diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2016-05-19 13:41:32 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2016-05-19 13:41:32 -0700 |
| commit | f4c80d5a16eb4b08a0d9ade154af1ebdc63f5752 (patch) | |
| tree | 5334acabf48210285333bc80d4a3e326efb36750 /sound | |
| parent | 7afd16f882887c9adc69cd1794f5e57777723217 (diff) | |
| parent | 17e1717c11a34f9b0956e33e0c4a4e4ae8c51a57 (diff) | |
Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This time was again a relatively calm development cycle; most of
updates are about drivers, and no radical changes are seen in any core
code. Here are some highlights:
ALSA core:
- Continued hardening of ALSA hrtimer
- A few leak fixes in timer interface
- Fix poll error handling in PCM and compress
- Add error propagation in compress API
- Removal of dead rtctimer driver
HD-audio:
- Native ELD notify support for i915 HDMI
- Realtek ALC234 & co support
- Code refactoring to standardize chmap support
- Continued development for SKL HDMI core support
Firewire:
- Apply delayed card registration to all drivers
- Improved / stabilized the handling of PCM stream start / stop
- Add tracepoints to dump a part of isochronous packet data
- Fixed incoming/outgoing packet parameter usages
- Add support for M-Audio profire series
USB-audio:
- Fixes for UAC2 clock source
- SS+ support
- Workaround for oft-seen repeated sample rate read errors
ASoC:
- Further slow progress on the topology code
- Substantial updates and improvements for the da7219, es8328,
fsl-ssi, Intel and rcar drivers.
- Compress error handling in WM ADSP driver"
* tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits)
ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type
sound: oss: Use setup_timer and mod_timer.
ASoC: hdac_hdmi: Remove the unused 'timeout' variable
ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex.
ASoC: fsl_ssi: Fix channel slipping in Playback at startup
ASoC: fsl_ssi: Fix samples being dropped at Playback startup
ASoC: fsl_ssi: Save a dev reference for dev_err() purpose.
ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk.
ASoC: fsl_ssi: Real hardware channels max number is 32
ASoC: pcm5102a: Add support for PCM5102A codec
ASoC: hdac_hdmi: add link management
ASoC: Intel: Skylake: add link management
ALSA: hdac: add link pm and ref counting
ALSA: au88x0: Fix zero clear of stream->resources
ASoC: rt298: Add DMI match for Broxton-P reference platform
ASoC: rt298: fix null deref on acpi driver data
ASoC: dapm: deprecate MICBIAS widget type
ALSA: firewire-lib: drop skip argument from helper functions to queue a packet
ALSA: firewire-lib: add context information to tracepoints
ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity
...
Diffstat (limited to 'sound')
125 files changed, 4505 insertions, 1492 deletions
diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 6d12ca9bcb80..9749f9e8b45c 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -141,35 +141,6 @@ config SND_SEQ_HRTIMER_DEFAULT Say Y here to use the HR-timer backend as the default sequencer timer. -config SND_RTCTIMER - tristate "RTC Timer support" - depends on RTC - select SND_TIMER - help - Say Y here to enable RTC timer support for ALSA. ALSA uses - the RTC timer as a precise timing source and maps the RTC - timer to ALSA's timer interface. The ALSA sequencer code also - can use this timing source. - - To compile this driver as a module, choose M here: the module - will be called snd-rtctimer. - - Note that this option is exclusive with the new RTC drivers - (CONFIG_RTC_CLASS) since this requires the old API. - -config SND_SEQ_RTCTIMER_DEFAULT - bool "Use RTC as default sequencer timer" - depends on SND_RTCTIMER && SND_SEQUENCER - depends on !SND_SEQ_HRTIMER_DEFAULT - default y - help - Say Y here to use the RTC timer as the default sequencer - timer. This is strongly recommended because it ensures - precise MIDI timing even when the system timer runs at less - than 1000 Hz. - - If in doubt, say Y. - config SND_DYNAMIC_MINORS bool "Dynamic device file minor numbers" help diff --git a/sound/core/Makefile b/sound/core/Makefile index 48ab4b8f8279..e85d9dd12c2d 100644 --- a/sound/core/Makefile +++ b/sound/core/Makefile @@ -37,7 +37,6 @@ obj-$(CONFIG_SND) += snd.o obj-$(CONFIG_SND_HWDEP) += snd-hwdep.o obj-$(CONFIG_SND_TIMER) += snd-timer.o obj-$(CONFIG_SND_HRTIMER) += snd-hrtimer.o -obj-$(CONFIG_SND_RTCTIMER) += snd-rtctimer.o obj-$(CONFIG_SND_PCM) += snd-pcm.o obj-$(CONFIG_SND_DMAENGINE_PCM) += snd-pcm-dmaengine.o obj-$(CONFIG_SND_RAWMIDI) += snd-rawmidi.o diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a9933c07a6bf..9b3334be9df2 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,9 +288,12 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); /* write is allowed when stream is running or has been steup */ - if (stream->runtime->state != SNDRV_PCM_STATE_SETUP && - stream->runtime->state != SNDRV_PCM_STATE_PREPARED && - stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_SETUP: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_RUNNING: + break; + default: mutex_unlock(&stream->device->lock); return -EBADFD; } @@ -391,14 +394,13 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) int retval = 0; if (snd_BUG_ON(!data)) - return -EFAULT; + return POLLERR; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; mutex_lock(&stream->device->lock); if (stream->runtime->state == SNDRV_PCM_STATE_OPEN) { - retval = -EBADFD; + retval = snd_compr_get_poll(stream) | POLLERR; goto out; } poll_wait(f, &stream->runtime->sleep, wait); @@ -421,10 +423,7 @@ static unsigned int snd_compr_poll(struct file *f, poll_table *wait) retval = snd_compr_get_poll(stream); break; default: - if (stream->direction == SND_COMPRESS_PLAYBACK) - retval = POLLOUT | POLLWRNORM | POLLERR; - else - retval = POLLIN | POLLRDNORM | POLLERR; + retval = snd_compr_get_poll(stream) | POLLERR; break; } out: @@ -802,9 +801,9 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) if (snd_BUG_ON(!data)) return -EFAULT; + stream = &data->stream; - if (snd_BUG_ON(!stream)) - return -EFAULT; + mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 656d9a9032dc..e2f27022b363 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -38,37 +38,53 @@ static unsigned int resolution; struct snd_hrtimer { struct snd_timer *timer; struct hrtimer hrt; - atomic_t running; + bool in_callback; }; static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) { struct snd_hrtimer *stime = container_of(hrt, struct snd_hrtimer, hrt); struct snd_timer *t = stime->timer; - unsigned long oruns; - - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - - oruns = hrtimer_forward_now(hrt, ns_to_ktime(t->sticks * resolution)); - snd_timer_interrupt(stime->timer, t->sticks * oruns); + ktime_t delta; + unsigned long ticks; + enum hrtimer_restart ret = HRTIMER_NORESTART; + + spin_lock(&t->lock); + if (!t->running) + goto out; /* fast path */ + stime->in_callback = true; + ticks = t->sticks; + spin_unlock(&t->lock); + + /* calculate the drift */ + delta = ktime_sub(hrt->base->get_time(), hrtimer_get_expires(hrt)); + if (delta.tv64 > 0) + ticks += ktime_divns(delta, ticks * resolution); + + snd_timer_interrupt(stime->timer, ticks); + + spin_lock(&t->lock); + if (t->running) { + hrtimer_add_expires_ns(hrt, t->sticks * resolution); + ret = HRTIMER_RESTART; + } - if (!atomic_read(&stime->running)) - return HRTIMER_NORESTART; - return HRTIMER_RESTART; + stime->in_callback = false; + out: + spin_unlock(&t->lock); + return ret; } static int snd_hrtimer_open(struct snd_timer *t) { struct snd_hrtimer *stime; - stime = kmalloc(sizeof(*stime), GFP_KERNEL); + stime = kzalloc(sizeof(*stime), GFP_KERNEL); if (!stime) return -ENOMEM; hrtimer_init(&stime->hrt, CLOCK_MONOTONIC, HRTIMER_MODE_REL); stime->timer = t; stime->hrt.function = snd_hrtimer_callback; - atomic_set(&stime->running, 0); t->private_data = stime; return 0; } @@ -78,6 +94,11 @@ static int snd_hrtimer_close(struct snd_timer *t) struct snd_hrtimer *stime = t->private_data; if (stime) { + spin_lock_irq(&t->lock); + t->running = 0; /* just to be sure */ + stime->in_callback = 1; /* skip start/stop */ + spin_unlock_irq(&t->lock); + hrtimer_cancel(&stime->hrt); kfree(stime); t->private_data = NULL; @@ -89,18 +110,19 @@ static int snd_hrtimer_start(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - atomic_set(&stime->running, 0); - hrtimer_try_to_cancel(&stime->hrt); + if (stime->in_callback) + return 0; hrtimer_start(&stime->hrt, ns_to_ktime(t->sticks * resolution), HRTIMER_MODE_REL); - atomic_set(&stime->running, 1); return 0; } static int snd_hrtimer_stop(struct snd_timer *t) { struct snd_hrtimer *stime = t->private_data; - atomic_set(&stime->running, 0); + + if (stime->in_callback) + return 0; hrtimer_try_to_cancel(&stime->hrt); return 0; } diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 697c166acf05..8eb58c709b14 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); * direction of the substream. If the substream is a playback stream the dst * fields will be initialized, if it is a capture stream the src fields will be * initialized. The {dst,src}_addr_width field will only be initialized if the - * addr_width field of the DAI DMA data struct is not equal to - * DMA_SLAVE_BUSWIDTH_UNDEFINED. + * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of + * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If + * both conditions are met the latter takes priority. */ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_pcm_substream *substream, @@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data( if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->dst_addr = dma_data->addr; slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->dst_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->dst_addr_width = dma_data->addr_width; } else { slave_config->src_addr = dma_data->addr; slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->src_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->src_addr_width = dma_data->addr_width; } diff --git a/sound/core/pcm_iec958.c b/sound/core/pcm_iec958.c index 36b2d7aca1bd..5e6aed64f451 100644 --- a/sound/core/pcm_iec958.c +++ b/sound/core/pcm_iec958.c @@ -9,30 +9,18 @@ #include <linux/types.h> #include <sound/asoundef.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include <sound/pcm_iec958.h> -/** - * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status - * @runtime: pcm runtime structure with ->rate filled in - * @cs: channel status buffer, at least four bytes - * @len: length of channel status buffer - * - * Create the consumer format channel status data in @cs of maximum size - * @len corresponding to the parameters of the PCM runtime @runtime. - * - * Drivers may wish to tweak the contents of the buffer after creation. - * - * Returns: length of buffer, or negative error code if something failed. - */ -int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, - size_t len) +static int create_iec958_consumer(uint rate, uint sample_width, + u8 *cs, size_t len) { unsigned int fs, ws; if (len < 4) return -EINVAL; - switch (runtime->rate) { + switch (rate) { case 32000: fs = IEC958_AES3_CON_FS_32000; break; @@ -59,7 +47,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, } if (len > 4) { - switch (snd_pcm_format_width(runtime->format)) { + switch (sample_width) { case 16: ws = IEC958_AES4_CON_WORDLEN_20_16; break; @@ -71,6 +59,7 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, IEC958_AES4_CON_MAX_WORDLEN_24; break; case 24: + case 32: /* Assume 24-bit width for 32-bit samples. */ ws = IEC958_AES4_CON_WORDLEN_24_20 | IEC958_AES4_CON_MAX_WORDLEN_24; break; @@ -92,4 +81,46 @@ int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, return len; } + +/** + * snd_pcm_create_iec958_consumer - create consumer format IEC958 channel status + * @runtime: pcm runtime structure with ->rate filled in + * @cs: channel status buffer, at least four bytes + * @len: length of channel status buffer + * + * Create the consumer format channel status data in @cs of maximum size + * @len corresponding to the parameters of the PCM runtime @runtime. + * + * Drivers may wish to tweak the contents of the buffer after creation. + * + * Returns: length of buffer, or negative error code if something failed. + */ +int snd_pcm_create_iec958_consumer(struct snd_pcm_runtime *runtime, u8 *cs, + size_t len) +{ + return create_iec958_consumer(runtime->rate, + snd_pcm_format_width(runtime->format), + cs, len); +} EXPORT_SYMBOL(snd_pcm_create_iec958_consumer); + +/** + * snd_pcm_create_iec958_consumer_hw_params - create IEC958 channel status + * @hw_params: the hw_params instance for extracting rate and sample format + * @cs: channel status buffer, at least four bytes + * @len: length of channel status buffer + * + * Create the consumer format channel status data in @cs of maximum size + * @len corresponding to the parameters of the PCM runtime @runtime. + * + * Drivers may wish to tweak the contents of the buffer after creation. + * + * Returns: length of buffer, or negative error code if something failed. + */ +int snd_pcm_create_iec958_consumer_hw_params(struct snd_pcm_hw_params *params, + u8 *cs, size_t len) +{ + return create_iec958_consumer(params_rate(params), params_width(params), + cs, len); +} +EXPORT_SYMBOL(snd_pcm_create_iec958_consumer_hw_params); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 3a9b66c6e09c..bb1261591a1f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1886,8 +1886,8 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_timer_interrupt(substream->timer, 1); #endif _end: - snd_pcm_stream_unlock_irqrestore(substream, flags); kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_pcm_stream_unlock_irqrestore(substream, flags); } EXPORT_SYMBOL(snd_pcm_period_elapsed); @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 9106d8e2300e..c61fd50f771f 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3161,7 +3161,7 @@ static unsigned int snd_pcm_playback_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLOUT | POLLWRNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); @@ -3200,7 +3200,7 @@ static unsigned int snd_pcm_capture_poll(struct file *file, poll_table * wait) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) - return -ENXIO; + return POLLIN | POLLRDNORM | POLLERR; runtime = substream->runtime; poll_wait(file, &runtime->sleep, wait); diff --git a/sound/core/rtctimer.c b/sound/core/rtctimer.c deleted file mode 100644 index f3420d11a12f..000000000000 --- a/sound/core/rtctimer.c +++ /dev/null @@ -1,187 +0,0 @@ -/* - * RTC based high-frequency timer - * - * Copyright (C) 2000 Takashi Iwai - * based on rtctimer.c by Steve Ratcliffe - * - * 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/init.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/log2.h> -#include <sound/core.h> -#include <sound/timer.h> - -#if IS_ENABLED(CONFIG_RTC) - -#include <linux/mc146818rtc.h> - -#define RTC_FREQ 1024 /* default frequency */ -#define NANO_SEC 1000000000L /* 10^9 in sec */ - -/* - * prototypes - */ -static int rtctimer_open(struct snd_timer *t); -static int rtctimer_close(struct snd_timer *t); -static int rtctimer_start(struct snd_timer *t); -static int rtctimer_stop(struct snd_timer *t); - - -/* - * The hardware dependent description for this timer. - */ -static struct snd_timer_hardware rtc_hw = { - .flags = SNDRV_TIMER_HW_AUTO | - SNDRV_TIMER_HW_FIRST | - SNDRV_TIMER_HW_TASKLET, - .ticks = 100000000L, /* FIXME: XXX */ - .open = rtctimer_open, - .close = rtctimer_close, - .start = rtctimer_start, - .stop = rtctimer_stop, -}; - -static int rtctimer_freq = RTC_FREQ; /* frequency */ -static struct snd_timer *rtctimer; -static struct tasklet_struct rtc_tasklet; -static rtc_task_t rtc_task; - - -static int -rtctimer_open(struct snd_timer *t) -{ - int err; - - err = rtc_register(&rtc_task); - if (err < 0) - return err; - t->private_data = &rtc_task; - return 0; -} - -static int -rtctimer_close(struct snd_timer *t) -{ - rtc_task_t *rtc = t->private_data; - if (rtc) { - rtc_unregister(rtc); - tasklet_kill(&rtc_tasklet); - t->private_data = NULL; - } - return 0; -} - -static int -rtctimer_start(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_IRQP_SET, rtctimer_freq); - rtc_control(rtc, RTC_PIE_ON, 0); - return 0; -} - -static int -rtctimer_stop(struct snd_timer *timer) -{ - rtc_task_t *rtc = timer->private_data; - if (snd_BUG_ON(!rtc)) - return -EINVAL; - rtc_control(rtc, RTC_PIE_OFF, 0); - return 0; -} - -static void rtctimer_tasklet(unsigned long data) -{ - snd_timer_interrupt((struct snd_timer *)data, 1); -} - -/* - * interrupt - */ -static void rtctimer_interrupt(void *private_data) -{ - tasklet_schedule(private_data); -} - - -/* - * ENTRY functions - */ -static int __init rtctimer_init(void) -{ - int err; - struct snd_timer *timer; - - if (rtctimer_freq < 2 || rtctimer_freq > 8192 || - !is_power_of_2(rtctimer_freq)) { - pr_err("ALSA: rtctimer: invalid frequency %d\n", rtctimer_freq); - return -EINVAL; - } - - /* Create a new timer and set up the fields */ - err = snd_timer_global_new("rtc", SNDRV_TIMER_GLOBAL_RTC, &timer); - if (err < 0) - return err; - - timer->module = THIS_MODULE; - strcpy(timer->name, "RTC timer"); - timer->hw = rtc_hw; - timer->hw.resolution = NANO_SEC / rtctimer_freq; - - tasklet_init(&rtc_tasklet, rtctimer_tasklet, (unsigned long)timer); - - /* set up RTC callback */ - rtc_task.func = rtctimer_interrupt; - rtc_task.private_data = &rtc_tasklet; - - err = snd_timer_global_register(timer); - if (err < 0) { - snd_timer_global_free(timer); - return err; - } - rtctimer = timer; /* remember this */ - - return 0; -} - -static void __exit rtctimer_exit(void) -{ - if (rtctimer) { - snd_timer_global_free(rtctimer); - rtctimer = NULL; - } -} - - -/* - * exported stuff - */ -module_init(rtctimer_init) -module_exit(rtctimer_exit) - -module_param(rtctimer_freq, int, 0444); -MODULE_PARM_DESC(rtctimer_freq, "timer frequency in Hz"); - -MODULE_LICENSE("GPL"); - -MODULE_ALIAS("snd-timer-" __stringify(SNDRV_TIMER_GLOBAL_RTC)); - -#endif /* IS_ENABLED(CONFIG_RTC) */ diff --git a/sound/core/seq/seq.c b/sound/core/seq/seq.c index 7e0aabb808a6..639544b4fb04 100644 --- a/sound/core/seq/seq.c +++ b/sound/core/seq/seq.c @@ -47,8 +47,6 @@ int seq_default_timer_card = -1; int seq_default_timer_device = #ifdef CONFIG_SND_SEQ_HRTIMER_DEFAULT SNDRV_TIMER_GLOBAL_HRTIMER -#elif defined(CONFIG_SND_SEQ_RTCTIMER_DEFAULT) - SNDRV_TIMER_GLOBAL_RTC #else SNDRV_TIMER_GLOBAL_SYSTEM #endif diff --git a/sound/core/timer.c b/sound/core/timer.c index 6469bedda2f3..e722022d325d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -37,8 +37,6 @@ #if IS_ENABLED(CONFIG_SND_HRTIMER) #define DEFAULT_TIMER_LIMIT 4 -#elif IS_ENABLED(CONFIG_SND_RTCTIMER) -#define DEFAULT_TIMER_LIMIT 2 #else #define DEFAULT_TIMER_LIMIT 1 #endif @@ -1225,6 +1223,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, tu->tstamp = *tstamp; if ((tu->filter & (1 << event)) == 0 || !tu->tread) return; + memset(&r1, 0, sizeof(r1)); r1.event = event; r1.tstamp = *tstamp; r1.val = resolution; @@ -1267,6 +1266,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, } if ((tu->filter & (1 << SNDRV_TIMER_EVENT_RESOLUTION)) && tu->last_resolution != resolution) { + memset(&r1, 0, sizeof(r1)); r1.event = SNDRV_TIMER_EVENT_RESOLUTION; r1.tstamp = tstamp; r1.val = resolution; @@ -1739,6 +1739,7 @@ static int snd_timer_user_params(struct file *file, if (tu->timeri->flags & SNDRV_TIMER_IFLG_EARLY_EVENT) { if (tu->tread) { struct snd_timer_tread tread; + memset(&tread, 0, sizeof(tread)); tread.event = SNDRV_TIMER_EVENT_EARLY; tread.tstamp.tv_sec = 0; tread.tstamp.tv_nsec = 0; diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 2a779c2f63ab..ab894ed1ff67 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -134,6 +134,7 @@ config SND_FIREWIRE_TASCAM Say Y here to include support for TASCAM. * FW-1884 * FW-1082 + * FW-1804 To compile this driver as a module, choose M here: the module will be called snd-firewire-tascam. diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile index 003c09029786..0ee1fb115d88 100644 --- a/sound/firewire/Makefile +++ b/sound/firewire/Makefile @@ -1,3 +1,6 @@ +# To find a header included by define_trace.h. +CFLAGS_amdtp-stream.o := -I$(src) + snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \ fcp.o cmp.o amdtp-stream.o amdtp-am824.o snd-isight-objs := isight.o diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h new file mode 100644 index 000000000000..9c04faf206b2 --- /dev/null +++ b/sound/firewire/amdtp-stream-trace.h @@ -0,0 +1,110 @@ +/* + * amdtp-stream-trace.h - tracepoint definitions to dump a part of packet data + * + * Copyright (c) 2016 Takashi Sakamoto + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#undef TRACE_SYSTEM +#define TRACE_SYSTEM snd_firewire_lib + +#if !defined(_AMDTP_STREAM_TRACE_H) || defined(TRACE_HEADER_MULTI_READ) +#define _AMDTP_STREAM_TRACE_H + +#include <linux/tracepoint.h> + +TRACE_EVENT(in_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_quadlets, index), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(unsigned int, irq) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->node_id; + __entry->dest = fw_parent_device(s->unit)->card->node_id; + __entry->cip_header0 = cip_header[0]; + __entry->cip_header1 = cip_header[1]; + __entry->payload_quadlets = payload_quadlets; + __entry->packet_index = s->packet_index; + __entry->irq = !!in_interrupt(); + __entry->index = index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, + __entry->index) +); + +TRACE_EVENT(out_packet, + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, index), + TP_STRUCT__entry( + __field(unsigned int, second) + __field(unsigned int, cycle) + __field(int, channel) + __field(int, src) + __field(int, dest) + __field(u32, cip_header0) + __field(u32, cip_header1) + __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(unsigned int, irq) + __field(unsigned int, index) + ), + TP_fast_assign( + __entry->second = cycles / CYCLES_PER_SECOND; + __entry->cycle = cycles % CYCLES_PER_SECOND; + __entry->channel = s->context->channel; + __entry->src = fw_parent_device(s->unit)->card->node_id; + __entry->dest = fw_parent_device(s->unit)->node_id; + __entry->cip_header0 = be32_to_cpu(cip_header[0]); + __entry->cip_header1 = be32_to_cpu(cip_header[1]); + __entry->payload_quadlets = payload_length / 4; + __entry->packet_index = s->packet_index; + __entry->irq = !!in_interrupt(); + __entry->index = index; + ), + TP_printk( + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", + __entry->second, + __entry->cycle, + __entry->src, + __entry->dest, + __entry->channel, + __entry->cip_header0, + __entry->cip_header1, + __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, + __entry->index) +); + +#endif + +#undef TRACE_INCLUDE_PATH +#define TRACE_INCLUDE_PATH . +#undef TRACE_INCLUDE_FILE +#define TRACE_INCLUDE_FILE amdtp-stream-trace +#include <trace/define_trace.h> diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ed2902609a4c..00060c4a9deb 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -19,6 +19,10 @@ #define CYCLES_PER_SECOND 8000 #define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND) +/* Always support Linux tracing subsystem. */ +#define CREATE_TRACE_POINTS +#include "amdtp-stream-trace.h" + #define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */ /* isochronous header parameters */ @@ -87,7 +91,6 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, init_waitqueue_head(&s->callback_wait); s->callbacked = false; - s->sync_slave = NULL; s->fmt = fmt; s->process_data_blocks = process_data_blocks; @@ -102,6 +105,10 @@ EXPORT_SYMBOL(amdtp_stream_init); */ void amdtp_stream_destroy(struct amdtp_stream *s) { + /* Not initialized. */ + if (s->protocol == NULL) + return; + WARN_ON(amdtp_stream_running(s)); kfree(s->protocol); mutex_destroy(&s->mutex); @@ -244,7 +251,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -349,7 +355,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -363,9 +368,8 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, - unsigned int header_length, - unsigned int payload_length, bool skip) +static int queue_packet(struct amdtp_stream *s, unsigned int header_length, + unsigned int payload_length) { struct fw_iso_packet p = {0}; int err = 0; @@ -376,8 +380,10 @@ static int queue_packet(struct amdtp_stream *s, p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); p.tag = TAG_CIP; p.header_length = header_length; - p.payload_length = (!skip) ? payload_length : 0; - p.skip = skip; + if (payload_length > 0) + p.payload_length = payload_length; + else + p.skip = true; err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer, s->buffer.packets[s->packet_index].offset); if (err < 0) { @@ -392,27 +398,30 @@ end: } static inline int queue_out_packet(struct amdtp_stream *s, - unsigned int payload_length, bool skip) + unsigned int payload_length) { - return queue_packet(s, OUT_PACKET_HEADER_SIZE, - payload_length, skip); + return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length); } static inline int queue_in_packet(struct amdtp_stream *s) { return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s), false); + amdtp_stream_get_max_payload(s)); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, - unsigned int syt) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, + unsigned int index) { __be32 *buffer; + unsigned int syt; + unsigned int data_blocks; unsigned int payload_length; unsigned int pcm_frames; struct snd_pcm_substream *pcm; buffer = s->buffer.packets[s->packet_index].buffer; + syt = calculate_syt(s, cycle); + data_blocks = calculate_data_blocks(s, syt); pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | @@ -424,9 +433,11 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, (syt & CIP_SYT_MASK)); s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; - payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; - if (queue_out_packet(s, payload_length, false) < 0) + + trace_out_packet(s, cycle, buffer, payload_length, index); + + if (queue_out_packet(s, payload_length) < 0) return -EIO; pcm = ACCESS_ONCE(s->pcm); @@ -438,19 +449,24 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks, } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, __be32 *buffer, - unsigned int *data_blocks, unsigned int syt) + unsigned int payload_quadlets, unsigned int cycle, + unsigned int index) { + __be32 *buffer; u32 cip_header[2]; - unsigned int fmt, fdf; + unsigned int fmt, fdf, syt; unsigned int data_block_quadlets, data_block_counter, dbc_interval; + unsigned int data_blocks; struct snd_pcm_substream *pcm; unsigned int pcm_frames; bool lost; + buffer = s->buffer.packets[s->packet_index].buffer; cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); + trace_in_packet(s, cycle, cip_header, payload_quadlets, index); + /* * This module supports 'Two-quadlet CIP header with SYT field'. * For convenience, also check FMT field is AM824 or not. @@ -460,7 +476,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Invalid CIP header for AMDTP: %08X:%08X\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -471,7 +487,7 @@ static int handle_in_packet(struct amdtp_stream *s, dev_info_ratelimited(&s->unit->device, "Detect unexpected protocol: %08x %08x\n", cip_header[0], cip_header[1]); - *data_blocks = 0; + data_blocks = 0; pcm_frames = 0; goto end; } @@ -480,7 +496,7 @@ static int handle_in_packet(struct amdtp_stream *s, fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT; if (payload_quadlets < 3 || (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) { - *data_blocks = 0; + data_blocks = 0; } else { data_block_quadlets = (cip_header[0] & CIP_DBS_MASK) >> CIP_DBS_SHIFT; @@ -494,12 +510,12 @@ static int handle_in_packet(struct amdtp_stream *s, if (s->flags & CIP_WRONG_DBS) data_block_quadlets = s->data_block_quadlets; - *data_blocks = (payload_quadlets - 2) / data_block_quadlets; + data_blocks = (payload_quadlets - 2) / data_block_quadlets; } /* Check data block counter continuity */ data_block_counter = cip_header[0] & CIP_DBC_MASK; - if (*data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && + if (data_blocks == 0 && (s->flags & CIP_EMPTY_HAS_WRONG_DBC) && s->data_block_counter != UINT_MAX) data_block_counter = s->data_block_counter; @@ -510,10 +526,10 @@ static int handle_in_packet(struct amdtp_stream *s, } else if (!