id == USB_ID(0x0a92, 0x1020)) /* ESI M4U */ ep->max_transfer = 4; else - ep->max_transfer = usb_maxpacket(umidi->chip->dev, pipe, 1); + ep->max_transfer = usb_maxpacket(umidi->dev, pipe, 1); for (i = 0; i < OUTPUT_URBS; ++i) { - buffer = usb_buffer_alloc(umidi->chip->dev, + buffer = usb_buffer_alloc(umidi->dev, ep->max_transfer, GFP_KERNEL, &ep->urbs[i].urb->transfer_dma); if (!buffer) { @@ -1129,12 +1175,12 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi* umidi, return -ENOMEM; } if (ep_info->out_interval) - usb_fill_int_urb(ep->urbs[i].urb, umidi->chip->dev, + usb_fill_int_urb(ep->urbs[i].urb, umidi->dev, pipe, buffer, ep->max_transfer, snd_usbmidi_out_urb_complete, &ep->urbs[i], ep_info->out_interval); else - usb_fill_bulk_urb(ep->urbs[i].urb, umidi->chip->dev, + usb_fill_bulk_urb(ep->urbs[i].urb, umidi->dev, pipe, buffer, ep->max_transfer, snd_usbmidi_out_urb_complete, &ep->urbs[i]); @@ -1172,6 +1218,7 @@ static void snd_usbmidi_free(struct snd_usb_midi* umidi) if (ep->in) snd_usbmidi_in_endpoint_delete(ep->in); } + mutex_destroy(&umidi->mutex); kfree(umidi); } @@ -1367,7 +1414,7 @@ static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) int i; for (i = 0; i < ARRAY_SIZE(snd_usbmidi_port_info); ++i) { - if (snd_usbmidi_port_info[i].id == umidi->chip->usb_id && + if (snd_usbmidi_port_info[i].id == umidi->usb_id && snd_usbmidi_port_info[i].port == number) return &snd_usbmidi_port_info[i]; } @@ -1405,7 +1452,7 @@ static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, port_info = find_port_info(umidi, number); name_format = port_info ? port_info->name : "%s MIDI %d"; snprintf(substream->name, sizeof(substream->name), - name_format, umidi->chip->card->shortname, number + 1); + name_format, umidi->card->shortname, number + 1); *rsubstream = substream; } @@ -1503,7 +1550,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, endpoints[epidx].out_ep = usb_endpoint_num(ep); if (usb_endpoint_xfer_int(ep)) endpoints[epidx].out_interval = ep->bInterval; - else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW) + else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW) /* * Low speed bulk transfers don't exist, so * force interrupt transfers for devices like @@ -1523,7 +1570,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, endpoints[epidx].in_ep = usb_endpoint_num(ep); if (usb_endpoint_xfer_int(ep)) endpoints[epidx].in_interval = ep->bInterval; - else if (snd_usb_get_speed(umidi->chip->dev) == USB_SPEED_LOW) + else if (snd_usb_get_speed(umidi->dev) == USB_SPEED_LOW) endpoints[epidx].in_interval = 1; endpoints[epidx].in_cables = (1 << ms_ep->bNumEmbMIDIJack) - 1; snd_printdd(KERN_INFO "EP %02X: %d jack(s)\n", @@ -1533,6 +1580,52 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, return 0; } +static int roland_load_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + static const char *const names[] = { "High Load", "Light Load" }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item > 1) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int roland_load_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + value->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +static int roland_load_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) +{ + struct snd_usb_midi* umidi = kcontrol->private_data; + int changed; + + if (value->value.enumerated.item[0] > 1) + return -EINVAL; + mutex_lock(&umidi->mutex); + changed = value->value.enumerated.item[0] != kcontrol->private_value; + if (changed) + kcontrol->private_value = value->value.enumerated.item[0]; + mutex_unlock(&umidi->mutex); + return changed; +} + +static struct snd_kcontrol_new roland_load_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "MIDI Input Mode", + .info = roland_load_info, + .get = roland_load_get, + .put = roland_load_put, + .private_value = 1, +}; + /* * On Roland devices, use the second alternate setting to be able to use * the interrupt input endpoint. @@ -1556,8 +1649,12 @@ static void snd_usbmidi_switch_roland_altsetting(struct