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-rw-r--r--include/linux/platform_data/asoc-palm27x.h9
-rw-r--r--include/linux/platform_data/asoc-poodle.h16
-rw-r--r--include/linux/platform_data/mmp_audio.h18
-rw-r--r--sound/soc/pxa/Kconfig176
-rw-r--r--sound/soc/pxa/Makefile33
-rw-r--r--sound/soc/pxa/brownstone.c133
-rw-r--r--sound/soc/pxa/corgi.c332
-rw-r--r--sound/soc/pxa/e740_wm9705.c168
-rw-r--r--sound/soc/pxa/e750_wm9705.c147
-rw-r--r--sound/soc/pxa/e800_wm9712.c147
-rw-r--r--sound/soc/pxa/em-x270.c92
-rw-r--r--sound/soc/pxa/hx4700.c207
-rw-r--r--sound/soc/pxa/magician.c366
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c201
-rw-r--r--sound/soc/pxa/mmp-pcm.c267
-rw-r--r--sound/soc/pxa/palm27x.c162
-rw-r--r--sound/soc/pxa/poodle.c291
-rw-r--r--sound/soc/pxa/tosa.c255
-rw-r--r--sound/soc/pxa/ttc-dkb.c143
-rw-r--r--sound/soc/pxa/z2.c218
-rw-r--r--sound/soc/pxa/zylonite.c266
21 files changed, 0 insertions, 3647 deletions
diff --git a/include/linux/platform_data/asoc-palm27x.h b/include/linux/platform_data/asoc-palm27x.h
deleted file mode 100644
index 22b69a393a57..000000000000
--- a/include/linux/platform_data/asoc-palm27x.h
+++ /dev/null
@@ -1,9 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0 */
-#ifndef _INCLUDE_PALMASOC_H_
-#define _INCLUDE_PALMASOC_H_
-
-struct palm27x_asoc_info {
- int jack_gpio;
-};
-
-#endif
diff --git a/include/linux/platform_data/asoc-poodle.h b/include/linux/platform_data/asoc-poodle.h
deleted file mode 100644
index 2052fad55c5c..000000000000
--- a/include/linux/platform_data/asoc-poodle.h
+++ /dev/null
@@ -1,16 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0 */
-#ifndef __LINUX_PLATFORM_DATA_POODLE_AUDIO
-#define __LINUX_PLATFORM_DATA_POODLE_AUDIO
-
-/* locomo is not a proper gpio driver, and uses its own api */
-struct poodle_audio_platform_data {
- struct device *locomo_dev;
-
- int gpio_amp_on;
- int gpio_mute_l;
- int gpio_mute_r;
- int gpio_232vcc_on;
- int gpio_jk_b;
-};
-
-#endif
diff --git a/include/linux/platform_data/mmp_audio.h b/include/linux/platform_data/mmp_audio.h
deleted file mode 100644
index 83428d8ee18d..000000000000
--- a/include/linux/platform_data/mmp_audio.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * MMP Platform AUDIO Management
- *
- * Copyright (c) 2011 Marvell Semiconductors Inc.
- */
-
-#ifndef MMP_AUDIO_H
-#define MMP_AUDIO_H
-
-struct mmp_audio_platdata {
- u32 period_max_capture;
- u32 buffer_max_capture;
- u32 period_max_playback;
- u32 buffer_max_playback;
-};
-
-#endif /* MMP_AUDIO_H */
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index a045693d5bc2..c26d1b36e8f7 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -8,10 +8,6 @@ config SND_PXA2XX_SOC
the PXA2xx AC97, I2S or SSP interface. You will also need
to select the audio interfaces to support below.
-config SND_MMP_SOC
- bool
- select MMP_SRAM
-
config SND_PXA2XX_AC97
tristate
@@ -41,15 +37,6 @@ config SND_MMP_SOC_SSPA
Say Y if you want to add support for codecs attached to
the MMP SSPA interface.
-config SND_PXA2XX_SOC_CORGI
- tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
- depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C
- select SND_PXA2XX_SOC_I2S
- select SND_SOC_WM8731_I2C
- help
- Say Y if you want to add support for SoC audio on Sharp
- Zaurus SL-C7x0 models (Corgi, Shepherd, Husky).
-
config SND_PXA2XX_SOC_SPITZ
tristate "SoC Audio support for Sharp Zaurus SL-Cxx00"
depends on SND_PXA2XX_SOC && PXA_SHARP_Cxx00 && I2C
@@ -59,101 +46,6 @@ config SND_PXA2XX_SOC_SPITZ
Say Y if you want to add support for SoC audio on Sharp
Zaurus SL-Cxx00 models (Spitz, Borzoi and Akita).
-config SND_PXA2XX_SOC_Z2
- tristate "SoC Audio support for Zipit Z2"
- depends on SND_PXA2XX_SOC && MACH_ZIPIT2 && I2C
- select SND_PXA2XX_SOC_I2S
- select SND_SOC_WM8750
- help
- Say Y if you want to add support for SoC audio on Zipit Z2.
-
-config SND_PXA2XX_SOC_POODLE
- tristate "SoC Audio support for Poodle"
- depends on SND_PXA2XX_SOC && MACH_POODLE && I2C
- select SND_PXA2XX_SOC_I2S
- select SND_SOC_WM8731_I2C
- help
- Say Y if you want to add support for SoC audio on Sharp
- Zaurus SL-5600 model (Poodle).
-
-config SND_PXA2XX_SOC_TOSA
- tristate "SoC AC97 Audio support for Tosa"
- depends on SND_PXA2XX_SOC && MACH_TOSA
- depends on MFD_TC6393XB
- depends on AC97_BUS=n
- select REGMAP
- select AC97_BUS_NEW
- select AC97_BUS_COMPAT
- select SND_PXA2XX_SOC_AC97
- select SND_SOC_WM9712
- help
- Say Y if you want to add support for SoC audio on Sharp
- Zaurus SL-C6000x models (Tosa).
-
-config SND_PXA2XX_SOC_E740
- tristate "SoC AC97 Audio support for e740"
- depends on SND_PXA2XX_SOC && MACH_E740
- depends on AC97_BUS=n
- select REGMAP
- select AC97_BUS_NEW
- select AC97_BUS_COMPAT
- select SND_SOC_WM9705
- select SND_PXA2XX_SOC_AC97
- help
- Say Y if you want to add support for SoC audio on the
- toshiba e740 PDA
-
-config SND_PXA2XX_SOC_E750
- tristate "SoC AC97 Audio support for e750"
- depends on SND_PXA2XX_SOC && MACH_E750
- depends on AC97_BUS=n
- select REGMAP
- select SND_SOC_WM9705
- select SND_PXA2XX_SOC_AC97
- help
- Say Y if you want to add support for SoC audio on the
- toshiba e750 PDA
-
-config SND_PXA2XX_SOC_E800
- tristate "SoC AC97 Audio support for e800"
- depends on SND_PXA2XX_SOC && MACH_E800
- depends on AC97_BUS=n
- select REGMAP
- select SND_SOC_WM9712
- select AC97_BUS_NEW
- select AC97_BUS_COMPAT
- select SND_PXA2XX_SOC_AC97
- help
- Say Y if you want to add support for SoC audio on the
- Toshiba e800 PDA
-
-config SND_PXA2XX_SOC_EM_X270
- tristate "SoC Audio support for CompuLab CM-X300"
- depends on SND_PXA2XX_SOC && MACH_CM_X300
- depends on AC97_BUS=n
- select REGMAP
- select AC97_BUS_NEW
- select AC97_BUS_COMPAT
- select SND_PXA2XX_SOC_AC97
- select SND_SOC_WM9712
- help
- Say Y if you want to add support for SoC audio on
- CompuLab EM-x270, eXeda and CM-X300 machines.
-
-config SND_PXA2XX_SOC_PALM27X
- bool "SoC Audio support for Palm T|X, T5, E2 and LifeDrive"
- depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || \
- MACH_PALMT5 || MACH_PALMTE2)
- depends on AC97_BUS=n
- select REGMAP
- select AC97_BUS_NEW
- select AC97_BUS_COMPAT
- select SND_PXA2XX_SOC_AC97
- select SND_SOC_WM9712
- help
- Say Y if you want to add support for SoC audio on
- Palm T|X, T5, E2 or LifeDrive handheld computer.
-
config SND_PXA910_SOC
tristate "SoC Audio for Marvell PXA910 chip"
depends on ARCH_MMP && SND
@@ -161,71 +53,3 @@ config SND_PXA910_SOC
help
Say Y if you want to add support for SoC audio on the
Marvell PXA910 reference platform.
-
-config SND_SOC_TTC_DKB
- tristate "SoC Audio support for TTC DKB"
- depends on SND_PXA910_SOC && MACH_TTC_DKB && I2C=y
- select PXA_SSP
- select SND_PXA_SOC_SSP
- select SND_MMP_SOC
- select MFD_88PM860X
- select SND_SOC_88PM860X
- help
- Say Y if you want to add support for SoC audio on TTC DKB
-
-
-config SND_SOC_ZYLONITE
- tristate "SoC Audio support for Marvell Zylonite"
- depends on SND_PXA2XX_SOC && MACH_ZYLONITE
- depends on AC97_BUS=n
- select AC97_BUS_NEW
- select AC97_BUS_COMPAT
- select SND_PXA2XX_SOC_AC97
- select REGMAP
- select SND_PXA_SOC_SSP
- select SND_SOC_WM9713
- help
- Say Y if you want to add support for SoC audio on the
- Marvell Zylonite reference platform.
-
-config SND_PXA2XX_SOC_HX4700
- tristate "SoC Audio support for HP iPAQ hx4700"
- depends on SND_PXA2XX_SOC && MACH_H4700 && I2C
- select SND_PXA2XX_SOC_I2S
- select SND_SOC_AK4641
- help
- Say Y if you want to add support for SoC audio on the
- HP iPAQ hx4700.
-
-config SND_PXA2XX_SOC_MAGICIAN
- tristate "SoC Audio support for HTC Magician"
- depends on SND_PXA2XX_SOC && MACH_MAGICIAN && I2C
- select SND_PXA2XX_SOC_I2S
- select SND_PXA_SOC_SSP
- select SND_SOC_UDA1380
- help
- Say Y if you want to add support for SoC audio on the
- HTC Magician.
-
-config SND_PXA2XX_SOC_MIOA701
- tristate "SoC Audio support for MIO A701"
- depends on SND_PXA2XX_SOC && MACH_MIOA701
- depends on AC97_BUS=n
- select REGMAP
- select AC97_BUS_NEW
- select AC97_BUS_COMPAT
- select SND_PXA2XX_SOC_AC97
- select SND_SOC_WM9713
- help
- Say Y if you want to add support for SoC audio on the
- MIO A701.
-
-config SND_MMP_SOC_BROWNSTONE
- tristate "SoC Audio support for Marvell Brownstone"
- depends on SND_MMP_SOC_SSPA && MACH_BROWNSTONE && I2C
- select SND_MMP_SOC
- select MFD_WM8994
- select SND_SOC_WM8994
- help
- Say Y if you want to add support for SoC audio on the
- Marvell Brownstone reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index b712eb894a61..406605fc7414 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -4,47 +4,14 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o
snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
snd-soc-pxa-ssp-objs := pxa-ssp.o
-snd-soc-mmp-objs := mmp-pcm.o
snd-soc-mmp-sspa-objs := mmp-sspa.o
obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
-obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
# PXA Machine Support
-snd-soc-corgi-objs := corgi.o
-snd-soc-poodle-objs := poodle.o
-snd-soc-tosa-objs := tosa.o
-snd-soc-e740-objs := e740_wm9705.o
-snd-soc-e750-objs := e750_wm9705.o
-snd-soc-e800-objs := e800_wm9712.o
snd-soc-spitz-objs := spitz.o
-snd-soc-em-x270-objs := em-x270.o
-snd-soc-palm27x-objs := palm27x.o
-snd-soc-zylonite-objs := zylonite.o
-snd-soc-hx4700-objs := hx4700.o
-snd-soc-magician-objs := magician.o
-snd-soc-mioa701-objs := mioa701_wm9713.o
-snd-soc-z2-objs := z2.o
-snd-soc-brownstone-objs := brownstone.o
-snd-soc-ttc-dkb-objs := ttc-dkb.o
-
-obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
-obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
-obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
-obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o
-obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o
-obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
-obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
-obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
-obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o
-obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
-obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
-obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o
-obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
-obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
-obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
deleted file mode 100644
index f310a8e91bbf..000000000000
--- a/sound/soc/pxa/brownstone.c
+++ /dev/null
@@ -1,133 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * linux/sound/soc/pxa/brownstone.c
- *
- * Copyright (C) 2011 Marvell International Ltd.