(s->flags & CIP_DBC_IS_END_EVENT)) { lost = data_block_counter != s->data_block_counter; } else { - if ((*data_blocks > 0) && (s->tx_dbc_interval > 0)) + if (data_blocks > 0 && s->tx_dbc_interval > 0) dbc_interval = s->tx_dbc_interval; else - dbc_interval = *data_blocks; + dbc_interval = data_blocks; lost = data_block_counter != ((s->data_block_counter + dbc_interval) & 0xff); @@ -526,13 +542,14 @@ static int handle_in_packet(struct amdtp_stream *s, return -EIO; } - pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt); + syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; + pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt); if (s->flags & CIP_DBC_IS_END_EVENT) s->data_block_counter = data_block_counter; else s->data_block_counter = - (data_block_counter + *data_blocks) & 0xff; + (data_block_counter + data_blocks) & 0xff; end: if (queue_in_packet(s) < 0) return -EIO; @@ -544,29 +561,50 @@ end: return 0; } -static void out_stream_callback(struct fw_iso_context *context, u32 cycle, +/* + * In CYCLE_TIMER register of IEEE 1394, 7 bits are used to represent second. On + * the other hand, in DMA descriptors of 1394 OHCI, 3 bits are used to represent + * it. Thus, via Linux firewire subsystem, we can get the 3 bits for second. + */ +static inline u32 compute_cycle_count(u32 tstamp) +{ + return (((tstamp >> 13) & 0x07) * 8000) + (tstamp & 0x1fff); +} + +static inline u32 increment_cycle_count(u32 cycle, unsigned int addend) +{ + cycle += addend; + if (cycle >= 8 * CYCLES_PER_SECOND) + cycle -= 8 * CYCLES_PER_SECOND; + return cycle; +} + +static inline u32 decrement_cycle_count(u32 cycle, unsigned int subtrahend) +{ + if (cycle < subtrahend) + cycle += 8 * CYCLES_PER_SECOND; + return cycle - subtrahend; +} + +static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int i, syt, packets = header_length / 4; - unsigned int data_blocks; + unsigned int i, packets = header_length / 4; + u32 cycle; if (s->packet_index < 0) return; - /* - * Compute the cycle of the last queued packet. - * (We need only the four lowest bits for the SYT, so we can ignore - * that bits 0-11 must wrap around at 3072.) - */ - cycle += QUEUE_LENGTH - packets; + cycle = compute_cycle_count(tstamp); - for (i = 0; i < packets; ++i) { - syt = calculate_syt(s, ++cycle); - data_blocks = calculate_data_blocks(s, syt); + /* Align to actual cycle count for the last packet. */ + cycle = increment_cycle_count(cycle, QUEUE_LENGTH - packets); - if (handle_out_packet(s, data_blocks, syt) < 0) { + for (i = 0; i < packets; ++i) { + cycle = increment_cycle_count(cycle, 1); + if (handle_out_packet(s, cycle, i) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -576,15 +614,15 @@ static void out_stream_callback(struct fw_iso_context *context, u32 cycle, fw_iso_context_queue_flush(s->context); } -static void in_stream_callback(struct fw_iso_context *context, u32 cycle, +static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; - unsigned int p, syt, packets; + unsigned int i, packets; unsigned int payload_quadlets, max_payload_quadlets; - unsigned int data_blocks; - __be32 *buffer, *headers = header; + __be32 *headers = header; + u32 cycle; if (s->packet_index < 0) return; @@ -592,70 +630,44 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle, /* The number of packets in buffer */ packets = header_length / IN_PACKET_HEADER_SIZE; + cycle = compute_cycle_count(tstamp); + + /* Align to actual cycle count for the last packet. */ + cycle = decrement_cycle_count(cycle, packets); + /* For buffer-over-run prevention. */ max_payload_quadlets = amdtp_stream_get_max_payload(s) / 4; - for (p = 0; p < packets; p++) { - buffer = s->buffer.packets[s->packet_index].buffer; + for (i = 0; i < packets; i++) { + cycle = increment_cycle_count(cycle, 1); /* The number of quadlets in this packet */ payload_quadlets = - (be32_to_cpu(headers[p]) >> ISO_DATA_LENGTH_SHIFT) / 4; + (be32_to_cpu(headers[i]) >> ISO_DATA_LENGTH_SHIFT) / 4; if (payload_quadlets > max_payload_quadlets) { dev_err(&s->unit->device, "Detect jumbo payload: %02x %02x\n", payload_quadlets, max_payload_quadlets); - s->packet_index = -1; break; } - syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK; - if (handle_in_packet(s, payload_quadlets, buffer, - &data_blocks, syt) < 0) { - s->packet_index = -1; + if (handle_in_packet(s, payload_quadlets, cycle, i) < 0) break; - } - - /* Process sync slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) { - if (handle_out_packet(s->sync_slave, - data_blocks, syt) < 0) { - s->packet_index = -1; - break; - } - } } - /* Queueing error or detecting discontinuity */ - if (s->packet_index < 0) { + /* Queueing error or detecting invalid payload. */ + if (i < packets) { + s->packet_index = -1; amdtp_stream_pcm_abort(s); - - /* Abort sync slave. */ - if (s->sync_slave) { - s->sync_slave->packet_index = -1; - amdtp_stream_pcm_abort(s->sync_slave); - } return; } - /* when sync to device, flush the packets for slave stream */ - if (s->sync_slave && s->sync_slave->callbacked) - fw_iso_context_queue_flush(s->sync_slave->context); - fw_iso_context_queue_flush(s->context); } -/* processing is done by master callback */ -static void slave_stream_callback(struct fw_iso_context *context, u32 cycle, - size_t header_length, void *header, - void *private_data) -{ - return; -} - /* this is executed one time */ static void amdtp_stream_first_callback(struct fw_iso_context *context, - u32 cycle, size_t header_length, + u32 tstamp, size_t header_length, void *header, void *private_data) { struct amdtp_stream *s = private_data; @@ -669,12 +681,10 @@ static void amdtp_stream_first_callback(struct fw_iso_context *context, if (s->direction == AMDTP_IN_STREAM) context->callback.sc = in_stream_callback; - else if (s->flags & CIP_SYNC_TO_DEVICE) - context->callback.sc = slave_stream_callback; else context->callback.sc = out_stream_callback; - context->callback.sc(context, cycle, header_length, header, s); + context->callback.sc(context, tstamp, header_length, header, s); } /** @@ -713,8 +723,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) goto err_unlock; } - if (s->direction == AMDTP_IN_STREAM && - s->flags & CIP_SKIP_INIT_DBC_CHECK) + if (s->direction == AMDTP_IN_STREAM) s->data_block_counter = UINT_MAX; else s->data_block_counter = 0; @@ -755,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) err = queue_in_packet(s); else - err = queue_out_packet(s, 0, true); + err = queue_out_packet(s, 0); if (err < 0) goto err_context; } while (s->packet_index > 0); @@ -794,11 +803,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets will be already + * queued in the buffer. These packets should be handled immediately + * to keep better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8775704a3665..c1bc7fad056e 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -17,8 +17,6 @@ * @CIP_BLOCKING: In blocking mode, each packet contains either zero or * SYT_INTERVAL samples, with these two types alternating so that * the overall sample rate comes out right. - * @CIP_SYNC_TO_DEVICE: In sync to device mode, time stamp in out packets is - * generated by in packets. Defaultly this driver generates timestamp. * @CIP_EMPTY_WITH_TAG0: Only for in-stream. Empty in-packets have TAG0. * @CIP_DBC_IS_END_EVENT: Only for in-stream. The value of dbc in an in-packet * corresponds to the end of event in the packet. Out of IEC 61883. @@ -26,8 +24,6 @@ * The value of data_block_quadlets is used instead of reported value. * @CIP_SKIP_DBC_ZERO_CHECK: Only for in-stream. Packets with zero in dbc is * skipped for detecting discontinuity. - * @CIP_SKIP_INIT_DBC_CHECK: Only for in-stream. The value of dbc in first - * packet is not continuous from an initial value. * @CIP_EMPTY_HAS_WRONG_DBC: Only for in-stream. The value of dbc in empty * packet is wrong but the others are correct. * @CIP_JUMBO_PAYLOAD: Only for in-stream. The number of data blocks in an @@ -37,14 +33,12 @@ enum cip_flags { CIP_NONBLOCKING = 0x00, CIP_BLOCKING = 0x01, - CIP_SYNC_TO_DEVICE = 0x02, - CIP_EMPTY_WITH_TAG0 = 0x04, - CIP_DBC_IS_END_EVENT = 0x08, - CIP_WRONG_DBS = 0x10, - CIP_SKIP_DBC_ZERO_CHECK = 0x20, - CIP_SKIP_INIT_DBC_CHECK = 0x40, - CIP_EMPTY_HAS_WRONG_DBC = 0x80, - CIP_JUMBO_PAYLOAD = 0x100, + CIP_EMPTY_WITH_TAG0 = 0x02, + CIP_DBC_IS_END_EVENT = 0x04, + CIP_WRONG_DBS = 0x08, + CIP_SKIP_DBC_ZERO_CHECK = 0x10, + CIP_EMPTY_HAS_WRONG_DBC = 0x20, + CIP_JUMBO_PAYLOAD = 0x40, }; /** @@ -132,12 +126,10 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; wait_queue_head_t callback_wait; - struct amdtp_stream *sync_slave; /* For backends to process data blocks. */ void *protocol; @@ -223,23 +215,6 @@ static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc) return sfc & 1; } -static inline void amdtp_stream_set_sync(enum cip_flags sync_mode, - struct amdtp_stream *master, - struct amdtp_stream *slave) -{ - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master->flags |= CIP_SYNC_TO_DEVICE; - slave->flags |= CIP_SYNC_TO_DEVICE; - master->sync_slave = slave; - } else { - master->flags &= ~CIP_SYNC_TO_DEVICE; - slave->flags &= ~CIP_SYNC_TO_DEVICE; - master->sync_slave = NULL; - } - - slave->sync_slave = NULL; -} - /** * amdtp_stream_wait_callback - sleep till callbacked or timeout * @s: the AMDTP stream diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 3e4e0756e3fe..f7e2cbd2a313 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -67,7 +67,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define MODEL_MAUDIO_PROJECTMIX 0x00010091 static int -name_device(struct snd_bebob *bebob, unsigned int vendor_id) +name_device(struct snd_bebob *bebob) { struct fw_device *fw_dev = fw_parent_device(bebob->unit); char vendor[24] = {0}; @@ -126,6 +126,17 @@ end: return err; } +static void bebob_free(struct snd_bebob *bebob) +{ + snd_bebob_stream_destroy_duplex(bebob); + fw_unit_put(bebob->unit); + + kfree(bebob->maudio_special_quirk); + + mutex_destroy(&bebob->mutex); + kfree(bebob); +} + /* * This module releases the FireWire unit data after all ALSA character devices * are released by applications. This is for releasing stream data or finishing @@ -137,18 +148,11 @@ bebob_card_free(struct snd_card *card) { struct snd_bebob *bebob = card->private_data; - snd_bebob_stream_destroy_duplex(bebob); - fw_unit_put(bebob->unit); - - kfree(bebob->maudio_special_quirk); - - if (bebob->card_index >= 0) { - mutex_lock(&devices_mutex); - clear_bit(bebob->card_index, devices_used); - mutex_unlock(&devices_mutex); - } + mutex_lock(&devices_mutex); + clear_bit(bebob->card_index, devices_used); + mutex_unlock(&devices_mutex); - mutex_destroy(&bebob->mutex); + bebob_free(card->private_data); } static const struct snd_bebob_spec * @@ -176,16 +180,17 @@ check_audiophile_booted(struct fw_unit *unit) return strncmp(name, "FW Audiophile Bootloader", 15) != 0; } -static int -bebob_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void +do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_bebob *bebob; - const struct snd_bebob_spec *spec; + struct snd_bebob *bebob = + container_of(work, struct snd_bebob, dwork.work); unsigned int card_index; int err; + if (bebob->registered) + return; + mutex_lock(&devices_mutex); for (card_index = 0; card_index < SNDRV_CARDS; card_index++) { @@ -193,64 +198,39 @@ bebob_probe(struct fw_unit *unit, break; } if (card_index >= SNDRV_CARDS) { - err = -ENOENT; - goto end; + mutex_unlock(&devices_mutex); + return; } - if ((entry->vendor_id == VEN_FOCUSRITE) && - (entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH)) - spec = get_saffire_spec(unit); - else if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH) && - !check_audiophile_booted(unit)) - spec = NULL; - else - spec = (const struct snd_bebob_spec *)entry->driver_data; - - if (spec == NULL) { - if ((entry->vendor_id == VEN_MAUDIO1) || - (entry->vendor_id == VEN_MAUDIO2)) - err = snd_bebob_maudio_load_firmware(unit); - else - err = -ENOSYS; - goto end; + err = snd_card_new(&bebob->unit->device, index[card_index], + id[card_index], THIS_MODULE, 0, &bebob->card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return; } - err = snd_card_new(&unit->device, index[card_index], id[card_index], - THIS_MODULE, sizeof(struct snd_bebob), &card); + err = name_device(bebob); if (err < 0) - goto end; - bebob = card->private_data; - bebob->card_index = card_index; - set_bit(card_index, devices_used); - card->private_free = bebob_card_free; - - bebob->card = card; - bebob->unit = fw_unit_get(unit); - bebob->spec = spec; - mutex_init(&bebob->mutex); - spin_lock_init(&bebob->lock); - init_waitqueue_head(&bebob->hwdep_wait); + goto error; - err = name_device(bebob, entry->vendor_id); + if (bebob->spec == &maudio_special_spec) { + if (bebob->entry->model_id == MODEL_MAUDIO_FW1814) + err = snd_bebob_maudio_special_discover(bebob, true); + else + err = snd_bebob_maudio_special_discover(bebob, false); + } else { + err = snd_bebob_stream_discover(bebob); + } if (err < 0) goto error; - if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_FW1814)) - err = snd_bebob_maudio_special_discover(bebob, true); - else if ((entry->vendor_id == VEN_MAUDIO1) && - (entry->model_id == MODEL_MAUDIO_PROJECTMIX)) - err = snd_bebob_maudio_special_discover(bebob, false); - else - err = snd_bebob_stream_discover(bebob); + err = snd_bebob_stream_init_duplex(bebob); if (err < 0) goto error; snd_bebob_proc_init(bebob); - if ((bebob->midi_input_ports > 0) || - (bebob->midi_output_ports > 0)) { + if (bebob->midi_input_ports > 0 || bebob->midi_output_ports > 0) { err = snd_bebob_create_midi_devices(bebob); if (err < 0) goto error; @@ -264,16 +244,75 @@ bebob_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_bebob_stream_init_duplex(bebob); + err = snd_card_register(bebob->card); if (err < 0) goto error; - if (!bebob->maudio_special_quirk) { - err = snd_card_register(card); - if (err < 0) { - snd_bebob_stream_destroy_duplex(bebob); - goto error; - } + set_bit(card_index, devices_used); + mutex_unlock(&devices_mutex); + + /* + * After registered, bebob instance can be released corresponding to + * releasing the sound card instance. + */ + bebob->card->private_free = bebob_card_free; + bebob->card->private_data = bebob; + bebob->registered = true; + + return; +error: + mutex_unlock(&devices_mutex); + snd_bebob_stream_destroy_duplex(bebob); + snd_card_free(bebob->card); + dev_info(&bebob->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int +bebob_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) +{ + struct snd_bebob *bebob; + const struct snd_bebob_spec *spec; + + if (entry->vendor_id == VEN_FOCUSRITE && + entry->model_id == MODEL_FOCUSRITE_SAFFIRE_BOTH) + spec = get_saffire_spec(unit); + else if (entry->vendor_id == VEN_MAUDIO1 && + entry->model_id == MODEL_MAUDIO_AUDIOPHILE_BOTH && + !check_audiophile_booted(unit)) + spec = NULL; + else + spec = (const struct snd_bebob_spec *)entry->driver_data; + + if (spec == NULL) { + if (entry->vendor_id == VEN_MAUDIO1 || + entry->vendor_id == VEN_MAUDIO2) + return snd_bebob_maudio_load_firmware(unit); + else + return -ENODEV; + } + + /* Allocate this independent of sound card instance. */ + bebob = kzalloc(sizeof(struct snd_bebob), GFP_KERNEL); + if (bebob == NULL) + return -ENOMEM; + + bebob->unit = fw_unit_get(unit); + bebob->entry = entry; + bebob->spec = spec; + dev_set_drvdata(&unit->device, bebob); + + mutex_init(&bebob->mutex); + spin_lock_init(&bebob->lock); + init_waitqueue_head(&bebob->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&bebob->dwork, do_registration); + + if (entry->vendor_id != VEN_MAUDIO1 || + (entry->model_id != MODEL_MAUDIO_FW1814 && + entry->model_id != MODEL_MAUDIO_PROJECTMIX)) { + snd_fw_schedule_registration(unit, &bebob->dwork); } else { /* * This is a workaround. This bus reset seems to have an effect @@ -285,19 +324,11 @@ bebob_probe(struct fw_unit *unit, * signals from dbus and starts I/Os. To avoid I/Os till the * future bus reset, registration is done in next update(). */ - bebob->deferred_registration = true; fw_schedule_bus_reset(fw_parent_device(bebob->unit)->card, false, true); } - dev_set_drvdata(&unit->device, bebob); -end: - mutex_unlock(&devices_mutex); - return err; -error: - mutex_unlock(&devices_mutex); - snd_card_free(card); - return err; + return 0; } /* @@ -324,15 +355,11 @@ bebob_update(struct fw_unit *unit) if (bebob == NULL) return; - fcp_bus_reset(bebob->unit); - - if (bebob->deferred_registration) { - if (snd_card_register(bebob->card) < 0) { - snd_bebob_stream_destroy_duplex(bebob); - snd_card_free(bebob->card); - } - bebob->deferred_registration = false; - } + /* Postpone a workqueue for deferred registration. */ + if (!bebob->registered) + snd_fw_schedule_registration(unit, &bebob->dwork); + else + fcp_bus_reset(bebob->unit); } static void bebob_remove(struct fw_unit *unit) @@ -342,8 +369,20 @@ static void bebob_remove(struct fw_unit *unit) if (bebob == NULL) return; - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(bebob->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&bebob->dwork); + + if (bebob->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(bebob->card); + } else { + /* Don't forget this case. */ + bebob_free(bebob); + } } static const struct snd_bebob_rate_spec normal_rate_spec = { diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h index b50bb33d9d46..e7f1bb925b12 100644 --- a/sound/firewire/bebob/bebob.h +++ b/sound/firewire/bebob/bebob.h @@ -83,6 +83,10 @@ struct snd_bebob { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + + const struct ieee1394_device_id *entry; const struct snd_bebob_spec *spec; unsigned int midi_input_ports; @@ -90,7 +94,6 @@ struct snd_bebob { bool connected; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; @@ -111,7 +114,6 @@ struct snd_bebob { /* for M-Audio special devices */ void *maudio_special_quirk; - bool deferred_registration; /* For BeBoB version quirk. */ unsigned int version; diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c index 77cbb02bff34..4d3034a68bdf 100644 --- a/sound/firewire/bebob/bebob_stream.c +++ b/sound/firewire/bebob/bebob_stream.c @@ -484,30 +484,6 @@ destroy_both_connections(struct snd_bebob *bebob) } static int -get_sync_mode(struct snd_bebob *bebob, enum cip_flags *sync_mode) -{ - enum snd_bebob_clock_type src; - int err; - - err = snd_bebob_stream_get_clock_src(bebob, &src); - if (err < 0) - return err; - - switch (src) { - case SND_BEBOB_CLOCK_TYPE_INTERNAL: - case SND_BEBOB_CLOCK_TYPE_EXTERNAL: - *sync_mode = CIP_SYNC_TO_DEVICE; - break; - default: - case SND_BEBOB_CLOCK_TYPE_SYT: - *sync_mode = 0; - break; - } - - return 0; -} - -static int start_stream(struct snd_bebob *bebob, struct amdtp_stream *stream, unsigned int rate) { @@ -550,8 +526,6 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob) goto end; } - bebob->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK; - /* * BeBoB v3 transfers packets with these qurks: * - In the beginning of streaming, the value of dbc is incremented @@ -584,8 +558,6 @@ end: int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) { const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate; - struct amdtp_stream *master, *slave; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -593,22 +565,11 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (bebob->substreams_counter == 0) goto end; - err = get_sync_mode(bebob, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &bebob->tx_stream; - slave = &bebob->rx_stream; - } else { - master = &bebob->rx_stream; - slave = &bebob->tx_stream; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(bebob, master); + err = check_connection_used_by_others(bebob, &bebob->rx_stream); if (err < 0) goto end; @@ -618,11 +579,12 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) * At bus reset, connections should not be broken here. So streams need * to be re-started. This is a reason to use SKIP_INIT_DBC_CHECK flag. */ - if (amdtp_streaming_error(master)) - amdtp_stream_stop(master); - if (amdtp_streaming_error(slave)) - amdtp_stream_stop(slave); - if (!amdtp_stream_running(master) && !amdtp_stream_running(slave)) + if (amdtp_streaming_error(&bebob->rx_stream)) + amdtp_stream_stop(&bebob->rx_stream); + if (amdtp_streaming_error(&bebob->tx_stream)) + amdtp_stream_stop(&bebob->tx_stream); + if (!amdtp_stream_running(&bebob->rx_stream) && + !amdtp_stream_running(&bebob->tx_stream)) break_both_connections(bebob); /* stop streams if rate is different */ @@ -635,16 +597,13 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - amdtp_stream_stop(master); - amdtp_stream_stop(slave); + amdtp_stream_stop(&bebob->rx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - bebob->master = master; - + if (!amdtp_stream_running(&bebob->rx_stream)) { /* * NOTE: * If establishing connections at first, Yamaha GO46 @@ -666,7 +625,7 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) if (err < 0) goto end; - err = start_stream(bebob, master, rate); + err = start_stream(bebob, &bebob->rx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP master stream:%d\n", err); @@ -685,15 +644,16 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) dev_err(&bebob->unit->device, "fail to ensure sampling rate: %d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } } /* wait first callback */ - if (!amdtp_stream_wait_callback(master, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->rx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; goto end; @@ -701,20 +661,21 @@ int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate) } /* start slave if needed */ - if (!amdtp_stream_running(slave)) { - err = start_stream(bebob, slave, rate); + if (!amdtp_stream_running(&bebob->tx_stream)) { + err = start_stream(bebob, &bebob->tx_stream, rate); if (err < 0) { dev_err(&bebob->unit->device, "fail to run AMDTP slave stream:%d\n", err); - amdtp_stream_stop(master); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); goto end; } /* wait first callback */ - if (!amdtp_stream_wait_callback(slave, CALLBACK_TIMEOUT)) { - amdtp_stream_stop(slave); - amdtp_stream_stop(master); + if (!amdtp_stream_wait_callback(&bebob->tx_stream, + CALLBACK_TIMEOUT)) { + amdtp_stream_stop(&bebob->tx_stream); + amdtp_stream_stop(&bebob->rx_stream); break_both_connections(bebob); err = -ETIMEDOUT; } @@ -725,22 +686,12 @@ end: void snd_bebob_stream_stop_duplex(struct snd_bebob *bebob) { - struct amdtp_stream *master, *slave; - - if (bebob->master == &bebob->rx_stream) { - slave = &bebob->tx_stream; - master = &bebob->rx_stream; - } else { - slave = &bebob->rx_stream; - master = &bebob->tx_stream; - } - if (bebob->substreams_counter == 0) { - amdtp_stream_pcm_abort(master); - amdtp_stream_stop(master); + amdtp_stream_pcm_abort(&bebob->rx_stream); + amdtp_stream_stop(&bebob->rx_stream); - amdtp_stream_pcm_abort(slave); - amdtp_stream_stop(slave); + amdtp_stream_pcm_abort(&bebob->tx_stream); + amdtp_stream_stop(&bebob->tx_stream); break_both_connections(bebob); } diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 8b64aef31a86..25e9f77275c4 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -20,8 +20,6 @@ MODULE_LICENSE("GPL v2"); #define WEISS_CATEGORY_ID 0x00 #define LOUD_CATEGORY_ID 0x10 -#define PROBE_DELAY_MS (2 * MSEC_PER_SEC) - /* * Some models support several isochronous channels, while these streams are not * always available. In this case, add the model name to this list. @@ -201,6 +199,10 @@ static void do_registration(struct work_struct *work) dice_card_strings(dice); + err = snd_dice_stream_init_duplex(dice); + if (err < 0) + goto error; + snd_dice_create_proc(dice); err = snd_dice_create_pcm(dice); @@ -229,28 +231,14 @@ static void do_registration(struct work_struct *work) return; error: + snd_dice_stream_destroy_duplex(dice); snd_dice_transaction_destroy(dice); + snd_dice_stream_destroy_duplex(dice); snd_card_free(dice->card); dev_info(&dice->unit->device, "Sound card registration failed: %d\n", err); } -static void schedule_registration(struct snd_dice *dice) -{ - struct fw_card *fw_card = fw_parent_device(dice->unit)->card; - u64 now, delay; - - now = get_jiffies_64(); - delay = fw_card->reset_jiffies + msecs_to_jiffies(PROBE_DELAY_MS); - - if (time_after64(delay, now)) - delay -= now; - else - delay = 0; - - mod_delayed_work(system_wq, &dice->dwork, delay); -} - static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { struct snd_dice *dice; @@ -273,15 +261,9 @@ static int dice_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) init_completion(&dice->clock_accepted); init_waitqueue_head(&dice->hwdep_wait); - err = snd_dice_stream_init_duplex(dice); - if (err < 0) { - dice_free(dice); - return err; - } - /* Allocate and register this sound card later. */ INIT_DEFERRABLE_WORK(&dice->dwork, do_registration); - schedule_registration(dice); + snd_fw_schedule_registration(unit, &dice->dwork); return 0; } @@ -312,7 +294,7 @@ static void dice_bus_reset(struct fw_unit *unit) /* Postpone a workqueue for deferred registration. */ if (!dice->registered) - schedule_registration(dice); + snd_fw_schedule_registration(unit, &dice->dwork); /* The handler address register becomes initialized. */ snd_dice_transaction_reinit(dice); @@ -335,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c index 0ac92aba5bc1..b3cffd01a19f 100644 --- a/sound/firewire/digi00x/amdtp-dot.c +++ b/sound/firewire/digi00x/amdtp-dot.c @@ -421,7 +421,7 @@ int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit, /* Use different mode between incoming/outgoing. */ if (dir == AMDTP_IN_STREAM) { - flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK; + flags = CIP_NONBLOCKING; process_data_blocks = process_tx_data_blocks; } else { flags = CIP_BLOCKING; diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c index 554324d8c602..735d35640807 100644 --- a/sound/firewire/digi00x/digi00x-transaction.c +++ b/sound/firewire/digi00x/digi00x-transaction.c @@ -126,12 +126,17 @@ int snd_dg00x_transaction_register(struct snd_dg00x *dg00x) return err; error: fw_core_remove_address_handler(&dg00x->async_handler); - dg00x->async_handler.address_callback = NULL; + dg00x->async_handler.callback_data = NULL; return err; } void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x) { + if (dg00x->async_handler.callback_data == NULL) + return; + snd_fw_async_midi_port_destroy(&dg00x->out_control); fw_core_remove_address_handler(&dg00x->async_handler); + + dg00x->async_handler.callback_data = NULL; } diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index 1f33b7a1fca4..cc4776c6ded3 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -40,10 +40,8 @@ static int name_card(struct snd_dg00x *dg00x) return 0; } -static void dg00x_card_free(struct snd_card *card) +static void dg00x_free(struct snd_dg00x *dg00x) { - struct snd_dg00x *dg00x = card->private_data; - snd_dg00x_stream_destroy_duplex(dg00x); snd_dg00x_transaction_unregister(dg00x); @@ -52,28 +50,24 @@ static void dg00x_card_free(struct snd_card *card) mutex_destroy(&dg00x->mutex); } -static int snd_dg00x_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void dg00x_card_free(struct snd_card *card) { - struct snd_card *card; - struct snd_dg00x *dg00x; - int err; + dg00x_free(card->private_data); +} - /* create card */ - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(struct snd_dg00x), &card); - if (err < 0) - return err; - card->private_free = dg00x_card_free; +static void do_registration(struct work_struct *work) +{ + struct snd_dg00x *dg00x = + container_of(work, struct snd_dg00x, dwork.work); + int err; - /* initialize myself */ - dg00x = card->private_data; - dg00x->card = card; - dg00x->unit = fw_unit_get(unit); + if (dg00x->registered) + return; - mutex_init(&dg00x->mutex); - spin_lock_init(&dg00x->lock); - init_waitqueue_head(&dg00x->hwdep_wait); + err = snd_card_new(&dg00x->unit->device, -1, NULL, THIS_MODULE, 0, + &dg00x->card); + if (err < 0) + return; err = name_card(dg00x); if (err < 0) @@ -101,35 +95,86 @@ static int snd_dg00x_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(dg00x->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, dg00x); + dg00x->card->private_free = dg00x_card_free; + dg00x->card->private_data = dg00x; + dg00x->registered = true; - return err; + return; error: - snd_card_free(card); - return err; + snd_dg00x_transaction_unregister(dg00x); + snd_dg00x_stream_destroy_duplex(dg00x); + snd_card_free(dg00x->card); + dev_info(&dg00x->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int snd_dg00x_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_dg00x *dg00x; + + /* Allocate this independent of sound card instance. */ + dg00x = kzalloc(sizeof(struct snd_dg00x), GFP_KERNEL); + if (dg00x == NULL) + return -ENOMEM; + + dg00x->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, dg00x); + + mutex_init(&dg00x->mutex); + spin_lock_init(&dg00x->lock); + init_waitqueue_head(&dg00x->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&dg00x->dwork, do_registration); + snd_fw_schedule_registration(unit, &dg00x->dwork); + + return 0; } static void snd_dg00x_update(struct fw_unit *unit) { struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!dg00x->registered) + snd_fw_schedule_registration(unit, &dg00x->dwork); + snd_dg00x_transaction_reregister(dg00x); - mutex_lock(&dg00x->mutex); - snd_dg00x_stream_update_duplex(dg00x); - mutex_unlock(&dg00x->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (dg00x->registered) { + mutex_lock(&dg00x->mutex); + snd_dg00x_stream_update_duplex(dg00x); + mutex_unlock(&dg00x->mutex); + } } static void snd_dg00x_remove(struct fw_unit *unit) { struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(dg00x->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&dg00x->dwork); + + if (dg00x->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(dg00x->card); + } else { + /* Don't forget this case. */ + dg00x_free(dg00x); + } } static const struct ieee1394_device_id snd_dg00x_id_table[] = { diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h index 907e73993677..2cd465c0caae 100644 --- a/sound/firewire/digi00x/digi00x.h +++ b/sound/firewire/digi00x/digi00x.h @@ -37,6 +37,9 @@ struct snd_dg00x { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + struct amdtp_stream tx_stream; struct fw_iso_resources tx_resources; diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c index 8f27b67503c8..71a0613d3da0 100644 --- a/sound/firewire/fireworks/fireworks.c +++ b/sound/firewire/fireworks/fireworks.c @@ -168,11 +168,34 @@ get_hardware_info(struct snd_efw *efw) sizeof(struct snd_efw_phys_grp) * hwinfo->phys_in_grp_count); memcpy(&efw->phys_out_grps, hwinfo->phys_out_grps, sizeof(struct snd_efw_phys_grp) * hwinfo->phys_out_grp_count); + + /* AudioFire8 (since 2009) and AudioFirePre8 */ + if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_9) + efw->is_af9 = true; + /* These models uses the same firmware. */ + if (hwinfo->type == MODEL_ECHO_AUDIOFIRE_2 || + hwinfo->type == MODEL_ECHO_AUDIOFIRE_4 || + hwinfo->type == MODEL_ECHO_AUDIOFIRE_9 || + hwinfo->type == MODEL_GIBSON_RIP || + hwinfo->type == MODEL_GIBSON_GOLDTOP) + efw->is_fireworks3 = true; end: kfree(hwinfo); return err; } +static void efw_free(struct snd_efw *efw) +{ + snd_efw_stream_destroy_duplex(efw); + snd_efw_transaction_remove_instance(efw); + fw_unit_put(efw->unit); + + kfree(efw->resp_buf); + + mutex_destroy(&efw->mutex); + kfree(efw); +} + /* * This module releases the FireWire unit data after all ALSA character devices * are released by applications. This is for releasing stream data or finishing @@ -184,28 +207,24 @@ efw_card_free(struct snd_card *card) { struct snd_efw *efw = card->private_data; - snd_efw_stream_destroy_duplex(efw); - snd_efw_transaction_remove_instance(efw); - fw_unit_put(efw->unit); - - kfree(efw->resp_buf); - if (efw->card_index >= 0) { mutex_lock(&devices_mutex); clear_bit(efw->card_index, devices_used); mutex_unlock(&devices_mutex); } - mutex_destroy(&efw->mutex); + efw_free(card->private_data); } -static int -efw_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void +do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_efw *efw; - int card_index, err; + struct snd_efw *efw = container_of(work, struct snd_efw, dwork.work); + unsigned int card_index; + int err; + + if (efw->registered) + return; mutex_lock(&devices_mutex); @@ -215,24 +234,16 @@ efw_probe(struct fw_unit *unit, break; } if (card_index >= SNDRV_CARDS) { - err = -ENOENT; - goto end; + mutex_unlock(&devices_mutex); + return; } - err = snd_card_new(&unit->device, index[card_index], id[card_index], - THIS_MODULE, sizeof(struct snd_efw), &card); - if (err < 0) - goto end; - efw = card->private_data; - efw->card_index = card_index; - set_bit(card_index, devices_used); - card->private_free = efw_card_free; - - efw->card = card; - efw->unit = fw_unit_get(unit); - mutex_init(&efw->mutex); - spin_lock_init(&efw->lock); - init_waitqueue_head(&efw->hwdep_wait); + err = snd_card_new(&efw->unit->device, index[card_index], + id[card_index], THIS_MODULE, 0, &efw->card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return; + } /* prepare response buffer */ snd_efw_resp_buf_size = clamp(snd_efw_resp_buf_size, @@ -248,16 +259,10 @@ efw_probe(struct fw_unit *unit, err = get_hardware_info(efw); if (err < 0) goto error; - /* AudioFire8 (since 2009) and AudioFirePre8 */ - if (entry->model_id == MODEL_ECHO_AUDIOFIRE_9) - efw->is_af9 = true; - /* These models uses the same firmware. */ - if (entry->model_id == MODEL_ECHO_AUDIOFIRE_2 || - entry->model_id == MODEL_ECHO_AUDIOFIRE_4 || - entry->model_id == MODEL_ECHO_AUDIOFIRE_9 || - entry->model_id == MODEL_GIBSON_RIP || - entry->model_id == MODEL_GIBSON_GOLDTOP) - efw->is_fireworks3 = true; + + err = snd_efw_stream_init_duplex(efw); + if (err < 0) + goto error; snd_efw_proc_init(efw); @@ -275,44 +280,93 @@ efw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_efw_stream_init_duplex(efw); + err = snd_card_register(efw->card); if (err < 0) goto error; - err = snd_card_register(card); - if (err < 0) { - snd_efw_stream_destroy_duplex(efw); - goto error; - } - - dev_set_drvdata(&unit->device, efw); -end: + set_bit(card_index, devices_used); mutex_unlock(&devices_mutex); - return err; + + /* + * After registered, efw instance can be released corresponding to + * releasing the sound card instance. + */ + efw->card->private_free = efw_card_free; + efw->card->private_data = efw; + efw->registered = true; + + return; error: - snd_efw_transaction_remove_instance(efw); mutex_unlock(&devices_mutex); - snd_card_free(card); - return err; + snd_efw_transaction_remove_instance(efw); + snd_efw_stream_destroy_duplex(efw); + snd_card_free(efw->card); + dev_info(&efw->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int +efw_probe(struct fw_unit *unit, const struct ieee1394_device_id *entry) +{ + struct snd_efw *efw; + + efw = kzalloc(sizeof(struct snd_efw), GFP_KERNEL); + if (efw == NULL) + return -ENOMEM; + + efw->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, efw); + + mutex_init(&efw->mutex); + spin_lock_init(&efw->lock); + init_waitqueue_head(&efw->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&efw->dwork, do_registration); + snd_fw_schedule_registration(unit, &efw->dwork); + + return 0; } static void