snd_usb_midi* umidi) snd_printdd(KERN_INFO "switching to altsetting %d with int ep\n", intfd->bAlternateSetting); - usb_set_interface(umidi->chip->dev, intfd->bInterfaceNumber, + usb_set_interface(umidi->dev, intfd->bInterfaceNumber, intfd->bAlternateSetting); + + umidi->roland_load_ctl = snd_ctl_new1(&roland_load_ctl, umidi); + if (snd_ctl_add(umidi->card, umidi->roland_load_ctl) < 0) + umidi->roland_load_ctl = NULL; } /* @@ -1573,7 +1670,7 @@ static int snd_usbmidi_detect_endpoints(struct snd_usb_midi* umidi, struct usb_endpoint_descriptor* epd; int i, out_eps = 0, in_eps = 0; - if (USB_ID_VENDOR(umidi->chip->usb_id) == 0x0582) + if (USB_ID_VENDOR(umidi->usb_id) == 0x0582) snd_usbmidi_switch_roland_altsetting(umidi); if (endpoint[0].out_ep || endpoint[0].in_ep) @@ -1760,12 +1857,12 @@ static int snd_usbmidi_create_rawmidi(struct snd_usb_midi* umidi, struct snd_rawmidi *rmidi; int err; - err = snd_rawmidi_new(umidi->chip->card, "USB MIDI", - umidi->chip->next_midi_device++, + err = snd_rawmidi_new(umidi->card, "USB MIDI", + umidi->next_midi_device++, out_ports, in_ports, &rmidi); if (err < 0) return err; - strcpy(rmidi->name, umidi->chip->card->shortname); + strcpy(rmidi->name, umidi->card->shortname); rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT | SNDRV_RAWMIDI_INFO_INPUT | SNDRV_RAWMIDI_INFO_DUPLEX; @@ -1804,7 +1901,7 @@ static void snd_usbmidi_input_start_ep(struct snd_usb_midi_in_endpoint* ep) return; for (i = 0; i < INPUT_URBS; ++i) { struct urb* urb = ep->urbs[i]; - urb->dev = ep->umidi->chip->dev; + urb->dev = ep->umidi->dev; snd_usbmidi_submit_urb(urb, GFP_KERNEL); } } @@ -1825,9 +1922,10 @@ void snd_usbmidi_input_start(struct list_head* p) /* * Creates and registers everything needed for a MIDI streaming interface. */ -int snd_usb_create_midi_interface(struct snd_usb_audio* chip, - struct usb_interface* iface, - const struct snd_usb_audio_quirk* quirk) +int snd_usbmidi_create(struct snd_card *card, + struct usb_interface* iface, + struct list_head *midi_list, + const struct snd_usb_audio_quirk* quirk) { struct snd_usb_midi* umidi; struct snd_usb_midi_endpoint_info endpoints[MIDI_MAX_ENDPOINTS]; @@ -1837,12 +1935,16 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, umidi = kzalloc(sizeof(*umidi), GFP_KERNEL); if (!umidi) return -ENOMEM; - umidi->chip = chip; + umidi->dev = interface_to_usbdev(iface); + umidi->card = card; umidi->iface = iface; umidi->quirk = quirk; umidi->usb_protocol_ops = &snd_usbmidi_standard_ops; init_timer(&umidi->error_timer); spin_lock_init(&umidi->disc_lock); + mutex_init(&umidi->mutex); + umidi->usb_id = USB_ID(le16_to_cpu(umidi->dev->descriptor.idVendor), + le16_to_cpu(umidi->dev->descriptor.idProduct)); umidi->error_timer.function = snd_usbmidi_error_timer; umidi->error_timer.data = (unsigned long)umidi; @@ -1851,7 +1953,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, switch (quirk ? quirk->type : QUIRK_MIDI_STANDARD_INTERFACE) { case QUIRK_MIDI_STANDARD_INTERFACE: err = snd_usbmidi_get_ms_info(umidi, endpoints); - if (chip->usb_id == USB_ID(0x0763, 0x0150)) /* M-Audio Uno */ + if (umidi->usb_id == USB_ID(0x0763, 0x0150)) /* M-Audio Uno */ umidi->usb_protocol_ops = &snd_usbmidi_maudio_broken_running_status_ops; break; @@ -1887,7 +1989,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, * interface 0, so we have to make sure that the USB core looks * again at interface 0 by calling usb_set_interface() on it. */ - usb_set_interface(umidi->chip->dev, 0, 0); + usb_set_interface(umidi->dev, 0, 0); err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; case QUIRK_MIDI_EMAGIC: @@ -1914,8 +2016,8 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, out_ports = 0; in_ports = 0; for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) { - out_ports += snd_usbmidi_count_bits(endpoints[i].out_cables); - in_ports += snd_usbmidi_count_bits(endpoints[i].in_cables); + out_ports += hweight16(endpoints[i].out_cables); + in_ports += hweight16(endpoints[i].in_cables); } err = snd_usbmidi_create_rawmidi(umidi, out_ports, in_ports); if (err < 0) { @@ -1933,14 +2035,14 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, return err; } - list_add(&umidi->list, &umidi->chip->midi_list); + list_add_tail(&umidi->list, midi_list); for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) snd_usbmidi_input_start_ep(umidi->endpoints[i].in); return 0; } -EXPORT_SYMBOL(snd_usb_create_midi_interface); +EXPORT_SYMBOL(snd_usbmidi_create); EXPORT_SYMBOL(snd_usbmidi_input_stop); EXPORT_SYMBOL(snd_usbmidi_input_start); EXPORT_SYMBOL(snd_usbmidi_disconnect); diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 3e5d66cf1f5..77c35885e21 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -277,6 +277,22 @@ static struct usbmix_name_map scratch_live_map[] = { { 0 } /* terminator */ }; +/* "Gamesurround Muse Pocket LT" looks same like "Sound Blaster MP3+" + * most importand difference is SU[8], it should be set to "Capture Source" + * to make alsamixer and PA working properly. + * FIXME: or mp3plus_map should use "Capture Source" too, + * so this maps can be merget + */ +static struct usbmix_name_map hercules_usb51_map[] = { + { 8, "Capture Source" }, /* SU, default "PCM Capture Source" */ + { 9, "Master Playback" }, /* FU, default "Speaker Playback" */ + { 10, "Mic Boost", 7 }, /* FU, default "Auto Gain Input" */ + { 11, "Line Capture" }, /* FU, default "PCM Capture" */ + { 13, "Mic Bypass Playback" }, /* FU, default "Mic Playback" */ + { 14, "Line Bypass Playback" }, /* FU, default "Line Playback" */ + { 0 } /* terminator */ +}; + /* * Control map entries */ @@ -316,6 +332,13 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .ignore_ctl_error = 1, }, { + /* Hercules Gamesurround Muse Pocket LT + * (USB 5.1 Channel Audio Adapter) + */ + .id = USB_ID(0x06f8, 0xc000), + .map = hercules_usb51_map, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f6f201eb24c..a892bda03df 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1563,6 +1563,29 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* has ID 0x00ea when not in Advanced Driver mode */ + USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e9), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "Roland", */ + /* .product_name = "UA-1G", */ + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, /* Guillemot devices */ { diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 99f33766cd5..f71cd28eca6 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -59,11 +59,33 @@ static int us122l_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_US122L, .data = &quirk_data }; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 1); - return snd_usb_create_midi_interface(&US122L(card)->chip, - iface, &quirk); + return snd_usbmidi_create(card, iface, + &US122L(card)->midi_list, &quirk); +} + +static int us144_create_usbmidi(struct snd_card *card) +{ + static struct snd_usb_midi_endpoint_info quirk_data = { + .out_ep = 4, + .in_ep = 3, + .out_cables = 0x001, + .in_cables = 0x001 + }; + static struct snd_usb_audio_quirk quirk = { + .vendor_name = "US144", + .product_name = NAME_ALLCAPS, + .ifnum = 0, + .type = QUIRK_MIDI_US122L, + .data = &quirk_data + }; + struct usb_device *dev = US122L(card)->dev; + struct usb_interface *iface = usb_ifnum_to_if(dev, 0); + + return snd_usbmidi_create(card, iface, + &US122L(card)->midi_list, &quirk); } /* @@ -171,7 +193,12 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) if (!us122l->first) us122l->first = file; - iface = usb_ifnum_to_if(us122l->chip.dev, 1); + + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->dev, 0); + usb_autopm_get_interface(iface); + } + iface = usb_ifnum_to_if(us122l->dev, 1); usb_autopm_get_interface(iface); return 0; } @@ -179,8 +206,14 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file) { struct us122l *us122l = hw->private_data; - struct usb_interface *iface = usb_ifnum_to_if(us122l->chip.dev, 1); + struct usb_interface *iface; snd_printdd(KERN_DEBUG "%p %p\n", hw, file); + + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + iface = usb_ifnum_to_if(us122l->dev, 