- */
-
-#include <linux/module.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include "../codecs/wm8994.h"
-#include "mmp-sspa.h"
-
-static const struct snd_kcontrol_new brownstone_dapm_control[] = {
- SOC_DAPM_PIN_SWITCH("Ext Spk"),
-};
-
-static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Ext Spk", NULL),
- SND_SOC_DAPM_HP("Headset Stereophone", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Main Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route brownstone_audio_map[] = {
- {"Ext Spk", NULL, "SPKOUTLP"},
- {"Ext Spk", NULL, "SPKOUTLN"},
- {"Ext Spk", NULL, "SPKOUTRP"},
- {"Ext Spk", NULL, "SPKOUTRN"},
-
- {"Headset Stereophone", NULL, "HPOUT1L"},
- {"Headset Stereophone", NULL, "HPOUT1R"},
-
- {"IN1RN", NULL, "Headset Mic"},
-
- {"DMIC1DAT", NULL, "MICBIAS1"},
- {"MICBIAS1", NULL, "Main Mic"},
-};
-
-static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- int freq_out, sspa_mclk, sysclk;
-
- if (params_rate(params) > 11025) {
- freq_out = params_rate(params) * 512;
- sysclk = params_rate(params) * 256;
- sspa_mclk = params_rate(params) * 64;
- } else {
- freq_out = params_rate(params) * 1024;
- sysclk = params_rate(params) * 512;
- sspa_mclk = params_rate(params) * 64;
- }
-
- snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
- snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
- snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
-
- /* set wm8994 sysclk */
- snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
-
- return 0;
-}
-
-/* machine stream operations */
-static const struct snd_soc_ops brownstone_ops = {
- .hw_params = brownstone_wm8994_hw_params,
-};
-
-SND_SOC_DAILINK_DEFS(wm8994,
- DAILINK_COMP_ARRAY(COMP_CPU("mmp-sspa-dai.0")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm8994-codec", "wm8994-aif1")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("mmp-pcm-audio")));
-
-static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
-{
- .name = "WM8994",
- .stream_name = "WM8994 HiFi",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &brownstone_ops,
- SND_SOC_DAILINK_REG(wm8994),
-},
-};
-
-/* audio machine driver */
-static struct snd_soc_card brownstone = {
- .name = "brownstone",
- .owner = THIS_MODULE,
- .dai_link = brownstone_wm8994_dai,
- .num_links = ARRAY_SIZE(brownstone_wm8994_dai),
-
- .controls = brownstone_dapm_control,
- .num_controls = ARRAY_SIZE(brownstone_dapm_control),
- .dapm_widgets = brownstone_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
- .dapm_routes = brownstone_audio_map,
- .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
- .fully_routed = true,
-};
-
-static int brownstone_probe(struct platform_device *pdev)
-{
- int ret;
-
- brownstone.dev = &pdev->dev;
- ret = devm_snd_soc_register_card(&pdev->dev, &brownstone);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static struct platform_driver mmp_driver = {
- .driver = {
- .name = "brownstone-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = brownstone_probe,
-};
-
-module_platform_driver(mmp_driver);
-
-MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
-MODULE_DESCRIPTION("ALSA SoC Brownstone");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:brownstone-audio");
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
deleted file mode 100644
index 4489d2c8b124..000000000000
--- a/sound/soc/pxa/corgi.c
+++ /dev/null
@@ -1,332 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * corgi.c -- SoC audio for Corgi
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
- * Richard Purdie <richard@openedhand.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/i2c.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/gpio.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-pxa.h>
-
-#include "../codecs/wm8731.h"
-#include "pxa2xx-i2s.h"
-
-#define CORGI_HP 0
-#define CORGI_MIC 1
-#define CORGI_LINE 2
-#define CORGI_HEADSET 3
-#define CORGI_HP_OFF 4
-#define CORGI_SPK_ON 0
-#define CORGI_SPK_OFF 1
-
- /* audio clock in Hz - rounded from 12.235MHz */
-#define CORGI_AUDIO_CLOCK 12288000
-
-static int corgi_jack_func;
-static int corgi_spk_func;
-
-static struct gpio_desc *gpiod_mute_l, *gpiod_mute_r,
- *gpiod_apm_on, *gpiod_mic_bias;
-
-static void corgi_ext_control(struct snd_soc_dapm_context *dapm)
-{
- snd_soc_dapm_mutex_lock(dapm);
-
- /* set up jack connection */
- switch (corgi_jack_func) {
- case CORGI_HP:
- /* set = unmute headphone */
- gpiod_set_value(gpiod_mute_l, 1);
- gpiod_set_value(gpiod_mute_r, 1);
- snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
- snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
- break;
- case CORGI_MIC:
- /* reset = mute headphone */
- gpiod_set_value(gpiod_mute_l, 0);
- gpiod_set_value(gpiod_mute_r, 0);
- snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
- break;
- case CORGI_LINE:
- gpiod_set_value(gpiod_mute_l, 0);
- gpiod_set_value(gpiod_mute_r, 0);
- snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack");
- snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
- break;
- case CORGI_HEADSET:
- gpiod_set_value(gpiod_mute_l, 0);
- gpiod_set_value(gpiod_mute_r, 1);
- snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
- break;
- }
-
- if (corgi_spk_func == CORGI_SPK_ON)
- snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
- else
- snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
-
- /* signal a DAPM event */
- snd_soc_dapm_sync_unlocked(dapm);
-
- snd_soc_dapm_mutex_unlock(dapm);
-}
-
-static int corgi_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
-
- /* check the jack status at stream startup */
- corgi_ext_control(&rtd->card->dapm);
-
- return 0;
-}
-
-/* we need to unmute the HP at shutdown as the mute burns power on corgi */
-static void corgi_shutdown(struct snd_pcm_substream *substream)
-{
- /* set = unmute headphone */
- gpiod_set_value(gpiod_mute_l, 1);
- gpiod_set_value(gpiod_mute_r, 1);
-}
-
-static int corgi_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- unsigned int clk = 0;
- int ret = 0;
-
- switch (params_rate(params)) {
- case 8000:
- case 16000:
- case 48000:
- case 96000:
- clk = 12288000;
- break;
- case 11025:
- case 22050:
- case 44100:
- clk = 11289600;
- break;
- }
-
- /* set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- /* set the I2S system clock as input (unused) */
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static const struct snd_soc_ops corgi_ops = {
- .startup = corgi_startup,
- .hw_params = corgi_hw_params,
- .shutdown = corgi_shutdown,
-};
-
-static int corgi_get_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = corgi_jack_func;
- return 0;
-}
-
-static int corgi_set_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (corgi_jack_func == ucontrol->value.enumerated.item[0])
- return 0;
-
- corgi_jack_func = ucontrol->value.enumerated.item[0];
- corgi_ext_control(&card->dapm);
- return 1;
-}
-
-static int corgi_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = corgi_spk_func;
- return 0;
-}
-
-static int corgi_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (corgi_spk_func == ucontrol->value.enumerated.item[0])
- return 0;
-
- corgi_spk_func = ucontrol->value.enumerated.item[0];
- corgi_ext_control(&card->dapm);
- return 1;
-}
-
-static int corgi_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(gpiod_apm_on, SND_SOC_DAPM_EVENT_ON(event));
- return 0;
-}
-
-static int corgi_mic_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(gpiod_mic_bias, SND_SOC_DAPM_EVENT_ON(event));
- return 0;
-}
-
-/* corgi machine dapm widgets */
-static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
-SND_SOC_DAPM_HP("Headphone Jack", NULL),
-SND_SOC_DAPM_MIC("Mic Jack", corgi_mic_event),
-SND_SOC_DAPM_SPK("Ext Spk", corgi_amp_event),
-SND_SOC_DAPM_LINE("Line Jack", NULL),
-SND_SOC_DAPM_HP("Headset Jack", NULL),
-};
-
-/* Corgi machine audio map (connections to the codec pins) */
-static const struct snd_soc_dapm_route corgi_audio_map[] = {
-
- /* headset Jack - in = micin, out = LHPOUT*/
- {"Headset Jack", NULL, "LHPOUT"},
-
- /* headphone connected to LHPOUT1, RHPOUT1 */
- {"Headphone Jack", NULL, "LHPOUT"},
- {"Headphone Jack", NULL, "RHPOUT"},
-
- /* speaker connected to LOUT, ROUT */
- {"Ext Spk", NULL, "ROUT"},
- {"Ext Spk", NULL, "LOUT"},
-
- /* mic is connected to MICIN (via right channel of headphone jack) */
- {"MICIN", NULL, "Mic Jack"},
-
- /* Same as the above but no mic bias for line signals */
- {"MICIN", NULL, "Line Jack"},
-};
-
-static const char * const jack_function[] = {"Headphone", "Mic", "Line",
- "Headset", "Off"};
-static const char * const spk_function[] = {"On", "Off"};
-static const struct soc_enum corgi_enum[] = {
- SOC_ENUM_SINGLE_EXT(5, jack_function),
- SOC_ENUM_SINGLE_EXT(2, spk_function),
-};
-
-static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
- SOC_ENUM_EXT("Jack Function", corgi_enum[0], corgi_get_jack,
- corgi_set_jack),
- SOC_ENUM_EXT("Speaker Function", corgi_enum[1], corgi_get_spk,
- corgi_set_spk),
-};
-
-/* corgi digital audio interface glue - connects codec <--> CPU */
-SND_SOC_DAILINK_DEFS(wm8731,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link corgi_dai = {
- .name = "WM8731",
- .stream_name = "WM8731",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &corgi_ops,
- SND_SOC_DAILINK_REG(wm8731),
-};
-
-/* corgi audio machine driver */
-static struct snd_soc_card corgi = {
- .name = "Corgi",
- .owner = THIS_MODULE,
- .dai_link = &corgi_dai,
- .num_links = 1,
-
- .controls = wm8731_corgi_controls,
- .num_controls = ARRAY_SIZE(wm8731_corgi_controls),
- .dapm_widgets = wm8731_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
- .dapm_routes = corgi_audio_map,
- .num_dapm_routes = ARRAY_SIZE(corgi_audio_map),
- .fully_routed = true,
-};
-
-static int corgi_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &corgi;
- int ret;
-
- card->dev = &pdev->dev;
-
- gpiod_mute_l = devm_gpiod_get(&pdev->dev, "mute-l", GPIOD_OUT_HIGH);
- if (IS_ERR(gpiod_mute_l))
- return PTR_ERR(gpiod_mute_l);
- gpiod_mute_r = devm_gpiod_get(&pdev->dev, "mute-r", GPIOD_OUT_HIGH);
- if (IS_ERR(gpiod_mute_r))
- return PTR_ERR(gpiod_mute_r);
- gpiod_apm_on = devm_gpiod_get(&pdev->dev, "apm-on", GPIOD_OUT_LOW);
- if (IS_ERR(gpiod_apm_on))
- return PTR_ERR(gpiod_apm_on);
- gpiod_mic_bias = devm_gpiod_get(&pdev->dev, "mic-bias", GPIOD_OUT_LOW);
- if (IS_ERR(gpiod_mic_bias))
- return PTR_ERR(gpiod_mic_bias);
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static struct platform_driver corgi_driver = {
- .driver = {
- .name = "corgi-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = corgi_probe,
-};
-
-module_platform_driver(corgi_driver);
-
-/* Module information */
-MODULE_AUTHOR("Richard Purdie");
-MODULE_DESCRIPTION("ALSA SoC Corgi");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:corgi-audio");
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
deleted file mode 100644
index 4e0e9b778d4c..000000000000
--- a/sound/soc/pxa/e740_wm9705.c
+++ /dev/null
@@ -1,168 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * e740-wm9705.c -- SoC audio for e740
- *
- * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/gpio/consumer.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <linux/platform_data/asoc-pxa.h>
-
-#include <asm/mach-types.h>
-
-static struct gpio_desc *gpiod_output_amp, *gpiod_input_amp;
-static struct gpio_desc *gpiod_audio_power;
-
-#define E740_AUDIO_OUT 1
-#define E740_AUDIO_IN 2
-
-static int e740_audio_power;
-
-static void e740_sync_audio_power(int status)
-{
- gpiod_set_value(gpiod_audio_power, !status);
- gpiod_set_value(gpiod_output_amp, (status & E740_AUDIO_OUT) ? 1 : 0);
- gpiod_set_value(gpiod_input_amp, (status & E740_AUDIO_IN) ? 1 : 0);
-}
-
-static int e740_mic_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- if (event & SND_SOC_DAPM_PRE_PMU)
- e740_audio_power |= E740_AUDIO_IN;
- else if (event & SND_SOC_DAPM_POST_PMD)
- e740_audio_power &= ~E740_AUDIO_IN;
-
- e740_sync_audio_power(e740_audio_power);
-
- return 0;
-}
-
-static int e740_output_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- if (event & SND_SOC_DAPM_PRE_PMU)
- e740_audio_power |= E740_AUDIO_OUT;