efw_update(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!efw->registered) + snd_fw_schedule_registration(unit, &efw->dwork); + snd_efw_transaction_bus_reset(efw->unit); - mutex_lock(&efw->mutex); - snd_efw_stream_update_duplex(efw); - mutex_unlock(&efw->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (efw->registered) { + mutex_lock(&efw->mutex); + snd_efw_stream_update_duplex(efw); + mutex_unlock(&efw->mutex); + } } static void efw_remove(struct fw_unit *unit) { struct snd_efw *efw = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(efw->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&efw->dwork); + + if (efw->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(efw->card); + } else { + /* Don't forget this case. */ + efw_free(efw); + } } static const struct ieee1394_device_id efw_id_table[] = { diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index 96c4e0c6a9bd..03ed35237e2b 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -65,6 +65,9 @@ struct snd_efw { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + /* for transaction */ u32 seqnum; bool resp_addr_changable; @@ -81,7 +84,6 @@ struct snd_efw { unsigned int pcm_capture_channels[SND_EFW_MULTIPLIER_MODES]; unsigned int pcm_playback_channels[SND_EFW_MULTIPLIER_MODES]; - struct amdtp_stream *master; struct amdtp_stream tx_stream; struct amdtp_stream rx_stream; struct cmp_connection out_conn; diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c index 425db8d88235..ee47924aef0d 100644 --- a/sound/firewire/fireworks/fireworks_stream.c +++ b/sound/firewire/fireworks/fireworks_stream.c @@ -121,23 +121,6 @@ destroy_stream(struct snd_efw *efw, struct amdtp_stream *stream) } static int -get_sync_mode(struct snd_efw *efw, enum cip_flags *sync_mode) -{ - enum snd_efw_clock_source clock_source; - int err; - - err = snd_efw_command_get_clock_source(efw, &clock_source); - if (err < 0) - return err; - - if (clock_source == SND_EFW_CLOCK_SOURCE_SYTMATCH) - return -ENOSYS; - - *sync_mode = CIP_SYNC_TO_DEVICE; - return 0; -} - -static int check_connection_used_by_others(struct snd_efw *efw, struct amdtp_stream *s) { struct cmp_connection *conn; @@ -208,9 +191,6 @@ end: int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) { - struct amdtp_stream *master, *slave; - unsigned int slave_substreams; - enum cip_flags sync_mode; unsigned int curr_rate; int err = 0; @@ -218,32 +198,19 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (efw->playback_substreams == 0 && efw->capture_substreams == 0) goto end; - err = get_sync_mode(efw, &sync_mode); - if (err < 0) - goto end; - if (sync_mode == CIP_SYNC_TO_DEVICE) { - master = &efw->tx_stream; - slave = &efw->rx_stream; - slave_substreams = efw->playback_substreams; - } else { - master = &efw->rx_stream; - slave = &efw->tx_stream; - slave_substreams = efw->capture_substreams; - } - /* * Considering JACK/FFADO streaming: * TODO: This can be removed hwdep functionality becomes popular. */ - err = check_connection_used_by_others(efw, master); + err = check_connection_used_by_others(efw, &efw->rx_stream); if (err < 0) goto end; /* packet queueing error */ - if (amdtp_streaming_error(slave)) - stop_stream(efw, slave); - if (amdtp_streaming_error(master)) - stop_stream(efw, master); + if (amdtp_streaming_error(&efw->tx_stream)) + stop_stream(efw, &efw->tx_stream); + if (amdtp_streaming_error(&efw->rx_stream)) + stop_stream(efw, &efw->rx_stream); /* stop streams if rate is different */ err = snd_efw_command_get_sampling_rate(efw, &curr_rate); @@ -252,20 +219,17 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) if (rate == 0) rate = curr_rate; if (rate != curr_rate) { - stop_stream(efw, slave); - stop_stream(efw, master); + stop_stream(efw, &efw->tx_stream); + stop_stream(efw, &efw->rx_stream); } /* master should be always running */ - if (!amdtp_stream_running(master)) { - amdtp_stream_set_sync(sync_mode, master, slave); - efw->master = master; - + if (!amdtp_stream_running(&efw->rx_stream)) { err = snd_efw_command_set_sampling_rate(efw, rate); if (err < 0) goto end; - err = start_stream(efw, master, rate); + err = start_stream(efw, &efw->rx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP master stream:%d\n", err); @@ -274,12 +238,13 @@ int snd_efw_stream_start_duplex(struct snd_efw *efw, unsigned int rate) } /* start slave if needed */ - if (slave_substreams > 0 && !amdtp_stream_running(slave)) { - err = start_stream(efw, slave, rate); + if (efw->capture_substreams > 0 && + !amdtp_stream_running(&efw->tx_stream)) { + err = start_stream(efw, &efw->tx_stream, rate); if (err < 0) { dev_err(&efw->unit->device, "fail to start AMDTP slave stream:%d\n", err); - stop_stream(efw, master); + stop_stream(efw, &efw->rx_stream); } } end: @@ -288,26 +253,11 @@ end: void snd_efw_stream_stop_duplex(struct snd_efw *efw) { - struct amdtp_stream *master, *slave; - unsigned int master_substreams, slave_substreams; - - if (efw->master == &efw->rx_stream) { - slave = &efw->tx_stream; - master = &efw->rx_stream; - slave_substreams = efw->capture_substreams; - master_substreams = efw->playback_substreams; - } else { - slave = &efw->rx_stream; - master = &efw->tx_stream; - slave_substreams = efw->playback_substreams; - master_substreams = efw->capture_substreams; - } - - if (slave_substreams == 0) { - stop_stream(efw, slave); + if (efw->capture_substreams == 0) { + stop_stream(efw, &efw->tx_stream); - if (master_substreams == 0) - stop_stream(efw, master); + if (efw->playback_substreams == 0) + stop_stream(efw, &efw->rx_stream); } } diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c index f80aafa44c89..ca4dfcf43175 100644 --- a/sound/firewire/lib.c +++ b/sound/firewire/lib.c @@ -67,6 +67,38 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode, } EXPORT_SYMBOL(snd_fw_transaction); +#define PROBE_DELAY_MS (2 * MSEC_PER_SEC) + +/** + * snd_fw_schedule_registration - schedule work for sound card registration + * @unit: an instance for unit on IEEE 1394 bus + * @dwork: delayed work with callback function + * + * This function is not designed for general purposes. When new unit is + * connected to IEEE 1394 bus, the bus is under bus-reset state because of + * topological change. In this state, units tend to fail both of asynchronous + * and isochronous communication. To avoid this problem, this function is used + * to postpone sound card registration after the state. The callers must + * set up instance of delayed work in advance. + */ +void snd_fw_schedule_registration(struct fw_unit *unit, + struct delayed_work *dwork) +{ + u64 now, delay; + + now = get_jiffies_64(); + delay = fw_parent_device(unit)->card->reset_jiffies + + msecs_to_jiffies(PROBE_DELAY_MS); + + if (time_after64(delay, now)) + delay -= now; + else + delay = 0; + + mod_delayed_work(system_wq, dwork, delay); +} +EXPORT_SYMBOL(snd_fw_schedule_registration); + static void async_midi_port_callback(struct fw_card *card, int rcode, void *data, size_t length, void *callback_data) diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h index f3f6f84c48d6..f6769312ebfc 100644 --- a/sound/firewire/lib.h +++ b/sound/firewire/lib.h @@ -22,6 +22,9 @@ static inline bool rcode_is_permanent_error(int rcode) return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR; } +void snd_fw_schedule_registration(struct fw_unit *unit, + struct delayed_work *dwork); + struct snd_fw_async_midi_port; typedef int (*snd_fw_async_midi_port_fill)( struct snd_rawmidi_substream *substream, diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 7cb5743c073b..d9361f352133 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -242,8 +242,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, * blocks than IEC 61883-6 defines. */ if (stream == &oxfw->tx_stream) { - oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK | - CIP_JUMBO_PAYLOAD; + oxfw->tx_stream.flags |= CIP_JUMBO_PAYLOAD; if (oxfw->wrong_dbs) oxfw->tx_stream.flags |= CIP_WRONG_DBS; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index abedc2207261..e629b88f7d93 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -118,15 +118,8 @@ end: return err; } -/* - * This module releases the FireWire unit data after all ALSA character devices - * are released by applications. This is for releasing stream data or finishing - * transactions safely. Thus at returning from .remove(), this module still keep - * references for the unit. - */ -static void oxfw_card_free(struct snd_card *card) +static void oxfw_free(struct snd_oxfw *oxfw) { - struct snd_oxfw *oxfw = card->private_data; unsigned int i; snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); @@ -144,6 +137,17 @@ static void oxfw_card_free(struct snd_card *card) mutex_destroy(&oxfw->mutex); } +/* + * This module releases the FireWire unit data after all ALSA character devices + * are released by applications. This is for releasing stream data or finishing + * transactions safely. Thus at returning from .remove(), this module still keep + * references for the unit. + */ +static void oxfw_card_free(struct snd_card *card) +{ + oxfw_free(card->private_data); +} + static int detect_quirks(struct snd_oxfw *oxfw) { struct fw_device *fw_dev = fw_parent_device(oxfw->unit); @@ -205,41 +209,39 @@ static int detect_quirks(struct snd_oxfw *oxfw) return 0; } -static int oxfw_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void do_registration(struct work_struct *work) { - struct snd_card *card; - struct snd_oxfw *oxfw; + struct snd_oxfw *oxfw = container_of(work, struct snd_oxfw, dwork.work); int err; - if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit)) - return -ENODEV; + if (oxfw->registered) + return; - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(*oxfw), &card); + err = snd_card_new(&oxfw->unit->device, -1, NULL, THIS_MODULE, 0, + &oxfw->card); if (err < 0) - return err; + return; - card->private_free = oxfw_card_free; - oxfw = card->private_data; - oxfw->card = card; - mutex_init(&oxfw->mutex); - oxfw->unit = fw_unit_get(unit); - oxfw->entry = entry; - spin_lock_init(&oxfw->lock); - init_waitqueue_head(&oxfw->hwdep_wait); + err = name_card(oxfw); + if (err < 0) + goto error; - err = snd_oxfw_stream_discover(oxfw); + err = detect_quirks(oxfw); if (err < 0) goto error; - err = name_card(oxfw); + err = snd_oxfw_stream_discover(oxfw); if (err < 0) goto error; - err = detect_quirks(oxfw); + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); if (err < 0) goto error; + if (oxfw->has_output) { + err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream); + if (err < 0) + goto error; + } err = snd_oxfw_create_pcm(oxfw); if (err < 0) @@ -255,54 +257,97 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->rx_stream); + err = snd_card_register(oxfw->card); if (err < 0) goto error; - if (oxfw->has_output) { - err = snd_oxfw_stream_init_simplex(oxfw, &oxfw->tx_stream); - if (err < 0) - goto error; - } - err = snd_card_register(card); - if (err < 0) { - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); - goto error; - } + /* + * After registered, oxfw instance can be released corresponding to + * releasing the sound card instance. + */ + oxfw->card->private_free = oxfw_card_free; + oxfw->card->private_data = oxfw; + oxfw->registered = true; + + return; +error: + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_destroy_simplex(oxfw, &oxfw->tx_stream); + snd_card_free(oxfw->card); + dev_info(&oxfw->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int oxfw_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_oxfw *oxfw; + + if (entry->vendor_id == VENDOR_LOUD && !detect_loud_models(unit)) + return -ENODEV; + + /* Allocate this independent of sound card instance. */ + oxfw = kzalloc(sizeof(struct snd_oxfw), GFP_KERNEL); + if (oxfw == NULL) + return -ENOMEM; + + oxfw->entry = entry; + oxfw->unit = fw_unit_get(unit); dev_set_drvdata(&unit->device, oxfw); + mutex_init(&oxfw->mutex); + spin_lock_init(&oxfw->lock); + init_waitqueue_head(&oxfw->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&oxfw->dwork, do_registration); + snd_fw_schedule_registration(unit, &oxfw->dwork); + return 0; -error: - snd_card_free(card); - return err; } static void oxfw_bus_reset(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); + if (!oxfw->registered) + snd_fw_schedule_registration(unit, &oxfw->dwork); + fcp_bus_reset(oxfw->unit); - mutex_lock(&oxfw->mutex); + if (oxfw->registered) { + mutex_lock(&oxfw->mutex); - snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream); - if (oxfw->has_output) - snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream); + snd_oxfw_stream_update_simplex(oxfw, &oxfw->rx_stream); + if (oxfw->has_output) + snd_oxfw_stream_update_simplex(oxfw, &oxfw->tx_stream); - mutex_unlock(&oxfw->mutex); + mutex_unlock(&oxfw->mutex); - if (oxfw->entry->vendor_id == OUI_STANTON) - snd_oxfw_scs1x_update(oxfw); + if (oxfw->entry->vendor_id == OUI_STANTON) + snd_oxfw_scs1x_update(oxfw); + } } static void oxfw_remove(struct fw_unit *unit) { struct snd_oxfw *oxfw = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(oxfw->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&oxfw->dwork); + + if (oxfw->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(oxfw->card); + } else { + /* Don't forget this case. */ + oxfw_free(oxfw); + } } static const struct compat_info griffin_firewave = { diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index 9beecc214767..2047dcb27625 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -36,10 +36,12 @@ struct snd_oxfw { struct snd_card *card; struct fw_unit *unit; - const struct device_info *device_info; struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; + bool wrong_dbs; bool has_output; u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c index 0e6dd5c61f53..4ad3bd7fd445 100644 --- a/sound/firewire/tascam/tascam-stream.c +++ b/sound/firewire/tascam/tascam-stream.c @@ -381,19 +381,17 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) return err; if (curr_rate != rate || - amdtp_streaming_error(&tscm->tx_stream) || - amdtp_streaming_error(&tscm->rx_stream)) { + amdtp_streaming_error(&tscm->rx_stream) || + amdtp_streaming_error(&tscm->tx_stream)) { finish_session(tscm); - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); release_resources(tscm); } - if (!amdtp_stream_running(&tscm->tx_stream)) { - amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE, - &tscm->tx_stream, &tscm->rx_stream); + if (!amdtp_stream_running(&tscm->rx_stream)) { err = keep_resources(tscm, rate); if (err < 0) goto error; @@ -406,27 +404,27 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) if (err < 0) goto error; - err = amdtp_stream_start(&tscm->tx_stream, - tscm->tx_resources.channel, + err = amdtp_stream_start(&tscm->rx_stream, + tscm->rx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->tx_stream, + if (!amdtp_stream_wait_callback(&tscm->rx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; } } - if (!amdtp_stream_running(&tscm->rx_stream)) { - err = amdtp_stream_start(&tscm->rx_stream, - tscm->rx_resources.channel, + if (!amdtp_stream_running(&tscm->tx_stream)) { + err = amdtp_stream_start(&tscm->tx_stream, + tscm->tx_resources.channel, fw_parent_device(tscm->unit)->max_speed); if (err < 0) goto error; - if (!amdtp_stream_wait_callback(&tscm->rx_stream, + if (!amdtp_stream_wait_callback(&tscm->tx_stream, CALLBACK_TIMEOUT)) { err = -ETIMEDOUT; goto error; @@ -435,8 +433,8 @@ int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate) return 0; error: - amdtp_stream_stop(&tscm->tx_stream); amdtp_stream_stop(&tscm->rx_stream); + amdtp_stream_stop(&tscm->tx_stream); finish_session(tscm); release_resources(tscm); diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index e281c338e562..9dc93a7eb9da 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -85,10 +85,8 @@ static int identify_model(struct snd_tscm *tscm) return 0; } -static void tscm_card_free(struct snd_card *card) +static void tscm_free(struct snd_tscm *tscm) { - struct snd_tscm *tscm = card->private_data; - snd_tscm_transaction_unregister(tscm); snd_tscm_stream_destroy_duplex(tscm); @@ -97,44 +95,36 @@ static void tscm_card_free(struct snd_card *card) mutex_destroy(&tscm->mutex); } -static int snd_tscm_probe(struct fw_unit *unit, - const struct ieee1394_device_id *entry) +static void tscm_card_free(struct snd_card *card) { - struct snd_card *card; - struct snd_tscm *tscm; + tscm_free(card->private_data); +} + +static void do_registration(struct work_struct *work) +{ + struct snd_tscm *tscm = container_of(work, struct snd_tscm, dwork.work); int err; - /* create card */ - err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE, - sizeof(struct snd_tscm), &card); + err = snd_card_new(&tscm->unit->device, -1, NULL, THIS_MODULE, 0, + &tscm->card); if (err < 0) - return err; - card->private_free = tscm_card_free; - - /* initialize myself */ - tscm = card->private_data; - tscm->card = card; - tscm->unit = fw_unit_get(unit); - - mutex_init(&tscm->mutex); - spin_lock_init(&tscm->lock); - init_waitqueue_head(&tscm->hwdep_wait); + return; err = identify_model(tscm); if (err < 0) goto error; - snd_tscm_proc_init(tscm); - - err = snd_tscm_stream_init_duplex(tscm); + err = snd_tscm_transaction_register(tscm); if (err < 0) goto error; - err = snd_tscm_create_pcm_devices(tscm); + err = snd_tscm_stream_init_duplex(tscm); if (err < 0) goto error; - err = snd_tscm_transaction_register(tscm); + snd_tscm_proc_init(tscm); + + err = snd_tscm_create_pcm_devices(tscm); if (err < 0) goto error; @@ -146,35 +136,91 @@ static int snd_tscm_probe(struct fw_unit *unit, if (err < 0) goto error; - err = snd_card_register(card); + err = snd_card_register(tscm->card); if (err < 0) goto error; - dev_set_drvdata(&unit->device, tscm); + /* + * After registered, tscm instance can be released corresponding to + * releasing the sound card instance. + */ + tscm->card->private_free = tscm_card_free; + tscm->card->private_data = tscm; + tscm->registered = true; - return err; + return; error: - snd_card_free(card); - return err; + snd_tscm_transaction_unregister(tscm); + snd_tscm_stream_destroy_duplex(tscm); + snd_card_free(tscm->card); + dev_info(&tscm->unit->device, + "Sound card registration failed: %d\n", err); +} + +static int snd_tscm_probe(struct fw_unit *unit, + const struct ieee1394_device_id *entry) +{ + struct snd_tscm *tscm; + + /* Allocate this independent of sound card instance. */ + tscm = kzalloc(sizeof(struct snd_tscm), GFP_KERNEL); + if (tscm == NULL) + return -ENOMEM; + + /* initialize myself */ + tscm->unit = fw_unit_get(unit); + dev_set_drvdata(&unit->device, tscm); + + mutex_init(&tscm->mutex); + spin_lock_init(&tscm->lock); + init_waitqueue_head(&tscm->hwdep_wait); + + /* Allocate and register this sound card later. */ + INIT_DEFERRABLE_WORK(&tscm->dwork, do_registration); + snd_fw_schedule_registration(unit, &tscm->dwork); + + return 0; } static void snd_tscm_update(struct fw_unit *unit) { struct snd_tscm *tscm = dev_get_drvdata(&unit->device); + /* Postpone a workqueue for deferred registration. */ + if (!tscm->registered) + snd_fw_schedule_registration(unit, &tscm->dwork); + snd_tscm_transaction_reregister(tscm); - mutex_lock(&tscm->mutex); - snd_tscm_stream_update_duplex(tscm); - mutex_unlock(&tscm->mutex); + /* + * After registration, userspace can start packet streaming, then this + * code block works fine. + */ + if (tscm->registered) { + mutex_lock(&tscm->mutex); + snd_tscm_stream_update_duplex(tscm); + mutex_unlock(&tscm->mutex); + } } static void snd_tscm_remove(struct fw_unit *unit) { struct snd_tscm *tscm = dev_get_drvdata(&unit->device); - /* No need to wait for releasing card object in this context. */ - snd_card_free_when_closed(tscm->card); + /* + * Confirm to stop the work for registration before the sound card is + * going to be released. The work is not scheduled again because bus + * reset handler is not called anymore. + */ + cancel_delayed_work_sync(&tscm->dwork); + + if (tscm->registered) { + /* No need to wait for releasing card object in this context. */ + snd_card_free_when_closed(tscm->card); + } else { + /* Don't forget this case. */ + tscm_free(tscm); + } } static const struct ieee1394_device_id snd_tscm_id_table[] = { diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h index 30ab77e924f7..1f61011579a7 100644 --- a/sound/firewire/tascam/tascam.h +++ b/sound/firewire/tascam/tascam.h @@ -51,6 +51,8 @@ struct snd_tscm { struct mutex mutex; spinlock_t lock; + bool registered; + struct delayed_work dwork; const struct snd_tscm_spec *spec; struct fw_iso_resources tx_resources; diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..31b510c5ca0b 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -105,6 +105,9 @@ int snd_hdac_ext_bus_init(struct hdac_ext_bus *ebus, struct device *dev, INIT_LIST_HEAD(&ebus->hlink_list); ebus->idx = idx++; + mutex_init(&ebus->lock); + ebus->cmd_dma_state = true; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_init); @@ -144,6 +147,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 548cc1e4114b..860f8cad6602 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -186,6 +186,9 @@ int snd_hdac_ext_bus_get_ml_capabilities(struct hdac_ext_bus *ebus) hlink->lcaps = readl(hlink->ml_addr + AZX_REG_ML_LCAP); hlink->lsdiid = readw(hlink->ml_addr + AZX_REG_ML_LSDIID); + /* since link in On, update the ref */ + hlink->ref_count = 1; + list_add_tail(&hlink->list, &ebus->hlink_list); } @@ -327,3 +330,66 @@ int snd_hdac_ext_bus_link_power_down_all(struct hdac_ext_bus *ebus) return 0; } EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); + +int snd_hdac_ext_bus_link_get(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + + mutex_lock(&ebus->lock); + + /* + * if we move from 0 to 1, count will be 1 so power up this link + * as well, also check the dma status and trigger that + */ + if (++link->ref_count == 1) { + if (!ebus->cmd_dma_state) { + snd_hdac_bus_init_cmd_io(&ebus->bus); + ebus->cmd_dma_state = true; + } + + ret = snd_hdac_ext_bus_link_power_up(link); + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_get); + +int snd_hdac_ext_bus_link_put(struct hdac_ext_bus *ebus, + struct hdac_ext_link *link) +{ + int ret = 0; + struct hdac_ext_link *hlink; + bool link_up = false; + + mutex_lock(&ebus->lock); + + /* + * if we move from 1 to 0, count will be 0 + * so power down this link as well + */ + if (--link->ref_count == 0) { + ret = snd_hdac_ext_bus_link_power_down(link); + + /* + * now check if all links are off, if so turn off + * cmd dma as well + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (hlink->ref_count) { + link_up = true; + break; + } + } + + if (!link_up) { + snd_hdac_bus_stop_cmd_io(&ebus->bus); + ebus->cmd_dma_state = false; + } + } + + mutex_unlock(&ebus->lock); + return ret; +} +EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_put); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 8c486235c905..9fee464e5d49 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -80,6 +80,22 @@ void snd_hdac_bus_init_cmd_io(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_cmd_io); +/* wait for cmd dmas till they are stopped */ +static void hdac_wait_for_cmd_dmas(struct hdac_bus *bus) +{ + unsigned long timeout; + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, RIRBCTL) & AZX_RBCTL_DMA_EN) + && time_before(jiffies, timeout)) + udelay(10); + + timeout = jiffies + msecs_to_jiffies(100); + while ((snd_hdac_chip_readb(bus, CORBCTL) & AZX_CORBCTL_RUN) + && time_before(jiffies, timeout)) + udelay(10); +} + /** * snd_hdac_bus_stop_cmd_io - clean up CORB/RIRB buffers * @bus: HD-audio core bus @@ -90,6 +106,7 @@ void snd_hdac_bus_stop_cmd_io(struct hdac_bus *bus) /* disable ringbuffer DMAs */ snd_hdac_chip_writeb(bus, RIRBCTL, 0); snd_hdac_chip_writeb(bus, CORBCTL, 0); + hdac_wait_for_cmd_dmas(bus); /* disable unsolicited responses */ snd_hdac_chip_updatel(bus, GCTL, AZX_GCTL_UNSOL, 0); spin_unlock_irq(&bus->reg_lock); diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 607bbeaebddf..c9af022676c2 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -158,22 +158,40 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); -/* There is a fixed mapping between audio pin node and display port - * on current Intel platforms: +/* There is a fixed mapping between audio pin node and display port. + * on SNB, IVY, HSW, BSW, SKL, BXT, KBL: * Pin Widget 5 - PORT B (port = 1 in i915 driver) * Pin Widget 6 - PORT C (port = 2 in i915 driver) * Pin Widget 7 - PORT D (port = 3 in i915 driver) + * + * on VLV, ILK: + * Pin Widget 4 - PORT B (port = 1 in i915 driver) + * Pin Widget 5 - PORT C (port = 2 in i915 driver) + * Pin Widget 6 - PORT D (port = 3 in i915 driver) */ -static int pin2port(hda_nid_t pin_nid) +static int pin2port(struct hdac_device *codec, hda_nid_t pin_nid) { - if (WARN_ON(pin_nid < 5 || pin_nid > 7)) + int base_nid; + + switch (codec->vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + base_nid = 3; + break; + default: + base_nid = 4; + break; + } + + if (WARN_ON(pin_nid <= base_nid || pin_nid > base_nid + 3)) return -1; - return pin_nid - 4; + return pin_nid - base_nid; } /** * snd_hdac_sync_audio_rate - Set N/CTS based on the sample rate - * @bus: HDA core bus + * @codec: HDA codec * @nid: the pin widget NID * @rate: the sample rate to set * @@ -183,14 +201,15 @@ static int pin2port(hda_nid_t pin_nid) * This function sets N/CTS value based on the given sample rate. * Returns zero for success, or a negative error code. */ -int snd_hdac_sync_audio_rate(struct hdac_bus *bus, hda_nid_t nid, int rate) +int snd_hdac_sync_audio_rate(struct hdac_device *codec, hda_nid_t nid, int rate) { + struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; int port; if (!acomp || !acomp->ops || !acomp->ops->sync_audio_rate) return -ENODEV; - port = pin2port(nid); + port = pin2port(codec, nid); if (port < 0) return -EINVAL; return acomp->ops->sync_audio_rate(acomp->dev, port, rate); @@ -199,7 +218,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); /** * snd_hdac_acomp_get_eld - Get the audio state and ELD via component - * @bus: HDA core bus + * @codec: HDA codec * @nid: the pin widget NID * @audio_enabled: the pointer to store the current audio state * @buffer: the buffer pointer to store ELD bytes @@ -217,16 +236,17 @@ EXPORT_SYMBOL_GPL(snd_hdac_sync_audio_rate); * thus it may be over @max_bytes. If it's over @max_bytes, it implies * that only a part of ELD bytes have been fetched. */ -int snd_hdac_acomp_get_eld(struct hdac_bus *bus, hda_nid_t nid, +int snd_hdac_acomp_get_eld(struct hdac_device *codec, hda_nid_t nid, bool *audio_enabled, char *buffer, int max_bytes) { + struct hdac_bus *bus = codec->bus; struct i915_audio_component *acomp = bus->audio_component; int port; if (!acomp || !acomp->ops || !acomp->ops->get_eld) return -ENODEV; - port = pin2port(nid); + port = pin2port(codec, nid); if (port < 0) return -EINVAL; return acomp->ops->get_eld(acomp->dev, port, audio_enabled, @@ -338,6 +358,9 @@ int snd_hdac_i915_init(struct hdac_bus *bus) struct i915_audio_component *acomp; int ret; + if (WARN_ON(hdac_acomp)) + return -EBUSY; + if (!i915_gfx_present()) return -ENODEV; @@ -371,6 +394,7 @@ out_master_del: out_err: kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; dev_info(dev, "failed to add i915 component master (%d)\n", ret); return ret; @@ -404,6 +428,7 @@ int snd_hdac_i915_exit(struct hdac_bus *bus) kfree(acomp); bus->audio_component = NULL; + hdac_acomp = NULL; return 0; } diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index d7ec86263828..c6c75e7e0981 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -625,13 +625,30 @@ static void hdmi_cea_alloc_to_tlv_chmap(struct hdac_chmap *hchmap, WARN_ON(count != channels); } +static int spk_mask_from_spk_alloc(int spk_alloc) +{ + int i; + int spk_mask = eld_speaker_allocation_bits[0]; + + for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) { + if (spk_alloc & (1 << i)) + spk_mask |= eld_speaker_allocation_bits[i]; + } + + return spk_mask; +} + static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, unsigned int size, unsigned int __user *tlv) { struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdac_chmap *chmap = info->private_data; + int pcm_idx = kcontrol->private_value; unsigned int __user *dst; int chs, count = 0; + unsigned long max_chs; + int type; + int spk_alloc, spk_mask; if (size < 8) return -ENOMEM; @@ -639,40 +656,59 @@ static int hdmi_chmap_ctl_tlv(struct snd_kcontrol *kcontrol, int op_flag, return -EFAULT; size -= 8; dst = tlv + 2; - for (chs = 2; chs <= chmap->channels_max; chs++) { + + spk_alloc = chmap->ops.get_spk_alloc(chmap->hdac, pcm_idx); + spk_mask = spk_mask_from_spk_alloc(spk_alloc); + + max_chs = hweight_long(spk_mask); + + for (chs = 2; chs <= max_chs; chs++) { int i; struct hdac_cea_channel_speaker_allocation *cap; cap = channel_allocations; for (i = 0; i < ARRAY_SIZE(channel_allocations); i++, cap++) { int chs_bytes = chs * 4; - int type = chmap->ops.chmap_cea_alloc_validate_get_type( - chmap, cap, chs); unsigned int tlv_chmap[8]; - if (type < 0) + if (cap->channels != chs) + continue; + + if (!