0); + usb_autopm_put_interface(iface); + } + iface = usb_ifnum_to_if(us122l->dev, 1); usb_autopm_put_interface(iface); if (us122l->first == file) us122l->first = NULL; @@ -264,7 +297,7 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, static void us122l_stop(struct us122l *us122l) { struct list_head *p; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_stop(p); usb_stream_stop(&us122l->sk); @@ -297,7 +330,7 @@ static bool us122l_start(struct us122l *us122l, unsigned use_packsize = 0; bool success = false; - if (us122l->chip.dev->speed == USB_SPEED_HIGH) { + if (us122l->dev->speed == USB_SPEED_HIGH) { /* The us-122l's descriptor defaults to iso max_packsize 78, which isn't needed for samplerates <= 48000. Lets save some memory: @@ -314,11 +347,11 @@ static bool us122l_start(struct us122l *us122l, break; } } - if (!usb_stream_new(&us122l->sk, us122l->chip.dev, 1, 2, + if (!usb_stream_new(&us122l->sk, us122l->dev, 1, 2, rate, use_packsize, period_frames, 6)) goto out; - err = us122l_set_sample_rate(us122l->chip.dev, rate); + err = us122l_set_sample_rate(us122l->dev, rate); if (err < 0) { us122l_stop(us122l); snd_printk(KERN_ERR "us122l_set_sample_rate error \n"); @@ -330,7 +363,7 @@ static bool us122l_start(struct us122l *us122l, snd_printk(KERN_ERR "us122l_start error %i \n", err); goto out; } - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_start(p); success = true; out: @@ -357,7 +390,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, err = -ENXIO; goto free; } - high_speed = us122l->chip.dev->speed == USB_SPEED_HIGH; + high_speed = us122l->dev->speed == USB_SPEED_HIGH; if ((cfg->sample_rate != 44100 && cfg->sample_rate != 48000 && (!high_speed || (cfg->sample_rate != 88200 && cfg->sample_rate != 96000))) || @@ -417,7 +450,7 @@ static int usb_stream_hwdep_new(struct snd_card *card) { int err; struct snd_hwdep *hw; - struct usb_device *dev = US122L(card)->chip.dev; + struct usb_device *dev = US122L(card)->dev; err = snd_hwdep_new(card, SND_USB_STREAM_ID, 0, &hw); if (err < 0) @@ -443,19 +476,29 @@ static bool us122l_create_card(struct snd_card *card) int err; struct us122l *us122l = US122L(card); - err = usb_set_interface(us122l->chip.dev, 1, 1); + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->dev, 0, 1); + if (err) { + snd_printk(KERN_ERR "usb_set_interface error \n"); + return false; + } + } + err = usb_set_interface(us122l->dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); return false; } - pt_info_set(us122l->chip.dev, 0x11); - pt_info_set(us122l->chip.dev, 0x10); + pt_info_set(us122l->dev, 0x11); + pt_info_set(us122l->dev, 0x10); if (!us122l_start(us122l, 44100, 256)) return false; - err = us122l_create_usbmidi(card); + if (us122l->dev->descriptor.idProduct == USB_ID_US144) + err = us144_create_usbmidi(card); + else + err = us122l_create_usbmidi(card); if (err < 0) { snd_printk(KERN_ERR "us122l_create_usbmidi error %i \n", err); us122l_stop(us122l); @@ -465,7 +508,7 @@ static bool us122l_create_card(struct snd_card *card) if (err < 0) { /* release the midi resources */ struct list_head *p; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_disconnect(p); us122l_stop(us122l); @@ -477,7 +520,7 @@ static bool us122l_create_card(struct snd_card *card) static void snd_us122l_free(struct snd_card *card) { struct us122l *us122l = US122L(card); - int index = us122l->chip.index; + int index = us122l->card_index; if (index >= 0 && index < SNDRV_CARDS) snd_us122l_card_used[index] = 0; } @@ -497,13 +540,12 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) sizeof(struct us122l), &card); if (err < 0) return err; - snd_us122l_card_used[US122L(card)->chip.index = dev] = 1; + snd_us122l_card_used[US122L(card)->card_index = dev] = 1; card->private_free = snd_us122l_free; - US122L(card)->chip.dev = device; - US122L(card)->chip.card = card; + US122L(card)->dev = device; mutex_init(&US122L(card)->mutex); init_waitqueue_head(&US122L(card)->sk.sleep); - INIT_LIST_HEAD(&US122L(card)->chip.midi_list); + INIT_LIST_HEAD(&US122L(card)->midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)", @@ -511,8 +553,8 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) le16_to_cpu(device->descriptor.idVendor), le16_to_cpu(device->descriptor.idProduct), 0, - US122L(card)->chip.dev->bus->busnum, - US122L(card)->chip.dev->devnum + US122L(card)->dev->bus->busnum, + US122L(card)->dev->devnum ); *cardp = card; return 0; @@ -542,6 +584,7 @@ static int us122l_usb_probe(struct usb_interface *intf, return err; } + usb_get_intf(usb_ifnum_to_if(device, 0)); usb_get_dev(device); *cardp = card; return 0; @@ -550,9 +593,16 @@ static int us122l_usb_probe(struct usb_interface *intf, static int snd_us122l_probe(struct usb_interface *intf, const struct usb_device_id *id) { + struct usb_device *device = interface_to_usbdev(intf); struct snd_card *card; int err; + if (device->descriptor.idProduct == USB_ID_US144 + && device->speed == USB_SPEED_HIGH) { + snd_printk(KERN_ERR "disable ehci-hcd to run US-144 \n"); + return -ENODEV; + } + snd_printdd(KERN_DEBUG"%p:%i\n", intf, intf->cur_altsetting->desc.bInterfaceNumber); if (intf->cur_altsetting->desc.bInterfaceNumber != 1) @@ -584,15 +634,15 @@ static void snd_us122l_disconnect(struct usb_interface *intf) mutex_lock(&us122l->mutex); us122l_stop(us122l); mutex_unlock(&us122l->mutex); - us122l->chip.shutdown = 1; /* release the midi resources */ - list_for_each(p, &us122l->chip.midi_list) { + list_for_each(p, &us122l->midi_list) { snd_usbmidi_disconnect(p); } - usb_put_intf(intf); - usb_put_dev(us122l->chip.dev); + usb_put_intf(usb_ifnum_to_if(us122l->dev, 0)); + usb_put_intf(usb_ifnum_to_if(us122l->dev, 1)); + usb_put_dev(us122l->dev); while (atomic_read(&us122l->mmap_count)) msleep(500); @@ -615,7 +665,7 @@ static int snd_us122l_suspend(struct usb_interface *intf, pm_message_t message) if (!us122l) return 0; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_stop(p); mutex_lock(&us122l->mutex); @@ -642,16 +692,23 @@ static int snd_us122l_resume(struct usb_interface *intf) mutex_lock(&us122l->mutex); /* needed, doesn't restart without: */ - err = usb_set_interface(us122l->chip.dev, 1, 1); + if (us122l->dev->descriptor.idProduct == USB_ID_US144) { + err = usb_set_interface(us122l->dev, 0, 1); + if (err) { + snd_printk(KERN_ERR "usb_set_interface error \n"); + goto unlock; + } + } + err = usb_set_interface(us122l->dev, 1, 1); if (err) { snd_printk(KERN_ERR "usb_set_interface error \n"); goto unlock; } - pt_info_set(us122l->chip.dev, 0x11); - pt_info_set(us122l->chip.dev, 0x10); + pt_info_set(us122l->dev, 0x11); + pt_info_set(us122l->dev, 0x10); - err = us122l_set_sample_rate(us122l->chip.dev, + err = us122l_set_sample_rate(us122l->dev, us122l->sk.s->cfg.sample_rate); if (err < 0) { snd_printk(KERN_ERR "us122l_set_sample_rate error \n"); @@ -661,7 +718,7 @@ static int snd_us122l_resume(struct usb_interface *intf) if (err) goto unlock; - list_for_each(p, &us122l->chip.midi_list) + list_for_each(p, &us122l->midi_list) snd_usbmidi_input_start(p); unlock: mutex_unlock(&us122l->mutex); @@ -675,11 +732,11 @@ static struct usb_device_id snd_us122l_usb_id_table[] = { .idVendor = 0x0644, .idProduct = USB_ID_US122L }, -/* { */ /* US-144 maybe works when @USB1.1. Untested. */ -/* .match_flags = USB_DEVICE_ID_MATCH_DEVICE, */ -/* .idVendor = 0x0644, */ -/* .idProduct = USB_ID_US144 */ -/* }, */ + { /* US-144 only works at USB1.1! Disable module ehci-hcd. */ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = 0x0644, + .idProduct = USB_ID_US144 + }, { /* terminator */ } }; diff --git a/sound/usb/usx2y/us122l.h b/sound/usb/usx2y/us122l.h index 3d10c4b2a0f..4daf1982e82 100644 --- a/sound/usb/usx2y/us122l.h +++ b/sound/usb/usx2y/us122l.h @@ -3,7 +3,8 @@ struct us122l { - struct snd_usb_audio chip; + struct usb_device *dev; + int card_index; int stride; struct usb_stream_kernel sk; @@ -12,6 +13,7 @@ struct us122l { unsigned second_periods_polled; struct file *master; struct file *slave; + struct list_head midi_list; atomic_t mmap_count; }; diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 52e04b2f35d..1879b72c40f 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -114,7 +114,7 @@ static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, struct usX2Ydev *us428 = hw->private_data; int id = -1; - switch (le16_to_cpu(us428->chip.dev->descriptor.idProduct)) { + switch (le16_to_cpu(us428->dev->descriptor.idProduct)) { case USB_ID_US122: id = USX2Y_TYPE_122; break; @@ -164,14 +164,14 @@ static int usX2Y_create_usbmidi(struct snd_card *card) .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &quirk_data_2 }; - struct usb_device *dev = usX2Y(card)->chip.dev; + struct usb_device *dev = usX2Y(card)->dev; struct usb_interface *iface = usb_ifnum_to_if(dev, 0); struct snd_usb_audio_quirk *quirk = le16_to_cpu(dev->descriptor.idProduct) == USB_ID_US428 ? &quirk_2 : &quirk_1; snd_printdd("usX2Y_create_usbmidi \n"); - return snd_usb_create_midi_interface(&usX2Y(card)->chip, iface, quirk); + return snd_usbmidi_create(card, iface, &usX2Y(card)->midi_list, quirk); } static int usX2Y_create_alsa_devices(struct snd_card *card) @@ -202,7 +202,7 @@ static int snd_usX2Y_hwdep_dsp_load(struct snd_hwdep *hw, snd_printdd( "dsp_load %s\n", dsp->name); if (access_ok(VERIFY_READ, dsp->image, dsp->length)) { - struct usb_device* dev = priv->chip.dev; + struct usb_device* dev = priv->dev; char *buf; buf = memdup_user(dsp->image, dsp->length); diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index cb4bb8373ca..c42350eed2e 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -239,8 +239,8 @@ static void i_usX2Y_In04Int(struct urb *urb) for (j = 0; j < URBS_AsyncSeq && !err; ++j) if (0 == usX2Y->AS04.urb[j]->status) { struct us428_p4out *p4out = us428ctls->p4out + send; // FIXME if more than 1 p4out is new, 1 gets lost. - usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->chip.dev, - usb_sndbulkpipe(usX2Y->chip.dev, 0x04), &p4out->val.vol, + usb_fill_bulk_urb(usX2Y->AS04.urb[j], usX2Y->dev, + usb_sndbulkpipe(usX2Y->dev, 0x04), &p4out->val.vol, p4out->type == eLT_Light ? sizeof(struct us428_lights) : 5, i_usX2Y_Out04Int, usX2Y); err = usb_submit_urb(usX2Y->AS04.urb[j], GFP_ATOMIC); @@ -253,7 +253,7 @@ static void i_usX2Y_In04Int(struct urb *urb) if (err) snd_printk(KERN_ERR "In04Int() usb_submit_urb err=%i\n", err); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; usb_submit_urb(urb, GFP_ATOMIC); } @@ -273,8 +273,8 @@ int usX2Y_AsyncSeq04_init(struct usX2Ydev *usX2Y) err = -ENOMEM; break; } - usb_fill_bulk_urb( usX2Y->AS04.urb[i], usX2Y->chip.dev, - usb_sndbulkpipe(usX2Y->chip.dev, 0x04), + usb_fill_bulk_urb( usX2Y->AS04.urb[i], usX2Y->dev, + usb_sndbulkpipe(usX2Y->dev, 0x04), usX2Y->AS04.buffer + URB_DataLen_AsyncSeq*i, 0, i_usX2Y_Out04Int, usX2Y ); @@ -293,7 +293,7 @@ int usX2Y_In04_init(struct usX2Ydev *usX2Y) } init_waitqueue_head(&usX2Y->In04WaitQueue); - usb_fill_int_urb(usX2Y->In04urb, usX2Y->chip.dev, usb_rcvintpipe(usX2Y->chip.dev, 0x4), + usb_fill_int_urb(usX2Y->In04urb, usX2Y->dev, usb_rcvintpipe(usX2Y->dev, 0x4), usX2Y->In04Buf, 21, i_usX2Y_In04Int, usX2Y, 10); @@ -348,13 +348,12 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) sizeof(struct usX2Ydev), &card); if (err < 0) return err; - snd_usX2Y_card_used[usX2Y(card)->chip.index = dev] = 1; + snd_usX2Y_card_used[usX2Y(card)->card_index = dev] = 1; card->private_free = snd_usX2Y_card_private_free; - usX2Y(card)->chip.dev = device; - usX2Y(card)->chip.card = card; + usX2Y(card)->dev = device; init_waitqueue_head(&usX2Y(card)->prepare_wait_queue); mutex_init(&usX2Y(card)->prepare_mutex); - INIT_LIST_HEAD(&usX2Y(card)->chip.midi_list); + INIT_LIST_HEAD(&usX2Y(card)->midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); sprintf(card->longname, "%s (%x:%x if %d at %03d/%03d)", @@ -362,7 +361,7 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) le16_to_cpu(device->descriptor.idVendor), le16_to_cpu(device->descriptor.idProduct), 0,//us428(card)->usbmidi.ifnum, - usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum + usX2Y(card)->dev->bus->busnum, usX2Y(card)->dev->devnum ); *cardp = card; return 0; @@ -432,8 +431,8 @@ static void snd_usX2Y_card_private_free(struct snd_card *card) usb_free_urb(usX2Y(card)->In04urb); if (usX2Y(card)->us428ctls_sharedmem) snd_free_pages(usX2Y(card)->us428ctls_sharedmem, sizeof(*usX2Y(card)->us428ctls_sharedmem)); - if (usX2Y(card)->chip.index >= 0 && usX2Y(card)->chip.index < SNDRV_CARDS) - snd_usX2Y_card_used[usX2Y(card)->chip.index] = 0; + if (usX2Y(card)->card_index >= 0 && usX2Y(card)->card_index < SNDRV_CARDS) + snd_usX2Y_card_used[usX2Y(card)->card_index] = 0; } /* @@ -445,13 +444,12 @@ static void usX2Y_usb_disconnect(struct usb_device *device, void* ptr) struct snd_card *card = ptr; struct usX2Ydev *usX2Y = usX2Y(card); struct list_head *p; - usX2Y->chip.shutdown = 1; usX2Y->chip_status = USX2Y_STAT_CHIP_HUP; usX2Y_unlinkSeq(&usX2Y->AS04); usb_kill_urb(usX2Y->In04urb); snd_card_disconnect(card); /* release the midi resources */ - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_disconnect(p); } if (usX2Y->us428ctls_sharedmem) diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 456b5fdbc33..1d174cea352 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -22,7 +22,8 @@ struct snd_usX2Y_urbSeq { #include "usx2yhwdeppcm.h" struct usX2Ydev { - struct snd_usb_audio chip; + struct usb_device *dev; + int card_index; int stride; struct urb *In04urb; void *In04Buf; @@ -42,6 +43,9 @@ struct usX2Ydev { struct snd_usX2Y_substream *subs[4]; struct snd_usX2Y_substream * volatile prepare_subs; wait_queue_head_t prepare_wait_queue; + struct list_head midi_list; + struct list_head pcm_list; + int pcm_devs; }; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 9efd27f6b52..74a67a85aa8 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -199,7 +199,7 @@ static int usX2Y_urb_submit(struct snd_usX2Y_substream *subs, struct urb *urb, i return -ENODEV; urb->start_frame = (frame + NRURBS * nr_of_packs()); // let hcd do rollover sanity checks urb->hcpriv = NULL; - urb->dev = subs->usX2Y->chip.dev; /* we need to set this at each time */ + urb->dev = subs->usX2Y->dev; /* we need to set this at each time */ if ((err = usb_submit_urb(urb, GFP_ATOMIC)) < 0) { snd_printk(KERN_ERR "usb_submit_urb() returned %i\n", err); return err; @@ -300,7 +300,7 @@ static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" "Most propably some urb of usb-frame %i is still missing.\n" "Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); usX2Y_clients_stop(usX2Y); @@ -313,7 +313,7 @@ static void i_usX2Y_urb_complete(struct urb *urb) if (unlikely(atomic_read(&subs->state) < state_PREPARED)) { snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", urb->status, urb->start_frame); return; @@ -424,7 +424,7 @@ static int usX2Y_urbs_allocate(struct snd_usX2Y_substream *subs) int i; unsigned int pipe; int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; - struct usb_device *dev = subs->usX2Y->chip.dev; + struct usb_device *dev = subs->usX2Y->dev; pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) : usb_rcvisocpipe(dev, subs->endpoint); @@ -500,7 +500,7 @@ static int usX2Y_urbs_start(struct snd_usX2Y_substream *subs) unsigned long pack; if (0 == i) atomic_set(&subs->state, state_STARTING3); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * pack; @@ -692,7 +692,7 @@ static int usX2Y_rate_set(struct usX2Ydev *usX2Y, int rate) } ((char*)(usbdata + i))[0] = ra[i].c1; ((char*)(usbdata + i))[1] = ra[i].c2; - usb_fill_bulk_urb(us->urb[i], usX2Y->chip.dev, usb_sndbulkpipe(usX2Y->chip.dev, 4), + usb_fill_bulk_urb(us->urb[i], usX2Y->dev, usb_sndbulkpipe(usX2Y->dev, 4), usbdata + i, 2, i_usX2Y_04Int, usX2Y); #ifdef OLD_USB us->urb[i]->transfer_flags = USB_QUEUE_BULK; @@ -740,17 +740,17 @@ static int usX2Y_format_set(struct usX2Ydev *usX2Y, snd_pcm_format_t format) alternate = 1; usX2Y->stride = 4; } - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_input_stop(p); } usb_kill_urb(usX2Y->In04urb); - if ((err = usb_set_interface(usX2Y->chip.dev, 0, alternate))) { + if ((err = usb_set_interface(usX2Y->dev, 0, alternate))) { snd_printk(KERN_ERR "usb_set_interface error \n"); return err; } - usX2Y->In04urb->dev = usX2Y->chip.dev; + usX2Y->In04urb->dev = usX2Y->dev; err = usb_submit_urb(usX2Y->In04urb, GFP_KERNEL); - list_for_each(p, &usX2Y->chip.midi_list) { + list_for_each(p, &usX2Y->midi_list) { snd_usbmidi_input_start(p); } usX2Y->format = format; @@ -955,7 +955,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, struct snd_pcm *pcm; int err, i; struct snd_usX2Y_substream **usX2Y_substream = - usX2Y(card)->subs + 2 * usX2Y(card)->chip.pcm_devs; + usX2Y(card)->subs + 2 * usX2Y(card)->pcm_devs; for (i = playback_endpoint ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; i <= SNDRV_PCM_STREAM_CAPTURE; ++i) { @@ -971,7 +971,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, usX2Y_substream[SNDRV_PCM_STREAM_PLAYBACK]->endpoint = playback_endpoint; usX2Y_substream[SNDRV_PCM_STREAM_CAPTURE]->endpoint = capture_endpoint; - err = snd_pcm_new(card, NAME_ALLCAPS" Audio", usX2Y(card)->chip.pcm_devs, + err = snd_pcm_new(card, NAME_ALLCAPS" Audio", usX2Y(card)->pcm_devs, playback_endpoint ? 1 : 0, 1, &pcm); if (err < 0) { @@ -987,7 +987,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, pcm->private_free = snd_usX2Y_pcm_private_free; pcm->info_flags = 0; - sprintf(pcm->name, NAME_ALLCAPS" Audio #%d", usX2Y(card)->chip.pcm_devs); + sprintf(pcm->name, NAME_ALLCAPS" Audio #%d", usX2Y(card)->pcm_devs); if ((playback_endpoint && 0 > (err = snd_pcm_lib_preallocate_pages(pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream, @@ -1001,7 +1001,7 @@ static int usX2Y_audio_stream_new(struct snd_card *card, int playback_endpoint, snd_usX2Y_pcm_private_free(pcm); return err; } - usX2Y(card)->chip.pcm_devs++; + usX2Y(card)->pcm_devs++; return 0; } @@ -1013,14 +1013,14 @@ int usX2Y_audio_create(struct snd_card *card) { int err = 0; - INIT_LIST_HEAD(&usX2Y(card)->chip.pcm_list); + INIT_LIST_HEAD(&usX2Y(card)->pcm_list); if (0 > (err = usX2Y_audio_stream_new(card, 0xA, 0x8))) return err; - if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) == USB_ID_US428) + if (le16_to_cpu(usX2Y(card)->dev->descriptor.idProduct) == USB_ID_US428) if (0 > (err = usX2Y_audio_stream_new(card, 0, 0xA))) return err; - if (le16_to_cpu(usX2Y(card)->chip.dev->descriptor.idProduct) != USB_ID_US122) + if (le16_to_cpu(usX2Y(card)->dev->descriptor.idProduct) != USB_ID_US122) err = usX2Y_rate_set(usX2Y(card), 44100); // Lets us428 recognize output-volume settings, disturbs us122. return err; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 4b2304c2e02..9ed6c3956ca 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -234,7 +234,7 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) if (unlikely(atomic_read(&subs->state) < state_PREPARED)) { snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usX2Y->chip.dev), + usb_get_current_frame_number(usX2Y->dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", urb->status, urb->start_frame); return; @@ -318,7 +318,7 @@ static int usX2Y_usbpcm_urbs_allocate(struct snd_usX2Y_substream *subs) int i; unsigned int pipe; int is_playback = subs == subs->usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; - struct usb_device *dev = subs->usX2Y->chip.dev; + struct usb_device *dev = subs->usX2Y->dev; pipe = is_playback ? usb_sndisocpipe(dev, subs->endpoint) : usb_rcvisocpipe(dev, subs->endpoint); @@ -441,7 +441,7 @@ static int usX2Y_usbpcm_urbs_start(struct snd_usX2Y_substream *subs) unsigned long pack; if (0 == u) atomic_set(&subs->state, state_STARTING3); - urb->dev = usX2Y->chip.dev; + urb->dev = usX2Y->dev; urb->transfer_flags = URB_ISO_ASAP; for (pack = 0; pack < nr_of_packs(); pack++) { urb->iso_frame_desc[pack].offset = subs->maxpacksize * (pack + u * nr_of_packs()); @@ -741,7 +741,7 @@ int usX2Y_hwdep_pcm_new(struct snd_card *card) int err; struct snd_hwdep *hw; struct snd_pcm *pcm; - struct usb_device *dev = usX2Y(card)->chip.dev; + struct usb_device *dev = usX2Y(card)->dev; if (1 != nr_of_packs()) return 0; |