- else if (event & SND_SOC_DAPM_POST_PMD)
- e740_audio_power &= ~E740_AUDIO_OUT;
-
- e740_sync_audio_power(e740_audio_power);
-
- return 0;
-}
-
-static const struct snd_soc_dapm_widget e740_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
- SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
- SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
- e740_output_amp_event, SND_SOC_DAPM_PRE_PMU |
- SND_SOC_DAPM_POST_PMD),
- SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
- e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU |
- SND_SOC_DAPM_POST_PMD),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- {"Output Amp", NULL, "LOUT"},
- {"Output Amp", NULL, "ROUT"},
- {"Output Amp", NULL, "MONOOUT"},
-
- {"Speaker", NULL, "Output Amp"},
- {"Headphone Jack", NULL, "Output Amp"},
-
- {"MIC1", NULL, "Mic Amp"},
- {"Mic Amp", NULL, "Mic (Internal)"},
-};
-
-SND_SOC_DAILINK_DEFS(ac97,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(ac97_aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link e740_dai[] = {
- {
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- SND_SOC_DAILINK_REG(ac97),
- },
- {
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- SND_SOC_DAILINK_REG(ac97_aux),
- },
-};
-
-static struct snd_soc_card e740 = {
- .name = "Toshiba e740",
- .owner = THIS_MODULE,
- .dai_link = e740_dai,
- .num_links = ARRAY_SIZE(e740_dai),
-
- .dapm_widgets = e740_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
- .fully_routed = true,
-};
-
-static int e740_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &e740;
- int ret;
-
- gpiod_input_amp = devm_gpiod_get(&pdev->dev, "Mic amp", GPIOD_OUT_LOW);
- ret = PTR_ERR_OR_ZERO(gpiod_input_amp);
- if (ret)
- return ret;
- gpiod_output_amp = devm_gpiod_get(&pdev->dev, "Output amp", GPIOD_OUT_LOW);
- ret = PTR_ERR_OR_ZERO(gpiod_output_amp);
- if (ret)
- return ret;
- gpiod_audio_power = devm_gpiod_get(&pdev->dev, "Audio power", GPIOD_OUT_HIGH);
- ret = PTR_ERR_OR_ZERO(gpiod_audio_power);
- if (ret)
- return ret;
-
- card->dev = &pdev->dev;
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static int e740_remove(struct platform_device *pdev)
-{
- return 0;
-}
-
-static struct platform_driver e740_driver = {
- .driver = {
- .name = "e740-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = e740_probe,
- .remove = e740_remove,
-};
-
-module_platform_driver(e740_driver);
-
-/* Module information */
-MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
-MODULE_DESCRIPTION("ALSA SoC driver for e740");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:e740-audio");
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
deleted file mode 100644
index 7a1e0d8bfd11..000000000000
--- a/sound/soc/pxa/e750_wm9705.c
+++ /dev/null
@@ -1,147 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * e750-wm9705.c -- SoC audio for e750
- *
- * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/gpio/consumer.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <linux/platform_data/asoc-pxa.h>
-
-#include <asm/mach-types.h>
-
-static struct gpio_desc *gpiod_spk_amp, *gpiod_hp_amp;
-
-static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- if (event & SND_SOC_DAPM_PRE_PMU)
- gpiod_set_value(gpiod_spk_amp, 1);
- else if (event & SND_SOC_DAPM_POST_PMD)
- gpiod_set_value(gpiod_spk_amp, 0);
-
- return 0;
-}
-
-static int e750_hp_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- if (event & SND_SOC_DAPM_PRE_PMU)
- gpiod_set_value(gpiod_hp_amp, 1);
- else if (event & SND_SOC_DAPM_POST_PMD)
- gpiod_set_value(gpiod_hp_amp, 0);
-
- return 0;
-}
-
-static const struct snd_soc_dapm_widget e750_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
- SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
- SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
- e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
- SND_SOC_DAPM_POST_PMD),
- SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
- e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
- SND_SOC_DAPM_POST_PMD),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- {"Headphone Amp", NULL, "HPOUTL"},
- {"Headphone Amp", NULL, "HPOUTR"},
- {"Headphone Jack", NULL, "Headphone Amp"},
-
- {"Speaker Amp", NULL, "MONOOUT"},
- {"Speaker", NULL, "Speaker Amp"},
-
- {"MIC1", NULL, "Mic (Internal)"},
-};
-
-SND_SOC_DAILINK_DEFS(ac97,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(ac97_aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9705-codec", "wm9705-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link e750_dai[] = {
- {
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- SND_SOC_DAILINK_REG(ac97),
- /* use ops to check startup state */
- },
- {
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- SND_SOC_DAILINK_REG(ac97_aux),
- },
-};
-
-static struct snd_soc_card e750 = {
- .name = "Toshiba e750",
- .owner = THIS_MODULE,
- .dai_link = e750_dai,
- .num_links = ARRAY_SIZE(e750_dai),
-
- .dapm_widgets = e750_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
- .fully_routed = true,
-};
-
-static int e750_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &e750;
- int ret;
-
- gpiod_hp_amp = devm_gpiod_get(&pdev->dev, "Headphone amp", GPIOD_OUT_LOW);
- ret = PTR_ERR_OR_ZERO(gpiod_hp_amp);
- if (ret)
- return ret;
- gpiod_spk_amp = devm_gpiod_get(&pdev->dev, "Speaker amp", GPIOD_OUT_LOW);
- ret = PTR_ERR_OR_ZERO(gpiod_spk_amp);
- if (ret)
- return ret;
-
- card->dev = &pdev->dev;
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static int e750_remove(struct platform_device *pdev)
-{
- return 0;
-}
-
-static struct platform_driver e750_driver = {
- .driver = {
- .name = "e750-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = e750_probe,
- .remove = e750_remove,
-};
-
-module_platform_driver(e750_driver);
-
-/* Module information */
-MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
-MODULE_DESCRIPTION("ALSA SoC driver for e750");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:e750-audio");
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
deleted file mode 100644
index a39c494127cf..000000000000
--- a/sound/soc/pxa/e800_wm9712.c
+++ /dev/null
@@ -1,147 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * e800-wm9712.c -- SoC audio for e800
- *
- * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/gpio/consumer.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-pxa.h>
-
-static struct gpio_desc *gpiod_spk_amp, *gpiod_hp_amp;
-
-static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- if (event & SND_SOC_DAPM_PRE_PMU)
- gpiod_set_value(gpiod_spk_amp, 1);
- else if (event & SND_SOC_DAPM_POST_PMD)
- gpiod_set_value(gpiod_spk_amp, 0);
-
- return 0;
-}
-
-static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
-{
- if (event & SND_SOC_DAPM_PRE_PMU)
- gpiod_set_value(gpiod_hp_amp, 1);
- else if (event & SND_SOC_DAPM_POST_PMD)
- gpiod_set_value(gpiod_hp_amp, 0);
-
- return 0;
-}
-
-static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
- SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
- SND_SOC_DAPM_SPK("Speaker", NULL),
- SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
- e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
- SND_SOC_DAPM_POST_PMD),
- SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
- e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
- SND_SOC_DAPM_POST_PMD),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- {"Headphone Jack", NULL, "HPOUTL"},
- {"Headphone Jack", NULL, "HPOUTR"},
- {"Headphone Jack", NULL, "Headphone Amp"},
-
- {"Speaker Amp", NULL, "MONOOUT"},
- {"Speaker", NULL, "Speaker Amp"},
-
- {"MIC1", NULL, "Mic (Internal1)"},
- {"MIC2", NULL, "Mic (Internal2)"},
-};
-
-
-SND_SOC_DAILINK_DEFS(ac97,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(ac97_aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link e800_dai[] = {
- {
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- SND_SOC_DAILINK_REG(ac97),
- },
- {
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- SND_SOC_DAILINK_REG(ac97_aux),
- },
-};
-
-static struct snd_soc_card e800 = {
- .name = "Toshiba e800",
- .owner = THIS_MODULE,
- .dai_link = e800_dai,
- .num_links = ARRAY_SIZE(e800_dai),
-
- .dapm_widgets = e800_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static int e800_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &e800;
- int ret;
-
- gpiod_hp_amp = devm_gpiod_get(&pdev->dev, "Headphone amp", GPIOD_OUT_LOW);
- ret = PTR_ERR_OR_ZERO(gpiod_hp_amp);
- if (ret)
- return ret;
- gpiod_spk_amp = devm_gpiod_get(&pdev->dev, "Speaker amp", GPIOD_OUT_LOW);
- ret = PTR_ERR_OR_ZERO(gpiod_spk_amp);
- if (ret)
- return ret;
-
- card->dev = &pdev->dev;
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static int e800_remove(struct platform_device *pdev)
-{
- return 0;
-}
-
-static struct platform_driver e800_driver = {
- .driver = {
- .name = "e800-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = e800_probe,
- .remove = e800_remove,
-};
-
-module_platform_driver(e800_driver);
-
-/* Module information */
-MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
-MODULE_DESCRIPTION("ALSA SoC driver for e800");
-MODULE_LICENSE("GPL v2");
-MODULE_ALIAS("platform:e800-audio");
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
deleted file mode 100644
index b59ec22e1e7e..000000000000
--- a/sound/soc/pxa/em-x270.c
+++ /dev/null
@@ -1,92 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * SoC audio driver for EM-X270, eXeda and CM-X300
- *
- * Copyright 2007, 2009 CompuLab, Ltd.
- *
- * Author: Mike Rapoport <mike@compulab.co.il>
- *
- * Copied from tosa.c:
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
- * Richard Purdie <richard@openedhand.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-pxa.h>
-
-SND_SOC_DAILINK_DEFS(ac97,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(ac97_aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link em_x270_dai[] = {
- {
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- SND_SOC_DAILINK_REG(ac97),
- },
- {
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- SND_SOC_DAILINK_REG(ac97_aux),
- },
-};
-
-static struct snd_soc_card em_x270 = {
- .name = "EM-X270",
- .owner = THIS_MODULE,
- .dai_link = em_x270_dai,
- .num_links = ARRAY_SIZE(em_x270_dai),
-};
-
-static struct platform_device *em_x270_snd_device;
-
-static int __init em_x270_init(void)
-{
- int ret;
-
- if (!(machine_is_em_x270() || machine_is_exeda()
- || machine_is_cm_x300()))
- return -ENODEV;
-
- em_x270_snd_device = platform_device_alloc("soc-audio", -1);
- if (!em_x270_snd_device)
- return -ENOMEM;
-
- platform_set_drvdata(em_x270_snd_device, &em_x270);
- ret = platform_device_add(em_x270_snd_device);
-
- if (ret)
- platform_device_put(em_x270_snd_device);
-
- return ret;
-}
-
-static void __exit em_x270_exit(void)
-{
- platform_device_unregister(em_x270_snd_device);
-}
-
-module_init(em_x270_init);
-module_exit(em_x270_exit);
-
-/* Module information */
-MODULE_AUTHOR("Mike Rapoport");
-MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c
deleted file mode 100644
index a323ddb8fc3e..000000000000
--- a/sound/soc/pxa/hx4700.c
+++ /dev/null
@@ -1,207 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * SoC audio for HP iPAQ hx4700
- *
- * Copyright (c) 2009 Philipp Zabel
- */
-
-#include <linux/module.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/delay.h>
-#include <linux/gpio/consumer.h>
-
-#include <sound/core.h>
-#include <sound/jack.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include "pxa2xx-i2s.h"
-
-static struct gpio_desc *gpiod_hp_driver, *gpiod_spk_sd;
-static struct snd_soc_jack hs_jack;
-
-/* Headphones jack detection DAPM pin */
-static struct snd_soc_jack_pin hs_jack_pin[] = {
- {
- .pin = "Headphone Jack",
- .mask = SND_JACK_HEADPHONE,
- .invert = 1,
- },
- {
- .pin = "Speaker",
- /* disable speaker when hp jack is inserted */
- .mask = SND_JACK_HEADPHONE,
- },
-};
-
-/* Headphones jack detection GPIO */
-static struct snd_soc_jack_gpio hs_jack_gpio = {
- .name = "earphone-det",
- .report = SND_JACK_HEADPHONE,
- .debounce_time = 200,
-};
-
-/*
- * iPAQ hx4700 uses I2S for capture and playback.
- */
-static int hx4700_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- int ret = 0;
-
- /* set the I2S system clock as output */
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
- SND_SOC_CLOCK_OUT);
- if (ret < 0)
- return ret;
-
- /* inform codec driver about clock freq *
- * (PXA I2S always uses divider 256) */
- ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params),
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static const struct snd_soc_ops hx4700_ops = {
- .hw_params = hx4700_hw_params,
-};
-
-static int hx4700_spk_power(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(gpiod_spk_sd, !SND_SOC_DAPM_EVENT_ON(event));
- return 0;
-}
-
-static int hx4700_hp_power(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(gpiod_hp_driver, !!SND_SOC_DAPM_EVENT_ON(event));
- return 0;
-}
-
-/* hx4700 machine dapm widgets */
-static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power),
- SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power),
- SND_SOC_DAPM_MIC("Built-in Microphone", NULL),
-};
-
-/* hx4700 machine audio_map */
-static const struct snd_soc_dapm_route hx4700_audio_map[] = {
-
- /* Headphone connected to LOUT, ROUT */
- {"Headphone Jack", NULL, "LOUT"},
- {"Headphone Jack", NULL, "ROUT"},
-
- /* Speaker connected to MOUT2 */
- {"Speaker", NULL, "MOUT2"},
-
- /* Microphone connected to MICIN */
- {"MICIN", NULL, "Built-in Microphone"},
- {"AIN", NULL, "MICOUT"},
-};
-
-/*
- * Logic for a ak4641 as connected on a HP iPAQ hx4700
- */
-static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
-{
- int err;
-
- /* Jack detection API stuff */
- err = snd_soc_card_jack_new_pins(rtd->card, "Headphone Jack",
- SND_JACK_HEADPHONE, &hs_jack,
- hs_jack_pin, ARRAY_SIZE(hs_jack_pin));
- if (err)
- return err;
-
- err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio);
-
- return err;
-}
-
-/* hx4700 digital audio interface glue - connects codec <--> CPU */
-SND_SOC_DAILINK_DEFS(ak4641,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")),
- DAILINK_COMP_ARRAY(COMP_CODEC("ak4641.0-0012", "ak4641-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link hx4700_dai = {
- .name = "ak4641",
- .stream_name = "AK4641",
- .init = hx4700_ak4641_init,
- .dai_fmt = SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &hx4700_ops,
- SND_SOC_DAILINK_REG(ak4641),
-};
-
-/* hx4700 audio machine driver */
-static struct snd_soc_card snd_soc_card_hx4700 = {
- .name = "iPAQ hx4700",
- .owner = THIS_MODULE,
- .dai_link = &hx4700_dai,
- .num_links = 1,
- .dapm_widgets = hx4700_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(hx4700_dapm_widgets),
- .dapm_routes = hx4700_audio_map,
- .num_dapm_routes = ARRAY_SIZE(hx4700_audio_map),
- .fully_routed = true,
-};
-
-static int hx4700_audio_probe(struct platform_device *pdev)
-{
- int ret;
-
- if (!machine_is_h4700())
- return -ENODEV;
-
- gpiod_hp_driver = devm_gpiod_get(&pdev->dev, "hp-driver", GPIOD_ASIS);
- ret = PTR_ERR_OR_ZERO(gpiod_hp_driver);
- if (ret)
- return ret;
- gpiod_spk_sd = devm_gpiod_get(&pdev->dev, "spk-sd", GPIOD_ASIS);
- ret = PTR_ERR_OR_ZERO(gpiod_spk_sd);
- if (ret)
- return ret;
-
- hs_jack_gpio.gpiod_dev = &pdev->dev;
- snd_soc_card_hx4700.dev = &pdev->dev;
- ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700);
-
- return ret;
-}
-
-static int hx4700_audio_remove(struct platform_device *pdev)
-{
- gpiod_set_value(gpiod_hp_driver, 0);
- gpiod_set_value(gpiod_spk_sd, 0);
- return 0;
-}
-
-static struct platform_driver hx4700_audio_driver = {
- .driver = {
- .name = "hx4700-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = hx4700_audio_probe,
- .remove = hx4700_audio_remove,
-};
-
-module_platform_driver(hx4700_audio_driver);
-
-MODULE_AUTHOR("Philipp Zabel");
-MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:hx4700-audio");
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
deleted file mode 100644
index b791a2ba5ce5..000000000000
--- a/sound/soc/pxa/magician.c
+++ /dev/null
@@ -1,366 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * SoC audio for HTC Magician
- *
- * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
- *
- * based on spitz.c,
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
- * Richard Purdie <richard@openedhand.com>
- */
-
-#include <linux/module.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/delay.h>
-#include <linux/gpio/consumer.h>
-#include <linux/i2c.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include "../codecs/uda1380.h"
-#include "pxa2xx-i2s.h"
-#include "pxa-ssp.h"
-
-#define MAGICIAN_MIC 0
-#define MAGICIAN_MIC_EXT 1
-
-static int magician_hp_switch;
-static int magician_spk_switch = 1;
-static int magician_in_sel = MAGICIAN_MIC;
-
-static struct gpio_desc *gpiod_spk_power, *gpiod_ep_power, *gpiod_mic_power;
-static struct gpio_desc *gpiod_in_sel0, *gpiod_in_sel1;
-
-static void magician_ext_control(struct snd_soc_dapm_context *dapm)
-{
-
- snd_soc_dapm_mutex_lock(dapm);
-
- if (magician_spk_switch)
- snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
- else
- snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
- if (magician_hp_switch)
- snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
- else
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
-
- switch (magician_in_sel) {
- case MAGICIAN_MIC:
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
- snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
- break;
- case MAGICIAN_MIC_EXT:
- snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
- snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
- break;
- }
-
- snd_soc_dapm_sync_unlocked(dapm);
-
- snd_soc_dapm_mutex_unlock(dapm);
-}
-
-static int magician_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
-
- /* check the jack status at stream startup */
- magician_ext_control(&rtd->card->dapm);
-
- return 0;
-}
-
-/*
- * Magician uses SSP port for playback.
- */
-static int magician_playback_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- unsigned int width;
- int ret = 0;
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
- SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_BC_FC);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
- SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_BP_FP);
- if (ret < 0)
- return ret;
-
- width = snd_pcm_format_physical_width(params_format(params));
- ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
- if (ret < 0)
- return ret;
-
- /* set audio clock as clock source */
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
- SND_SOC_CLOCK_OUT);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-/*
- * Magician uses I2S for capture.
- */
-static int magician_capture_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- int ret = 0;
-
- /* set codec DAI configuration */
- ret = snd_soc_dai_set_fmt(codec_dai,
- SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_BC_FC);
- if (ret < 0)
- return ret;
-
- /* set cpu DAI configuration */
- ret = snd_soc_dai_set_fmt(cpu_dai,
- SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_BP_FP);
- if (ret < 0)
- return ret;
-
- /* set the I2S system clock as output */
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
- SND_SOC_CLOCK_OUT);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static const struct snd_soc_ops magician_capture_ops = {
- .startup = magician_startup,
- .hw_params = magician_capture_hw_params,
-};
-
-static const struct snd_soc_ops magician_playback_ops = {
- .startup = magician_startup,
- .hw_params = magician_playback_hw_params,
-};
-
-static int magician_get_hp(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = magician_hp_switch;
- return 0;
-}
-
-static int magician_set_hp(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (magician_hp_switch == ucontrol->value.integer.value[0])
- return 0;
-
- magician_hp_switch = ucontrol->value.integer.value[0];
- magician_ext_control(&card->dapm);
- return 1;
-}
-
-static int magician_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.integer.value[0] = magician_spk_switch;
- return 0;
-}
-
-static int magician_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (magician_spk_switch == ucontrol->value.integer.value[0])
- return 0;
-
- magician_spk_switch = ucontrol->value.integer.value[0];
- magician_ext_control(&card->dapm);
- return 1;
-}
-
-static int magician_get_input(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = magician_in_sel;
- return 0;
-}
-
-static int magician_set_input(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- if (magician_in_sel == ucontrol->value.enumerated.item[0])
- return 0;
-
- magician_in_sel = ucontrol->value.enumerated.item[0];
-
- switch (magician_in_sel) {
- case MAGICIAN_MIC:
- gpiod_set_value(gpiod_in_sel1, 1);
- break;
- case MAGICIAN_MIC_EXT:
- gpiod_set_value(gpiod_in_sel1, 0);
- }
-
- return 1;
-}
-
-static int magician_spk_power(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(gpiod_spk_power, SND_SOC_DAPM_EVENT_ON(event));
- return 0;
-}
-
-static int magician_hp_power(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(gpiod_ep_power, SND_SOC_DAPM_EVENT_ON(event));
- return 0;
-}
-
-static int magician_mic_bias(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(gpiod_mic_power, SND_SOC_DAPM_EVENT_ON(event));
- return 0;
-}
-
-/* magician machine dapm widgets */
-static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
- SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
- SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
- SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
-};
-
-/* magician machine audio_map */
-static const struct snd_soc_dapm_route audio_map[] = {
-
- /* Headphone connected to VOUTL, VOUTR */
- {"Headphone Jack", NULL, "VOUTL"},
- {"Headphone Jack", NULL, "VOUTR"},
-
- /* Speaker connected to VOUTL, VOUTR */
- {"Speaker", NULL, "VOUTL"},
- {"Speaker", NULL, "VOUTR"},
-
- /* Mics are connected to VINM */
- {"VINM", NULL, "Headset Mic"},
- {"VINM", NULL, "Call Mic"},
-};
-
-static const char * const input_select[] = {"Call Mic", "Headset Mic"};
-static const struct soc_enum magician_in_sel_enum =
- SOC_ENUM_SINGLE_EXT(2, input_select);
-
-static const struct snd_kcontrol_new uda1380_magician_controls[] = {
- SOC_SINGLE_BOOL_EXT("Headphone Switch",
- (unsigned long)&magician_hp_switch,
- magician_get_hp, magician_set_hp),
- SOC_SINGLE_BOOL_EXT("Speaker Switch",
- (unsigned long)&magician_spk_switch,
- magician_get_spk, magician_set_spk),
- SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
- magician_get_input, magician_set_input),
-};
-
-/* magician digital audio interface glue - connects codec <--> CPU */
-SND_SOC_DAILINK_DEFS(playback,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa-ssp-dai.0")),
- DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-0018",
- "uda1380-hifi-playback")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(capture,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")),
- DAILINK_COMP_ARRAY(COMP_CODEC("uda1380-codec.0-0018",
- "uda1380-hifi-capture")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link magician_dai[] = {
-{
- .name = "uda1380",
- .stream_name = "UDA1380 Playback",
- .ops = &magician_playback_ops,
- SND_SOC_DAILINK_REG(playback),
-},
-{
- .name = "uda1380",
- .stream_name = "UDA1380 Capture",
- .ops = &magician_capture_ops,
- SND_SOC_DAILINK_REG(capture),
-}
-};
-
-/* magician audio machine driver */
-static struct snd_soc_card snd_soc_card_magician = {
- .name = "Magician",
- .owner = THIS_MODULE,
- .dai_link = magician_dai,
- .num_links = ARRAY_SIZE(magician_dai),
-
- .controls = uda1380_magician_controls,
- .num_controls = ARRAY_SIZE(uda1380_magician_controls),
- .dapm_widgets = uda1380_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
- .fully_routed = true,
-};
-
-static int magician_audio_probe(struct platform_device *pdev)