(cap->spk_mask == (spk_mask & cap->spk_mask))) continue; + + type = chmap->ops.chmap_cea_alloc_validate_get_type( + chmap, cap, chs); + if (type < 0) + return -ENODEV; if (size < 8) return -ENOMEM; + if (put_user(type, dst) || put_user(chs_bytes, dst + 1)) return -EFAULT; + dst += 2; size -= 8; count += 8; + if (size < chs_bytes) return -ENOMEM; + size -= chs_bytes; count += chs_bytes; chmap->ops.cea_alloc_to_tlv_chmap(chmap, cap, tlv_chmap, chs); + if (copy_to_user(dst, tlv_chmap, chs_bytes)) return -EFAULT; dst += chs; } } + if (put_user(count, tlv + 1)) return -EFAULT; + return 0; } diff --git a/sound/hda/local.h b/sound/hda/local.h index d692f417ddc0..0d5bb159d538 100644 --- a/sound/hda/local.h +++ b/sound/hda/local.h @@ -16,6 +16,16 @@ static inline int get_wcaps_type(unsigned int wcaps) return (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; } +static inline unsigned int get_wcaps_channels(u32 wcaps) +{ + unsigned int chans; + + chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = (chans + 1) * 2; + + return chans; +} + extern const struct attribute_group *hdac_dev_attr_groups[]; int hda_widget_sysfs_init(struct hdac_device *codec); void hda_widget_sysfs_exit(struct hdac_device *codec); diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff5693d..718d5e3b7806 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index b36ea47527e8..0b8d0de87273 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -1414,11 +1414,9 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m else { #ifdef CONFIG_ARCH_NETWINDER if (machine_is_netwinder()) { - init_timer(&vnc_timer); - vnc_timer.function = vnc_slider_tick; - vnc_timer.expires = jiffies; - vnc_timer.data = nr_waveartist_devs; - add_timer(&vnc_timer); + setup_timer(&vnc_timer, vnc_slider_tick, + nr_waveartist_devs); + mod_timer(&vnc_timer, jiffies); vnc_configure_mixer(devc, 0); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4667c3232b7f..4a054d720112 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2151,8 +2151,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if (stream->type != VORTEX_PCM_A3D) { @@ -2162,7 +2161,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXIN)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } @@ -2175,8 +2174,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_A3D)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); dev_err(vortex->card->dev, "out of A3D sources. Sorry\n"); return -EBUSY; @@ -2290,8 +2288,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXOUT)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if ((src[i] = @@ -2299,8 +2296,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index a6d6d8d0867a..df5741a78fd2 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -432,7 +432,10 @@ static snd_pcm_uframes_t snd_vortex_pcm_pointer(struct snd_pcm_substream *substr #endif //printk(KERN_INFO "vortex: pointer = 0x%x\n", current_ptr); spin_unlock(&chip->lock); - return (bytes_to_frames(substream->runtime, current_ptr)); + current_ptr = bytes_to_frames(substream->runtime, current_ptr); + if (current_ptr >= substream->runtime->buffer_size) + current_ptr = 0; + return current_ptr; } /* operators */ diff --git a/sound/pci/ctxfi/cttimer.c b/sound/pci/ctxfi/cttimer.c index a5d460453d7b..8f945341720b 100644 --- a/sound/pci/ctxfi/cttimer.c +++ b/sound/pci/ctxfi/cttimer.c @@ -49,7 +49,7 @@ struct ct_timer { spinlock_t lock; /* global timer lock (for xfitimer) */ spinlock_t list_lock; /* lock for instance list */ struct ct_atc *atc; - struct ct_timer_ops *ops; + const struct ct_timer_ops *ops; struct list_head instance_head; struct list_head running_head; unsigned int wc; /* current wallclock */ @@ -128,7 +128,7 @@ static void ct_systimer_prepare(struct ct_timer_instance *ti) #define ct_systimer_free ct_systimer_prepare -static struct ct_timer_ops ct_systimer_ops = { +static const struct ct_timer_ops ct_systimer_ops = { .init = ct_systimer_init, .free_instance = ct_systimer_free, .prepare = ct_systimer_prepare, @@ -322,7 +322,7 @@ static void ct_xfitimer_free_global(struct ct_timer *atimer) ct_xfitimer_irq_stop(atimer); } -static struct ct_timer_ops ct_xfitimer_ops = { +static const struct ct_timer_ops ct_xfitimer_ops = { .prepare = ct_xfitimer_prepare, .start = ct_xfitimer_start, .stop = ct_xfitimer_stop, diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 0dc44ebb0032..626cd2167d29 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1548,7 +1548,7 @@ static int snd_es1373_line_get(struct snd_kcontrol *kcontrol, int val = 0; spin_lock_irq(&ensoniq->reg_lock); - if ((ensoniq->ctrl & ES_1371_GPIO_OUTM) >= 4) + if (ensoniq->ctrl & ES_1371_GPIO_OUT(4)) val = 1; ucontrol->value.integer.value[0] = val; spin_unlock_irq(&ensoniq->reg_lock); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bb02c2d48fd5..7f3b5ed81995 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -50,9 +50,13 @@ config SND_HDA_RECONFIG bool "Allow dynamic codec reconfiguration" help Say Y here to enable the HD-audio codec re-configuration feature. - This adds the sysfs interfaces to allow user to clear the whole - codec configuration, change the codec setup, add extra verbs, - and re-configure the codec dynamically. + It allows user to clear the whole codec configuration, change the + codec setup, add extra verbs, and re-configure the codec dynamically. + + Note that this item alone doesn't provide the sysfs interface, but + enables the feature just for the patch loader below. + If you need the traditional sysfs entries for the manual interaction, + turn on CONFIG_SND_HDA_HWDEP as well. config SND_HDA_INPUT_BEEP bool "Support digital beep via input layer" diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index dfaf1a93fb8a..320445f3bf73 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -5434,6 +5434,7 @@ static int dyn_adc_capture_pcm_prepare(struct hda_pcm_stream *hinfo, spec->cur_adc_stream_tag = stream_tag; spec->cur_adc_format = format; snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format); + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_PREPARE); return 0; } @@ -5444,6 +5445,7 @@ static int dyn_adc_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct hda_gen_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->cur_adc); spec->cur_adc = 0; + call_pcm_capture_hook(hinfo, codec, substream, HDA_GEN_PCM_ACT_CLEANUP); return 0; } diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index a010d704e0e2..d0d5ad8beac5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -114,6 +114,9 @@ struct hdmi_ops { int (*setup_stream)(struct hda_codec *codec, hda_nid_t cvt_nid, hda_nid_t pin_nid, u32 stream_tag, int format); + void (*pin_cvt_fixup)(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid); }; struct hdmi_pcm { @@ -684,7 +687,8 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, if (!channels) return; - if (is_haswell_plus(codec)) + /* some HW (e.g. HSW+) needs reprogramming the amp at each time */ + if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); @@ -864,9 +868,6 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, struct hdmi_spec *spec = codec->spec; int err; - if (is_haswell_plus(codec)) - haswell_verify_D0(codec, cvt_nid, pin_nid); - err = spec->ops.pin_hbr_setup(codec, pin_nid, is_hbr_format(format)); if (err) { @@ -884,7 +885,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, * of the pin. */ static int hdmi_choose_cvt(struct hda_codec *codec, - int pin_idx, int *cvt_id, int *mux_id) + int pin_idx, int *cvt_id) { struct hdmi_spec *spec = codec->spec; struct hdmi_spec_per_pin *per_pin; @@ -925,8 +926,6 @@ static int hdmi_choose_cvt(struct hda_codec *codec, if (cvt_id) *cvt_id = cvt_idx; - if (mux_id) - *mux_id = mux_idx; return 0; } @@ -1019,9 +1018,6 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, int mux_idx; struct hdmi_spec *spec = codec->spec; - if (!is_haswell_plus(codec) && !is_valleyview_plus(codec)) - return; - /* On Intel platform, the mapping of converter nid to * mux index of the pins are always the same. * The pin nid may be 0, this means all pins will not @@ -1032,6 +1028,17 @@ static void intel_not_share_assigned_cvt_nid(struct hda_codec *codec, intel_not_share_assigned_cvt(codec, pin_nid, mux_idx); } +/* skeleton caller of pin_cvt_fixup ops */ +static void pin_cvt_fixup(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec->ops.pin_cvt_fixup) + spec->ops.pin_cvt_fixup(codec, per_pin, cvt_nid); +} + /* called in hdmi_pcm_open when no pin is assigned to the PCM * in dyn_pcm_assign mode. */ @@ -1049,7 +1056,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, if (pcm_idx < 0) return -EINVAL; - err = hdmi_choose_cvt(codec, -1, &cvt_idx, NULL); + err = hdmi_choose_cvt(codec, -1, &cvt_idx); if (err) return err; @@ -1057,7 +1064,7 @@ static int hdmi_pcm_open_no_pin(struct hda_pcm_stream *hinfo, per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - intel_not_share_assigned_cvt_nid(codec, 0, per_cvt->cvt_nid); + pin_cvt_fixup(codec, NULL, per_cvt->cvt_nid); set_bit(pcm_idx, &spec->pcm_in_use); /* todo: setup spdif ctls assign */ @@ -1089,7 +1096,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, { struct hdmi_spec *spec = codec->spec; struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, pcm_idx, mux_idx = 0; + int pin_idx, cvt_idx, pcm_idx; struct hdmi_spec_per_pin *per_pin; struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; @@ -1118,7 +1125,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, } } - err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx); if (err < 0) { mutex_unlock(&spec->pcm_lock); return err; @@ -1135,11 +1142,10 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + per_pin->mux_idx); /* configure unused pins to choose other converters */ - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) - intel_not_share_assigned_cvt(codec, per_pin->pin_nid, mux_idx); + pin_cvt_fixup(codec, per_pin, 0); snd_hda_spdif_ctls_assign(codec, pcm_idx, per_cvt->cvt_nid); @@ -1372,12 +1378,7 @@ static void update_eld(struct hda_codec *codec, * and this can make HW reset converter selection on a pin. */ if (eld->eld_valid && !old_eld_valid && per_pin->setup) { - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { - intel_verify_pin_cvt_connect(codec, per_pin); - intel_not_share_assigned_cvt(codec, per_pin->pin_nid, - per_pin->mux_idx); - } - + pin_cvt_fixup(codec, per_pin, 0); hdmi_setup_audio_infoframe(codec, per_pin, per_pin->non_pcm); } @@ -1484,7 +1485,7 @@ static void sync_eld_via_acomp(struct hda_codec *codec, mutex_lock(&per_pin->lock); eld->monitor_present = false; - size = snd_hdac_acomp_get_eld(&codec->bus->core, per_pin->pin_nid, + size = snd_hdac_acomp_get_eld(&codec->core, per_pin->pin_nid, &eld->monitor_present, eld->eld_buffer, ELD_MAX_SIZE); if (size > 0) { @@ -1711,7 +1712,7 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, * skip pin setup and return 0 to make audio playback * be ongoing */ - intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid); + pin_cvt_fixup(codec, NULL, cvt_nid); snd_hda_codec_setup_stream(codec, cvt_nid, stream_tag, 0, format); mutex_unlock(&spec->pcm_lock); @@ -1724,23 +1725,21 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, } per_pin = get_pin(spec, pin_idx); pin_nid = per_pin->pin_nid; - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) { - /* Verify pin:cvt selections to avoid silent audio after S3. - * After S3, the audio driver restores pin:cvt selections - * but this can happen before gfx is ready and such selection - * is overlooked by HW. Thus multiple pins can share a same - * default convertor and mute control will affect each other, - * which can cause a resumed audio playback become silent - * after S3. - */ - intel_verify_pin_cvt_connect(codec, per_pin); - intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx); - } + + /* Verify pin:cvt selections to avoid silent audio after S3. + * After S3, the audio driver restores pin:cvt selections + * but this can happen before gfx is ready and such selection + * is overlooked by HW. Thus multiple pins can share a same + * default convertor and mute control will affect each other, + * which can cause a resumed audio playback become silent + * after S3. + */ + pin_cvt_fixup(codec, per_pin, 0); /* Call sync_audio_rate to set the N/CTS/M manually if necessary */ /* Todo: add DP1.2 MST audio support later */ if (codec_has_acomp(codec)) - snd_hdac_sync_audio_rate(&codec->bus->core, pin_nid, runtime->rate); + snd_hdac_sync_audio_rate(&codec->core, pin_nid, runtime->rate); non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); mutex_lock(&per_pin->lock); @@ -1837,6 +1836,18 @@ static const struct hda_pcm_ops generic_ops = { .cleanup = generic_hdmi_playback_pcm_cleanup, }; +static int hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hda_codec *codec = container_of(hdac, struct hda_codec, core); + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin = pcm_idx_to_pin(spec, pcm_idx); + + if (!per_pin) + return 0; + + return per_pin->sink_eld.info.spk_alloc; +} + static void hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, unsigned char *chmap) { @@ -2075,6 +2086,20 @@ static void hdmi_array_free(struct hdmi_spec *spec) snd_array_free(&spec->cvts); } +static void generic_spec_free(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + if (spec) { + if (spec->i915_bound) + snd_hdac_i915_exit(&codec->bus->core); + hdmi_array_free(spec); + kfree(spec); + codec->spec = NULL; + } + codec->dp_mst = false; +} + static void generic_hdmi_free(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -2099,10 +2124,7 @@ static void generic_hdmi_free(struct hda_codec *codec) spec->pcm_rec[pcm_idx].jack = NULL; } - if (spec->i915_bound) - snd_hdac_i915_exit(&codec->bus->core); - hdmi_array_free(spec); - kfree(spec); + generic_spec_free(codec); } #ifdef CONFIG_PM @@ -2140,6 +2162,55 @@ static const struct hdmi_ops generic_standard_hdmi_ops = { .setup_stream = hdmi_setup_stream, }; +/* allocate codec->spec and assign/initialize generic parser ops */ +static int alloc_generic_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + + spec->ops = generic_standard_hdmi_ops; + mutex_init(&spec->pcm_lock); + snd_hdac_register_chmap_ops(&codec->core, &spec->chmap); + + spec->chmap.ops.get_chmap = hdmi_get_chmap; + spec->chmap.ops.set_chmap = hdmi_set_chmap; + spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; + spec->chmap.ops.get_spk_alloc = hdmi_get_spk_alloc, + + codec->spec = spec; + hdmi_array_init(spec, 4); + + codec->patch_ops = generic_hdmi_patch_ops; + + return 0; +} + +/* generic HDMI parser */ +static int patch_generic_hdmi(struct hda_codec *codec) +{ + int err; + + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; + } + + generic_hdmi_init_per_pins(codec); + return 0; +} + +/* + * Intel codec parsers and helpers + */ + static void intel_haswell_fixup_connect_list(struct hda_codec *codec, hda_nid_t nid) { @@ -2217,12 +2288,23 @@ static void haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, static void intel_pin_eld_notify(void *audio_ptr, int port) { struct hda_codec *codec = audio_ptr; - int pin_nid = port + 0x04; + int pin_nid; /* we assume only from port-B to port-D */ if (port < 1 || port > 3) return; + switch (codec->core.vendor_id) { + case 0x80860054: /* ILK */ + case 0x80862804: /* ILK */ + case 0x80862882: /* VLV */ + pin_nid = port + 0x03; + break; + default: + pin_nid = port + 0x04; + break; + } + /* skip notification during system suspend (but not in runtime PM); * the state will be updated at resume */ @@ -2236,93 +2318,159 @@ static void intel_pin_eld_notify(void *audio_ptr, int port) check_presence_and_report(codec, pin_nid); } -static int patch_generic_hdmi(struct hda_codec *codec) +/* register i915 component pin_eld_notify callback */ +static void register_i915_notifier(struct hda_codec *codec) { - struct hdmi_spec *spec; + struct hdmi_spec *spec = codec->spec; - spec = kzalloc(sizeof(*spec), GFP_KERNEL); - if (spec == NULL) - return -ENOMEM; + spec->use_acomp_notifier = true; + spec->i915_audio_ops.audio_ptr = codec; + /* intel_audio_codec_enable() or intel_audio_codec_disable() + * will call pin_eld_notify with using audio_ptr pointer + * We need make sure audio_ptr is really setup + */ + wmb(); + spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; + snd_hdac_i915_register_notifier(&spec->i915_audio_ops); +} - spec->ops = generic_standard_hdmi_ops; - mutex_init(&spec->pcm_lock); - snd_hdac_register_chmap_ops(&codec->core, &spec->chmap); +/* setup_stream ops override for HSW+ */ +static int i915_hsw_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, + hda_nid_t pin_nid, u32 stream_tag, int format) +{ + haswell_verify_D0(codec, cvt_nid, pin_nid); + return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); +} - spec->chmap.ops.get_chmap = hdmi_get_chmap; - spec->chmap.ops.set_chmap = hdmi_set_chmap; - spec->chmap.ops.is_pcm_attached = is_hdmi_pcm_attached; +/* pin_cvt_fixup ops override for HSW+ and VLV+ */ +static void i915_pin_cvt_fixup(struct hda_codec *codec, + struct hdmi_spec_per_pin *per_pin, + hda_nid_t cvt_nid) +{ + if (per_pin) { + intel_verify_pin_cvt_connect(codec, per_pin); + intel_not_share_assigned_cvt(codec, per_pin->pin_nid, + per_pin->mux_idx); + } else { + intel_not_share_assigned_cvt_nid(codec, 0, cvt_nid); + } +} - codec->spec = spec; - hdmi_array_init(spec, 4); +/* Intel Haswell and onwards; audio component with eld notifier */ +static int patch_i915_hsw_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; -#ifdef CONFIG_SND_HDA_I915 - /* Try to bind with i915 for Intel HSW+ codecs (if not done yet) */ - if ((codec->core.vendor_id >> 16) == 0x8086 && - is_haswell_plus(codec)) { -#if 0 - /* on-demand binding leads to an unbalanced refcount when - * both i915 and hda drivers are probed concurrently; - * disabled temporarily for now - */ - if (!codec->bus->core.audio_component) - if (!snd_hdac_i915_init(&codec->bus->core)) - spec->i915_bound = true; -#endif - /* use i915 audio component notifier for hotplug */ - if (codec->bus->core.audio_component) - spec->use_acomp_notifier = true; + /* HSW+ requires i915 binding */ + if (!codec->bus->core.audio_component) { + codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + return -ENODEV; } -#endif - if (is_haswell_plus(codec)) { - intel_haswell_enable_all_pins(codec, true); - intel_haswell_fixup_enable_dp12(codec); - } + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; - /* For Valleyview/Cherryview, only the display codec is in the display - * power well and can use link_power ops to request/release the power. - * For Haswell/Broadwell, the controller is also in the power well and + intel_haswell_enable_all_pins(codec, true); + intel_haswell_fixup_enable_dp12(codec); + + /* For Haswell/Broadwell, the controller is also in the power well and * can cover the codec power request, and so need not set this flag. - * For previous platforms, there is no such power well feature. */ - if (is_valleyview_plus(codec) || is_skylake(codec) || - is_broxton(codec)) + if (!is_haswell(codec) && !is_broadwell(codec)) codec->core.link_power_control = 1; - if (hdmi_parse_codec(codec) < 0) { - if (spec->i915_bound) - snd_hdac_i915_exit(&codec->bus->core); - codec->spec = NULL; - kfree(spec); - return -EINVAL; + codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + codec->depop_delay = 0; + codec->auto_runtime_pm = 1; + + spec->ops.setup_stream = i915_hsw_setup_stream; + spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; } - codec->patch_ops = generic_hdmi_patch_ops; - if (is_haswell_plus(codec)) { - codec->patch_ops.set_power_state = haswell_set_power_state; - codec->dp_mst = true; + + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); + return 0; +} + +/* Intel Baytrail and Braswell; with eld notifier */ +static int patch_i915_byt_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + /* requires i915 binding */ + if (!codec->bus->core.audio_component) { + codec_info(codec, "No i915 binding for Intel HDMI/DP codec\n"); + return -ENODEV; } - /* Enable runtime pm for HDMI audio codec of HSW/BDW/SKL/BYT/BSW */ - if (is_haswell_plus(codec) || is_valleyview_plus(codec)) - codec->auto_runtime_pm = 1; + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; - generic_hdmi_init_per_pins(codec); + /* For Valleyview/Cherryview, only the display codec is in the display + * power well and can use link_power ops to request/release the power. + */ + codec->core.link_power_control = 1; + codec->depop_delay = 0; + codec->auto_runtime_pm = 1; - if (codec_has_acomp(codec)) { - codec->depop_delay = 0; - spec->i915_audio_ops.audio_ptr = codec; - /* intel_audio_codec_enable() or intel_audio_codec_disable() - * will call pin_eld_notify with using audio_ptr pointer - * We need make sure audio_ptr is really setup - */ - wmb(); - spec->i915_audio_ops.pin_eld_notify = intel_pin_eld_notify; - snd_hdac_i915_register_notifier(&spec->i915_audio_ops); + spec->ops.pin_cvt_fixup = i915_pin_cvt_fixup; + + err = hdmi_parse_codec(codec); + if (err < 0) { + generic_spec_free(codec); + return err; } - WARN_ON(spec->dyn_pcm_assign && !codec_has_acomp(codec)); + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); + return 0; +} + +/* Intel IronLake, SandyBridge and IvyBridge; with eld notifier */ +static int patch_i915_cpt_hdmi(struct hda_codec *codec) +{ + struct hdmi_spec *spec; + int err; + + /* no i915 component should have been bound before this */ + if (WARN_ON(codec->bus->core.audio_component)) + return -EBUSY; + + err = alloc_generic_hdmi(codec); + if (err < 0) + return err; + spec = codec->spec; + + /* Try to bind with i915 now */ + err = snd_hdac_i915_init(&codec->bus->core); + if (err < 0) + goto error; + spec->i915_bound = true; + + err = hdmi_parse_codec(codec); + if (err < 0) + goto error; + + generic_hdmi_init_per_pins(codec); + register_i915_notifier(codec); return 0; + + error: + generic_spec_free(codec); + return err; } /* @@ -3492,21 +3640,21 @@ HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi), HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi), HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_i915_cpt_hdmi), HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_i915_cpt_hdmi), +HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_i915_hsw_hdmi), +HDA_CODEC_ENTRY(0x8086280b, "Kabylake HDMI", patch_i915_hsw_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_generic_hdmi), -HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_generic_hdmi), +HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), +HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi), /* special ID for generic HDMI */ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4918ffa5ba68..002f153bc659 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -342,6 +342,11 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) case 0x10ec0293: alc_update_coef_idx(codec, 0xa, 1<<13, 0); break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + alc_update_coef_idx(codec, 0x10, 1<<15, 0); + break; case 0x10ec0662: if ((coef & 0x00f0) == 0x0030) alc_update_coef_idx(codec, 0x4, 1<<10, 0); /* EAPD Ctrl */ @@ -2647,6 +2652,7 @@ enum { ALC269_TYPE_ALC255, ALC269_TYPE_ALC256, ALC269_TYPE_ALC225, + ALC269_TYPE_ALC294, }; /* @@ -2677,6 +2683,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC255: case ALC269_TYPE_ALC256: case ALC269_TYPE_ALC225: + case ALC269_TYPE_ALC294: ssids = alc269_ssids; break; default: @@ -6028,6 +6035,11 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0225: spec->codec_variant = ALC269_TYPE_ALC225; break; + case 0x10ec0234: + case 0x10ec0274: + case 0x10ec0294: + spec->codec_variant = ALC269_TYPE_ALC294; + break; } if (snd_hda_codec_read(codec, 0x51, 0, AC_VERB_PARAMETERS, 0) == 0x10ec5505) { @@ -6942,6 +6954,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0225, "ALC225", patch_alc269), HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269), HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0234, "ALC234", patch_alc269), HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269), HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269), HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269), @@ -6952,6 +6965,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269), HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269), HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662), + HDA_CODEC_ENTRY(0x10ec0274, "ALC274", patch_alc269), HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269), HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269), HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269), @@ -6964,6 +6978,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = { HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269), HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269), HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269), + HDA_CODEC_ENTRY(0x10ec0294, "ALC294", patch_alc269), HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269), HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861), HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd), diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8151318a69a2..9720a30dbfff 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -42,12 +42,6 @@ #include <asm/pgtable.h> #include <asm/cacheflush.h> -#ifdef CONFIG_KVM_GUEST -#include <linux/kvm_para.h> -#else -#define kvm_para_available() (0) -#endif - MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Intel 82801AA,82901AB,i810,i820,i830,i840,i845,MX440; SiS 7012; Ali 5455"); MODULE_LICENSE("GPL"); @@ -2972,25 +2966,17 @@ static int snd_intel8x0_inside_vm(struct pci_dev *pci) goto fini; } - /* detect KVM and Parallels virtual environments */ - result = kvm_para_available(); -#ifdef X86_FEATURE_HYPERVISOR - result = result || boot_cpu_has(X86_FEATURE_HYPERVISOR); -#endif - if (!result) - goto fini; - /* check for known (emulated) devices */ + result = 0; if (pci->subsystem_vendor == PCI_SUBVENDOR_ID_REDHAT_QUMRANET && pci->subsystem_device == PCI_SUBDEVICE_ID_QEMU) { /* KVM emulated sound, PCI SSID: 1af4:1100 */ msg = "enable KVM"; + result = 1; } else if (pci->subsystem_vendor == 0x1ab8) { /* Parallels VM emulated sound, PCI SSID: 1ab8:xxxx */ msg = "enable Parallels VM"; - } else { - msg = "disable (unknown or VT-d) VM"; - result = 0; + result = 1; } fini: diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index f3d62020ef66..a80684bdc30d 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -644,7 +644,7 @@ static int lx_pipe_wait_for_state(struct lx6464es *chip, u32 pipe, if (err < 0) return err; - if (current_state == state) + if (!err && current_state == state) return 0; mdelay(1); diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 276897033639..1267e1af0fae 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 5741c0aa6c03..b5d1caa04d8e 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -206,8 +206,8 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, stype = substream->stream; pcd = to_dmadata(substream); - DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %d " - "runtime->min_align %d\n", + DBG("runtime->dma_area = 0x%08lx dma_addr_t = 0x%08lx dma_size = %zu " + "runtime->min_align %lu\n", (unsigned long)runtime->dma_area, (unsigned long)runtime->dma_addr, runtime->dma_bytes, runtime->min_align); diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 1c1f2210387b..6ba20498202e 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -259,6 +259,9 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: data_length = 16; break; + case SNDRV_PCM_FORMAT_S24_LE: + data_length = 24; + break; case SNDRV_PCM_FORMAT_S32_LE: data_length = 32; break; @@ -273,13 +276,20 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* otherwise calculate a fitting block ratio */ bclk_ratio = 2 * data_length; - /* set target clock rate*/ - clk_set_rate(dev->clk, sampling_rate * bclk_ratio); + /* Clock should only be set up here if CPU is clock master */ + switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + clk_set_rate(dev->clk, sampling_rate * bclk_ratio); + break; + default: + break; + } /* Setup the frame format */ format = BCM2835_I2S_CHEN; - if (data_length > 24) + if (data_length >= 24) format |= BCM2835_I2S_CHWEX; format |= BCM2835_I2S_CHWID((data_length-8)&0xf); @@ -570,6 +580,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .capture = { @@ -577,6 +588,7 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE + | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, .ops = &bcm2835_i2s_dai_ops, @@ -678,6 +690,15 @@ static int bcm2835_i2s_probe(struct platform_device *pdev) dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].maxburst = 2; dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].maxburst = 2; + /* + * Set the PACK flag to enable S16_LE support (2 S16_LE values + * packed into 32-bit transfers). + */ + dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].flags = + SND_DMAENGINE_PCM_DAI_FLAG_PACK; + dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].flags = + SND_DMAENGINE_PCM_DAI_FLAG_PACK; + /* BCLK ratio - use default */ dev->bclk_ratio = 0; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c16478..b3afae990e39 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -88,12 +88,14 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_NAU8825 if I2C + select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM179X_I2C if I2C select SND_SOC_PCM179X_SPI if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_PCM3168A_I2C if I2C select SND_SOC_PCM3168A_SPI if SPI_MASTER + select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C @@ -477,6 +479,11 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate + select SND_PCM_ELD + select SND_PCM_IEC958 + config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" @@ -575,6 +582,9 @@ config SND_SOC_PCM3168A_SPI select SND_SOC_PCM3168A select REGMAP_SPI +config SND_SOC_PCM5102A + tristate + config SND_SOC_PCM512x tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..b7b99416537f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -81,6 +81,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o +snd-soc-hdmi-codec-objs := hdmi-codec.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm179x-codec-objs := pcm179x.o snd-soc-pcm179x-i2c-objs := pcm179x-i2c.o @@ -89,6 +90,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm3168a-objs := pcm3168a.o snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o +snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o @@ -290,6 +292,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM179X) += snd-soc-pcm179x-codec.o obj-$(CONFIG_SND_SOC_PCM179X_I2C) += snd-soc-pcm179x-i2c.o @@ -298,6 +301,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o +obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index cda27c22812a..1ee8506c06c7 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -608,9 +608,7 @@ static struct clk *ak4642_of_parse_mcko(struct device *dev) of_property_read_string(np, "clock-output-names", &clk_name); - clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, - (parent_clk_name) ? 0 : CLK_IS_ROOT, - rate); + clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name, 0, rate); if (!IS_ERR(clk)) of_clk_add_provider(np, of_clk_src_simple_get, clk); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 83959312f7a0..664a8c044ffb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -221,6 +221,8 @@ int arizona_init_spk(struct snd_soc_codec *codec) switch (arizona->type) { case WM8997: + case CS47L24: + case WM1831: break; default: ret = snd_soc_dapm_new_controls(dapm, &arizona_spkr, 1); @@ -1134,7 +1136,6 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w, int event) { struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - unsigned int mask = 0x3 << w->shift; unsigned int val; switch (event) { @@ -1148,7 +1149,7 @@ int arizona_anc_ev(struct snd_soc_dapm_widget *w, return 0; } - snd_soc_update_bits(codec, ARIZONA_CLOCK_CONTROL, mask, val); + snd_soc_write(codec, ARIZONA_CLOCK_CONTROL, val); return 0; } @@ -2047,7 +2048,21 @@ static int arizona_calc_fratio(struct arizona_fll *fll, init_ratio, Fref, refdiv); while (div <= ARIZONA_FLL_MAX_REFDIV) { - for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + /* start from init_ratio because this may already give a + * fractional N.K + */ + for (ratio = init_ratio; ratio > 0; ratio--) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + arizona_fll_dbg(fll, + "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", + Fref, refdiv, div, ratio); + return ratio; + } + } + + for (ratio = init_ratio + 1; ratio <= ARIZONA_FLL_MAX_FRATIO; ratio++) { if ((ARIZONA_FLL_VCO_CORNER / 2) / (fll->vco_mult * ratio) < Fref) { @@ -2073,17 +2088,6 @@ static int arizona_calc_fratio(struct arizona_fll *fll, } } - for (ratio = init_ratio - 1; ratio > 0; ratio--) { - if (target % (ratio * Fref)) { - cfg->refdiv = refdiv; - cfg->fratio = ratio - 1; - arizona_fll_dbg(fll, - "pseudo: found fref=%u refdiv=%d(%d) ratio=%d\n", - Fref, refdiv, div, ratio); - return ratio; - } - } - div *= 2; Fref /= 2; refdiv++; diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 7cd5f769bb61..eec1ff853b98 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -56,7 +56,7 @@ struct cs42l56_private { u8 iface; u8 iface_fmt; u8 iface_inv; -#if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) +#if IS_ENABLED(CONFIG_INPUT) struct input_dev *beep; struct work_struct beep_work; int beep_rate; diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 00e9b6fc1b5c..5ec5a682d186 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -807,6 +807,9 @@ static const struct snd_soc_dapm_route cs47l24_dapm_routes[] = { { "IN2L PGA", NULL, "IN2L" }, { "IN2R PGA", NULL, "IN2R" }, + { "Audio Trace DSP", NULL, "DSP2" }, + { "Audio Trace DSP", NULL, "SYSCLK" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), @@ -1016,6 +1019,27 @@ static struct snd_soc_dai_driver cs47l24_dai[] = { .formats = CS47L24_FORMATS, }, }, + { + .name = "cs47l24-cpu-trace", + .capture = { + .stream_name = "Audio Trace CPU", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + .compress_new = snd_soc_new_compress, + }, + { + .name = "cs47l24-dsp-trace", + .capture = { + .stream_name = "Audio Trace DSP", + .channels_min = 1, + .channels_max = 6, + .rates = CS47L24_RATES, + .formats = CS47L24_FORMATS, + }, + }, }; static int cs47l24_open(struct snd_compr_stream *stream) @@ -1027,6 +1051,8 @@ static int cs47l24_open(struct snd_compr_stream *stream) if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-voicectrl") == 0) { n_adsp = 2; + } else if (strcmp(rtd->codec_dai->name, "cs47l24-dsp-trace") == 0) { + n_adsp = 1; } else { dev_err(arizona->dev, "No suitable compressed stream for DAI '%s'\n", @@ -1041,10 +1067,16 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data) { struct cs47l24_priv *priv = data; struct arizona *arizona = priv->core.arizona; - int ret; + int serviced = 0; + int i, ret; - ret = wm_adsp_compr_handle_irq(&priv->core.adsp[2]); - if (ret == -ENODEV) { + for (i = 1; i <= 2; ++i) { + ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]); + if (ret != -ENODEV) + serviced++; + } + + if (!serviced) { dev_err(arizona->dev, "Spurious compressed data IRQ\n"); return IRQ_NONE; } @@ -1160,6 +1192,7 @@ static struct snd_compr_ops cs47l24_compr_ops = { static struct snd_soc_platform_driver cs47l24_compr_platform = { .compr_ops = &cs47l24_compr_ops, }; + static int cs47l24_probe(struct platform_device *pdev) { struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); @@ -1228,9 +1261,9 @@ static int cs47l24_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Failed to register platform: %d\n", ret); return ret; } + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cs47l24, cs47l24_dai, ARRAY_SIZE(cs47l24_dai)); - if (ret < 0) { dev_err(&pdev->dev, "Failed to register codec: %d\n", ret); snd_soc_unregister_platform(&pdev->dev); @@ -1241,10 +1274,15 @@ static int cs47l24_probe(struct platform_device *pdev) static int cs47l24_remove(struct platform_device *pdev) { + struct cs47l24_priv *cs47l24 = platform_get_drvdata(pdev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + wm_adsp2_remove(&cs47l24->core.adsp[1]); + wm_adsp2_remove(&cs47l24->core.adsp[2]); + return 0; } diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 7278f93460c1..e5527bc570ae 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -726,6 +726,68 @@ static const struct snd_kcontrol_new da7213_dapm_mixoutr_controls[] = { /* + * DAPM Events + */ + +static int da7213_dai_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable DAI clks for master mode */ + if (da7213->master) + snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, + DA7213_DAI_CLK_EN_MASK, + DA7213_DAI_CLK_EN_MASK); + + /* PC synchronised to DAI */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, + DA7213_PC_FREERUN_MASK, 0); + + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_read(codec, DA7213_PLL_CTRL); + if (!(pll_ctrl & DA7213_PLL_SRM_EN)) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_read(codec, DA7213_PLL_STATUS); + if (pll_status & DA7219_PLL_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7213_SRM_CHECK_RETRIES) & (!srm_lock)); + + if (!srm_lock) + dev_warn(codec->dev, "SRM failed to lock\n"); + + return 0; + case SND_SOC_DAPM_POST_PMD: + /* PC free-running */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, + DA7213_PC_FREERUN_MASK, + DA7213_PC_FREERUN_MASK); + + /* Disable DAI clks if in master mode */ + if (da7213->master) + snd_soc_update_bits(codec, DA7213_DAI_CLK_MODE, + DA7213_DAI_CLK_EN_MASK, 0); + return 0; + default: + return -EINVAL; + } +} + + +/* * DAPM widgets */ @@ -736,7 +798,8 @@ static const struct snd_soc_dapm_widget da7213_dapm_widgets[] = { /* Use a supply here as this controls both input & output DAIs */ SND_SOC_DAPM_SUPPLY("DAI", DA7213_DAI_CTRL, DA7213_DAI_EN_SHIFT, - DA7213_NO_INVERT, NULL, 0), + DA7213_NO_INVERT, da7213_dai_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), /* * Input @@ -1143,11 +1206,9 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* Set master/slave mode */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - dai_clk_mode |= DA7213_DAI_CLK_EN_MASTER_MODE; da7213->master = true; break; case SND_SOC_DAIFMT_CBS_CFS: - dai_clk_mode |= DA7213_DAI_CLK_EN_SLAVE_MODE; da7213->master = false; break; default: @@ -1281,28 +1342,28 @@ static int da7213_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, pll_ctrl = 0; /* Workout input divider based on MCLK rate */ - if ((da7213->mclk_rate == 32768) && (source == DA7213_SYSCLK_PLL)) { + if (da7213->mclk_rate == 32768) { /* 32KHz PLL Mode */ - indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; - indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; + indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL; freq_ref = 3750000; pll_ctrl |= DA7213_PLL_32K_MODE; } else { /* 5 - 54MHz MCLK */ if (da7213->mclk_rate < 5000000) { goto pll_err; - } else if (da7213->mclk_rate <= 10000000) { - indiv_bits = DA7213_PLL_INDIV_5_10_MHZ; - indiv = DA7213_PLL_INDIV_5_10_MHZ_VAL; - } else if (da7213->mclk_rate <= 20000000) { - indiv_bits = DA7213_PLL_INDIV_10_20_MHZ; - indiv = DA7213_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7213->mclk_rate <= 40000000) { - indiv_bits = DA7213_PLL_INDIV_20_40_MHZ; - indiv = DA7213_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7213->mclk_rate <= 9000000) { + indiv_bits = DA7213_PLL_INDIV_5_TO_9_MHZ; + indiv = DA7213_PLL_INDIV_5_TO_9_MHZ_VAL; + } else if (da7213->mclk_rate <= 18000000) { + indiv_bits = DA7213_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7213_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7213->mclk_rate <= 36000000) { + indiv_bits = DA7213_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7213_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7213->mclk_rate <= 54000000) { - indiv_bits = DA7213_PLL_INDIV_40_54_MHZ; - indiv = DA7213_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7213_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7213_PLL_INDIV_36_TO_54_MHZ_VAL; } else { goto pll_err; } @@ -1547,6 +1608,10 @@ static int da7213_probe(struct snd_soc_codec *codec) /* Default to using SRM for slave mode */ da7213->srm_en = true; + /* Default PC counter to free-running */ + snd_soc_update_bits(codec, DA7213_PC_COUNT, DA7213_PC_FREERUN_MASK, + DA7213_PC_FREERUN_MASK); + /* Enable all Gain Ramps */ snd_soc_update_bits(codec, DA7213_AUX_L_CTRL, DA7213_GAIN_RAMP_EN, DA7213_GAIN_RAMP_EN); diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h index 030fd691b076..fbb7a356a501 100644 --- a/sound/soc/codecs/da7213.h +++ b/sound/soc/codecs/da7213.h @@ -142,6 +142,9 @@ * Bit fields */ +/* DA7213_PLL_STATUS = 0x03 */ +#define DA7219_PLL_SRM_LOCK (0x1 << 1) + /* DA7213_SR = 0x22 */ #define DA7213_SR_8000 (0x1 << 0) #define DA7213_SR_11025 (0x2 << 0) @@ -160,10 +163,10 @@ #define DA7213_VMID_EN (0x1 << 7) /* DA7213_PLL_CTRL = 0x27 */ -#define DA7213_PLL_INDIV_5_10_MHZ (0x0 << 2) -#define DA7213_PLL_INDIV_10_20_MHZ (0x1 << 2) -#define DA7213_PLL_INDIV_20_40_MHZ (0x2 << 2) -#define DA7213_PLL_INDIV_40_54_MHZ (0x3 << 2) +#define DA7213_PLL_INDIV_5_TO_9_MHZ (0x0 << 2) +#define DA7213_PLL_INDIV_9_TO_18_MHZ (0x1 << 2) +#define DA7213_PLL_INDIV_18_TO_36_MHZ (0x2 << 2) +#define DA7213_PLL_INDIV_36_TO_54_MHZ (0x3 << 2) #define DA7213_PLL_INDIV_MASK (0x3 << 2) #define DA7213_PLL_MCLK_SQR_EN (0x1 << 4) #define DA7213_PLL_32K_MODE (0x1 << 5) @@ -178,8 +181,6 @@ #define DA7213_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0) #define DA7213_DAI_CLK_POL_INV (0x1 << 2) #define DA7213_DAI_WCLK_POL_INV (0x1 << 3) -#define DA7213_DAI_CLK_EN_SLAVE_MODE (0x0 << 7) -#define DA7213_DAI_CLK_EN_MASTER_MODE (0x1 << 7) #define DA7213_DAI_CLK_EN_MASK (0x1 << 7) /* DA7213_DAI_CTRL = 0x29 */ @@ -412,6 +413,9 @@ #define DA7213_DMIC_CLK_RATE_SHIFT 2 #define DA7213_DMIC_CLK_RATE_MASK (0x1 << 2) +/* DA7213_PC_COUNT = 0x94 */ +#define DA7213_PC_FREERUN_MASK (0x1 << 0) + /* DA7213_DIG_CTRL = 0x99 */ #define DA7213_DAC_L_INV_SHIFT 3 #define DA7213_DAC_R_INV_SHIFT 7 @@ -495,15 +499,16 @@ #define DA7213_ALC_AVG_ITERATIONS 5 /* PLL related */ -#define DA7213_SYSCLK_MCLK 0 -#define DA7213_SYSCLK_PLL 1 -#define DA7213_PLL_FREQ_OUT_90316800 90316800 -#define DA7213_PLL_FREQ_OUT_98304000 98304000 -#define DA7213_PLL_FREQ_OUT_94310400 94310400 -#define DA7213_PLL_INDIV_5_10_MHZ_VAL 2 -#define DA7213_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7213_SYSCLK_MCLK 0 +#define DA7213_SYSCLK_PLL 1 +#define DA7213_PLL_FREQ_OUT_90316800 90316800 +#define DA7213_PLL_FREQ_OUT_98304000 98304000 +#define DA7213_PLL_FREQ_OUT_94310400 94310400 +#define DA7213_PLL_INDIV_5_TO_9_MHZ_VAL 2 +#define DA7213_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7213_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7213_PLL_INDIV_36_TO_54_MHZ_VAL 16 +#define DA7213_SRM_CHECK_RETRIES 8 enum da7213_clk_src { DA7213_CLKSRC_MCLK = 0, diff --git a/sound/soc/codecs/da7218.c b/sound/soc/codecs/da7218.c index 93575f251866..99ce23e113bf 100644 --- a/sound/soc/codecs/da7218.c +++ b/sound/soc/codecs/da7218.c @@ -1868,27 +1868,27 @@ static int da7218_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */ if (da7218->mclk_rate == 32768) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; } else if (da7218->mclk_rate < 2000000) { dev_err(codec->dev, "PLL input clock %d below valid range\n", da7218->mclk_rate); return -EINVAL; - } else if (da7218->mclk_rate <= 5000000) { - indiv_bits = DA7218_PLL_INDIV_2_5_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 10000000) { - indiv_bits = DA7218_PLL_INDIV_5_10_MHZ; - indiv = DA7218_PLL_INDIV_2_10_MHZ_VAL; - } else if (da7218->mclk_rate <= 20000000) { - indiv_bits = DA7218_PLL_INDIV_10_20_MHZ; - indiv = DA7218_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7218->mclk_rate <= 40000000) { - indiv_bits = DA7218_PLL_INDIV_20_40_MHZ; - indiv = DA7218_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7218->mclk_rate <= 4500000) { + indiv_bits = DA7218_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7218->mclk_rate <= 9000000) { + indiv_bits = DA7218_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7218->mclk_rate <= 18000000) { + indiv_bits = DA7218_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7218_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7218->mclk_rate <= 36000000) { + indiv_bits = DA7218_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7218_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7218->mclk_rate <= 54000000) { - indiv_bits = DA7218_PLL_INDIV_40_54_MHZ; - indiv = DA7218_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7218_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7218_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7218->mclk_rate); diff --git a/sound/soc/codecs/da7218.h b/sound/soc/codecs/da7218.h index c2c59049a2ad..477cd37723cf 100644 --- a/sound/soc/codecs/da7218.h +++ b/sound/soc/codecs/da7218.h @@ -876,15 +876,11 @@ /* DA7218_PLL_CTRL = 0x91 */ #define DA7218_PLL_INDIV_SHIFT 0 #define DA7218_PLL_INDIV_MASK (0x7 << 0) -#define DA7218_PLL_INDIV_2_5_MHZ (0x0 << 0) -#define DA7218_PLL_INDIV_5_10_MHZ (0x1 << 0) -#define DA7218_PLL_INDIV_10_20_MHZ (0x2 << 0) -#define DA7218_PLL_INDIV_20_40_MHZ (0x3 << 0) -#define DA7218_PLL_INDIV_40_54_MHZ (0x4 << 0) -#define DA7218_PLL_INDIV_2_10_MHZ_VAL 2 -#define DA7218_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7218_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7218_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 0) +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 0) +#define DA7218_PLL_INDIV_9_TO_18_MHZ (0x2 << 0) +#define DA7218_PLL_INDIV_18_TO_36_MHZ (0x3 << 0) +#define DA7218_PLL_INDIV_36_TO_54_MHZ (0x4 << 0) #define DA7218_PLL_MCLK_SQR_EN_SHIFT 4 #define DA7218_PLL_MCLK_SQR_EN_MASK (0x1 << 4) #define DA7218_PLL_MODE_SHIFT 6 @@ -1336,6 +1332,13 @@ #define DA7218_PLL_FREQ_OUT_90316 90316800 #define DA7218_PLL_FREQ_OUT_98304 98304000 +/* PLL Frequency Dividers */ +#define DA7218_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7218_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7218_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7218_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7218_PLL_INDIV_36_TO_54_MHZ_VAL 16 + /* ALC Calibration */ #define DA7218_ALC_CALIB_DELAY_MIN 2500 #define DA7218_ALC_CALIB_DELAY_MAX 5000 diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 81c0708b85c1..5c93899f1f0e 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -11,6 +11,7 @@ * option) any later version. */ +#include <linux/acpi.h> #include <linux/clk.h> #include <linux/i2c.h> #include <linux/of_device.h> @@ -1025,7 +1026,7 @@ static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq)) return 0; - if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) { + if ((freq < 2000000) || (freq > 54000000)) { dev_err(codec_dai->dev, "Unsupported MCLK value %d\n", freq); return -EINVAL; @@ -1079,21 +1080,21 @@ static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, dev_err(codec->dev, "PLL input clock %d below valid range\n", da7219->mclk_rate); return -EINVAL; - } else if (da7219->mclk_rate <= 5000000) { - indiv_bits = DA7219_PLL_INDIV_2_5_MHZ; - indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL; - } else if (da7219->mclk_rate <= 10000000) { - indiv_bits = DA7219_PLL_INDIV_5_10_MHZ; - indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL; - } else if (da7219->mclk_rate <= 20000000) { - indiv_bits = DA7219_PLL_INDIV_10_20_MHZ; - indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL; - } else if (da7219->mclk_rate <= 40000000) { - indiv_bits = DA7219_PLL_INDIV_20_40_MHZ; - indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL; + } else if (da7219->mclk_rate <= 4500000) { + indiv_bits = DA7219_PLL_INDIV_2_TO_4_5_MHZ; + indiv = DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL; + } else if (da7219->mclk_rate <= 9000000) { + indiv_bits = DA7219_PLL_INDIV_4_5_TO_9_MHZ; + indiv = DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL; + } else if (da7219->mclk_rate <= 18000000) { + indiv_bits = DA7219_PLL_INDIV_9_TO_18_MHZ; + indiv = DA7219_PLL_INDIV_9_TO_18_MHZ_VAL; + } else if (da7219->mclk_rate <= 36000000) { + indiv_bits = DA7219_PLL_INDIV_18_TO_36_MHZ; + indiv = DA7219_PLL_INDIV_18_TO_36_MHZ_VAL; } else if (da7219->mclk_rate <= 54000000) { - indiv_bits = DA7219_PLL_INDIV_40_54_MHZ; - indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL; + indiv_bits = DA7219_PLL_INDIV_36_TO_54_MHZ; + indiv = DA7219_PLL_INDIV_36_TO_54_MHZ_VAL; } else { dev_err(codec->dev, "PLL input clock %d above valid range\n", da7219->mclk_rate); @@ -1426,6 +1427,12 @@ static const struct of_device_id da7219_of_match[] = { }; MODULE_DEVICE_TABLE(of, da7219_of_match); +static const struct acpi_device_id da7219_acpi_match[] = { + { .id = "DLGS7219", }, + { } +}; +MODULE_DEVICE_TABLE(acpi, da7219_acpi_match); + static enum da7219_micbias_voltage da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val) { @@ -1955,6 +1962,7 @@ static struct i2c_driver da7219_i2c_driver = { .driver = { .name = "da7219", .of_match_table = of_match_ptr(da7219_of_match), + .acpi_match_table = ACPI_PTR(da7219_acpi_match), }, .probe = da7219_i2c_probe, .remove = da7219_i2c_remove, diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 5a787e738084..ff2a2f02ce40 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -194,11 +194,11 @@ /* DA7219_PLL_CTRL = 0x20 */ #define DA7219_PLL_INDIV_SHIFT 2 #define DA7219_PLL_INDIV_MASK (0x7 << 2) -#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2) -#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2) -#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2) -#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2) -#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2) +#define DA7219_PLL_INDIV_2_TO_4_5_MHZ (0x0 << 2) +#define DA7219_PLL_INDIV_4_5_TO_9_MHZ (0x1 << 2) +#define DA7219_PLL_INDIV_9_TO_18_MHZ (0x2 << 2) +#define DA7219_PLL_INDIV_18_TO_36_MHZ (0x3 << 2) +#define DA7219_PLL_INDIV_36_TO_54_MHZ (0x4 << 2) #define DA7219_PLL_MCLK_SQR_EN_SHIFT 5 #define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5) #define DA7219_PLL_MODE_SHIFT 6 @@ -761,11 +761,11 @@ #define DA7219_PLL_FREQ_OUT_98304 98304000 /* PLL Frequency Dividers */ -#define DA7219_PLL_INDIV_2_5_MHZ_VAL 1 -#define DA7219_PLL_INDIV_5_10_MHZ_VAL 2 -#define DA7219_PLL_INDIV_10_20_MHZ_VAL 4 -#define DA7219_PLL_INDIV_20_40_MHZ_VAL 8 -#define DA7219_PLL_INDIV_40_54_MHZ_VAL 16 +#define DA7219_PLL_INDIV_2_TO_4_5_MHZ_VAL 1 +#define DA7219_PLL_INDIV_4_5_TO_9_MHZ_VAL 2 +#define DA7219_PLL_INDIV_9_TO_18_MHZ_VAL 4 +#define DA7219_PLL_INDIV_18_TO_36_MHZ_VAL 8 +#define DA7219_PLL_INDIV_36_TO_54_MHZ_VAL 16 /* SRM */ #define DA7219_SRM_CHECK_RETRIES 8 diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index afa6c5db9dcc..2086d7107622 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -26,18 +26,30 @@ #include <sound/tlv.h> #include "es8328.h" -#define ES8328_SYSCLK_RATE_1X 11289600 -#define ES8328_SYSCLK_RATE_2X 22579200 +static const unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 32000, 48000, 96000, +}; -/* Run the codec at 22.5792 or 11.2896 MHz to support these rates */ -static struct { - int rate; - u8 ratio; -} mclk_ratios[] = { - { 8000, 9 }, - {11025, 7 }, - {22050, 4 }, - {44100, 2 }, +static const int ratios_12288[] = { + 10, 7, 6, 4, 3, 2, 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static const unsigned int rates_11289[] = { + 8018, 11025, 22050, 44100, 88200, +}; + +static const int ratios_11289[] = { + 9, 7, 4, 2, 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_11289 = { + .count = ARRAY_SIZE(rates_11289), + .list = rates_11289, }; /* regulator supplies for sgtl5000, VDDD is an optional external supply */ @@ -57,16 +69,28 @@ static const char * const supply_names[ES8328_SUPPLY_NUM] = { "HPVDD", }; -#define ES8328_RATES (SNDRV_PCM_RATE_44100 | \ +#define ES8328_RATES (SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_11025) -#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_8000) +#define ES8328_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) struct es8328_priv { struct regmap *regmap; struct clk *clk; int playback_fs; bool deemph; + int mclkdiv2; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; + const int *mclk_ratios; struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; }; @@ -439,54 +463,131 @@ static int es8328_mute(struct snd_soc_dai *dai, int mute) mute ? ES8328_DACCONTROL3_DACMUTE : 0); } +static int es8328_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + + if (es8328->sysclk_constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + es8328->sysclk_constraints); + + return 0; +} + static int es8328_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 es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int clk_rate; int i; int reg; - u8 ratio; + int wl; + int ratio; + + if (!es8328->sysclk_constraints) { + dev_err(codec->dev, "No MCLK configured\n"); + return -EINVAL; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = ES8328_DACCONTROL2; else reg = ES8328_ADCCONTROL5; - clk_rate = clk_get_rate(es8328->clk); + for (i = 0; i < es8328->sysclk_constraints->count; i++) + if (es8328->sysclk_constraints->list[i] == params_rate(params)) + break; - if ((clk_rate != ES8328_SYSCLK_RATE_1X) && - (clk_rate != ES8328_SYSCLK_RATE_2X)) { - dev_err(codec->dev, - "%s: clock is running at %d Hz, not %d or %d Hz\n", - __func__, clk_rate, - ES8328_SYSCLK_RATE_1X, ES8328_SYSCLK_RATE_2X); + if (i == es8328->sysclk_constraints->count) { + dev_err(codec->dev, "LRCLK %d unsupported with current clock\n", + params_rate(params)); return -EINVAL; } - /* find master mode MCLK to sampling frequency ratio */ - ratio = mclk_ratios[0].rate; - for (i = 1; i < ARRAY_SIZE(mclk_ratios); i++) - if (params_rate(params) <= mclk_ratios[i].rate) - ratio = mclk_ratios[i].ratio; + ratio = es8328->mclk_ratios[i]; + snd_soc_update_bits(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MCLKDIV2, + es8328->mclkdiv2 ? ES8328_MASTERMODE_MCLKDIV2 : 0); + + switch (params_width(params)) { + case 16: + wl = 3; + break; + case 18: + wl = 2; + break; + case 20: + wl = 1; + break; + case 24: + wl = 0; + break; + case 32: + wl = 4; + break; + default: + return -EINVAL; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_update_bits(codec, ES8328_DACCONTROL1, + ES8328_DACCONTROL1_DACWL_MASK, + wl << ES8328_DACCONTROL1_DACWL_SHIFT); + es8328->playback_fs = params_rate(params); es8328_set_deemph(codec); - } + } else + snd_soc_update_bits(codec, ES8328_ADCCONTROL4, + ES8328_ADCCONTROL4_ADCWL_MASK, + wl << ES8328_ADCCONTROL4_ADCWL_SHIFT); return snd_soc_update_bits(codec, reg, ES8328_RATEMASK, ratio); } +static int es8328_set_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); + int mclkdiv2 = 0; + + switch (freq) { + case 0: + es8328->sysclk_constraints = NULL; + es8328->mclk_ratios = NULL; + break; + case 22579200: + mclkdiv2 = 1; + /* fallthru */ + case 11289600: + es8328->sysclk_constraints = &constraints_11289; + es8328->mclk_ratios = ratios_11289; + break; + case 24576000: + mclkdiv2 = 1; + /* fallthru */ + case 12288000: + es8328->sysclk_constraints = &constraints_12288; + es8328->mclk_ratios = ratios_12288; + break; + default: + return -EINVAL; + } + + es8328->mclkdiv2 = mclkdiv2; + return 0; +} + static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { struct snd_soc_codec *codec = codec_dai->codec; - struct es8328_priv *es8328 = snd_soc_codec_get_drvdata(codec); - int clk_rate; - u8 mode = ES8328_DACCONTROL1_DACWL_16; + u8 dac_mode = 0; + u8 adc_mode = 0; /* set master/slave audio interface */ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBM_CFM) @@ -495,13 +596,16 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_I2S; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_I2S; break; case SND_SOC_DAIFMT_RIGHT_J: - mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_RJUST; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_RJUST; break; case SND_SOC_DAIFMT_LEFT_J: - mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + dac_mode |= ES8328_DACCONTROL1_DACFORMAT_LJUST; + adc_mode |= ES8328_ADCCONTROL4_ADCFORMAT_LJUST; break; default: return -EINVAL; @@ -511,18 +615,14 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) return -EINVAL; - snd_soc_write(codec, ES8328_DACCONTROL1, mode); - snd_soc_write(codec, ES8328_ADCCONTROL4, mode); + snd_soc_update_bits(codec, ES8328_DACCONTROL1, + ES8328_DACCONTROL1_DACFORMAT_MASK, dac_mode); + snd_soc_update_bits(codec, ES8328_ADCCONTROL4, + ES8328_ADCCONTROL4_ADCFORMAT_MASK, adc_mode); /* Master serial port mode, with BCLK generated automatically */ - clk_rate = clk_get_rate(es8328->clk); - if (clk_rate == ES8328_SYSCLK_RATE_1X) - snd_soc_write(codec, ES8328_MASTERMODE, - ES8328_MASTERMODE_MSC); - else - snd_soc_write(codec, ES8328_MASTERMODE, - ES8328_MASTERMODE_MCLKDIV2 | - ES8328_MASTERMODE_MSC); + snd_soc_update_bits(codec, ES8328_MASTERMODE, + ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC); return 0; } @@ -579,8 +679,10 @@ static int es8328_set_bias_level(struct snd_soc_codec *codec, } static const struct snd_soc_dai_ops es8328_dai_ops = { + .startup = es8328_startup, .hw_params = es8328_hw_params, .digital_mute = es8328_mute, + .set_sysclk = es8328_set_sysclk, .set_fmt = es8328_set_dai_fmt, }; @@ -601,6 +703,7 @@ static struct snd_soc_dai_driver es8328_dai = { .formats = ES8328_FORMATS, }, .ops = &es8328_dai_ops, + .symmetric_rates = 1, }; static int es8328_suspend(struct snd_soc_codec *codec) @@ -708,6 +811,7 @@ const struct regmap_config es8328_regmap_config = { .val_bits = 8, .max_register = ES8328_REG_MAX, .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, }; EXPORT_SYMBOL_GPL(es8328_regmap_config); diff --git a/sound/soc/codecs/es8328.h b/sound/soc/codecs/es8328.h index 156c748c89c7..1a736e72a929 100644 --- a/sound/soc/codecs/es8328.h +++ b/sound/soc/codecs/es8328.h @@ -22,7 +22,7 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_CONTROL1_VMIDSEL_50k (1 << 0) #define ES8328_CONTROL1_VMIDSEL_500k (2 << 0) #define ES8328_CONTROL1_VMIDSEL_5k (3 << 0) -#define ES8328_CONTROL1_VMIDSEL_MASK (7 << 0) +#define ES8328_CONTROL1_VMIDSEL_MASK (3 << 0) #define ES8328_CONTROL1_ENREF (1 << 2) #define ES8328_CONTROL1_SEQEN (1 << 3) #define ES8328_CONTROL1_SAMEFS (1 << 4) @@ -84,7 +84,20 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL1 0x09 #define ES8328_ADCCONTROL2 0x0a #define ES8328_ADCCONTROL3 0x0b + #define ES8328_ADCCONTROL4 0x0c +#define ES8328_ADCCONTROL4_ADCFORMAT_MASK (3 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_I2S (0 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_LJUST (1 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_RJUST (2 << 0) +#define ES8328_ADCCONTROL4_ADCFORMAT_PCM (3 << 0) +#define ES8328_ADCCONTROL4_ADCWL_SHIFT 2 +#define ES8328_ADCCONTROL4_ADCWL_MASK (7 << 2) +#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_NORMAL (0 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_I2S_POL_INV (1 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK2 (0 << 5) +#define ES8328_ADCCONTROL4_ADCLRP_PCM_MSB_CLK1 (1 << 5) + #define ES8328_ADCCONTROL5 0x0d #define ES8328_ADCCONTROL5_RATEMASK (0x1f << 0) @@ -109,15 +122,13 @@ int es8328_probe(struct device *dev, struct regmap *regmap); #define ES8328_ADCCONTROL14 0x16 #define ES8328_DACCONTROL1 0x17 +#define ES8328_DACCONTROL1_DACFORMAT_MASK (3 << 1) #define ES8328_DACCONTROL1_DACFORMAT_I2S (0 << 1) #define ES8328_DACCONTROL1_DACFORMAT_LJUST (1 << 1) #define ES8328_DACCONTROL1_DACFORMAT_RJUST (2 << 1) #define ES8328_DACCONTROL1_DACFORMAT_PCM (3 << 1) -#define ES8328_DACCONTROL1_DACWL_24 (0 << 3) -#define ES8328_DACCONTROL1_DACWL_20 (1 << 3) -#define ES8328_DACCONTROL1_DACWL_18 (2 << 3) -#define ES8328_DACCONTROL1_DACWL_16 (3 << 3) -#define ES8328_DACCONTROL1_DACWL_32 (4 << 3) +#define ES8328_DACCONTROL1_DACWL_SHIFT 3 +#define ES8328_DACCONTROL1_DACWL_MASK (7 << 3) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_NORMAL (0 << 6) #define ES8328_DACCONTROL1_DACLRP_I2S_POL_INV (1 << 6) #define ES8328_DACCONTROL1_DACLRP_PCM_MSB_CLK2 (0 << 6) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index aaa038ffc8a5..181cd3bf0b92 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -29,6 +29,7 @@ #include <sound/hdaudio_ext.h> #include <sound/hda_i915.h> #include <sound/pcm_drm_eld.h> +#include <sound/hda_chmap.h> #include "../../hda/local.h" #include "hdac_hdmi.h" @@ -60,11 +61,17 @@ struct hdac_hdmi_cvt { struct hdac_hdmi_cvt_params params; }; +/* Currently only spk_alloc, more to be added */ +struct hdac_hdmi_parsed_eld { + u8 spk_alloc; +}; + struct hdac_hdmi_eld { bool monitor_present; bool eld_valid; int eld_size; char eld_buffer[ELD_MAX_SIZE]; + struct hdac_hdmi_parsed_eld info; }; struct hdac_hdmi_pin { @@ -76,6 +83,10 @@ struct hdac_hdmi_pin { struct hdac_ext_device *edev; int repoll_count; struct delayed_work work; + struct mutex lock; + bool chmap_set; + unsigned char chmap[8]; /* ALSA API channel-map */ + int channels; /* current number of channels */ }; struct hdac_hdmi_pcm { @@ -100,8 +111,22 @@ struct hdac_hdmi_priv { int num_pin; int num_cvt; struct mutex pin_mutex; + struct hdac_chmap chmap; }; +static struct hdac_hdmi_pcm *get_hdmi_pcm_from_id(struct hdac_hdmi_priv *hdmi, + int pcm_idx) +{ + struct hdac_hdmi_pcm *pcm; + + list_for_each_entry(pcm, &hdmi->pcm_list, head) { + if (pcm->pcm_id == pcm_idx) + return pcm; + } + + return NULL; +} + static inline struct hdac_ext_device *to_hda_ext_device(struct device *dev) { struct hdac_device *hdac = dev_to_hdac_dev(dev); @@ -278,26 +303,31 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, int i; const u8 *eld_buf; u8 conn_type; - int channels = 2; + int channels, ca; list_for_each_entry(pin, &hdmi->pin_list, head) { if (pin->nid == pin_nid) break; } + ca = snd_hdac_channel_allocation(&hdac->hdac, pin->eld.info.spk_alloc, + pin->channels, pin->chmap_set, true, pin->chmap); + + channels = snd_hdac_get_active_channels(ca); + hdmi->chmap.ops.set_channel_count(&hdac->hdac, cvt_nid, channels); + + snd_hdac_setup_channel_mapping(&hdmi->chmap, pin->nid, false, ca, + pin->channels, pin->chmap, pin->chmap_set); + eld_buf = pin->eld.eld_buffer; conn_type = drm_eld_get_conn_type(eld_buf); - /* setup channel count */ - snd_hdac_codec_write(&hdac->hdac, cvt_nid, 0, - AC_VERB_SET_CVT_CHAN_COUNT, channels - 1); - switch (conn_type) { case DRM_ELD_CONN_TYPE_HDMI: hdmi_audio_infoframe_init(&frame); - /* Default stereo for now */ frame.channels = channels; + frame.channel_allocation = ca; ret = hdmi_audio_infoframe_pack(&frame, buffer, sizeof(buffer)); if (ret < 0) @@ -311,7 +341,7 @@ static int hdac_hdmi_setup_audio_infoframe(struct hdac_ext_device *hdac, dp_ai.len = 0x1b; dp_ai.ver = 0x11 << 2; dp_ai.CC02_CT47 = channels - 1; - dp_ai.CA = 0; + dp_ai.CA = ca; dip = (u8 *)&dp_ai; break; @@ -370,17 +400,23 @@ static int hdac_hdmi_playback_prepare(struct snd_pcm_substream *substream, struct hdac_ext_device *hdac = snd_soc_dai_get_drvdata(dai); struct hdac_hdmi_priv *hdmi = hdac->private_data; struct hdac_hdmi_dai_pin_map *dai_map; + struct hdac_hdmi_pin *pin; struct hdac_ext_dma_params *dd; int ret; dai_map = &hdmi->dai_map[dai->id]; + pin = dai_map->pin; dd = (struct hdac_ext_dma_params *)snd_soc_dai_get_dma_data(dai, substream); dev_dbg(&hdac->hdac.dev, "stream tag from cpu dai %d format in cvt 0x%x\n", dd->stream_tag, dd->format); + mutex_lock(&pin->lock); + pin->channels = substream->runtime->channels; + ret = hdac_hdmi_setup_audio_infoframe(hdac, dai_map->cvt->nid, dai_map->pin->nid); + mutex_unlock(&pin->lock); if (ret < 0) return ret; @@ -640,6 +676,12 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, snd_hdac_codec_write(&hdac->hdac, dai_map->pin->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + mutex_lock(&dai_map->pin->lock); + dai_map->pin->chmap_set = false; + memset(dai_map->pin->chmap, 0, sizeof(dai_map->pin->chmap)); + dai_map->pin->channels = 0; + mutex_unlock(&dai_map->pin->lock); + dai_map->pin = NULL; } } @@ -647,10 +689,19 @@ static void hdac_hdmi_pcm_close(struct snd_pcm_substream *substream, static int hdac_hdmi_query_cvt_params(struct hdac_device *hdac, struct hdac_hdmi_cvt *cvt) { + unsigned int chans; + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; int err; - /* Only stereo supported as of now */ - cvt->params.channels_min = cvt->params.channels_max = 2; + chans = get_wcaps(hdac, cvt->nid); + chans = get_wcaps_channels(chans); + + cvt->params.channels_min = 2; + + cvt->params.channels_max = chans; + if (chans > hdmi->chmap.channels_max) + hdmi->chmap.channels_max = chans; err = snd_hdac_query_supported_pcm(hdac, cvt->nid, &cvt->params.rates, @@ -1008,6 +1059,12 @@ static int hdac_hdmi_add_cvt(struct hdac_ext_device *edev, hda_nid_t nid) return hdac_hdmi_query_cvt_params(&edev->hdac, cvt); } +static void hdac_hdmi_parse_eld(struct hdac_ext_device *edev, + struct hdac_hdmi_pin *pin) +{ + pin->eld.info.spk_alloc = pin->eld.eld_buffer[DRM_ELD_SPEAKER]; +} + static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) { struct hdac_ext_device *edev = pin->edev; @@ -1065,6 +1122,7 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, int repoll) snd_jack_report(pcm->jack, SND_JACK_AVOUT); } + hdac_hdmi_parse_eld(edev, pin); print_hex_dump_bytes("ELD: ", DUMP_PREFIX_OFFSET, pin->eld.eld_buffer, pin->eld.eld_size); @@ -1123,6 +1181,7 @@ static int hdac_hdmi_add_pin(struct hdac_ext_device *edev, hda_nid_t nid) hdmi->num_pin++; pin->edev = edev; + mutex_init(&pin->lock); INIT_DELAYED_WORK(&pin->work, hdac_hdmi_repoll_eld); return 0; @@ -1342,6 +1401,19 @@ static struct i915_audio_component_audio_ops aops = { .pin_eld_notify = hdac_hdmi_eld_notify_cb, }; +static struct snd_pcm *hdac_hdmi_get_pcm_from_id(struct snd_soc_card *card, + int device) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + if (rtd->pcm && (rtd->pcm->device == device)) + return rtd->pcm; + } + + return NULL; +} + int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) { char jack_name[NAME_SIZE]; @@ -1351,6 +1423,8 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) snd_soc_component_get_dapm(&codec->component); struct hdac_hdmi_priv *hdmi = edev->private_data; struct hdac_hdmi_pcm *pcm; + struct snd_pcm *snd_pcm; + int err; /* * this is a new PCM device, create new pcm and @@ -1362,6 +1436,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device) pcm->pcm_id = device; pcm->cvt = hdmi->dai_map[dai->id].cvt; + snd_pcm = hdac_hdmi_get_pcm_from_id(dai->component->card, device); + if (snd_pcm) { + err = snd_hdac_add_chmap_ctls(snd_pcm, device, &hdmi->chmap); + if (err < 0) { + dev_err(&edev->hdac.dev, + "chmap control add failed with err: %d for pcm: %d\n", + err, device); + kfree(pcm); + return err; + } + } + list_add_tail(&pcm->head, &hdmi->pcm_list); sprintf(jack_name, "HDMI/DP, pcm=%d Jack", device); @@ -1378,10 +1464,18 @@ static int hdmi_codec_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(&codec->component); struct hdac_hdmi_pin *pin; + struct hdac_ext_link *hlink = NULL; int ret; edev->scodec = codec; + /* + * hold the ref while we probe, also no need to drop the ref on + * exit, we call pm_runtime_suspend() so that will do for us + */ + hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + snd_hdac_ext_bus_link_get(edev->ebus, hlink); + ret = create_fill_widget_route_map(dapm); if (ret < 0) return ret; @@ -1475,19 +1569,83 @@ static struct snd_soc_codec_driver hdmi_hda_codec = { .idle_bias_off = true, }; +static void hdac_hdmi_get_chmap(struct hdac_device *hdac, int pcm_idx, + unsigned char *chmap) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + /* chmap is already set to 0 in caller */ + if (!pin) + return; + + memcpy(chmap, pin->chmap, ARRAY_SIZE(pin->chmap)); +} + +static void hdac_hdmi_set_chmap(struct hdac_device *hdac, int pcm_idx, + unsigned char *chmap, int prepared) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + mutex_lock(&pin->lock); + pin->chmap_set = true; + memcpy(pin->chmap, chmap, ARRAY_SIZE(pin->chmap)); + if (prepared) + hdac_hdmi_setup_audio_infoframe(edev, pcm->cvt->nid, pin->nid); + mutex_unlock(&pin->lock); +} + +static bool is_hdac_hdmi_pcm_attached(struct hdac_device *hdac, int pcm_idx) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + return pin ? true:false; +} + +static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) +{ + struct hdac_ext_device *edev = to_ehdac_device(hdac); + struct hdac_hdmi_priv *hdmi = edev->private_data; + struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); + struct hdac_hdmi_pin *pin = pcm->pin; + + if (!pin || !pin->eld.eld_valid) + return 0; + + return pin->eld.info.spk_alloc; +} + static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) { struct hdac_device *codec = &edev->hdac; struct hdac_hdmi_priv *hdmi_priv; struct snd_soc_dai_driver *hdmi_dais = NULL; + struct hdac_ext_link *hlink = NULL; int num_dais = 0; int ret = 0; + /* hold the ref while we probe */ + hlink = snd_hdac_ext_bus_get_link(edev->ebus, dev_name(&edev->hdac.dev)); + snd_hdac_ext_bus_link_get(edev->ebus, hlink); + hdmi_priv = devm_kzalloc(&codec->dev, sizeof(*hdmi_priv), GFP_KERNEL); if (hdmi_priv == NULL) return -ENOMEM; edev->private_data = hdmi_priv; + snd_hdac_register_chmap_ops(codec, &hdmi_priv->chmap); + hdmi_priv->chmap.ops.get_chmap = hdac_hdmi_get_chmap; + hdmi_priv->chmap.ops.set_chmap = hdac_hdmi_set_chmap; + hdmi_priv->chmap.ops.is_pcm_attached = is_hdac_hdmi_pcm_attached; + hdmi_priv->chmap.ops.get_spk_alloc = hdac_hdmi_get_spk_alloc; dev_set_drvdata(&codec->dev, edev); @@ -1516,8 +1674,12 @@ static int hdac_hdmi_dev_probe(struct hdac_ext_device *edev) } /* ASoC specific initialization */ - return snd_soc_register_codec(&codec->dev, &hdmi_hda_codec, - hdmi_dais, num_dais); + ret = snd_soc_register_codec(&codec->dev, &hdmi_hda_codec, + hdmi_dais, num_dais); + + snd_hdac_ext_bus_link_put(edev->ebus, hlink); + + return ret; } static int hdac_hdmi_dev_remove(struct hdac_ext_device *edev) @@ -1556,6 +1718,8 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + struct hdac_ext_bus *ebus = hbus_to_ebus(bus); + struct hdac_ext_link *hlink = NULL; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1579,6 +1743,9 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) return err; } + hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + snd_hdac_ext_bus_link_put(ebus, hlink); + return 0; } @@ -1587,6 +1754,8 @@ static int hdac_hdmi_runtime_resume(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; + struct hdac_ext_bus *ebus = hbus_to_ebus(bus); + struct hdac_ext_link *hlink = NULL; int err; dev_dbg(dev, "Enter: %s\n", __func__); @@ -1595,6 +1764,9 @@ static int hdac_hdmi_runtime_resume(struct device *dev) if (!bus) return 0; + hlink = snd_hdac_ext_bus_get_link(ebus, dev_name(dev)); + snd_hdac_ext_bus_link_get(ebus, hlink); + err = snd_hdac_display_power(bus, true); if (err < 0) { dev_err(bus->dev, "Cannot turn on display power on i915\n"); diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c new file mode 100644 index 000000000000..8e36e883e453 --- /dev/null +++ b/sound/soc/codecs/hdmi-codec.c @@ -0,0 +1,432 @@ +/* + * ALSA SoC codec for HDMI encoder drivers + * Copyright (C) 2015 Texas Instruments Incorporated - http://www.ti.com/ + * Author: Jyri Sarha <jsarha@ti.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. + */ +#include <linux/module.h> +#include <linux/string.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/pcm_drm_eld.h> +#include <sound/hdmi-codec.h> +#include <sound/pcm_iec958.h> + +#include <drm/drm_crtc.h> /* This is only to get MAX_ELD_BYTES */ + +struct hdmi_codec_priv { + struct hdmi_codec_pdata hcd; + struct snd_soc_dai_driver *daidrv; + struct hdmi_codec_daifmt daifmt[2]; + struct mutex current_stream_lock; + struct snd_pcm_substream *current_stream; + struct snd_pcm_hw_constraint_list ratec; + uint8_t eld[MAX_ELD_BYTES]; +}; + +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "TX", NULL, "Playback" }, +}; + +enum { + DAI_ID_I2S = 0, + DAI_ID_SPDIF, +}; + +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = sizeof(hcp->eld); + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + + memcpy(ucontrol->value.bytes.data, hcp->eld, sizeof(hcp->eld)); + + return 0; +} + +static const struct snd_kcontrol_new hdmi_controls[] = { + { + .access = SNDRV_CTL_ELEM_ACCESS_READ | + SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, + }, +}; + +static int hdmi_codec_new_stream(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + mutex_lock(&hcp->current_stream_lock); + if (!hcp->current_stream) { + hcp->current_stream = substream; + } else if (hcp->current_stream != substream) { + dev_err(dai->dev, "Only one simultaneous stream supported!\n"); + ret = -EINVAL; + } + mutex_unlock(&hcp->current_stream_lock); + + return ret; +} + +static int hdmi_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + if (hcp->hcd.ops->audio_startup) { + ret = hcp->hcd.ops->audio_startup(dai->dev->parent); + if (ret) { + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); + return ret; + } + } + + if (hcp->hcd.ops->get_eld) { + ret = hcp->hcd.ops->get_eld(dai->dev->parent, hcp->eld, + sizeof(hcp->eld)); + + if (!ret) { + ret = snd_pcm_hw_constraint_eld(substream->runtime, + hcp->eld); + if (ret) + return ret; + } + } + return 0; +} + +static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + WARN_ON(hcp->current_stream != substream); + + hcp->hcd.ops->audio_shutdown(dai->dev->parent); + + mutex_lock(&hcp->current_stream_lock); + hcp->current_stream = NULL; + mutex_unlock(&hcp->current_stream_lock); +} + +static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_params hp = { + .iec = { + .status = { 0 }, + .subcode = { 0 }, + .pad = 0, + .dig_subframe = { 0 }, + } + }; + int ret; + + dev_dbg(dai->dev, "%s() width %d rate %d channels %d\n", __func__, + params_width(params), params_rate(params), + params_channels(params)); + + if (params_width(params) > 24) + params->msbits = 24; + + ret = snd_pcm_create_iec958_consumer_hw_params(params, hp.iec.status, + sizeof(hp.iec.status)); + if (ret < 0) { + dev_err(dai->dev, "Creating IEC958 channel status failed %d\n", + ret); + return ret; + } + + ret = hdmi_codec_new_stream(substream, dai); + if (ret) + return ret; + + hdmi_audio_infoframe_init(&hp.cea); + hp.cea.channels = params_channels(params); + hp.cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; + hp.cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; + hp.cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; + + hp.sample_width = params_width(params); + hp.sample_rate = params_rate(params); + hp.channels = params_channels(params); + + return hcp->hcd.ops->hw_params(dai->dev->parent, &hcp->daifmt[dai->id], + &hp); +} + +static int hdmi_codec_set_fmt(struct snd_soc_dai *dai, + unsigned int fmt) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + struct hdmi_codec_daifmt cf = { 0 }; + int ret = 0; + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (dai->id == DAI_ID_SPDIF) { + cf.fmt = HDMI_SPDIF; + } else { + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + cf.bit_clk_master = 1; + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFM: + cf.frame_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBM_CFS: + cf.bit_clk_master = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + cf.frame_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_NF: + cf.bit_clk_inv = 1; + break; + case SND_SOC_DAIFMT_IB_IF: + cf.frame_clk_inv = 1; + cf.bit_clk_inv = 1; + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + cf.fmt = HDMI_I2S; + break; + case SND_SOC_DAIFMT_DSP_A: + cf.fmt = HDMI_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + cf.fmt = HDMI_DSP_B; + break; + case SND_SOC_DAIFMT_RIGHT_J: + cf.fmt = HDMI_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + cf.fmt = HDMI_LEFT_J; + break; + case SND_SOC_DAIFMT_AC97: + cf.fmt = HDMI_AC97; + break; + default: + dev_err(dai->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + } + + hcp->daifmt[dai->id] = cf; + + return ret; +} + +static int hdmi_codec_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + + dev_dbg(dai->dev, "%s()\n", __func__); + + if (hcp->hcd.ops->digital_mute) + return hcp->hcd.ops->digital_mute(dai->dev->parent, mute); + + return 0; +} + +static const struct snd_soc_dai_ops hdmi_dai_ops = { + .startup = hdmi_codec_startup, + .shutdown = hdmi_codec_shutdown, + .hw_params = hdmi_codec_hw_params, + .set_fmt = hdmi_codec_set_fmt, + .digital_mute = hdmi_codec_digital_mute, +}; + + +#define HDMI_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) + +#define SPDIF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +/* + * This list is only for formats allowed on the I2S bus. So there is + * some formats listed that are not supported by HDMI interface. For + * instance allowing the 32-bit formats enables 24-precision with CPU + * DAIs that do not support 24-bit formats. If the extra formats cause + * problems, we should add the video side driver an option to disable + * them. + */ +#define I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE |\ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE |\ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +static struct snd_soc_dai_driver hdmi_i2s_dai = { + .name = "i2s-hifi", + .id = DAI_ID_I2S, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = HDMI_RATES, + .formats = I2S_FORMATS, + .sig_bits = 24, + }, + .ops = &hdmi_dai_ops, +}; + +static const struct snd_soc_dai_driver hdmi_spdif_dai = { + .name = "spdif-hifi", + .id = DAI_ID_SPDIF, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = HDMI_RATES, + .formats = SPDIF_FORMATS, + }, + .ops = &hdmi_dai_ops, +}; + +static struct snd_soc_codec_driver hdmi_codec = { + .controls = hdmi_controls, + .num_controls = ARRAY_SIZE(hdmi_controls), + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + +static int hdmi_codec_probe(struct platform_device *pdev) +{ + struct hdmi_codec_pdata *hcd = pdev->dev.platform_data; + struct device *dev = &pdev->dev; + struct hdmi_codec_priv *hcp; + int dai_count, i = 0; + int ret; + + dev_dbg(dev, "%s()\n", __func__); + + if (!hcd) { + dev_err(dev, "%s: No plalform data\n", __func__); + return -EINVAL; + } + + dai_count = hcd->i2s + hcd->spdif; + if (dai_count < 1 || !hcd->ops || !hcd->ops->hw_params || + !hcd->ops->audio_shutdown) { + dev_err(dev, "%s: Invalid parameters\n", __func__); + return -EINVAL; + } + + hcp = devm_kzalloc(dev, sizeof(*hcp), GFP_KERNEL); + if (!hcp) + return -ENOMEM; + + hcp->hcd = *hcd; + mutex_init(&hcp->current_stream_lock); + + hcp->daidrv = devm_kzalloc(dev, dai_count * sizeof(*hcp->daidrv), + GFP_KERNEL); + if (!hcp->daidrv) + return -ENOMEM; + + if (hcd->i2s) { + hcp->daidrv[i] = hdmi_i2s_dai; + hcp->daidrv[i].playback.channels_max = + hcd->max_i2s_channels; + i++; + } + + if (hcd->spdif) + hcp->daidrv[i] = hdmi_spdif_dai; + + ret = snd_soc_register_codec(dev, &hdmi_codec, hcp->daidrv, + dai_count); + if (ret) { + dev_err(dev, "%s: snd_soc_register_codec() failed (%d)\n", + __func__, ret); + return ret; + } + + dev_set_drvdata(dev, hcp); + return 0; +} + +static int hdmi_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver hdmi_codec_driver = { + .driver = { + .name = HDMI_CODEC_DRV_NAME, + }, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, +}; + +module_platform_driver(hdmi_codec_driver); + +MODULE_AUTHOR("Jyri Sarha <jsarha@ti.com>"); +MODULE_DESCRIPTION("HDMI Audio Codec Driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" HDMI_CODEC_DRV_NAME); diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c new file mode 100644 index 000000000000..ed515677409b --- /dev/null +++ b/sound/soc/codecs/pcm5102a.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM5102A codec + * + * Author: Florian Meier <florian.meier@koalo.de> + * Copyright 2013 + * + * 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. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/soc.h> + +static struct snd_soc_dai_driver pcm5102a_dai = { + .name = "pcm5102a-hifi", + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm5102a; + +static int pcm5102a_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm5102a, + &pcm5102a_dai, 1); +} + +static int pcm5102a_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id pcm5102a_of_match[] = { + { .compatible = "ti,pcm5102a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm5102a_of_match); + +static struct platform_driver pcm5102a_codec_driver = { + .probe = pcm5102a_probe, + .remove = pcm5102a_remove, + .driver = { + .name = "pcm5102a-codec", + .owner = THIS_MODULE, + .of_match_table = pcm5102a_of_match, + }, +}; + +module_platform_driver(pcm5102a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM5102A codec driver"); +MODULE_AUTHOR("Florian Meier <florian.meier@koalo.de>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f0e6c06e89ac..a1aaffc20862 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -17,6 +17,7 @@ #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/dmi.h> #include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> @@ -1132,6 +1133,17 @@ static const struct acpi_device_id rt298_acpi_match[] = { }; MODULE_DEVICE_TABLE(acpi, rt298_acpi_match); +static const struct dmi_system_id force_combo_jack_table[] = { + { + .ident = "Intel Broxton P", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corp"), + DMI_MATCH(DMI_PRODUCT_NAME, "Broxton P") + } + }, + { } +}; + static int rt298_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1184,11 +1196,16 @@ static int rt298_i2c_probe(struct i2c_client *i2c, /* enable jack combo mode on supported devices */ acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); - if (acpiid) { + if (acpiid && acpiid->driver_data) { rt298->pdata = *(struct rt298_platform_data *) acpiid->driver_data; } + if (dmi_check_system(force_combo_jack_table)) { + rt298->pdata.cbj_en = true; + rt298->pdata.gpio2_en = false; + } + /* VREF Charging */ regmap_update_bits(rt298->regmap, 0x04, 0x80, 0x80); regmap_update_bits(rt298->regmap, 0x1b, 0x860, 0x860); diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7af5e7380d61..3c6594da6c9c 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (btn_type == 0)/* button release */ report = rt5645->jack_type; else { - if (rt5645->pdata.jd_invert) { - mod_timer(&rt5645->btn_check_timer, - msecs_to_jiffies(100)); - } + mod_timer(&rt5645->btn_check_timer, + msecs_to_jiffies(100)); } break; @@ -3557,6 +3555,12 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_SYS_VENDOR, "GOOGLE"), }, }, + { + .ident = "Google Setzer", + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Setzer"), + }, + }, { } }; @@ -3810,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); - setup_timer(&rt5645->btn_check_timer, - rt5645_btn_check_callback, (unsigned long)rt5645); } + setup_timer(&rt5645->btn_check_timer, + rt5645_btn_check_callback, (unsigned long)rt5645); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 1bae17ee8817..da60e3fe5ee7 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2098,10 +2098,14 @@ static int wm5102_probe(struct platform_device *pdev) static int wm5102_remove(struct platform_device *pdev) { + struct wm5102_priv *wm5102 = platform_get_drvdata(pdev); + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + wm_adsp2_remove(&wm5102->core.adsp[0]); + return 0; } diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2728ac545ffe..b5820e4d5471 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2437,10 +2437,16 @@ static int wm5110_probe(struct platform_device *pdev) static int wm5110_remove(struct platform_device *pdev) { + struct wm5110_priv *wm5110 = platform_get_drvdata(pdev); + int i; + snd_soc_unregister_platform(&pdev->dev); snd_soc_unregister_codec(&pdev->dev); pm_runtime_disable(&pdev->dev); + for (i = 0; i < WM5110_NUM_ADSP; i++) + wm_adsp2_remove(&wm5110->core.adsp[i]); + return 0; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d3b1cb15e7f0..a07bd7c2c587 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -160,6 +160,8 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +#define ADSP_MAX_STD_CTRL_SIZE 512 + struct wm_adsp_buf { struct list_head list; void *buf; @@ -271,8 +273,11 @@ struct wm_adsp_buffer { __be32 words_written[2]; /* total words written (64 bit) */ }; +struct wm_adsp_compr; + struct wm_adsp_compr_buf { struct wm_adsp *dsp; + struct wm_adsp_compr *compr; struct wm_adsp_buffer_region *regions; u32 host_buf_ptr; @@ -435,6 +440,7 @@ struct wm_coeff_ctl { size_t len; unsigned int set:1; struct snd_kcontrol *kcontrol; + struct soc_bytes_ext bytes_ext; unsigned int flags; }; @@ -711,10 +717,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } +static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) +{ + return container_of(ext, struct wm_coeff_ctl, bytes_ext); +} + static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -763,7 +776,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -780,6 +795,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, + const unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (copy_from_user(ctl->cache, bytes, size)) { + ret = -EFAULT; + } else { + ctl->set = 1; + if (ctl->enabled) + ret = wm_coeff_write_control(ctl, ctl->cache, size); + } + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, void *buf, size_t len) { @@ -822,7 +860,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -845,12 +885,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, + unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { + if (ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + else + ret = -EPERM; + } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + } + + if (!ret && copy_to_user(bytes, ctl->cache, size)) + ret = -EFAULT; + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + struct wmfw_ctl_work { struct wm_adsp *dsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; +static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) +{ + unsigned int out, rd, wr, vol; + + if (len > ADSP_MAX_STD_CTRL_SIZE) { + rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ; + wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } else { + rd = SNDRV_CTL_ELEM_ACCESS_READ; + wr = SNDRV_CTL_ELEM_ACCESS_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = 0; + } + + if (in) { + if (in & WMFW_CTL_FLAG_READABLE) + out |= rd; + if (in & WMFW_CTL_FLAG_WRITEABLE) + out |= wr; + if (in & WMFW_CTL_FLAG_VOLATILE) + out |= vol; + } else { + out |= rd | wr | vol; + } + + return out; +} + static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; @@ -868,19 +968,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; - kcontrol->private_value = (unsigned long)ctl; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kcontrol->tlv.c = snd_soc_bytes_tlv_callback; + kcontrol->private_value = (unsigned long)&ctl->bytes_ext; - if (ctl->flags) { - if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE; - if (ctl->flags & WMFW_CTL_FLAG_READABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ; - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } else { - kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE; - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + + kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len); ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1); if (ret < 0) @@ -944,6 +1040,13 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } +static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) +{ + kfree(ctl->cache); + kfree(ctl->name); + kfree(ctl); +} + static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region, unsigned int offset, unsigned int len, @@ -1032,11 +1135,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->flags = flags; ctl->offset = offset; - if (len > 512) { - adsp_warn(dsp, "Truncating control %s from %d\n", - ctl->name, len); - len = 512; - } ctl->len = len; ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { @@ -1564,6 +1662,19 @@ static struct wm_adsp_alg_region *wm_adsp_create_region(struct wm_adsp *dsp, return alg_region; } +static void wm_adsp_free_alg_regions(struct wm_adsp *dsp) +{ + struct wm_adsp_alg_region *alg_region; + + while (!list_empty(&dsp->alg_regions)) { + alg_region = list_first_entry(&dsp->alg_regions, + struct wm_adsp_alg_region, + list); + list_del(&alg_region->list); + kfree(alg_region); + } +} + static int wm_adsp1_setup_algs(struct wm_adsp *dsp) { struct wmfw_adsp1_id_hdr adsp1_id; @@ -1994,7 +2105,6 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; unsigned int val; @@ -2074,13 +2184,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - while (!list_empty(&dsp->alg_regions)) { - alg_region = list_first_entry(&dsp->alg_regions, - struct wm_adsp_alg_region, - list); - list_del(&alg_region->list); - kfree(alg_region); - } + + wm_adsp_free_alg_regions(dsp); break; default: @@ -2222,7 +2327,6 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; - struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; @@ -2240,9 +2344,13 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; + mutex_lock(&dsp->pwr_lock); + if (wm_adsp_fw[dsp->fw].num_caps != 0) ret = wm_adsp_buffer_init(dsp); + mutex_unlock(&dsp->pwr_lock); + break; case SND_SOC_DAPM_PRE_PMD: @@ -2269,13 +2377,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - while (!list_empty(&dsp->alg_regions)) { - alg_region = list_first_entry(&dsp->alg_regions, - struct wm_adsp_alg_region, - list); - list_del(&alg_region->list); - kfree(alg_region); - } + wm_adsp_free_alg_regions(dsp); if (wm_adsp_fw[dsp->fw].num_caps != 0) wm_adsp_buffer_free(dsp); @@ -2340,6 +2442,54 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +void wm_adsp2_remove(struct wm_adsp *dsp) +{ + struct wm_coeff_ctl *ctl; + + while (!list_empty(&dsp->ctl_list)) { + ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl, + list); + list_del(&ctl->list); + wm_adsp_free_ctl_blk(ctl); + } +} +EXPORT_SYMBOL_GPL(wm_adsp2_remove); + +static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) +{ + return compr->buf != NULL; +} + +static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) +{ + /* + * Note this will be more complex once each DSP can support multiple + * streams + */ + if (!compr->dsp->buffer) + return -EINVAL; + + compr->buf = compr->dsp->buffer; + compr->buf->compr = compr; + + return 0; +} + +static void wm_adsp_compr_detach(struct wm_adsp_compr *compr) +{ + if (!compr) + return; + + /* Wake the poll so it can see buffer is no longer attached */ + if (compr->stream) + snd_compr_fragment_elapsed(compr->stream); + + if (wm_adsp_compr_attached(compr)) { + compr->buf->compr = NULL; + compr->buf = NULL; + } +} + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; @@ -2393,6 +2543,7 @@ int wm_adsp_compr_free(struct snd_compr_stream *stream) mutex_lock(&dsp->pwr_lock); + wm_adsp_compr_detach(compr); dsp->compr = NULL; kfree(compr->raw_buf); @@ -2689,6 +2840,8 @@ err_buffer: static int wm_adsp_buffer_free(struct wm_adsp *dsp) { if (dsp->buffer) { + wm_adsp_compr_detach(dsp->buffer->compr); + kfree(dsp->buffer->regions); kfree(dsp->buffer); @@ -2698,25 +2851,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp) return 0; } -static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) -{ - return compr->buf != NULL; -} - -static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) -{ - /* - * Note this will be more complex once each DSP can support multiple - * streams - */ - if (!compr->dsp->buffer) - return -EINVAL; - - compr->buf = compr->dsp->buffer; - - return 0; -} - int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) { struct wm_adsp_compr *compr = stream->runtime->private_data; @@ -2805,21 +2939,41 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) avail += wm_adsp_buffer_size(buf); adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", - buf->read_index, write_index, avail); + buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); buf->avail = avail; return 0; } +static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf) +{ + int ret; + + ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); + if (ret < 0) { + adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret); + return ret; + } + if (buf->error != 0) { + adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error); + return -EIO; + } + + return 0; +} + int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) { - struct wm_adsp_compr_buf *buf = dsp->buffer; - struct wm_adsp_compr *compr = dsp->compr; + struct wm_adsp_compr_buf *buf; + struct wm_adsp_compr *compr; int ret = 0; mutex_lock(&dsp->pwr_lock); + buf = dsp->buffer; + compr = dsp->compr; + if (!buf) { ret = -ENODEV; goto out; @@ -2827,16 +2981,9 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) adsp_dbg(dsp, "Handling buffer IRQ\n"); - ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); - if (ret < 0) { - adsp_err(dsp, "Failed to check buffer error: %d\n", ret); - goto out; - } - if (buf->error != 0) { - adsp_err(dsp, "Buffer error occurred: %d\n", buf->error); - ret = -EIO; - goto out; - } + ret = wm_adsp_buffer_get_error(buf); + if (ret < 0) + goto out_notify; /* Wake poll to report error */ ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), &buf->irq_count); @@ -2851,6 +2998,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) goto out; } +out_notify: if (compr && compr->stream) snd_compr_fragment_elapsed(compr->stream); @@ -2879,14 +3027,16 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct snd_compr_tstamp *tstamp) { struct wm_adsp_compr *compr = stream->runtime->private_data; - struct wm_adsp_compr_buf *buf = compr->buf; struct wm_adsp *dsp = compr->dsp; + struct wm_adsp_compr_buf *buf; int ret = 0; adsp_dbg(dsp, "Pointer request\n"); mutex_lock(&dsp->pwr_lock); + buf = compr->buf; + if (!compr->buf) { ret = -ENXIO; goto out; @@ -2909,6 +3059,10 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, * DSP to inform us once a whole fragment is available. */ if (buf->avail < wm_adsp_compr_frag_words(compr)) { + ret = wm_adsp_buffer_get_error(buf); + if (ret < 0) + goto out; + ret = wm_adsp_buffer_reenable_irq(buf); if (ret < 0) { adsp_err(dsp, diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index b61cb57e600f..feb61e2c4bb4 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +void wm_adsp2_remove(struct wm_adsp *dsp); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 50ca291cc225..6b732d8e5896 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -16,7 +16,11 @@ config SND_EDMA_SOC - DRA7xx family config SND_DAVINCI_SOC_I2S - tristate + tristate "DaVinci Multichannel Buffered Serial Port (McBSP) support" + depends on SND_EDMA_SOC + help + Say Y or M here if you want to have support for McBSP IP found in + Texas Instruments DaVinci DA850 SoCs. config SND_DAVINCI_SOC_MCASP tristate "Multichannel Audio Serial Port (McASP) support" diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index ec98548a5fc9..384961651904 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -4,9 +4,15 @@ * Author: Vladimir Barinov, <vbarinov@embeddedalley.com> * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com> * + * DT support (c) 2016 Petr Kulhavy, Barix AG <petr@barix.com> + * based on davinci-mcasp.c DT support + * * 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. + * + * TODO: + * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers */ #include <linux/init.h> @@ -650,13 +656,24 @@ static const struct snd_soc_component_driver davinci_i2s_component = { static int davinci_i2s_probe(struct platform_device *pdev) { + struct snd_dmaengine_dai_dma_data *dma_data; struct davinci_mcbsp_dev *dev; struct resource *mem, *res; void __iomem *io_base; int *dma; int ret; - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); + if (!mem) { + dev_warn(&pdev->dev, + "\"mpu\" mem resource not found, using index 0\n"); + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + } + io_base = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(io_base)) return PTR_ERR(io_base); @@ -666,39 +683,43 @@ static int davinci_i2s_probe(struct platform_device *pdev) if (!dev) return -ENOMEM; - dev->clk = clk_get(&pdev->dev, NULL); - if (IS_ERR(dev->clk)) - return -ENODEV; - clk_enable(dev->clk); - dev->base = io_base; - dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = - (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); + /* setup DMA, first TX, then RX */ + dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG); - dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = - (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); - - /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENXIO; - goto err_release_clk; + if (res) { + dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; + *dma = res->start; + dma_data->filter_data = dma; + } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) { + dma_data->filter_data = "tx"; + } else { + dev_err(&pdev->dev, "Missing DMA tx resource\n"); + return -ENODEV; } - dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; - *dma = res->start; - dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = dma; + + dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG); res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENXIO; - goto err_release_clk; + if (res) { + dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; + *dma = res->start; + dma_data->filter_data = dma; + } else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) { + dma_data->filter_data = "rx"; + } else { + dev_err(&pdev->dev, "Missing DMA rx resource\n"); + return -ENODEV; } - dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE]; - *dma = res->start; - dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = dma; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return -ENODEV; + clk_enable(dev->clk); dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -737,11 +758,18 @@ static int davinci_i2s_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id davinci_i2s_match[] = { + { .compatible = "ti,da850-mcbsp" }, + {}, +}; +MODULE_DEVICE_TABLE(of, davinci_i2s_match); + static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .driver = { .name = "davinci-mcbsp", + .of_match_table = of_match_ptr(davinci_i2s_match), }, }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e1324989bd6b..0f66fda2c772 