-{
- struct device *dev = &pdev->dev;
-
- gpiod_spk_power = devm_gpiod_get(dev, "SPK_POWER", GPIOD_OUT_LOW);
- if (IS_ERR(gpiod_spk_power))
- return PTR_ERR(gpiod_spk_power);
- gpiod_ep_power = devm_gpiod_get(dev, "EP_POWER", GPIOD_OUT_LOW);
- if (IS_ERR(gpiod_ep_power))
- return PTR_ERR(gpiod_ep_power);
- gpiod_mic_power = devm_gpiod_get(dev, "MIC_POWER", GPIOD_OUT_LOW);
- if (IS_ERR(gpiod_mic_power))
- return PTR_ERR(gpiod_mic_power);
- gpiod_in_sel0 = devm_gpiod_get(dev, "IN_SEL0", GPIOD_OUT_HIGH);
- if (IS_ERR(gpiod_in_sel0))
- return PTR_ERR(gpiod_in_sel0);
- gpiod_in_sel1 = devm_gpiod_get(dev, "IN_SEL1", GPIOD_OUT_LOW);
- if (IS_ERR(gpiod_in_sel1))
- return PTR_ERR(gpiod_in_sel1);
-
- snd_soc_card_magician.dev = &pdev->dev;
- return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_magician);
-}
-
-static struct platform_driver magician_audio_driver = {
- .driver.name = "magician-audio",
- .driver.pm = &snd_soc_pm_ops,
- .probe = magician_audio_probe,
-};
-module_platform_driver(magician_audio_driver);
-
-MODULE_AUTHOR("Philipp Zabel");
-MODULE_DESCRIPTION("ALSA SoC Magician");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:magician-audio");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
deleted file mode 100644
index 0fa37637eca9..000000000000
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ /dev/null
@@ -1,201 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * Handles the Mitac mioa701 SoC system
- *
- * Copyright (C) 2008 Robert Jarzmik
- *
- * This is a little schema of the sound interconnections :
- *
- * Sagem X200 Wolfson WM9713
- * +--------+ +-------------------+ Rear Speaker
- * | | | | /-+
- * | +--->----->---+MONOIN SPKL+--->----+-+ |
- * | GSM | | | | | |
- * | +--->----->---+PCBEEP SPKR+--->----+-+ |
- * | CHIP | | | \-+
- * | +---<-----<---+MONO |
- * | | | | Front Speaker
- * +--------+ | | /-+
- * | HPL+--->----+-+ |
- * | | | | |
- * | OUT3+--->----+-+ |
- * | | \-+
- * | |
- * | | Front Micro
- * | | +
- * | MIC1+-----<--+o+
- * | | +
- * +-------------------+ ---
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/platform_device.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-pxa.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/initval.h>
-#include <sound/ac97_codec.h>
-
-#include "../codecs/wm9713.h"
-
-#define AC97_GPIO_PULL 0x58
-
-/* Use GPIO8 for rear speaker amplifier */
-static int rear_amp_power(struct snd_soc_component *component, int power)
-{
- unsigned short reg;
-
- if (power) {
- reg = snd_soc_component_read(component, AC97_GPIO_CFG);
- snd_soc_component_write(component, AC97_GPIO_CFG, reg | 0x0100);
- reg = snd_soc_component_read(component, AC97_GPIO_PULL);
- snd_soc_component_write(component, AC97_GPIO_PULL, reg | (1<<15));
- } else {
- reg = snd_soc_component_read(component, AC97_GPIO_CFG);
- snd_soc_component_write(component, AC97_GPIO_CFG, reg & ~0x0100);
- reg = snd_soc_component_read(component, AC97_GPIO_PULL);
- snd_soc_component_write(component, AC97_GPIO_PULL, reg & ~(1<<15));
- }
-
- return 0;
-}
-
-static int rear_amp_event(struct snd_soc_dapm_widget *widget,
- struct snd_kcontrol *kctl, int event)
-{
- struct snd_soc_card *card = widget->dapm->card;
- struct snd_soc_pcm_runtime *rtd;
- struct snd_soc_component *component;
-
- rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]);
- component = asoc_rtd_to_codec(rtd, 0)->component;
- return rear_amp_power(component, SND_SOC_DAPM_EVENT_ON(event));
-}
-
-/* mioa701 machine dapm widgets */
-static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = {
- SND_SOC_DAPM_SPK("Front Speaker", NULL),
- SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event),
- SND_SOC_DAPM_MIC("Headset", NULL),
- SND_SOC_DAPM_LINE("GSM Line Out", NULL),
- SND_SOC_DAPM_LINE("GSM Line In", NULL),
- SND_SOC_DAPM_MIC("Headset Mic", NULL),
- SND_SOC_DAPM_MIC("Front Mic", NULL),
-};
-
-static const struct snd_soc_dapm_route audio_map[] = {
- /* Call Mic */
- {"Mic Bias", NULL, "Front Mic"},
- {"MIC1", NULL, "Mic Bias"},
-
- /* Headset Mic */
- {"LINEL", NULL, "Headset Mic"},
- {"LINER", NULL, "Headset Mic"},
-
- /* GSM Module */
- {"MONOIN", NULL, "GSM Line Out"},
- {"PCBEEP", NULL, "GSM Line Out"},
- {"GSM Line In", NULL, "MONO"},
-
- /* headphone connected to HPL, HPR */
- {"Headset", NULL, "HPL"},
- {"Headset", NULL, "HPR"},
-
- /* front speaker connected to HPL, OUT3 */
- {"Front Speaker", NULL, "HPL"},
- {"Front Speaker", NULL, "OUT3"},
-
- /* rear speaker connected to SPKL, SPKR */
- {"Rear Speaker", NULL, "SPKL"},
- {"Rear Speaker", NULL, "SPKR"},
-};
-
-static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
-
- /* Prepare GPIO8 for rear speaker amplifier */
- snd_soc_component_update_bits(component, AC97_GPIO_CFG, 0x100, 0x100);
-
- /* Prepare MIC input */
- snd_soc_component_update_bits(component, AC97_3D_CONTROL, 0xc000, 0xc000);
-
- return 0;
-}
-
-static struct snd_soc_ops mioa701_ops;
-
-SND_SOC_DAILINK_DEFS(ac97,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(ac97_aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link mioa701_dai[] = {
- {
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- .init = mioa701_wm9713_init,
- .ops = &mioa701_ops,
- SND_SOC_DAILINK_REG(ac97),
- },
- {
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- .ops = &mioa701_ops,
- SND_SOC_DAILINK_REG(ac97_aux),
- },
-};
-
-static struct snd_soc_card mioa701 = {
- .name = "MioA701",
- .owner = THIS_MODULE,
- .dai_link = mioa701_dai,
- .num_links = ARRAY_SIZE(mioa701_dai),
-
- .dapm_widgets = mioa701_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static int mioa701_wm9713_probe(struct platform_device *pdev)
-{
- int rc;
-
- if (!machine_is_mioa701())
- return -ENODEV;
-
- mioa701.dev = &pdev->dev;
- rc = devm_snd_soc_register_card(&pdev->dev, &mioa701);
- if (!rc)
- dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will "
- "lead to overheating and possible destruction of your device."
- " Do not use without a good knowledge of mio's board design!\n");
- return rc;
-}
-
-static struct platform_driver mioa701_wm9713_driver = {
- .probe = mioa701_wm9713_probe,
- .driver = {
- .name = "mioa701-wm9713",
- .pm = &snd_soc_pm_ops,
- },
-};
-
-module_platform_driver(mioa701_wm9713_driver);
-
-/* Module information */
-MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
-MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:mioa701-wm9713");
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
deleted file mode 100644
index 99b245e3079a..000000000000
--- a/sound/soc/pxa/mmp-pcm.c
+++ /dev/null
@@ -1,267 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * linux/sound/soc/pxa/mmp-pcm.c
- *
- * Copyright (C) 2011 Marvell International Ltd.
- */
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/platform_device.h>
-#include <linux/slab.h>
-#include <linux/dma-mapping.h>
-#include <linux/dmaengine.h>
-#include <linux/platform_data/dma-mmp_tdma.h>
-#include <linux/platform_data/mmp_audio.h>
-
-#include <sound/pxa2xx-lib.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-#include <sound/dmaengine_pcm.h>
-
-#define DRV_NAME "mmp-pcm"
-
-struct mmp_dma_data {
- int ssp_id;
- struct resource *dma_res;
-};
-
-#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \
- SNDRV_PCM_INFO_MMAP_VALID | \
- SNDRV_PCM_INFO_INTERLEAVED | \
- SNDRV_PCM_INFO_PAUSE | \
- SNDRV_PCM_INFO_RESUME | \
- SNDRV_PCM_INFO_NO_PERIOD_WAKEUP)
-
-static struct snd_pcm_hardware mmp_pcm_hardware[] = {
- {
- .info = MMP_PCM_INFO,
- .period_bytes_min = 1024,
- .period_bytes_max = 2048,
- .periods_min = 2,
- .periods_max = 32,
- .buffer_bytes_max = 4096,
- .fifo_size = 32,
- },
- {
- .info = MMP_PCM_INFO,
- .period_bytes_min = 1024,
- .period_bytes_max = 2048,
- .periods_min = 2,
- .periods_max = 32,
- .buffer_bytes_max = 4096,
- .fifo_size = 32,
- },
-};
-
-static int mmp_pcm_hw_params(struct snd_soc_component *component,
- struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
- struct dma_slave_config slave_config;
- int ret;
-
- ret =
- snd_dmaengine_pcm_prepare_slave_config(substream, params,
- &slave_config);
- if (ret)
- return ret;
-
- ret = dmaengine_slave_config(chan, &slave_config);
- if (ret)
- return ret;
-
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
-
- return 0;
-}
-
-static int mmp_pcm_trigger(struct snd_soc_component *component,
- struct snd_pcm_substream *substream, int cmd)
-{
- return snd_dmaengine_pcm_trigger(substream, cmd);
-}
-
-static snd_pcm_uframes_t mmp_pcm_pointer(struct snd_soc_component *component,
- struct snd_pcm_substream *substream)
-{
- return snd_dmaengine_pcm_pointer(substream);
-}
-
-static bool filter(struct dma_chan *chan, void *param)
-{
- struct mmp_dma_data *dma_data = param;
- bool found = false;
- char *devname;
-
- devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
- dma_data->ssp_id);
- if (devname && (strcmp(dev_name(chan->device->dev), devname) == 0) &&
- (chan->chan_id == dma_data->dma_res->start)) {
- found = true;
- }
-
- kfree(devname);
- return found;
-}
-
-static int mmp_pcm_open(struct snd_soc_component *component,
- struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct platform_device *pdev = to_platform_device(component->dev);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- struct mmp_dma_data dma_data;
- struct resource *r;
-
- r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
- if (!r)
- return -EBUSY;
-
- snd_soc_set_runtime_hwparams(substream,
- &mmp_pcm_hardware[substream->stream]);
-
- dma_data.dma_res = r;
- dma_data.ssp_id = cpu_dai->id;
-
- return snd_dmaengine_pcm_open_request_chan(substream, filter,
- &dma_data);
-}
-
-static int mmp_pcm_close(struct snd_soc_component *component,
- struct snd_pcm_substream *substream)
-{
- return snd_dmaengine_pcm_close_release_chan(substream);
-}
-
-static int mmp_pcm_mmap(struct snd_soc_component *component,
- struct snd_pcm_substream *substream,
- struct vm_area_struct *vma)
-{
- struct snd_pcm_runtime *runtime = substream->runtime;
- unsigned long off = vma->vm_pgoff;
-
- vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
- return remap_pfn_range(vma, vma->vm_start,
- __phys_to_pfn(runtime->dma_addr) + off,
- vma->vm_end - vma->vm_start, vma->vm_page_prot);
-}
-
-static void mmp_pcm_free_dma_buffers(struct snd_soc_component *component,
- struct snd_pcm *pcm)
-{
- struct snd_pcm_substream *substream;
- struct snd_dma_buffer *buf;
- int stream;
- struct gen_pool *gpool;
-
- gpool = sram_get_gpool("asram");
- if (!gpool)
- return;
-
- for (stream = 0; stream < 2; stream++) {
- size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
-
- substream = pcm->streams[stream].substream;
- if (!substream)
- continue;
-
- buf = &substream->dma_buffer;
- if (!buf->area)
- continue;
- gen_pool_free(gpool, (unsigned long)buf->area, size);
- buf->area = NULL;
- }
-
-}
-
-static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
- int stream)
-{
- struct snd_dma_buffer *buf = &substream->dma_buffer;
- size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
- struct gen_pool *gpool;
-
- buf->dev.type = SNDRV_DMA_TYPE_DEV;
- buf->dev.dev = substream->pcm->card->dev;
- buf->private_data = NULL;
-
- gpool = sram_get_gpool("asram");
- if (!gpool)
- return -ENOMEM;
-
- buf->area = gen_pool_dma_alloc(gpool, size, &buf->addr);
- if (!buf->area)
- return -ENOMEM;
- buf->bytes = size;
- return 0;
-}
-
-static int mmp_pcm_new(struct snd_soc_component *component,
- struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_pcm_substream *substream;
- struct snd_pcm *pcm = rtd->pcm;
- int ret, stream;
-
- for (stream = 0; stream < 2; stream++) {
- substream = pcm->streams[stream].substream;
-
- ret = mmp_pcm_preallocate_dma_buffer(substream, stream);
- if (ret)
- goto err;
- }
-
- return 0;
-
-err:
- mmp_pcm_free_dma_buffers(component, pcm);
- return ret;
-}
-
-static const struct snd_soc_component_driver mmp_soc_component = {
- .name = DRV_NAME,
- .open = mmp_pcm_open,
- .close = mmp_pcm_close,
- .hw_params = mmp_pcm_hw_params,
- .trigger = mmp_pcm_trigger,
- .pointer = mmp_pcm_pointer,
- .mmap = mmp_pcm_mmap,
- .pcm_construct = mmp_pcm_new,
- .pcm_destruct = mmp_pcm_free_dma_buffers,
-};
-
-static int mmp_pcm_probe(struct platform_device *pdev)
-{
- struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
-
- if (pdata) {
- mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
- pdata->buffer_max_playback;
- mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
- pdata->period_max_playback;
- mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
- pdata->buffer_max_capture;
- mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
- pdata->period_max_capture;
- }
- return devm_snd_soc_register_component(&pdev->dev, &mmp_soc_component,
- NULL, 0);
-}
-
-static struct platform_driver mmp_pcm_driver = {
- .driver = {
- .name = "mmp-pcm-audio",
- },
-
- .probe = mmp_pcm_probe,
-};
-
-module_platform_driver(mmp_pcm_driver);
-
-MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
-MODULE_DESCRIPTION("MMP Soc Audio DMA module");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:mmp-pcm-audio");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
deleted file mode 100644
index a2321c01c160..000000000000
--- a/sound/soc/pxa/palm27x.c
+++ /dev/null
@@ -1,162 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * linux/sound/soc/pxa/palm27x.c
- *
- * SoC Audio driver for Palm T|X, T5 and LifeDrive
- *
- * based on tosa.c
- *
- * Copyright (C) 2008 Marek Vasut <marek.vasut@gmail.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/gpio.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-pxa.h>
-#include <linux/platform_data/asoc-palm27x.h>
-
-static struct snd_soc_jack hs_jack;
-
-/* Headphones jack detection DAPM pins */
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- {
- .pin = "Headphone Jack",
- .mask = SND_JACK_HEADPHONE,
- },
-};
-
-/* Headphones jack detection gpios */
-static struct snd_soc_jack_gpio hs_jack_gpios[] = {
- [0] = {
- /* gpio is set on per-platform basis */
- .name = "hp-gpio",
- .report = SND_JACK_HEADPHONE,
- .debounce_time = 200,
- },
-};
-
-/* Palm27x machine dapm widgets */
-static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_SPK("Ext. Speaker", NULL),
- SND_SOC_DAPM_MIC("Ext. Microphone", NULL),
-};
-
-/* PalmTX audio map */
-static const struct snd_soc_dapm_route audio_map[] = {
- /* headphone connected to HPOUTL, HPOUTR */
- {"Headphone Jack", NULL, "HPOUTL"},
- {"Headphone Jack", NULL, "HPOUTR"},
-
- /* ext speaker connected to ROUT2, LOUT2 */
- {"Ext. Speaker", NULL, "LOUT2"},
- {"Ext. Speaker", NULL, "ROUT2"},
-
- /* mic connected to MIC1 */
- {"MIC1", NULL, "Ext. Microphone"},
-};
-
-static struct snd_soc_card palm27x_asoc;
-
-static int palm27x_ac97_init(struct snd_soc_pcm_runtime *rtd)
-{
- int err;
-
- /* Jack detection API stuff */
- err = snd_soc_card_jack_new_pins(rtd->card, "Headphone Jack",
- SND_JACK_HEADPHONE, &hs_jack,
- hs_jack_pins,
- ARRAY_SIZE(hs_jack_pins));
- if (err)
- return err;
-
- err = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
-
- return err;
-}
-
-SND_SOC_DAILINK_DEFS(hifi,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link palm27x_dai[] = {
-{
- .name = "AC97 HiFi",
- .stream_name = "AC97 HiFi",
- .init = palm27x_ac97_init,
- SND_SOC_DAILINK_REG(hifi),
-},
-{
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- SND_SOC_DAILINK_REG(aux),
-},
-};
-
-static struct snd_soc_card palm27x_asoc = {
- .name = "Palm/PXA27x",
- .owner = THIS_MODULE,
- .dai_link = palm27x_dai,
- .num_links = ARRAY_SIZE(palm27x_dai),
- .dapm_widgets = palm27x_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(palm27x_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
- .fully_routed = true,
-};
-
-static int palm27x_asoc_probe(struct platform_device *pdev)
-{
- int ret;
-
- if (!(machine_is_palmtx() || machine_is_palmt5() ||
- machine_is_palmld() || machine_is_palmte2()))
- return -ENODEV;
-
- if (!pdev->dev.platform_data) {
- dev_err(&pdev->dev, "please supply platform_data\n");
- return -ENODEV;
- }
-
- hs_jack_gpios[0].gpio = ((struct palm27x_asoc_info *)
- (pdev->dev.platform_data))->jack_gpio;
-
- palm27x_asoc.dev = &pdev->dev;
-
- ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static struct platform_driver palm27x_wm9712_driver = {
- .probe = palm27x_asoc_probe,
- .driver = {
- .name = "palm27x-asoc",
- .pm = &snd_soc_pm_ops,
- },
-};
-
-module_platform_driver(palm27x_wm9712_driver);
-
-/* Module information */
-MODULE_AUTHOR("Marek Vasut <marek.vasut@gmail.com>");
-MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:palm27x-asoc");
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
deleted file mode 100644
index 5fdaa477e85d..000000000000
--- a/sound/soc/pxa/poodle.c
+++ /dev/null
@@ -1,291 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * poodle.c -- SoC audio for Poodle
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
- * Richard Purdie <richard@openedhand.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/i2c.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <asm/hardware/locomo.h>
-#include <linux/platform_data/asoc-pxa.h>
-#include <linux/platform_data/asoc-poodle.h>
-
-#include "../codecs/wm8731.h"
-#include "pxa2xx-i2s.h"
-
-#define POODLE_HP 1
-#define POODLE_HP_OFF 0
-#define POODLE_SPK_ON 1
-#define POODLE_SPK_OFF 0
-
- /* audio clock in Hz - rounded from 12.235MHz */
-#define POODLE_AUDIO_CLOCK 12288000
-
-static int poodle_jack_func;
-static int poodle_spk_func;
-
-static struct poodle_audio_platform_data *poodle_pdata;
-
-static void poodle_ext_control(struct snd_soc_dapm_context *dapm)
-{
- /* set up jack connection */
- if (poodle_jack_func == POODLE_HP) {
- /* set = unmute headphone */
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_l, 1);
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_r, 1);
- snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
- } else {
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_l, 0);
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_r, 0);
- snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
- }
-
- /* set the endpoints to their new connection states */
- if (poodle_spk_func == POODLE_SPK_ON)
- snd_soc_dapm_enable_pin(dapm, "Ext Spk");
- else
- snd_soc_dapm_disable_pin(dapm, "Ext Spk");
-
- /* signal a DAPM event */
- snd_soc_dapm_sync(dapm);
-}
-
-static int poodle_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
-
- /* check the jack status at stream startup */
- poodle_ext_control(&rtd->card->dapm);
-
- return 0;
-}
-
-/* we need to unmute the HP at shutdown as the mute burns power on poodle */
-static void poodle_shutdown(struct snd_pcm_substream *substream)
-{
- /* set = unmute headphone */
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_l, 1);
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_r, 1);
-}
-
-static int poodle_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- unsigned int clk = 0;
- int ret = 0;
-
- switch (params_rate(params)) {
- case 8000:
- case 16000:
- case 48000:
- case 96000:
- clk = 12288000;
- break;
- case 11025:
- case 22050:
- case 44100:
- clk = 11289600;
- break;
- }
-
- /* set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- /* set the I2S system clock as input (unused) */
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static const struct snd_soc_ops poodle_ops = {
- .startup = poodle_startup,
- .hw_params = poodle_hw_params,
- .shutdown = poodle_shutdown,
-};
-
-static int poodle_get_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = poodle_jack_func;
- return 0;
-}
-
-static int poodle_set_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (poodle_jack_func == ucontrol->value.enumerated.item[0])
- return 0;
-
- poodle_jack_func = ucontrol->value.enumerated.item[0];
- poodle_ext_control(&card->dapm);
- return 1;
-}
-
-static int poodle_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = poodle_spk_func;
- return 0;
-}
-
-static int poodle_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (poodle_spk_func == ucontrol->value.enumerated.item[0])
- return 0;
-
- poodle_spk_func = ucontrol->value.enumerated.item[0];
- poodle_ext_control(&card->dapm);
- return 1;
-}
-
-static int poodle_amp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- if (SND_SOC_DAPM_EVENT_ON(event))
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_amp_on, 0);
- else
- locomo_gpio_write(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_amp_on, 1);
-
- return 0;
-}
-
-/* poodle machine dapm widgets */
-static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
-SND_SOC_DAPM_HP("Headphone Jack", NULL),
-SND_SOC_DAPM_SPK("Ext Spk", poodle_amp_event),
-SND_SOC_DAPM_MIC("Microphone", NULL),
-};
-
-/* Corgi machine connections to the codec pins */
-static const struct snd_soc_dapm_route poodle_audio_map[] = {
-
- /* headphone connected to LHPOUT1, RHPOUT1 */
- {"Headphone Jack", NULL, "LHPOUT"},
- {"Headphone Jack", NULL, "RHPOUT"},
-
- /* speaker connected to LOUT, ROUT */
- {"Ext Spk", NULL, "ROUT"},
- {"Ext Spk", NULL, "LOUT"},
-
- {"MICIN", NULL, "Microphone"},
-};
-
-static const char * const jack_function[] = {"Off", "Headphone"};
-static const char * const spk_function[] = {"Off", "On"};
-static const struct soc_enum poodle_enum[] = {
- SOC_ENUM_SINGLE_EXT(2, jack_function),
- SOC_ENUM_SINGLE_EXT(2, spk_function),
-};
-
-static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
- SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack,
- poodle_set_jack),
- SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk,
- poodle_set_spk),
-};
-
-/* poodle digital audio interface glue - connects codec <--> CPU */
-SND_SOC_DAILINK_DEFS(wm8731,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm8731.0-001b", "wm8731-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link poodle_dai = {
- .name = "WM8731",
- .stream_name = "WM8731",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &poodle_ops,
- SND_SOC_DAILINK_REG(wm8731),
-};
-
-/* poodle audio machine driver */
-static struct snd_soc_card poodle = {
- .name = "Poodle",
- .dai_link = &poodle_dai,
- .num_links = 1,
- .owner = THIS_MODULE,
-
- .controls = wm8731_poodle_controls,
- .num_controls = ARRAY_SIZE(wm8731_poodle_controls),
- .dapm_widgets = wm8731_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(wm8731_dapm_widgets),
- .dapm_routes = poodle_audio_map,
- .num_dapm_routes = ARRAY_SIZE(poodle_audio_map),
- .fully_routed = true,
-};
-
-static int poodle_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &poodle;
- int ret;
-
- poodle_pdata = pdev->dev.platform_data;
- locomo_gpio_set_dir(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_amp_on, 0);
- /* should we mute HP at startup - burning power ?*/
- locomo_gpio_set_dir(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_l, 0);
- locomo_gpio_set_dir(poodle_pdata->locomo_dev,
- poodle_pdata->gpio_mute_r, 0);
-
- card->dev = &pdev->dev;
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- return ret;
-}
-
-static struct platform_driver poodle_driver = {
- .driver = {
- .name = "poodle-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = poodle_probe,
-};
-
-module_platform_driver(poodle_driver);
-
-/* Module information */
-MODULE_AUTHOR("Richard Purdie");
-MODULE_DESCRIPTION("ALSA SoC Poodle");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:poodle-audio");
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
deleted file mode 100644
index 30f83cab0c32..000000000000
--- a/sound/soc/pxa/tosa.c
+++ /dev/null
@@ -1,255 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * tosa.c -- SoC audio for Tosa
- *
- * Copyright 2005 Wolfson Microelectronics PLC.
- * Copyright 2005 Openedhand Ltd.
- *
- * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
- * Richard Purdie <richard@openedhand.com>
- *
- * GPIO's
- * 1 - Jack Insertion
- * 5 - Hookswitch (headset answer/hang up switch)
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/gpio/consumer.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-pxa.h>
-
-#define TOSA_HP 0
-#define TOSA_MIC_INT 1
-#define TOSA_HEADSET 2
-#define TOSA_HP_OFF 3
-#define TOSA_SPK_ON 0
-#define TOSA_SPK_OFF 1
-
-static struct gpio_desc *tosa_mute;
-static int tosa_jack_func;
-static int tosa_spk_func;
-
-static void tosa_ext_control(struct snd_soc_dapm_context *dapm)
-{
-
- snd_soc_dapm_mutex_lock(dapm);
-
- /* set up jack connection */
- switch (tosa_jack_func) {
- case TOSA_HP:
- snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
- snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
- break;
- case TOSA_MIC_INT:
- snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack");
- break;
- case TOSA_HEADSET:
- snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)");
- snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
- snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack");
- break;
- }
-
- if (tosa_spk_func == TOSA_SPK_ON)
- snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
- else
- snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
-
- snd_soc_dapm_sync_unlocked(dapm);
-
- snd_soc_dapm_mutex_unlock(dapm);
-}
-
-static int tosa_startup(struct snd_pcm_substream *substream)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
-
- /* check the jack status at stream startup */
- tosa_ext_control(&rtd->card->dapm);
-
- return 0;
-}
-
-static const struct snd_soc_ops tosa_ops = {
- .startup = tosa_startup,
-};
-
-static int tosa_get_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = tosa_jack_func;
- return 0;
-}
-
-static int tosa_set_jack(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (tosa_jack_func == ucontrol->value.enumerated.item[0])
- return 0;
-
- tosa_jack_func = ucontrol->value.enumerated.item[0];
- tosa_ext_control(&card->dapm);
- return 1;
-}
-
-static int tosa_get_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- ucontrol->value.enumerated.item[0] = tosa_spk_func;
- return 0;
-}
-
-static int tosa_set_spk(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
-
- if (tosa_spk_func == ucontrol->value.enumerated.item[0])
- return 0;
-
- tosa_spk_func = ucontrol->value.enumerated.item[0];
- tosa_ext_control(&card->dapm);
- return 1;
-}
-
-/* tosa dapm event handlers */
-static int tosa_hp_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *k, int event)
-{
- gpiod_set_value(tosa_mute, SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0);
- return 0;
-}
-
-/* tosa machine dapm widgets */
-static const struct snd_soc_dapm_widget tosa_dapm_widgets[] = {
-SND_SOC_DAPM_HP("Headphone Jack", tosa_hp_event),
-SND_SOC_DAPM_HP("Headset Jack", NULL),
-SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
-SND_SOC_DAPM_SPK("Speaker", NULL),
-};
-
-/* tosa audio map */
-static const struct snd_soc_dapm_route audio_map[] = {
-
- /* headphone connected to HPOUTL, HPOUTR */
- {"Headphone Jack", NULL, "HPOUTL"},
- {"Headphone Jack", NULL, "HPOUTR"},
-
- /* ext speaker connected to LOUT2, ROUT2 */
- {"Speaker", NULL, "LOUT2"},
- {"Speaker", NULL, "ROUT2"},
-
- /* internal mic is connected to mic1, mic2 differential - with bias */
- {"MIC1", NULL, "Mic Bias"},
- {"MIC2", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Mic (Internal)"},
-
- /* headset is connected to HPOUTR, and LINEINR with bias */
- {"Headset Jack", NULL, "HPOUTR"},
- {"LINEINR", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Headset Jack"},
-};
-
-static const char * const jack_function[] = {"Headphone", "Mic", "Line",
- "Headset", "Off"};
-static const char * const spk_function[] = {"On", "Off"};
-static const struct soc_enum tosa_enum[] = {
- SOC_ENUM_SINGLE_EXT(5, jack_function),
- SOC_ENUM_SINGLE_EXT(2, spk_function),
-};
-
-static const struct snd_kcontrol_new tosa_controls[] = {
- SOC_ENUM_EXT("Jack Function", tosa_enum[0], tosa_get_jack,
- tosa_set_jack),
- SOC_ENUM_EXT("Speaker Function", tosa_enum[1], tosa_get_spk,
- tosa_set_spk),
-};
-
-SND_SOC_DAILINK_DEFS(ac97,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(ac97_aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9712-codec", "wm9712-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link tosa_dai[] = {
-{
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- .ops = &tosa_ops,
- SND_SOC_DAILINK_REG(ac97),
-},
-{
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- .ops = &tosa_ops,
- SND_SOC_DAILINK_REG(ac97_aux),
-},
-};
-
-static struct snd_soc_card tosa = {
- .name = "Tosa",
- .owner = THIS_MODULE,
- .dai_link = tosa_dai,
- .num_links = ARRAY_SIZE(tosa_dai),
-
- .controls = tosa_controls,
- .num_controls = ARRAY_SIZE(tosa_controls),
- .dapm_widgets = tosa_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
- .fully_routed = true,
-};
-
-static int tosa_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &tosa;
- int ret;
-
- tosa_mute = devm_gpiod_get(&pdev->dev, NULL, GPIOD_OUT_LOW);
- if (IS_ERR(tosa_mute))
- return dev_err_probe(&pdev->dev, PTR_ERR(tosa_mute),
- "failed to get L_MUTE GPIO\n");
- gpiod_set_consumer_name(tosa_mute, "Headphone Jack");
-
- card->dev = &pdev->dev;
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret) {
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
- }
- return ret;
-}
-
-static struct platform_driver tosa_driver = {
- .driver = {
- .name = "tosa-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = tosa_probe,
-};
-
-module_platform_driver(tosa_driver);
-
-/* Module information */
-MODULE_AUTHOR("Richard Purdie");
-MODULE_DESCRIPTION("ALSA SoC Tosa");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:tosa-audio");
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
deleted file mode 100644
index 6cc970bb2aac..000000000000
--- a/sound/soc/pxa/ttc-dkb.c
+++ /dev/null
@@ -1,143 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * linux/sound/soc/pxa/ttc_dkb.c
- *
- * Copyright (C) 2012 Marvell International Ltd.
- */
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-#include <asm/mach-types.h>
-#include <sound/pcm_params.h>
-#include "../codecs/88pm860x-codec.h"
-
-static struct snd_soc_jack hs_jack, mic_jack;
-
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, },
-};
-
-static struct snd_soc_jack_pin mic_jack_pins[] = {
- { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, },
-};
-
-/* ttc machine dapm widgets */
-static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headset Stereophone", NULL),
- SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
- SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
- SND_SOC_DAPM_SPK("Ext Speaker", NULL),
- SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
- SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
- SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
-};
-
-/* ttc machine audio map */
-static const struct snd_soc_dapm_route ttc_audio_map[] = {
- {"Headset Stereophone", NULL, "HS1"},
- {"Headset Stereophone", NULL, "HS2"},
-
- {"Ext Speaker", NULL, "LSP"},
- {"Ext Speaker", NULL, "LSN"},
-
- {"Lineout Out 1", NULL, "LINEOUT1"},
- {"Lineout Out 2", NULL, "LINEOUT2"},
-
- {"MIC1P", NULL, "Mic1 Bias"},
- {"MIC1N", NULL, "Mic1 Bias"},
- {"Mic1 Bias", NULL, "Ext Mic 1"},
-
- {"MIC2P", NULL, "Mic1 Bias"},
- {"MIC2N", NULL, "Mic1 Bias"},
- {"Mic1 Bias", NULL, "Headset Mic 2"},
-
- {"MIC3P", NULL, "Mic3 Bias"},
- {"MIC3N", NULL, "Mic3 Bias"},
- {"Mic3 Bias", NULL, "Ext Mic 3"},
-};
-
-static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component;
-
- /* Headset jack detection */
- snd_soc_card_jack_new_pins(rtd->card, "Headphone Jack",
- SND_JACK_HEADPHONE | SND_JACK_BTN_0 |
- SND_JACK_BTN_1 | SND_JACK_BTN_2,
- &hs_jack,
- hs_jack_pins, ARRAY_SIZE(hs_jack_pins));
- snd_soc_card_jack_new_pins(rtd->card, "Microphone Jack",
- SND_JACK_MICROPHONE, &mic_jack,
- mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
-
- /* headphone, microphone detection & headset short detection */
- pm860x_hs_jack_detect(component, &hs_jack, SND_JACK_HEADPHONE,
- SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
- pm860x_mic_jack_detect(component, &hs_jack, SND_JACK_MICROPHONE);
-
- return 0;
-}
-
-/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
-SND_SOC_DAILINK_DEFS(i2s,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa-ssp-dai.1")),
- DAILINK_COMP_ARRAY(COMP_CODEC("88pm860x-codec", "88pm860x-i2s")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("mmp-pcm-audio")));
-
-static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
-{
- .name = "88pm860x i2s",
- .stream_name = "audio playback",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBM_CFM,
- .init = ttc_pm860x_init,
- SND_SOC_DAILINK_REG(i2s),
-},
-};
-
-/* ttc/td audio machine driver */
-static struct snd_soc_card ttc_dkb_card = {
- .name = "ttc-dkb-hifi",
- .owner = THIS_MODULE,
- .dai_link = ttc_pm860x_hifi_dai,
- .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
-
- .dapm_widgets = ttc_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
- .dapm_routes = ttc_audio_map,
- .num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
-};
-
-static int ttc_dkb_probe(struct platform_device *pdev)
-{
- struct snd_soc_card *card = &ttc_dkb_card;
- int ret;
-
- card->dev = &pdev->dev;
-
- ret = devm_snd_soc_register_card(&pdev->dev, card);
- if (ret)
- dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
- ret);
-
- return ret;
-}
-
-static struct platform_driver ttc_dkb_driver = {
- .driver = {
- .name = "ttc-dkb-audio",
- .pm = &snd_soc_pm_ops,
- },
- .probe = ttc_dkb_probe,
-};
-
-module_platform_driver(ttc_dkb_driver);
-
-/* Module information */
-MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
-MODULE_DESCRIPTION("ALSA SoC TTC DKB");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:ttc-dkb-audio");
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
deleted file mode 100644
index 020dcce1df1f..000000000000
--- a/sound/soc/pxa/z2.c
+++ /dev/null
@@ -1,218 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-only
-/*
- * linux/sound/soc/pxa/z2.c
- *
- * SoC Audio driver for Aeronix Zipit Z2
- *
- * Copyright (C) 2009 Ken McGuire <kenm@desertweyr.com>
- * Copyright (C) 2010 Marek Vasut <marek.vasut@gmail.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
-#include <linux/platform_device.h>
-#include <linux/gpio/consumer.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/soc.h>
-#include <sound/jack.h>
-
-#include <asm/mach-types.h>
-#include <linux/platform_data/asoc-pxa.h>
-
-#include "../codecs/wm8750.h"
-#include "pxa2xx-i2s.h"
-
-static struct snd_soc_card snd_soc_z2;
-
-static int z2_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- unsigned int clk = 0;
- int ret = 0;
-
- switch (params_rate(params)) {
- case 8000:
- case 16000:
- case 48000:
- case 96000:
- clk = 12288000;
- break;
- case 11025:
- case 22050:
- case 44100:
- clk = 11289600;
- break;
- }
-
- /* set the codec system clock for DAC and ADC */
- ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- /* set the I2S system clock as input (unused) */
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
- SND_SOC_CLOCK_IN);
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static struct snd_soc_jack hs_jack;
-
-/* Headset jack detection DAPM pins */
-static struct snd_soc_jack_pin hs_jack_pins[] = {
- {
- .pin = "Mic Jack",
- .mask = SND_JACK_MICROPHONE,
- },
- {
- .pin = "Headphone Jack",
- .mask = SND_JACK_HEADPHONE,
- },
- {
- .pin = "Ext Spk",
- .mask = SND_JACK_HEADPHONE,
- .invert = 1
- },
-};
-
-/* Headset jack detection gpios */
-static struct snd_soc_jack_gpio hs_jack_gpios[] = {
- {
- .name = "hsdet-gpio",
- .report = SND_JACK_HEADSET,
- .debounce_time = 200,
- .invert = 1,
- },
-};
-
-/* z2 machine dapm widgets */
-static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone Jack", NULL),
- SND_SOC_DAPM_MIC("Mic Jack", NULL),
- SND_SOC_DAPM_SPK("Ext Spk", NULL),
-
- /* headset is a mic and mono headphone */
- SND_SOC_DAPM_HP("Headset Jack", NULL),
-};
-
-/* Z2 machine audio_map */
-static const struct snd_soc_dapm_route z2_audio_map[] = {
-
- /* headphone connected to LOUT1, ROUT1 */
- {"Headphone Jack", NULL, "LOUT1"},
- {"Headphone Jack", NULL, "ROUT1"},
-
- /* ext speaker connected to LOUT2, ROUT2 */
- {"Ext Spk", NULL, "ROUT2"},
- {"Ext Spk", NULL, "LOUT2"},
-
- /* mic is connected to R input 2 - with bias */
- {"RINPUT2", NULL, "Mic Bias"},
- {"Mic Bias", NULL, "Mic Jack"},
-};
-
-/*
- * Logic for a wm8750 as connected on a Z2 Device
- */
-static int z2_wm8750_init(struct snd_soc_pcm_runtime *rtd)
-{
- int ret;
-
- /* Jack detection API stuff */
- ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack",
- SND_JACK_HEADSET, &hs_jack,
- hs_jack_pins,
- ARRAY_SIZE(hs_jack_pins));
- if (ret)
- goto err;
-
- ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
- hs_jack_gpios);
- if (ret)
- goto err;
-
- return 0;
-
-err:
- return ret;
-}
-
-static const struct snd_soc_ops z2_ops = {
- .hw_params = z2_hw_params,
-};
-
-/* z2 digital audio interface glue - connects codec <--> CPU */
-SND_SOC_DAILINK_DEFS(wm8750,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-i2s")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm8750.0-001b", "wm8750-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link z2_dai = {
- .name = "wm8750",
- .stream_name = "WM8750",
- .init = z2_wm8750_init,
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &z2_ops,
- SND_SOC_DAILINK_REG(wm8750),
-};
-
-/* z2 audio machine driver */
-static struct snd_soc_card snd_soc_z2 = {
- .name = "Z2",
- .owner = THIS_MODULE,
- .dai_link = &z2_dai,
- .num_links = 1,
-
- .dapm_widgets = wm8750_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
- .dapm_routes = z2_audio_map,
- .num_dapm_routes = ARRAY_SIZE(z2_audio_map),
- .fully_routed = true,
-};
-
-static struct platform_device *z2_snd_device;
-
-static int __init z2_init(void)
-{
- int ret;
-
- if (!machine_is_zipit2())
- return -ENODEV;
-
- z2_snd_device = platform_device_alloc("soc-audio", -1);
- if (!z2_snd_device)
- return -ENOMEM;
-
- hs_jack_gpios[0].gpiod_dev = &z2_snd_device->dev;
- platform_set_drvdata(z2_snd_device, &snd_soc_z2);
- ret = platform_device_add(z2_snd_device);
-
- if (ret)
- platform_device_put(z2_snd_device);
-
- return ret;
-}
-
-static void __exit z2_exit(void)
-{
- platform_device_unregister(z2_snd_device);
-}
-
-module_init(z2_init);
-module_exit(z2_exit);
-
-MODULE_AUTHOR("Ken McGuire <kenm@desertweyr.com>, "
- "Marek Vasut <marek.vasut@gmail.com>");
-MODULE_DESCRIPTION("ALSA SoC ZipitZ2");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
deleted file mode 100644
index bb89a53f4ab1..000000000000
--- a/sound/soc/pxa/zylonite.c
+++ /dev/null
@@ -1,266 +0,0 @@
-// SPDX-License-Identifier: GPL-2.0-or-later
-/*
- * zylonite.c -- SoC audio for Zylonite
- *
- * Copyright 2008 Wolfson Microelectronics PLC.
- * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
- */
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/device.h>
-#include <linux/clk.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/pcm_params.h>
-#include <sound/soc.h>
-
-#include "../codecs/wm9713.h"
-#include "pxa-ssp.h"
-
-/*
- * There is a physical switch SW15 on the board which changes the MCLK
- * for the WM9713 between the standard AC97 master clock and the
- * output of the CLK_POUT signal from the PXA.
- */
-static int clk_pout;
-module_param(clk_pout, int, 0);
-MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
-
-static struct clk *pout;
-
-static struct snd_soc_card zylonite;
-
-static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
- SND_SOC_DAPM_HP("Headphone", NULL),
- SND_SOC_DAPM_MIC("Headset Microphone", NULL),
- SND_SOC_DAPM_MIC("Handset Microphone", NULL),
- SND_SOC_DAPM_SPK("Multiactor", NULL),
- SND_SOC_DAPM_SPK("Headset Earpiece", NULL),
-};
-
-/* Currently supported audio map */
-static const struct snd_soc_dapm_route audio_map[] = {
-
- /* Headphone output connected to HPL/HPR */
- { "Headphone", NULL, "HPL" },
- { "Headphone", NULL, "HPR" },
-
- /* On-board earpiece */
- { "Headset Earpiece", NULL, "OUT3" },
-
- /* Headphone mic */
- { "MIC2A", NULL, "Mic Bias" },
- { "Mic Bias", NULL, "Headset Microphone" },
-
- /* On-board mic */
- { "MIC1", NULL, "Mic Bias" },
- { "Mic Bias", NULL, "Handset Microphone" },
-
- /* Multiactor differentially connected over SPKL/SPKR */
- { "Multiactor", NULL, "SPKL" },
- { "Multiactor", NULL, "SPKR" },
-};
-
-static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd)
-{
- if (clk_pout)
- snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), 0, 0,
- clk_get_rate(pout), 0);
-
- return 0;
-}
-
-static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
- struct snd_pcm_hw_params *params)
-{
- struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
- struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
- struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
- unsigned int wm9713_div = 0;
- int ret = 0;
- int rate = params_rate(params);
-
- /* Only support ratios that we can generate neatly from the AC97
- * based master clock - in particular, this excludes 44.1kHz.
- * In most applications the voice DAC will be used for telephony
- * data so multiples of 8kHz will be the common case.
- */
- switch (rate) {
- case 8000:
- wm9713_div = 12;
- break;
- case 16000:
- wm9713_div = 6;
- break;
- case 48000:
- wm9713_div = 2;
- break;
- default:
- /* Don't support OSS emulation */
- return -EINVAL;
- }
-
- ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
- if (ret < 0)
- return ret;
-
- if (clk_pout)
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
- WM9713_PCMDIV(wm9713_div));
- else
- ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
- WM9713_PCMDIV(wm9713_div));
- if (ret < 0)
- return ret;
-
- return 0;
-}
-
-static const struct snd_soc_ops zylonite_voice_ops = {
- .hw_params = zylonite_voice_hw_params,
-};
-
-SND_SOC_DAILINK_DEFS(ac97,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-hifi")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(ac97_aux,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa2xx-ac97-aux")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-aux")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-SND_SOC_DAILINK_DEFS(voice,
- DAILINK_COMP_ARRAY(COMP_CPU("pxa-ssp-dai.2")),
- DAILINK_COMP_ARRAY(COMP_CODEC("wm9713-codec", "wm9713-voice")),
- DAILINK_COMP_ARRAY(COMP_PLATFORM("pxa-pcm-audio")));
-
-static struct snd_soc_dai_link zylonite_dai[] = {
-{
- .name = "AC97",
- .stream_name = "AC97 HiFi",
- .init = zylonite_wm9713_init,
- SND_SOC_DAILINK_REG(ac97),
-},
-{
- .name = "AC97 Aux",
- .stream_name = "AC97 Aux",
- SND_SOC_DAILINK_REG(ac97_aux),
-},
-{
- .name = "WM9713 Voice",
- .stream_name = "WM9713 Voice",
- .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
- SND_SOC_DAIFMT_CBS_CFS,
- .ops = &zylonite_voice_ops,
- SND_SOC_DAILINK_REG(voice),
-},
-};
-
-static int zylonite_probe(struct snd_soc_card *card)
-{
- int ret;
-
- if (clk_pout) {
- pout = clk_get(NULL, "CLK_POUT");
- if (IS_ERR(pout)) {
- dev_err(card->dev, "Unable to obtain CLK_POUT: %ld\n",
- PTR_ERR(pout));
- return PTR_ERR(pout);
- }
-
- ret = clk_enable(pout);
- if (ret != 0) {
- dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
- ret);
- clk_put(pout);
- return ret;
- }
-
- dev_dbg(card->dev, "MCLK enabled at %luHz\n",
- clk_get_rate(pout));
- }
-
- return 0;
-}
-
-static int zylonite_remove(struct snd_soc_card *card)
-{
- if (clk_pout) {
- clk_disable(pout);
- clk_put(pout);
- }
-
- return 0;
-}
-
-static int zylonite_suspend_post(struct snd_soc_card *card)
-{
- if (clk_pout)
- clk_disable(pout);
-
- return 0;
-}
-
-static int zylonite_resume_pre(struct snd_soc_card *card)
-{
- int ret = 0;
-
- if (clk_pout) {
- ret = clk_enable(pout);
- if (ret != 0)
- dev_err(card->dev, "Unable to enable CLK_POUT: %d\n",
- ret);
- }
-
- return ret;
-}
-
-static struct snd_soc_card zylonite = {
- .name = "Zylonite",
- .owner = THIS_MODULE,
- .probe = &zylonite_probe,
- .remove = &zylonite_remove,
- .suspend_post = &zylonite_suspend_post,
- .resume_pre = &zylonite_resume_pre,
- .dai_link = zylonite_dai,
- .num_links = ARRAY_SIZE(zylonite_dai),
-
- .dapm_widgets = zylonite_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets),
- .dapm_routes = audio_map,
- .num_dapm_routes = ARRAY_SIZE(audio_map),
-};
-
-static struct platform_device *zylonite_snd_ac97_device;
-
-static int __init zylonite_init(void)
-{
- int ret;
-
- zylonite_snd_ac97_device = platform_device_alloc("soc-audio", -1);
- if (!zylonite_snd_ac97_device)
- return -ENOMEM;
-
- platform_set_drvdata(zylonite_snd_ac97_device, &zylonite);
-
- ret = platform_device_add(zylonite_snd_ac97_device);
- if (ret != 0)
- platform_device_put(zylonite_snd_ac97_device);
-
- return ret;
-}
-
-static void __exit zylonite_exit(void)
-{
- platform_device_unregister(zylonite_snd_ac97_device);
-}
-
-module_init(zylonite_init);
-module_exit(zylonite_exit);
-
-MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
-MODULE_DESCRIPTION("ALSA SoC WM9713 Zylonite");
-MODULE_LICENSE("GPL");
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-22 18:42:32 +0000