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + ACLKX | AFSX | ACLKR | AHCLKR | AFSR); mcasp->bclk_master = 0; break; default: @@ -540,21 +540,19 @@ out: return ret; } -static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, +static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id, int div, bool explicit) { - struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); - pm_runtime_get_sync(mcasp->dev); switch (div_id) { - case 0: /* MCLK divider */ + case MCASP_CLKDIV_AUXCLK: /* MCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(div - 1), AHCLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRDIV(div - 1), AHCLKRDIV_MASK); break; - case 1: /* BCLK divider */ + case MCASP_CLKDIV_BCLK: /* BCLK divider */ mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXDIV(div - 1), ACLKXDIV_MASK); mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, @@ -563,7 +561,8 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, mcasp->bclk_div = div; break; - case 2: /* + case MCASP_CLKDIV_BCLK_FS_RATIO: + /* * BCLK/LRCLK ratio descries how many bit-clock cycles * fit into one frame. The clock ratio is given for a * full period of data (for I2S format both left and @@ -591,7 +590,9 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - return __davinci_mcasp_set_clkdiv(dai, div_id, div, 1); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + + return __davinci_mcasp_set_clkdiv(mcasp, div_id, div, 1); } static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, @@ -999,27 +1000,53 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, } static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp, - unsigned int bclk_freq, - int *error_ppm) + unsigned int bclk_freq, bool set) { - int div = mcasp->sysclk_freq / bclk_freq; - int rem = mcasp->sysclk_freq % bclk_freq; + int error_ppm; + unsigned int sysclk_freq = mcasp->sysclk_freq; + u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG); + int div = sysclk_freq / bclk_freq; + int rem = sysclk_freq % bclk_freq; + int aux_div = 1; + + if (div > (ACLKXDIV_MASK + 1)) { + if (reg & AHCLKXE) { + aux_div = div / (ACLKXDIV_MASK + 1); + if (div % (ACLKXDIV_MASK + 1)) + aux_div++; + + sysclk_freq /= aux_div; + div = sysclk_freq / bclk_freq; + rem = sysclk_freq % bclk_freq; + } else if (set) { + dev_warn(mcasp->dev, "Too fast reference clock (%u)\n", + sysclk_freq); + } + } if (rem != 0) { if (div == 0 || - ((mcasp->sysclk_freq / div) - bclk_freq) > - (bclk_freq - (mcasp->sysclk_freq / (div+1)))) { + ((sysclk_freq / div) - bclk_freq) > + (bclk_freq - (sysclk_freq / (div+1)))) { div++; rem = rem - bclk_freq; } } - if (error_ppm) - *error_ppm = - (div*1000000 + (int)div64_long(1000000LL*rem, - (int)bclk_freq)) - /div - 1000000; + error_ppm = (div*1000000 + (int)div64_long(1000000LL*rem, + (int)bclk_freq)) / div - 1000000; + + if (set) { + if (error_ppm) + dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", + error_ppm); + + __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0); + if (reg & AHCLKXE) + __davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_AUXCLK, + aux_div, 0); + } - return div; + return error_ppm; } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -1044,18 +1071,11 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); - int ppm, div; if (mcasp->slot_width) sbits = mcasp->slot_width; - div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots, - &ppm); - if (ppm) - dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", - ppm); - - __davinci_mcasp_set_clkdiv(cpu_dai, 1, div, 0); + davinci_mcasp_calc_clk_div(mcasp, rate * sbits * slots, true); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -1166,7 +1186,8 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, davinci_mcasp_dai_rates[i]; int ppm; - davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); + ppm = davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, + false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { if (range.empty) { range.min = davinci_mcasp_dai_rates[i]; @@ -1205,8 +1226,9 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, if (rd->mcasp->slot_width) sbits = rd->mcasp->slot_width; - davinci_mcasp_calc_clk_div(rd->mcasp, sbits*slots*rate, - &ppm); + ppm = davinci_mcasp_calc_clk_div(rd->mcasp, + sbits * slots * rate, + false); if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { snd_mask_set(&nfmt, i); count++; @@ -1230,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, int i, dir; int tdm_slots = mcasp->tdm_slots; - if (mcasp->tdm_mask[substream->stream]) - tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + /* Do not allow more then one stream per direction */ + if (mcasp->substreams[substream->stream]) + return -EBUSY; mcasp->substreams[substream->stream] = substream; + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a3be108a8c17..1e8787fb3fb7 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -306,4 +306,9 @@ #define NUMEVT(x) (((x) & 0xFF) << 8) #define NUMDMA_MASK (0xFF) +/* clock divider IDs */ +#define MCASP_CLKDIV_AUXCLK 0 /* HCLK divider from AUXCLK */ +#define MCASP_CLKDIV_BCLK 1 /* BCLK divider from HCLK */ +#define MCASP_CLKDIV_BCLK_FS_RATIO 2 /* to set BCLK FS ration */ + #endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index bff258d7bcea..0db69b7e9617 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -100,6 +100,7 @@ struct dw_i2s_dev { struct device *dev; u32 ccr; u32 xfer_resolution; + u32 fifo_th; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; @@ -147,17 +148,18 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { + struct i2s_clk_config_data *config = &dev->config; u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - for (i = 0; i < 4; i++) { + for (i = 0; i < (config->chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); } i2s_write_reg(dev->i2s_base, ITER, 1); } else { - for (i = 0; i < 4; i++) { + for (i = 0; i < (config->chan_nr / 2); i++) { irq = i2s_read_reg(dev->i2s_base, IMR(i)); i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); } @@ -231,14 +233,16 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) if (stream == SNDRV_PCM_STREAM_PLAYBACK) { i2s_write_reg(dev->i2s_base, TCR(ch_reg), dev->xfer_resolution); - i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), + dev->fifo_th - 1); irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); } else { i2s_write_reg(dev->i2s_base, RCR(ch_reg), dev->xfer_resolution); - i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), + dev->fifo_th - 1); irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); @@ -498,6 +502,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, */ u32 comp1 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp1); u32 comp2 = i2s_read_reg(dev->i2s_base, dev->i2s_reg_comp2); + u32 fifo_depth = 1 << (1 + COMP1_FIFO_DEPTH_GLOBAL(comp1)); u32 idx; if (dev->capability & DWC_I2S_RECORD && @@ -536,6 +541,7 @@ static int dw_configure_dai(struct dw_i2s_dev *dev, dev->capability |= DW_I2S_SLAVE; } + dev->fifo_th = fifo_depth / 2; return 0; } diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df771e3b..2147994ab46f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> #include "fsl_sai.h" #include "imx-pcm.h" @@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + int index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -797,7 +801,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -877,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) { + gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + return PTR_ERR(gpr); + } + + index = of_alias_get_id(np, "sai"); + if (index < 0) + return index; + + regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index), + MCLK_DIR(index)); + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; @@ -898,6 +919,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index ed8de1035cda..632ecc0e3956 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -137,6 +137,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) case CCSR_SSI_SACDAT: case CCSR_SSI_SATAG: case CCSR_SSI_SACCST: + case CCSR_SSI_SOR: return true; default: return false; @@ -261,6 +262,7 @@ struct fsl_ssi_private { struct fsl_ssi_dbg dbg_stats; const struct fsl_ssi_soc_data *soc; + struct device *dev; }; /* @@ -400,6 +402,26 @@ static void fsl_ssi_rxtx_config(struct fsl_ssi_private *ssi_private, } /* + * Clear RX or TX FIFO to remove samples from the previous + * stream session which may be still present in the FIFO and + * may introduce bad samples and/or channel slipping. + * + * Note: The SOR is not documented in recent IMX datasheet, but + * is described in IMX51 reference manual at section 56.3.3.15. + */ +static void fsl_ssi_fifo_clear(struct fsl_ssi_private *ssi_private, + bool is_rx) +{ + if (is_rx) { + regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR, + CCSR_SSI_SOR_RX_CLR, CCSR_SSI_SOR_RX_CLR); + } else { + regmap_update_bits(ssi_private->regs, CCSR_SSI_SOR, + CCSR_SSI_SOR_TX_CLR, CCSR_SSI_SOR_TX_CLR); + } +} + +/* * Calculate the bits that have to be disabled for the current stream that is * getting disabled. This keeps the bits enabled that are necessary for the * second stream to work if 'stream_active' is true. @@ -474,9 +496,11 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, * (online configuration) */ if (enable) { - regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); + fsl_ssi_fifo_clear(ssi_private, vals->scr & CCSR_SSI_SCR_RE); + regmap_update_bits(regs, CCSR_SSI_SRCR, vals->srcr, vals->srcr); regmap_update_bits(regs, CCSR_SSI_STCR, vals->stcr, vals->stcr); + regmap_update_bits(regs, CCSR_SSI_SIER, vals->sier, vals->sier); } else { u32 sier; u32 srcr; @@ -506,8 +530,40 @@ static void fsl_ssi_config(struct fsl_ssi_private *ssi_private, bool enable, config_done: /* Enabling of subunits is done after configuration */ - if (enable) + if (enable) { + if (ssi_private->use_dma && (vals->scr & CCSR_SSI_SCR_TE)) { + /* + * Be sure the Tx FIFO is filled when TE is set. + * Otherwise, there are some chances to start the + * playback with some void samples inserted first, + * generating a channel slip. + * + * First, SSIEN must be set, to let the FIFO be filled. + * + * Notes: + * - Limit this fix to the DMA case until FIQ cases can + * be tested. + * - Limit the length of the busy loop to not lock the + * system too long, even if 1-2 loops are sufficient + * in general. + */ + int i; + int max_loop = 100; + regmap_update_bits(regs, CCSR_SSI_SCR, + CCSR_SSI_SCR_SSIEN, CCSR_SSI_SCR_SSIEN); + for (i = 0; i < max_loop; i++) { + u32 sfcsr; + regmap_read(regs, CCSR_SSI_SFCSR, &sfcsr); + if (CCSR_SSI_SFCSR_TFCNT0(sfcsr)) + break; + } + if (i == max_loop) { + dev_err(ssi_private->dev, + "Timeout waiting TX FIFO filling\n"); + } + } regmap_update_bits(regs, CCSR_SSI_SCR, vals->scr, vals->scr); + } } @@ -670,6 +726,15 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, if (IS_ERR(ssi_private->baudclk)) return -EINVAL; + /* + * Hardware limitation: The bclk rate must be + * never greater than 1/5 IPG clock rate + */ + if (freq * 5 > clk_get_rate(ssi_private->clk)) { + dev_err(cpu_dai->dev, "bitclk > ipgclk/5\n"); + return -EINVAL; + } + baudclk_is_used = ssi_private->baudclk_streams & ~(BIT(substream->stream)); /* It should be already enough to divide clock by setting pm alone */ @@ -686,13 +751,6 @@ static int fsl_ssi_set_bclk(struct snd_pcm_substream *substream, else clkrate = clk_round_rate(ssi_private->baudclk, tmprate); - /* - * Hardware limitation: The bclk rate must be - * never greater than 1/5 IPG clock rate - */ - if (clkrate * 5 > clk_get_rate(ssi_private->clk)) - continue; - clkrate /= factor; afreq = clkrate / (i + 1); @@ -1158,14 +1216,14 @@ static struct snd_soc_dai_driver fsl_ssi_dai_template = { .playback = { .stream_name = "CPU-Playback", .channels_min = 1, - .channels_max = 2, + .channels_max = 32, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, .capture = { .stream_name = "CPU-Capture", .channels_min = 1, - .channels_max = 2, + .channels_max = 32, .rates = FSLSSI_I2S_RATES, .formats = FSLSSI_I2S_FORMATS, }, @@ -1402,6 +1460,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->soc = of_id->data; + ssi_private->dev = &pdev->dev; sprop = of_get_property(np, "fsl,mode", NULL); if (sprop) { diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index e63cd5ecfd8f..dac6688540dc 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -220,7 +220,7 @@ static int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, ret = dma_mmap_wc(substream->pcm->card->dev, vma, runtime->dma_area, runtime->dma_addr, runtime->dma_bytes); - pr_debug("%s: ret: %d %p %pad 0x%08x\n", __func__, ret, + pr_debug("%s: ret: %d %p %pad 0x%08zx\n", __func__, ret, runtime->dma_area, &runtime->dma_addr, runtime->dma_bytes); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 1120f4f4d011..91c15abb625e 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,6 +58,21 @@ config SND_SOC_INTEL_HASWELL_MACH Say Y if you have such a device If unsure select "N". +config SND_SOC_INTEL_BXT_RT298_MACH + tristate "ASoC Audio driver for Broxton with RT298 I2S mode" + depends on X86 && ACPI && I2C + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT298 + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Broxton platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C @@ -162,6 +177,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH config SND_SOC_INTEL_SKYLAKE tristate select SND_HDA_EXT_CORE + select SND_HDA_DSP_LOADER select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index b97e6adcf1b2..98720a93de8a 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -195,7 +195,7 @@ static int sst_check_and_send_slot_map(struct sst_data *drv, struct snd_kcontrol if (e->w && e->w->power) ret = sst_send_slot_map(drv); - else + else if (!e->w) dev_err(&drv->pdev->dev, "Slot control: %s doesn't have DAPM widget!!!\n", kcontrol->id.name); return ret; diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 3310c0f9c356..a8506774f510 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o +snd-soc-sst-bxt-rt298-objs := bxt_rt298.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -14,6 +15,7 @@ snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 3f8a1e10bed0..7486a0022fde 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -201,7 +201,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c new file mode 100644 index 000000000000..f4787515c0ed --- /dev/null +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -0,0 +1,353 @@ +/* + * Intel Broxton-P I2S Machine Driver + * + * Copyright (C) 2014-2016, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Skylake I2S Machine driver + * + * 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. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/hdac_hdmi.h" +#include "../../codecs/rt298.h" + +static struct snd_soc_jack broxton_headset; +/* Headset jack detection DAPM pins */ + +enum { + BXT_DPCM_AUDIO_PB = 0, + BXT_DPCM_AUDIO_CP, + BXT_DPCM_AUDIO_REF_CP, + BXT_DPCM_AUDIO_HDMI1_PB, + BXT_DPCM_AUDIO_HDMI2_PB, + BXT_DPCM_AUDIO_HDMI3_PB, +}; + +static struct snd_soc_jack_pin broxton_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new broxton_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget broxton_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), +}; + +static const struct snd_soc_dapm_route broxton_rt298_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack detect */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMic", NULL, "SoC DMIC"}, + + {"HDMI1", NULL, "hif5 Output"}, + {"HDMI2", NULL, "hif6 Output"}, + {"HDMI3", NULL, "hif7 Output"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp5 Tx"}, + { "ssp5 Tx", NULL, "codec0_out"}, + + { "codec0_in", NULL, "ssp5 Rx" }, + { "ssp5 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hifi3", NULL, "iDisp3 Tx"}, + { "iDisp3 Tx", NULL, "iDisp3_out"}, + { "hifi2", NULL, "iDisp2 Tx"}, + { "iDisp2 Tx", NULL, "iDisp2_out"}, + { "hifi1", NULL, "iDisp1 Tx"}, + { "iDisp1 Tx", NULL, "iDisp1_out"}, + +}; + +static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret = 0; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &broxton_headset, + broxton_headset_pins, ARRAY_SIZE(broxton_headset_pins)); + + if (ret) + return ret; + + rt298_mic_detect(codec, &broxton_headset); + return 0; +} + +static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *dai = rtd->codec_dai; + + return hdac_hdmi_jack_init(dai, BXT_DPCM_AUDIO_HDMI1_PB + dai->id); +} + +static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP5 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int broxton_rt298_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->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, + 19200000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broxton_rt298_ops = { + .hw_params = broxton_rt298_hw_params, +}; + +/* broxton digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broxton_rt298_dais[] = { + /* Front End DAI links */ + [BXT_DPCM_AUDIO_PB] + { + .name = "Bxt Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [BXT_DPCM_AUDIO_CP] + { + .name = "Bxt Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [BXT_DPCM_AUDIO_REF_CP] + { + .name = "Bxt Audio Reference cap", + .stream_name = "refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI1_PB] + { + .name = "Bxt HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI2_PB] + { + .name = "Bxt HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + [BXT_DPCM_AUDIO_HDMI3_PB] + { + .name = "Bxt HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP5 - Codec */ + .name = "SSP5-Codec", + .id = 0, + .cpu_dai_name = "SSP5 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt298-aif1", + .init = broxton_rt298_codec_init, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broxton_ssp5_fixup, + .ops = &broxton_rt298_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = broxton_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +/* broxton audio machine driver for SPT + RT298S */ +static struct snd_soc_card broxton_rt298 = { + .name = "broxton-rt298", + .owner = THIS_MODULE, + .dai_link = broxton_rt298_dais, + .num_links = ARRAY_SIZE(broxton_rt298_dais), + .controls = broxton_controls, + .num_controls = ARRAY_SIZE(broxton_controls), + .dapm_widgets = broxton_widgets, + .num_dapm_widgets = ARRAY_SIZE(broxton_widgets), + .dapm_routes = broxton_rt298_map, + .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), + .fully_routed = true, +}; + +static int broxton_audio_probe(struct platform_device *pdev) +{ + broxton_rt298.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &broxton_rt298); +} + +static struct platform_driver broxton_audio = { + .probe = broxton_audio_probe, + .driver = { + .name = "bxt_alc298s_i2s", + }, +}; +module_platform_driver(broxton_audio) + +/* Module information */ +MODULE_AUTHOR("Ramesh Babu <Ramesh.Babu@intel.com>"); +MODULE_AUTHOR("Senthilnathan Veppur <senthilnathanx.veppur@intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Broxton"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bxt_alc298s_i2s"); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 032a2e753f0b..88efb62439ba 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1c95ccc886c4..35f591eab3c9 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -267,7 +267,7 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index ac60b04540fd..cdcced9f32b6 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -255,7 +255,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 3f2c1ea3a83e..d7ef292c402d 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -295,7 +295,7 @@ static struct snd_soc_dai_link cht_dailink[] = { /* back ends */ { .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index 2e5347f8f96c..df9d254baa18 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -273,7 +273,7 @@ static struct snd_soc_dai_link cht_dailink[] = { { /* SSP2 - Codec */ .name = "SSP2-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "ssp2-port", .platform_name = "sst-mfld-platform", .no_pcm = 1, diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c index 22558572cb9c..863f1d5e2a2c 100644 --- a/sound/soc/intel/boards/haswell.c +++ b/sound/soc/intel/boards/haswell.c @@ -156,7 +156,7 @@ static struct snd_soc_dai_link haswell_rt5640_dais[] = { { /* SSP0 - Codec */ .name = "Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "snd-soc-dummy-dai", .platform_name = "snd-soc-dummy", .no_pcm = 1, diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 72176b79a18d..d2808652b974 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -30,6 +30,16 @@ static struct snd_soc_jack skylake_headset; static struct snd_soc_card skylake_audio_card; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_nau8825_private { + struct list_head hdmi_pcm_list; +}; + enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -192,23 +202,56 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB); + pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) @@ -391,7 +434,6 @@ static struct snd_soc_dai_link skylake_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, .ops = &skylaye_refcap_ops, @@ -456,7 +498,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -472,7 +514,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -489,7 +531,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -501,7 +543,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -512,7 +554,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -523,7 +565,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", @@ -534,6 +576,21 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825max", @@ -547,11 +604,21 @@ static struct snd_soc_card skylake_audio_card = { .dapm_routes = skylake_map, .num_dapm_routes = ARRAY_SIZE(skylake_map), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_nau8825_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_audio_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_audio_card, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); } diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 5f1ca99ae9b0..e19aa99c4f72 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -34,6 +34,15 @@ static struct snd_soc_jack skylake_headset; static struct snd_soc_card skylake_audio_card; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_nau88125_private { + struct list_head hdmi_pcm_list; +}; enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -222,24 +231,57 @@ static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB); + return 0; } static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; + pcm->codec_dai = dai; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI2_PB); + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI3_PB); + pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) @@ -440,7 +482,6 @@ static struct snd_soc_dai_link skylake_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, .ops = &skylaye_refcap_ops, @@ -505,7 +546,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -523,7 +564,7 @@ static struct snd_soc_dai_link skylake_dais[] = { { /* SSP1 - Codec */ .name = "SSP1-Codec", - .be_id = 1, + .id = 1, .cpu_dai_name = "SSP1 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -540,7 +581,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "dmic01", - .be_id = 2, + .id = 2, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -552,7 +593,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp1", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -563,7 +604,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp2", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -574,7 +615,7 @@ static struct snd_soc_dai_link skylake_dais[] = { }, { .name = "iDisp3", - .be_id = 5, + .id = 5, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", @@ -585,6 +626,21 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825adi", @@ -600,11 +656,21 @@ static struct snd_soc_card skylake_audio_card = { .codec_conf = ssm4567_codec_conf, .num_configs = ARRAY_SIZE(ssm4567_codec_conf), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_nau88125_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_audio_card.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_audio_card, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); } diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 2016397a8e75..426b48233fdb 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -30,6 +30,16 @@ static struct snd_soc_jack skylake_headset; +struct skl_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct skl_rt286_private { + struct list_head hdmi_pcm_list; +}; + enum { SKL_DPCM_AUDIO_PB = 0, SKL_DPCM_AUDIO_CP, @@ -142,9 +152,20 @@ static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd) { + struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *dai = rtd->codec_dai; + struct skl_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; - return hdac_hdmi_jack_init(dai, SKL_DPCM_AUDIO_HDMI1_PB + dai->id); + pcm->device = SKL_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; } static unsigned int rates[] = { @@ -317,7 +338,6 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { .platform_name = "0000:00:1f.3", .init = NULL, .dpcm_capture = 1, - .ignore_suspend = 1, .nonatomic = 1, .dynamic = 1, }, @@ -375,7 +395,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { { /* SSP0 - Codec */ .name = "SSP0-Codec", - .be_id = 0, + .id = 0, .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, @@ -393,7 +413,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "dmic01", - .be_id = 1, + .id = 1, .cpu_dai_name = "DMIC01 Pin", .codec_name = "dmic-codec", .codec_dai_name = "dmic-hifi", @@ -405,7 +425,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp1", - .be_id = 2, + .id = 2, .cpu_dai_name = "iDisp1 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi1", @@ -416,7 +436,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp2", - .be_id = 3, + .id = 3, .cpu_dai_name = "iDisp2 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi2", @@ -427,7 +447,7 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, { .name = "iDisp3", - .be_id = 4, + .id = 4, .cpu_dai_name = "iDisp3 Pin", .codec_name = "ehdaudio0D2", .codec_dai_name = "intel-hdmi-hifi3", @@ -438,6 +458,21 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, }; +static int skylake_card_late_probe(struct snd_soc_card *card) +{ + struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card); + struct skl_hdmi_pcm *pcm; + int err; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device); + if (err < 0) + return err; + } + + return 0; +} + /* skylake audio machine driver for SPT + RT286S */ static struct snd_soc_card skylake_rt286 = { .name = "skylake-rt286", @@ -451,11 +486,21 @@ static struct snd_soc_card skylake_rt286 = { .dapm_routes = skylake_rt286_map, .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), .fully_routed = true, + .late_probe = skylake_card_late_probe, }; static int skylake_audio_probe(struct platform_device *pdev) { + struct skl_rt286_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + skylake_rt286.dev = &pdev->dev; + snd_soc_card_set_drvdata(&skylake_rt286, ctx); return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286); } diff --git a/sound/soc/intel/common/sst-acpi.h b/sound/soc/intel/common/sst-acpi.h index 4dcfb7e5ed70..8398cb227ba9 100644 --- a/sound/soc/intel/common/sst-acpi.h +++ b/sound/soc/intel/common/sst-acpi.h @@ -12,10 +12,19 @@ * */ +#include <linux/kconfig.h> +#include <linux/stddef.h> #include <linux/acpi.h> /* translation fron HID to I2C name, needed for DAI codec_name */ +#if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); +#else +inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +{ + return NULL; +} +#endif /* acpi match */ struct sst_acpi_mach *sst_acpi_find_machine(struct sst_acpi_mach *machines); diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1aa819c7e09b..994256b39b9c 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n", dma_area, size, pages); for (i = 0; i < pages; i++) { diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index 914b6dab9bea..c28f5d0e1d99 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o + skl-sst.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c new file mode 100644 index 000000000000..965ce40ce752 --- /dev/null +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -0,0 +1,328 @@ +/* + * bxt-sst.c - DSP library functions for BXT platform + * + * Copyright (C) 2015-16 Intel Corp + * Author:Rafal Redzimski <rafal.f.redzimski@intel.com> + * Jeeja KP <jeeja.kp@intel.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; version 2 of the License. + * + * 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. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/firmware.h> +#include <linux/device.h> + +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-sst-ipc.h" + +#define BXT_BASEFW_TIMEOUT 3000 +#define BXT_INIT_TIMEOUT 500 +#define BXT_IPC_PURGE_FW 0x01004000 + +#define BXT_ROM_INIT 0x5 +#define BXT_ADSP_SRAM0_BASE 0x80000 + +/* Firmware status window */ +#define BXT_ADSP_FW_STATUS BXT_ADSP_SRAM0_BASE +#define BXT_ADSP_ERROR_CODE (BXT_ADSP_FW_STATUS + 0x4) + +#define BXT_ADSP_SRAM1_BASE 0xA0000 + +static unsigned int bxt_get_errorcode(struct sst_dsp *ctx) +{ + return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE); +} + +static int sst_bxt_prepare_fw(struct sst_dsp *ctx, + const void *fwdata, u32 fwsize) +{ + int stream_tag, ret, i; + u32 reg; + + stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab); + if (stream_tag < 0) { + dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n", + stream_tag); + return stream_tag; + } + + ctx->dsp_ops.stream_tag = stream_tag; + memcpy(ctx->dmab.area, fwdata, fwsize); + + /* Purge FW request */ + sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY | + BXT_IPC_PURGE_FW | (stream_tag - 1)); + + ret = skl_dsp_enable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret); + ret = -EIO; + goto base_fw_load_failed; + } + + for (i = BXT_INIT_TIMEOUT; i > 0; --i) { + reg = sst_dsp_shim_read(ctx, SKL_ADSP_REG_HIPCIE); + + if (reg & SKL_ADSP_REG_HIPCIE_DONE) { + sst_dsp_shim_update_bits_forced(ctx, + SKL_ADSP_REG_HIPCIE, + SKL_ADSP_REG_HIPCIE_DONE, + SKL_ADSP_REG_HIPCIE_DONE); + break; + } + mdelay(1); + } + if (!i) { + dev_info(ctx->dev, "Waiting for HIPCIE done, reg: 0x%x\n", reg); + sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCIE, + SKL_ADSP_REG_HIPCIE_DONE, + SKL_ADSP_REG_HIPCIE_DONE); + } + + /* enable Interrupt */ + skl_ipc_int_enable(ctx); + skl_ipc_op_int_enable(ctx); + + for (i = BXT_INIT_TIMEOUT; i > 0; --i) { + if (SKL_FW_INIT == + (sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS) & + SKL_FW_STS_MASK)) { + + dev_info(ctx->dev, "ROM loaded, continue FW loading\n"); + break; + } + mdelay(1); + } + if (!i) { + dev_err(ctx->dev, "Timeout for ROM init, HIPCIE: 0x%x\n", reg); + ret = -EIO; + goto base_fw_load_failed; + } + + return ret; + +base_fw_load_failed: + ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag); + skl_dsp_disable_core(ctx); + return ret; +} + +static int sst_transfer_fw_host_dma(struct sst_dsp *ctx) +{ + int ret; + + ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag); + ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK, + BXT_ROM_INIT, BXT_BASEFW_TIMEOUT, "Firmware boot"); + + ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag); + ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag); + + return ret; +} + +static int bxt_load_base_firmware(struct sst_dsp *ctx) +{ + const struct firmware *fw = NULL; + struct skl_sst *skl = ctx->thread_context; + int ret; + + ret = request_firmware(&fw, ctx->fw_name, ctx->dev); + if (ret < 0) { + dev_err(ctx->dev, "Request firmware failed %d\n", ret); + goto sst_load_base_firmware_failed; + } + + ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size); + /* Retry Enabling core and ROM load. Retry seemed to help */ + if (ret < 0) { + ret = sst_bxt_prepare_fw(ctx, fw->data, fw->size); + if (ret < 0) { + dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret); + goto sst_load_base_firmware_failed; + } + } + + ret = sst_transfer_fw_host_dma(ctx); + if (ret < 0) { + dev_err(ctx->dev, "Transfer firmware failed %d\n", ret); + dev_info(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), + sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); + + skl_dsp_disable_core(ctx); + } else { + dev_dbg(ctx->dev, "Firmware download successful\n"); + ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, + msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n"); + skl_dsp_disable_core(ctx); + ret = -EIO; + } else { + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + ret = 0; + } + } + +sst_load_base_firmware_failed: + release_firmware(fw); + return ret; +} + +static int bxt_set_dsp_D0(struct sst_dsp *ctx) +{ + struct skl_sst *skl = ctx->thread_context; + int ret; + + skl->boot_complete = false; + + ret = skl_dsp_enable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "enable dsp core failed ret: %d\n", ret); + return ret; + } + + /* enable interrupt */ + skl_ipc_int_enable(ctx); + skl_ipc_op_int_enable(ctx); + + ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, + msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); + if (ret == 0) { + dev_err(ctx->dev, "ipc: error DSP boot timeout\n"); + dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), + sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); + return -EIO; + } + + skl_dsp_set_state_locked(ctx, SKL_DSP_RUNNING); + return 0; +} + +static int bxt_set_dsp_D3(struct sst_dsp *ctx) +{ + struct skl_ipc_dxstate_info dx; + struct skl_sst *skl = ctx->thread_context; + int ret = 0; + + if (!is_skl_dsp_running(ctx)) + return ret; + + dx.core_mask = SKL_DSP_CORE0_MASK; + dx.dx_mask = SKL_IPC_D3_MASK; + + ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID, + SKL_BASE_FW_MODULE_ID, &dx); + if (ret < 0) { + dev_err(ctx->dev, "Failed to set DSP to D3 state: %d\n", ret); + return ret; + } + + ret = skl_dsp_disable_core(ctx); + if (ret < 0) { + dev_err(ctx->dev, "disbale dsp core failed: %d\n", ret); + ret = -EIO; + } + + skl_dsp_set_state_locked(ctx, SKL_DSP_RESET); + return 0; +} + +static struct skl_dsp_fw_ops bxt_fw_ops = { + .set_state_D0 = bxt_set_dsp_D0, + .set_state_D3 = bxt_set_dsp_D3, + .load_fw = bxt_load_base_firmware, + .get_fw_errcode = bxt_get_errorcode, +}; + +static struct sst_ops skl_ops = { + .irq_handler = skl_dsp_sst_interrupt, + .write = sst_shim32_write, + .read = sst_shim32_read, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .free = skl_dsp_free, +}; + +static struct sst_dsp_device skl_dev = { + .thread = skl_dsp_irq_thread_handler, + .ops = &skl_ops, +}; + +int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + const char *fw_name, struct skl_dsp_loader_ops dsp_ops, + struct skl_sst **dsp) +{ + struct skl_sst *skl; + struct sst_dsp *sst; + int ret; + + skl = devm_kzalloc(dev, sizeof(*skl), GFP_KERNEL); + if (skl == NULL) + return -ENOMEM; + + skl->dev = dev; + skl_dev.thread_context = skl; + + skl->dsp = skl_dsp_ctx_init(dev, &skl_dev, irq); + if (!skl->dsp) { + dev_err(skl->dev, "skl_dsp_ctx_init failed\n"); + return -ENODEV; + } + + sst = skl->dsp; + sst->fw_name = fw_name; + sst->dsp_ops = dsp_ops; + sst->fw_ops = bxt_fw_ops; + sst->addr.lpe = mmio_base; + sst->addr.shim = mmio_base; + + sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), + SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); + + ret = skl_ipc_init(dev, skl); + if (ret) + return ret; + + skl->boot_complete = false; + init_waitqueue_head(&skl->boot_wait); + + ret = sst->fw_ops.load_fw(sst); + if (ret < 0) { + dev_err(dev, "Load base fw failed: %x", ret); + return ret; + } + + if (dsp) + *dsp = skl; + + return 0; +} +EXPORT_SYMBOL_GPL(bxt_sst_dsp_init); + + +void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) +{ + skl_ipc_free(&ctx->ipc); + ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp); + + if (ctx->dsp->addr.lpe) + iounmap(ctx->dsp->addr.lpe); + + ctx->dsp->ops->free(ctx->dsp); +} +EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel Broxton IPC driver"); diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 79c5089b85d6..226db84ba20f 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -72,6 +72,105 @@ static void skl_dsp_enable_notification(struct skl_sst *ctx, bool enable) skl_ipc_set_large_config(&ctx->ipc, &msg, (u32 *)&mask); } +static int skl_dsp_setup_spib(struct device *dev, unsigned int size, + int stream_tag, int enable) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_stream *stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + struct hdac_ext_stream *estream; + + if (!stream) + return -EINVAL; + + estream = stream_to_hdac_ext_stream(stream); + /* enable/disable SPIB for this hdac stream */ + snd_hdac_ext_stream_spbcap_enable(ebus, enable, stream->index); + + /* set the spib value */ + snd_hdac_ext_stream_set_spib(ebus, estream, size); + + return 0; +} + +static int skl_dsp_prepare(struct device *dev, unsigned int format, + unsigned int size, struct snd_dma_buffer *dmab) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_stream *estream; + struct hdac_stream *stream; + struct snd_pcm_substream substream; + int ret; + + if (!bus) + return -ENODEV; + + memset(&substream, 0, sizeof(substream)); + substream.stream = SNDRV_PCM_STREAM_PLAYBACK; + + estream = snd_hdac_ext_stream_assign(ebus, &substream, + HDAC_EXT_STREAM_TYPE_HOST); + if (!estream) + return -ENODEV; + + stream = hdac_stream(estream); + + /* assign decouple host dma channel */ + ret = snd_hdac_dsp_prepare(stream, format, size, dmab); + if (ret < 0) + return ret; + + skl_dsp_setup_spib(dev, size, stream->stream_tag, true); + + return stream->stream_tag; +} + +static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_stream *stream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + if (!stream) + return -EINVAL; + + snd_hdac_dsp_trigger(stream, start); + + return 0; +} + +static int skl_dsp_cleanup(struct device *dev, + struct snd_dma_buffer *dmab, int stream_tag) +{ + struct hdac_ext_bus *ebus = dev_get_drvdata(dev); + struct hdac_stream *stream; + struct hdac_ext_stream *estream; + struct hdac_bus *bus = ebus_to_hbus(ebus); + + if (!bus) + return -ENODEV; + + stream = snd_hdac_get_stream(bus, + SNDRV_PCM_STREAM_PLAYBACK, stream_tag); + if (!stream) + return -EINVAL; + + estream = stream_to_hdac_ext_stream(stream); + skl_dsp_setup_spib(dev, 0, stream_tag, false); + snd_hdac_ext_stream_release(estream, HDAC_EXT_STREAM_TYPE_HOST); + + snd_hdac_dsp_cleanup(stream, dmab); + + return 0; +} + static struct skl_dsp_loader_ops skl_get_loader_ops(void) { struct skl_dsp_loader_ops loader_ops; @@ -84,6 +183,21 @@ static struct skl_dsp_loader_ops skl_get_loader_ops(void) return loader_ops; }; +static struct skl_dsp_loader_ops bxt_get_loader_ops(void) +{ + struct skl_dsp_loader_ops loader_ops; + + memset(&loader_ops, 0, sizeof(loader_ops)); + + loader_ops.alloc_dma_buf = skl_alloc_dma_buf; + loader_ops.free_dma_buf = skl_free_dma_buf; + loader_ops.prepare = skl_dsp_prepare; + loader_ops.trigger = skl_dsp_trigger; + loader_ops.cleanup = skl_dsp_cleanup; + + return loader_ops; +}; + static const struct skl_dsp_ops dsp_ops[] = { { .id = 0x9d70, @@ -91,6 +205,12 @@ static const struct skl_dsp_ops dsp_ops[] = { .init = skl_sst_dsp_init, .cleanup = skl_sst_dsp_cleanup }, + { + .id = 0x5a98, + .loader_ops = bxt_get_loader_ops, + .init = bxt_sst_dsp_init, + .cleanup = bxt_sst_dsp_cleanup + }, }; static int skl_get_dsp_ops(int pci_id) @@ -744,7 +864,7 @@ int skl_init_module(struct skl_sst *ctx, return ret; } mconfig->m_state = SKL_MODULE_INIT_DONE; - + kfree(param_data); return ret; } diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 14d1916ea9f8..7d73648e5f9a 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -25,11 +25,12 @@ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, #define DSDT_NHLT_PATH "\\_SB.PCI0.HDAS" -void *skl_nhlt_init(struct device *dev) +struct nhlt_acpi_table *skl_nhlt_init(struct device *dev) { acpi_handle handle; union acpi_object *obj; struct nhlt_resource_desc *nhlt_ptr = NULL; + struct nhlt_acpi_table *nhlt_table = NULL; if (ACPI_FAILURE(acpi_get_handle(NULL, DSDT_NHLT_PATH, &handle))) { dev_err(dev, "Requested NHLT device not found\n"); @@ -39,18 +40,20 @@ void *skl_nhlt_init(struct device *dev) obj = acpi_evaluate_dsm(handle, OSC_UUID, 1, 1, NULL); if (obj && obj->type == ACPI_TYPE_BUFFER) { nhlt_ptr = (struct nhlt_resource_desc *)obj->buffer.pointer; - - return memremap(nhlt_ptr->min_addr, nhlt_ptr->length, + nhlt_table = (struct nhlt_acpi_table *) + memremap(nhlt_ptr->min_addr, nhlt_ptr->length, MEMREMAP_WB); + ACPI_FREE(obj); + return nhlt_table; } dev_err(dev, "device specific method to extract NHLT blob failed\n"); return NULL; } -void skl_nhlt_free(void *addr) +void skl_nhlt_free(struct nhlt_acpi_table *nhlt) { - memunmap(addr); + memunmap((void *) nhlt); } static struct nhlt_specific_cfg *skl_get_specific_cfg( @@ -120,7 +123,7 @@ struct nhlt_specific_cfg struct hdac_bus *bus = ebus_to_hbus(&skl->ebus); struct device *dev = bus->dev; struct nhlt_specific_cfg *sp_config; - struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + struct nhlt_acpi_table *nhlt = skl->nhlt; u16 bps = (s_fmt == 16) ? 16 : 32; u8 j; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index dab0900eef26..7c81b31748ff 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -51,7 +51,7 @@ static struct snd_pcm_hardware azx_pcm_hw = { .rate_min = 8000, .rate_max = 48000, .channels_min = 1, - .channels_max = HDA_QUAD, + .channels_max = 8, .buffer_bytes_max = AZX_MAX_BUF_SIZE, .period_bytes_min = 128, .period_bytes_max = AZX_MAX_BUF_SIZE / 2, @@ -213,7 +213,7 @@ static int skl_be_prepare(struct snd_pcm_substream *substream, struct skl_sst *ctx = skl->skl_sst; struct skl_module_cfg *mconfig; - if ((dai->playback_active > 1) || (dai->capture_active > 1)) + if (dai->playback_widget->power || dai->capture_widget->power) return 0; mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream); @@ -402,23 +402,33 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, struct skl_module_cfg *mconfig; struct hdac_ext_bus *ebus = get_bus_ctx(substream); struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); + struct snd_soc_dapm_widget *w; int ret; mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); if (!mconfig) return -EIO; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + w = dai->playback_widget; + else + w = dai->capture_widget; + switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - skl_pcm_prepare(substream, dai); - /* - * enable DMA Resume enable bit for the stream, set the dpib - * & lpib position to resune before starting the DMA - */ - snd_hdac_ext_stream_drsm_enable(ebus, true, - hdac_stream(stream)->index); - snd_hdac_ext_stream_set_dpibr(ebus, stream, stream->dpib); - snd_hdac_ext_stream_set_lpib(stream, stream->lpib); + if (!w->ignore_suspend) { + skl_pcm_prepare(substream, dai); + /* + * enable DMA Resume enable bit for the stream, set the + * dpib & lpib position to resume before starting the + * DMA + */ + snd_hdac_ext_stream_drsm_enable(ebus, true, + hdac_stream(stream)->index); + snd_hdac_ext_stream_set_dpibr(ebus, stream, + stream->dpib); + snd_hdac_ext_stream_set_lpib(stream, stream->lpib); + } case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: @@ -448,7 +458,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, return ret; ret = skl_decoupled_trigger(substream, cmd); - if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) { + if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { /* save the dpib and lpib positions */ stream->dpib = readl(ebus->bus.remap_addr + AZX_REG_VS_SDXDPIB_XBASE + @@ -523,7 +533,6 @@ static int skl_link_pcm_prepare(struct snd_pcm_substream *substream, if (!link) return -EINVAL; - snd_hdac_ext_bus_link_power_up(link); snd_hdac_ext_link_stream_reset(link_dev); snd_hdac_ext_link_stream_setup(link_dev, format_val); @@ -682,7 +691,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI1 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -697,7 +706,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI2 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -712,7 +721,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "HDMI3 Playback", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | @@ -760,12 +769,84 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }, { + .name = "SSP2 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp2 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp2 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP3 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp3 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp3 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP4 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp4 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp4 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ + .name = "SSP5 Pin", + .ops = &skl_be_ssp_dai_ops, + .playback = { + .stream_name = "ssp5 Tx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "ssp5 Rx", + .channels_min = HDA_STEREO, + .channels_max = HDA_STEREO, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}, +{ .name = "iDisp1 Pin", .ops = &skl_link_dai_ops, .playback = { .stream_name = "iDisp1 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE, @@ -777,7 +858,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "iDisp2 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | @@ -790,7 +871,7 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { .playback = { .stream_name = "iDisp3 Tx", .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, + .channels_max = 8, .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c index 2962ef22fc84..13c19855ee1a 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.c +++ b/sound/soc/intel/skylake/skl-sst-dsp.c @@ -336,8 +336,6 @@ void skl_dsp_free(struct sst_dsp *dsp) skl_ipc_int_disable(dsp); free_irq(dsp->irq, dsp); - dsp->cl_dev.ops.cl_cleanup_controller(dsp); - skl_cldma_int_disable(dsp); skl_ipc_op_int_disable(dsp); skl_ipc_int_disable(dsp); diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h index b6e310d49dd6..deabe7308d3b 100644 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ b/sound/soc/intel/skylake/skl-sst-dsp.h @@ -118,16 +118,25 @@ struct skl_dsp_fw_ops { int (*set_state_D0)(struct sst_dsp *ctx); int (*set_state_D3)(struct sst_dsp *ctx); unsigned int (*get_fw_errcode)(struct sst_dsp *ctx); - int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, char *mod_name); + int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name); int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id); }; struct skl_dsp_loader_ops { + int stream_tag; + int (*alloc_dma_buf)(struct device *dev, struct snd_dma_buffer *dmab, size_t size); int (*free_dma_buf)(struct device *dev, struct snd_dma_buffer *dmab); + int (*prepare)(struct device *dev, unsigned int format, + unsigned int byte_size, + struct snd_dma_buffer *bufp); + int (*trigger)(struct device *dev, bool start, int stream_tag); + + int (*cleanup)(struct device *dev, struct snd_dma_buffer *dmab, + int stream_tag); }; struct skl_load_module_info { @@ -160,6 +169,10 @@ int skl_dsp_boot(struct sst_dsp *ctx); int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, const char *fw_name, struct skl_dsp_loader_ops dsp_ops, struct skl_sst **dsp); +int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, + const char *fw_name, struct skl_dsp_loader_ops dsp_ops, + struct skl_sst **dsp); void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); +void bxt_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx); #endif /*__SKL_SST_DSP_H__*/ diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 348a734f8e24..13ec8d53b526 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -20,6 +20,7 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/err.h> +#include <linux/uuid.h> #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" @@ -304,14 +305,16 @@ static int skl_transfer_module(struct sst_dsp *ctx, return ret; } -static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, char *guid) +static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) { struct skl_module_table *module_entry = NULL; int ret = 0; char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ + uuid_le *uuid_mod; - snprintf(mod_name, sizeof(mod_name), "%s%s%s", - "intel/dsp_fw_", guid, ".bin"); + uuid_mod = (uuid_le *)guid; + snprintf(mod_name, sizeof(mod_name), "%s%pUL%s", + "intel/dsp_fw_", uuid_mod, ".bin"); module_entry = skl_module_get_from_id(ctx, mod_id); if (module_entry == NULL) { @@ -451,6 +454,10 @@ void skl_sst_dsp_cleanup(struct device *dev, struct skl_sst *ctx) skl_clear_module_table(ctx->dsp); skl_ipc_free(&ctx->ipc); ctx->dsp->ops->free(ctx->dsp); + if (ctx->boot_complete) { + ctx->dsp->cl_dev.ops.cl_cleanup_controller(ctx->dsp); + skl_cldma_int_disable(ctx->dsp); + } } EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index cdb78b7e5a14..3e036b0349b9 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -154,13 +154,32 @@ static void skl_dump_mconfig(struct skl_sst *ctx, dev_dbg(ctx->dev, "ch_cfg = %d\n", mcfg->out_fmt[0].ch_cfg); } +static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs) +{ + int slot_map = 0xFFFFFFFF; + int start_slot = 0; + int i; + + for (i = 0; i < chs; i++) { + /* + * For 2 channels with starting slot as 0, slot map will + * look like 0xFFFFFF10. + */ + slot_map &= (~(0xF << (4 * i)) | (start_slot << (4 * i))); + start_slot++; + } + fmt->ch_map = slot_map; +} + static void skl_tplg_update_params(struct skl_module_fmt *fmt, struct skl_pipe_params *params, int fixup) { if (fixup & SKL_RATE_FIXUP_MASK) fmt->s_freq = params->s_freq; - if (fixup & SKL_CH_FIXUP_MASK) + if (fixup & SKL_CH_FIXUP_MASK) { fmt->channels = params->ch; + skl_tplg_update_chmap(fmt, fmt->channels); + } if (fixup & SKL_FMT_FIXUP_MASK) { fmt->valid_bit_depth = skl_get_bit_depth(params->s_fmt); @@ -1564,6 +1583,8 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, return -ENOMEM; w->priv = mconfig; + memcpy(&mconfig->guid, &dfw_config->uuid, 16); + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; @@ -1593,10 +1614,6 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, mconfig->time_slot = dfw_config->time_slot; mconfig->formats_config.caps_size = dfw_config->caps.caps_size; - if (dfw_config->is_loadable) - memcpy(mconfig->guid, dfw_config->uuid, - ARRAY_SIZE(dfw_config->uuid)); - mconfig->m_in_pin = devm_kzalloc(bus->dev, (mconfig->max_in_queue) * sizeof(*mconfig->m_in_pin), GFP_KERNEL); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index d2d923002d5c..e4b399cd7868 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -281,7 +281,7 @@ enum skl_module_state { }; struct skl_module_cfg { - char guid[SKL_UUID_STR_SZ]; + u8 guid[16]; struct skl_module_inst_id id; u8 domain; bool homogenous_inputs; diff --git a/sound/soc/intel/skylake/skl-tplg-interface.h b/sound/soc/intel/skylake/skl-tplg-interface.h index 1db88a63ac17..a32e5e9cc530 100644 --- a/sound/soc/intel/skylake/skl-tplg-interface.h +++ b/sound/soc/intel/skylake/skl-tplg-interface.h @@ -181,7 +181,7 @@ struct skl_dfw_pipe { } __packed; struct skl_dfw_module { - char uuid[SKL_UUID_STR_SZ]; + u8 uuid[16]; u16 module_id; u16 instance_id; diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 3982f5536f2d..06d8c263c68f 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -229,7 +229,12 @@ static int skl_suspend(struct device *dev) * running, we need to save the state for these and continue */ if (skl->supend_active) { + /* turn off the links and stop the CORB/RIRB DMA if it is On */ snd_hdac_ext_bus_link_power_down_all(ebus); + + if (ebus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(&ebus->bus); + enable_irq_wake(bus->irq); pci_save_state(pci); pci_disable_device(pci); @@ -255,6 +260,7 @@ static int skl_resume(struct device *dev) struct hdac_ext_bus *ebus = pci_get_drvdata(pci); struct skl *skl = ebus_to_skl(ebus); struct hdac_bus *bus = ebus_to_hbus(ebus); + struct hdac_ext_link *hlink = NULL; int ret; /* Turned OFF in HDMI codec driver after codec reconfiguration */ @@ -276,8 +282,29 @@ static int skl_resume(struct device *dev) ret = pci_enable_device(pci); snd_hdac_ext_bus_link_power_up_all(ebus); disable_irq_wake(bus->irq); + /* + * turn On the links which are On before active suspend + * and start the CORB/RIRB DMA if On before + * active suspend. + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (hlink->ref_count) + snd_hdac_ext_bus_link_power_up(hlink); + } + + if (ebus->cmd_dma_state) + snd_hdac_bus_init_cmd_io(&ebus->bus); } else { ret = _skl_resume(ebus); + + /* turn off the links which are off before suspend */ + list_for_each_entry(hlink, &ebus->hlink_list, list) { + if (!hlink->ref_count) + snd_hdac_ext_bus_link_power_down(hlink); + } + + if (!ebus->cmd_dma_state) + snd_hdac_bus_stop_cmd_io(&ebus->bus); } return ret; @@ -613,6 +640,7 @@ static int skl_probe(struct pci_dev *pci, struct skl *skl; struct hdac_ext_bus *ebus = NULL; struct hdac_bus *bus = NULL; + struct hdac_ext_link *hlink = NULL; int err; /* we use ext core ops, so provide NULL for ops here */ @@ -643,7 +671,7 @@ static int skl_probe(struct pci_dev *pci, err = skl_machine_device_register(skl, (void *)pci_id->driver_data); if (err < 0) - goto out_free; + goto out_nhlt_free; err = skl_init_dsp(skl); if (err < 0) { @@ -679,6 +707,12 @@ static int skl_probe(struct pci_dev *pci, } } + /* + * we are done probling so decrement link counts + */ + list_for_each_entry(hlink, &ebus->hlink_list, list) + snd_hdac_ext_bus_link_put(ebus, hlink); + /*configure PM */ pm_runtime_put_noidle(bus->dev); pm_runtime_allow(bus->dev); @@ -693,6 +727,8 @@ out_dsp_free: skl_free_dsp(skl); out_mach_free: skl_machine_device_unregister(skl); +out_nhlt_free: + skl_nhlt_free(skl->nhlt); out_free: skl->init_failed = 1; skl_free(ebus); @@ -743,6 +779,7 @@ static void skl_remove(struct pci_dev *pci) skl_free_dsp(skl); skl_machine_device_unregister(skl); skl_dmic_device_unregister(skl); + skl_nhlt_free(skl->nhlt); skl_free(ebus); dev_set_drvdata(&pci->dev, NULL); } diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 39e16fa7a92b..4b4b3876aea9 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -66,7 +66,7 @@ struct skl { struct platform_device *dmic_dev; struct platform_device *i2s_dev; - void *nhlt; /* nhlt ptr */ + struct nhlt_acpi_table *nhlt; /* nhlt ptr */ struct skl_sst *skl_sst; /* sst skl ctx */ struct skl_dsp_resource resource; @@ -103,8 +103,8 @@ struct skl_dsp_ops { int skl_platform_unregister(struct device *dev); int skl_platform_register(struct device *dev); -void *skl_nhlt_init(struct device *dev); -void skl_nhlt_free(void *addr); +struct nhlt_acpi_table *skl_nhlt_init(struct device *dev); +void skl_nhlt_free(struct nhlt_acpi_table *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e20416bd3..574c6af28c06 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -75,29 +82,37 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START, - I2S_XFER_TXS_START); + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START, - I2S_XFER_TXS_STOP); + if (!i2s->rx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC, - I2S_CLR_TXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); - - /* Should wait for clear operation to finish */ - while (val & I2S_CLR_TXC) { regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } @@ -113,29 +128,37 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_RXS_START, - I2S_XFER_RXS_START); + I2S_XFER_TXS_START | I2S_XFER_RXS_START, + I2S_XFER_TXS_START | I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_RXS_START, - I2S_XFER_RXS_STOP); + if (!i2s->tx_start) { + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START | + I2S_XFER_RXS_START, + I2S_XFER_TXS_STOP | + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_RXC, - I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC | I2S_CLR_RXC, + I2S_CLR_TXC | I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); - - /* Should wait for clear operation to finish */ - while (val & I2S_CLR_RXC) { regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; + + /* Should wait for clear operation to finish */ + while (val) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; + } } } } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b159610..16369cad4803 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -930,7 +930,18 @@ static struct snd_soc_component *soc_find_component( return NULL; } -static struct snd_soc_dai *snd_soc_find_dai( +/** + * snd_soc_find_dai - Find a registered DAI + * + * @dlc: name of the DAI and optional component info to match + * + * This function will search all regsitered components and their DAIs to + * find the DAI of the same name. The component's of_node and name + * should also match if being specified. + * + * Return: pointer of DAI, or NULL if not found. + */ +struct snd_soc_dai *snd_soc_find_dai( const struct snd_soc_dai_link_component *dlc) { struct snd_soc_component *component; @@ -959,6 +970,7 @@ static struct snd_soc_dai *snd_soc_find_dai( return NULL; } +EXPORT_SYMBOL_GPL(snd_soc_find_dai); static bool soc_is_dai_link_bound(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6fd1906af387..6cef3977507a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea } /* - * Prepare formats mask for valid/allowed sample types. If the dma does - * not have support for the given physical word size, it needs to be - * masked out so user space can not use the format which produces - * corrupted audio. - * In case the dma driver does not implement the slave_caps the default - * assumption is that it supports 1, 2 and 4 bytes widths. + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. */ - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* Enable only samples with DMA supported physical widths */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); - break; - default: - /* Unsupported types */ - break; + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } - } return snd_soc_set_runtime_hwparams(substream, &hw); } diff --git a/sound/usb/card.c b/sound/usb/card.c index 3fc63583a537..69860da473ea 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -350,6 +350,7 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: break; default: dev_err(&dev->dev, "unknown device speed %d\n", snd_usb_get_speed(dev)); @@ -450,6 +451,9 @@ static int snd_usb_audio_create(struct usb_interface *intf, case USB_SPEED_SUPER: strlcat(card->longname, ", super speed", sizeof(card->longname)); break; + case USB_SPEED_SUPER_PLUS: + strlcat(card->longname, ", super speed plus", sizeof(card->longname)); + break; default: break; } diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7ccbcaf6a147..26dd5f20f149 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -309,6 +309,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, * support reading */ if (snd_usb_get_sample_rate_quirk(chip)) return 0; + /* the firmware is likely buggy, don't repeat to fail too many times */ + if (chip->sample_rate_read_error > 2) + return 0; if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, @@ -316,6 +319,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data, sizeof(data))) < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); + chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 51ed1ac825fd..7712e2b84183 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -120,6 +120,7 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, case USB_SPEED_HIGH: case USB_SPEED_WIRELESS: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: if (get_endpoint(alts, 0)->bInterval >= 1 && get_endpoint(alts, 0)->bInterval <= 4) return get_endpoint(alts, 0)->bInterval - 1; diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 47de8af42f16..7ba92921bf28 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -911,6 +911,7 @@ static void snd_usbmidi_us122l_output(struct snd_usb_midi_out_endpoint *ep, switch (snd_usb_get_speed(ep->umidi->dev)) { case USB_SPEED_HIGH: case USB_SPEED_SUPER: + case USB_SPEED_SUPER_PLUS: count = 1; break; default: diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 4f85757009b3..2f8c388ef84f 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -45,6 +45,7 @@ #include <linux/bitops.h> #include <linux/init.h> #include <linux/list.h> +#include <linux/log2.h> #include <linux/slab.h> #include <linux/string.h> #include <linux/usb.h> @@ -1378,6 +1379,71 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_add_control(&cval->head, kctl); } +static int parse_clock_source_unit(struct mixer_build *state, int unitid, + void *_ftr) +{ + struct uac_clock_source_descriptor *hdr = _ftr; + struct usb_mixer_elem_info *cval; + struct snd_kcontrol *kctl; + char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; + int ret; + + if (state->mixer->protocol != UAC_VERSION_2) + return -EINVAL; + + if (hdr->bLength != sizeof(*hdr)) { + usb_audio_dbg(state->chip, + "Bogus clock source descriptor length of %d, ignoring.\n", + hdr->bLength); + return 0; + } + + /* + * The only property of this unit we are interested in is the + * clock source validity. If that isn't readable, just bail out. + */ + if (!uac2_control_is_readable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + return 0; + + cval = kzalloc(sizeof(*cval), GFP_KERNEL); + if (!cval) + return -ENOMEM; + + snd_usb_mixer_elem_init_std(&cval->head, state->mixer, hdr->bClockID); + + cval->min = 0; + cval->max = 1; + cval->channels = 1; + cval->val_type = USB_MIXER_BOOLEAN; + cval->control = UAC2_CS_CONTROL_CLOCK_VALID; + + if (uac2_control_is_writeable(hdr->bmControls, + ilog2(UAC2_CS_CONTROL_CLOCK_VALID))) + kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval); + else { + cval->master_readonly = 1; + kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval); + } + + if (!kctl) { + kfree(cval); + return -ENOMEM; + } + + kctl->private_free = snd_usb_mixer_elem_free; + ret = snd_usb_copy_string_desc(state, hdr->iClockSource, + name, sizeof(name)); + if (ret > 0) + snprintf(kctl->id.name, sizeof(kctl->id.name), + "%s Validity", name); + else + snprintf(kctl->id.name, sizeof(kctl->id.name), + "Clock Source %d Validity", hdr->bClockID); + + return snd_usb_mixer_add_control(&cval->head, kctl); +} + /* * parse a feature unit * @@ -2126,10 +2192,11 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) switch (p1[2]) { case UAC_INPUT_TERMINAL: - case UAC2_CLOCK_SOURCE: return 0; /* NOP */ case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); + case UAC2_CLOCK_SOURCE: + return parse_clock_source_unit(state, unitid, p1); case UAC_SELECTOR_UNIT: case UAC2_CLOCK_SELECTOR: return parse_audio_selector_unit(state, unitid, p1); @@ -2307,6 +2374,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, __u8 unitid = (index >> 8) & 0xff; __u8 control = (value >> 8) & 0xff; __u8 channel = value & 0xff; + unsigned int count = 0; if (channel >= MAX_CHANNELS) { usb_audio_dbg(mixer->chip, @@ -2315,6 +2383,12 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, return; } + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) + count++; + + if (count == 0) + return; + for (list = mixer->id_elems[unitid]; list; list = list->next_id_elem) { struct usb_mixer_elem_info *info; @@ -2322,7 +2396,7 @@ static void snd_usb_mixer_interrupt_v2(struct usb_mixer_interface *mixer, continue; info = (struct usb_mixer_elem_info *)list; - if (info->control != control) + if (count > 1 && info->control != control) continue; switch (attribute) { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b665d85555cb..4d5c89a7ba2b 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -47,6 +47,7 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + int sample_rate_read_error; struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ |