From e394fe55f7cf5a4f6c20fbd02ab37b1d5c3dd364 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Mon, 1 Jul 2013 16:49:04 +0800 Subject: ASoC: adav80x: Add module device table for adav801 This driver can be built as module, thus add module device table for adav801 to support module auto loading. To make the naming consistent, also rename adav80x_id to adav80x_i2c_id. Signed-off-by: Axel Lin Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 3c839cc4e00e..15b012d0f226 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -868,6 +868,12 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct spi_device_id adav80x_spi_id[] = { + { "adav801", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adav80x_spi_id); + static int adav80x_spi_probe(struct spi_device *spi) { return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); @@ -885,15 +891,16 @@ static struct spi_driver adav80x_spi_driver = { }, .probe = adav80x_spi_probe, .remove = adav80x_spi_remove, + .id_table = adav80x_spi_id, }; #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static const struct i2c_device_id adav80x_id[] = { +static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } }; -MODULE_DEVICE_TABLE(i2c, adav80x_id); +MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) @@ -913,7 +920,7 @@ static struct i2c_driver adav80x_i2c_driver = { }, .probe = adav80x_i2c_probe, .remove = adav80x_i2c_remove, - .id_table = adav80x_id, + .id_table = adav80x_i2c_id, }; #endif -- cgit v1.2.3 From a2911cdb1fd09de7c0da3938ffab1bc5cedda4f9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 3 Jul 2013 21:15:13 -0700 Subject: ASoC: add ak4554 driver ak4554 is very simple DA/AD converter which has no setting register. Note that it has hard coded asymmetric data format playback : SND_SOC_DAIFMT_RIGHT_J capture : SND_SOC_DAIFMT_LEFT_J This driver has single DAI and doesn't have set_fmt. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 3 ++ sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/ak4554.c | 79 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 84 insertions(+) create mode 100644 sound/soc/codecs/ak4554.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..ffb9adb18647 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -198,6 +198,9 @@ config SND_SOC_AK4104 config SND_SOC_AK4535 tristate +config SND_SOC_AK4554 + tristate + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..fab4086796a0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o +snd-soc-ak4554-objs := ak4554.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -138,6 +139,7 @@ obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o +obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c new file mode 100644 index 000000000000..c1a1733f8a35 --- /dev/null +++ b/sound/soc/codecs/ak4554.c @@ -0,0 +1,79 @@ +/* + * ak4554.c + * + * Copyright (C) 2013 Renesas Solutions Corp. + * Kuninori Morimoto + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include + +/* + * ak4554 is very simple DA/AD converter which has no setting register. + * + * CAUTION + * + * ak4554 playback format is SND_SOC_DAIFMT_RIGHT_J, + * and, capture format is SND_SOC_DAIFMT_LEFT_J + * on same bit clock, LR clock. + * But, this driver doesn't have snd_soc_dai_ops :: set_fmt + * + * CPU/Codec DAI image + * + * CPU-DAI1 (plaback only fmt = RIGHT_J) --+-- ak4554 + * | + * CPU-DAI2 (capture only fmt = LEFT_J) ---+ + */ + +static struct snd_soc_dai_driver ak4554_dai = { + .name = "ak4554-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .symmetric_rates = 1, +}; + +static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { +}; + +static int ak4554_soc_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_ak4554, + &ak4554_dai, 1); +} + +static int ak4554_soc_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak4554_driver = { + .driver = { + .name = "ak4554-adc-dac", + .owner = THIS_MODULE, + }, + .probe = ak4554_soc_probe, + .remove = ak4554_soc_remove, +}; +module_platform_driver(ak4554_driver); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SoC AK4554 driver"); +MODULE_AUTHOR("Kuninori Morimoto "); -- cgit v1.2.3 From b25f77815021ec6e7400a82c4984b9c80699ce80 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 4 Jul 2013 19:42:49 -0700 Subject: ASoC: ak4554: add DT support Support for loading the ak4554 codec module via devicetree. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/ak4554.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index c1a1733f8a35..6aed9c4d06d6 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -64,10 +64,17 @@ static int ak4554_soc_remove(struct platform_device *pdev) return 0; } +static struct of_device_id ak4554_of_match[] = { + { .compatible = "asahi-kasei,ak4554" }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4554_of_match); + static struct platform_driver ak4554_driver = { .driver = { .name = "ak4554-adc-dac", .owner = THIS_MODULE, + .of_match_table = ak4554_of_match, }, .probe = ak4554_soc_probe, .remove = ak4554_soc_remove, -- cgit v1.2.3 From b63144e6c6097486e7678f9ecc2769b68d2ec83e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 4 Jul 2013 08:56:28 +0100 Subject: ASoC: arizona: Add signal activity output for DRC When doing signal activity detection, the only output from the DRC will often be a GPIO pin. This patch adds a signal activity output that is activated when a GPIO is configured to output the DRC signal activity detection. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5110.c | 9 +++++++++ 4 files changed, 49 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index de625813c0e6..c9116ac0d4a7 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -19,6 +19,7 @@ #include #include +#include #include #include "arizona.h" @@ -223,6 +224,38 @@ int arizona_init_spk(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_spk); +int arizona_init_gpio(struct snd_soc_codec *codec) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + int i; + + switch (arizona->type) { + case WM5110: + snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + } + + snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); + + for (i = 0; i < ARRAY_SIZE(arizona->pdata.gpio_defaults); i++) { + switch (arizona->pdata.gpio_defaults[i] & ARIZONA_GPN_FN_MASK) { + case ARIZONA_GP_FN_DRC1_SIGNAL_DETECT: + snd_soc_dapm_enable_pin(&codec->dapm, + "DRC1 Signal Activity"); + break; + case ARIZONA_GP_FN_DRC2_SIGNAL_DETECT: + snd_soc_dapm_enable_pin(&codec->dapm, + "DRC2 Signal Activity"); + break; + default: + break; + } + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_gpio); + const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b60b08ccc1d0..fe1b794bd5f0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -242,6 +242,7 @@ extern int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout); extern int arizona_init_spk(struct snd_soc_codec *codec); +extern int arizona_init_gpio(struct snd_soc_codec *codec); extern int arizona_init_dai(struct arizona_priv *priv, int dai); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 282fd232cdf7..a6cbdb4b5c0f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -998,6 +998,8 @@ SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), + SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | @@ -1614,6 +1616,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "SPKDAT1R", NULL, "OUT5R" }, { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, }; static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, @@ -1781,6 +1786,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_spk(codec); + arizona_init_gpio(codec); snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2e7cb4ba161a..fc410377277f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -432,6 +432,9 @@ SND_SOC_DAPM_INPUT("IN3R"), SND_SOC_DAPM_INPUT("IN4L"), SND_SOC_DAPM_INPUT("IN4R"), +SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), +SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), + SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | @@ -1006,6 +1009,11 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "SPKDAT2R", NULL, "OUT6R" }, { "MICSUPP", NULL, "SYSCLK" }, + + { "DRC1 Signal Activity", NULL, "DRC1L" }, + { "DRC1 Signal Activity", NULL, "DRC1R" }, + { "DRC2 Signal Activity", NULL, "DRC2L" }, + { "DRC2 Signal Activity", NULL, "DRC2R" }, }; static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, @@ -1170,6 +1178,7 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_spk(codec); + arizona_init_gpio(codec); snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); -- cgit v1.2.3 From b79fae606c921522577f3000b6b9a807cd733d2e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 4 Jul 2013 16:53:03 +0100 Subject: ASoC: arizona: Add default case to silence build warning Reported-by: Fengguang Wu Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index c9116ac0d4a7..8dc6881496de 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -233,6 +233,9 @@ int arizona_init_gpio(struct snd_soc_codec *codec) switch (arizona->type) { case WM5110: snd_soc_dapm_disable_pin(&codec->dapm, "DRC2 Signal Activity"); + break; + default: + break; } snd_soc_dapm_disable_pin(&codec->dapm, "DRC1 Signal Activity"); -- cgit v1.2.3 From 64b0c282e87948bc0c5a1b94ca3d4dd9f6415c6f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 25 Jun 2013 11:00:06 +0100 Subject: ASoC: codecs: Make ALL_CODECS depend on COMPILE_TEST The main function of the option is to enable compile testing. There is still an option since COMPILE_TEST is intended to enable selection of extra drivers rather than forcing them on. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..01d112b48e7e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + depends on COMPILE_TEST select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE -- cgit v1.2.3 From b641edfbf253a67a87698df275e3a35a7deac5d4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:31:58 +0100 Subject: ASoC: pcm3008: Remove noisy version print The version number has never been updated and the printk isn't based on any interaction with the device. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f2a6282b41f4..32e5a591f921 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,8 +28,6 @@ #include "pcm3008.h" -#define PCM3008_VERSION "0.2" - #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -64,8 +62,6 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) struct pcm3008_setup_data *setup = codec->dev->platform_data; int ret = 0; - printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); - /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF -- cgit v1.2.3 From d7f184958e5292280c1048df951932bbfcd95f60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:48:24 +0100 Subject: ASoC: pcm3008: Move gpio allocation to probe This is better from a device model point of view since we don't try to do things like instantiate the card until the required resources appear. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 87 ++++++++++++++++++++-------------------------- 1 file changed, 38 insertions(+), 49 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 32e5a591f921..8ab1c03d26bd 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -57,10 +57,40 @@ static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) gpio_free(setup->pdda_pin); } -static int pcm3008_soc_probe(struct snd_soc_codec *codec) +#ifdef CONFIG_PM +static int pcm3008_soc_suspend(struct snd_soc_codec *codec) +{ + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value(setup->pdad_pin, 0); + gpio_set_value(setup->pdda_pin, 0); + + return 0; +} + +static int pcm3008_soc_resume(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; - int ret = 0; + + gpio_set_value(setup->pdad_pin, 1); + gpio_set_value(setup->pdda_pin, 1); + + return 0; +} +#else +#define pcm3008_soc_suspend NULL +#define pcm3008_soc_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { + .suspend = pcm3008_soc_suspend, + .resume = pcm3008_soc_resume, +}; + +static int pcm3008_codec_probe(struct platform_device *pdev) +{ + struct pcm3008_setup_data *setup = pdev->dev.platform_data; + int ret; /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON @@ -97,63 +127,22 @@ static int pcm3008_soc_probe(struct snd_soc_codec *codec) if (ret != 0) goto gpio_err; - return ret; + return snd_soc_register_codec(&pdev->dev, + &soc_codec_dev_pcm3008, &pcm3008_dai, 1); gpio_err: pcm3008_gpio_free(setup); - return ret; } -static int pcm3008_soc_remove(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - pcm3008_gpio_free(setup); - return 0; -} - -#ifdef CONFIG_PM -static int pcm3008_soc_suspend(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value(setup->pdad_pin, 0); - gpio_set_value(setup->pdda_pin, 0); - - return 0; -} - -static int pcm3008_soc_resume(struct snd_soc_codec *codec) +static int pcm3008_codec_remove(struct platform_device *pdev) { - struct pcm3008_setup_data *setup = codec->dev->platform_data; + struct pcm3008_setup_data *setup = pdev->dev.platform_data; - gpio_set_value(setup->pdad_pin, 1); - gpio_set_value(setup->pdda_pin, 1); - - return 0; -} -#else -#define pcm3008_soc_suspend NULL -#define pcm3008_soc_resume NULL -#endif - -static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { - .probe = pcm3008_soc_probe, - .remove = pcm3008_soc_remove, - .suspend = pcm3008_soc_suspend, - .resume = pcm3008_soc_resume, -}; + snd_soc_unregister_codec(&pdev->dev); -static int pcm3008_codec_probe(struct platform_device *pdev) -{ - return snd_soc_register_codec(&pdev->dev, - &soc_codec_dev_pcm3008, &pcm3008_dai, 1); -} + pcm3008_gpio_free(setup); -static int pcm3008_codec_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); return 0; } -- cgit v1.2.3 From 33319a2fcc4482a97a8924e777bc0036d1d96696 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:49:16 +0100 Subject: ASoC: pcm3008: Check for platform data The driver will crash if none is provided. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8ab1c03d26bd..4fa4ded30407 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -92,6 +92,9 @@ static int pcm3008_codec_probe(struct platform_device *pdev) struct pcm3008_setup_data *setup = pdev->dev.platform_data; int ret; + if (!setup) + return -EINVAL; + /* DEM1 DEM0 DE-EMPHASIS_MODE * Low Low De-emphasis 44.1 kHz ON * Low High De-emphasis OFF -- cgit v1.2.3 From d57a79acc77160089e191b6c78e9d42bed517a62 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Jul 2013 16:53:55 +0100 Subject: ASoC: pcm3008: Convert to devm_gpio_request_one() Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 44 ++++++++++++-------------------------------- 1 file changed, 12 insertions(+), 32 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 4fa4ded30407..b883f99d6f9f 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -49,14 +49,6 @@ static struct snd_soc_dai_driver pcm3008_dai = { }, }; -static void pcm3008_gpio_free(struct pcm3008_setup_data *setup) -{ - gpio_free(setup->dem0_pin); - gpio_free(setup->dem1_pin); - gpio_free(setup->pdad_pin); - gpio_free(setup->pdda_pin); -} - #ifdef CONFIG_PM static int pcm3008_soc_suspend(struct snd_soc_codec *codec) { @@ -103,49 +95,37 @@ static int pcm3008_codec_probe(struct platform_device *pdev) */ /* Configure DEM0 GPIO (turning OFF DAC De-emphasis). */ - ret = gpio_request(setup->dem0_pin, "codec_dem0"); - if (ret == 0) - ret = gpio_direction_output(setup->dem0_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->dem0_pin, + GPIOF_OUT_INIT_HIGH, "codec_dem0"); if (ret != 0) - goto gpio_err; + return ret; /* Configure DEM1 GPIO (turning OFF DAC De-emphasis). */ - ret = gpio_request(setup->dem1_pin, "codec_dem1"); - if (ret == 0) - ret = gpio_direction_output(setup->dem1_pin, 0); + ret = devm_gpio_request_one(&pdev->dev, setup->dem1_pin, + GPIOF_OUT_INIT_LOW, "codec_dem1"); if (ret != 0) - goto gpio_err; + return ret; /* Configure PDAD GPIO. */ - ret = gpio_request(setup->pdad_pin, "codec_pdad"); - if (ret == 0) - ret = gpio_direction_output(setup->pdad_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->pdad_pin, + GPIOF_OUT_INIT_HIGH, "codec_pdad"); if (ret != 0) - goto gpio_err; + return ret; /* Configure PDDA GPIO. */ - ret = gpio_request(setup->pdda_pin, "codec_pdda"); - if (ret == 0) - ret = gpio_direction_output(setup->pdda_pin, 1); + ret = devm_gpio_request_one(&pdev->dev, setup->pdda_pin, + GPIOF_OUT_INIT_HIGH, "codec_pdda"); if (ret != 0) - goto gpio_err; + return ret; return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm3008, &pcm3008_dai, 1); - -gpio_err: - pcm3008_gpio_free(setup); - return ret; } static int pcm3008_codec_remove(struct platform_device *pdev) { - struct pcm3008_setup_data *setup = pdev->dev.platform_data; - snd_soc_unregister_codec(&pdev->dev); - pcm3008_gpio_free(setup); - return 0; } -- cgit v1.2.3 From 57e265c8d71fb94c130bfb31f589cc9e97fb3928 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Jul 2013 20:46:34 +0100 Subject: ASoC: wm8994: Move runtime PM init to platform device init As well as being better style this allows the device to idle when there is no audio card instantaited which is probably what we want. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1d4b1ec66e36..02c320f71cdf 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4014,9 +4014,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->micdet_irq = control->pdata.micdet_irq; - pm_runtime_enable(codec->dev); - pm_runtime_idle(codec->dev); - /* By default use idle_bias_off, will override for WM8994 */ codec->dapm.idle_bias_off = 1; @@ -4389,8 +4386,6 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); - pm_runtime_disable(codec->dev); - for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i, &wm8994->fll_locked[i]); @@ -4449,6 +4444,9 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994, wm8994_dai, ARRAY_SIZE(wm8994_dai)); } @@ -4456,6 +4454,8 @@ static int wm8994_probe(struct platform_device *pdev) static int wm8994_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + return 0; } -- cgit v1.2.3 From cb23e852aabb50f5083fb734c2067220d087d26e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 4 Jul 2013 20:01:01 -0300 Subject: ASoC: sglt5000: Provide the reg_stride field sgtl5000 has 16-bit registers, and only even numbers are valid for its registers addresses. Let regmap knows about this feature by specifying the 'reg_stride' field, so that it can access only the valid registers. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559dc92c..7c99f3ccb1c6 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1470,6 +1470,7 @@ static struct snd_soc_codec_driver sgtl5000_driver = { static const struct regmap_config sgtl5000_regmap = { .reg_bits = 16, .val_bits = 16, + .reg_stride = 2, .max_register = SGTL5000_MAX_REG_OFFSET, .volatile_reg = sgtl5000_volatile, -- cgit v1.2.3 From f2c4fa655f7139a181a6d6db99a49cab96ed0337 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 13:36:05 +0100 Subject: ASoC: tlv320aic3x: Add compatible strings for specific devices The driver supports a range of devices but currently doesn't allow those device names to be used for enumeration on DT. Add the currently listed I2C IDs as compatible strings. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e5b926883131..c9bb760f405f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1582,6 +1582,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) #if defined(CONFIG_OF) static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, + { .compatible = "ti,tlv320aic33" }, + { .compatible = "ti,tlv320aic3007" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- cgit v1.2.3 From cbaa56896146cbb5ab54bd65f98d020af282e6c6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Jul 2013 13:39:52 +0100 Subject: ASoC: tlv320aic3x: List tlv320aic3106 as a supported device Currently there is no specific handling for it but the tlv320aic3106 is supported using this driver. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c9bb760f405f..cad4fb17a933 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1492,6 +1492,7 @@ static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic3x", AIC3X_MODEL_3X }, { "tlv320aic33", AIC3X_MODEL_33 }, { "tlv320aic3007", AIC3X_MODEL_3007 }, + { "tlv320aic3106", AIC3X_MODEL_3X }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1584,6 +1585,7 @@ static const struct of_device_id tlv320aic3x_of_match[] = { { .compatible = "ti,tlv320aic3x", }, { .compatible = "ti,tlv320aic33" }, { .compatible = "ti,tlv320aic3007" }, + { .compatible = "ti,tlv320aic3106" }, {}, }; MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match); -- cgit v1.2.3 From c6c0925ea32d37696da7d71631a4a0c999f2094f Mon Sep 17 00:00:00 2001 From: Rongjun Ying Date: Wed, 17 Jul 2013 14:12:16 +0800 Subject: ASoC: hdmi-codec: let the driver support HDMI sink Devices like mobilephones, computers are typically used as HDMI sources, but devices like TV, navigators will be HDMI sinks. for auto scenerios, In-Vehicle Infotainment(IVI) can be HDMI sink to display movies from mobilephones. Signed-off-by: Rongjun Ying Signed-off-by: Barry Song Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index 2bcae2b40c92..f0986b9f1939 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -37,6 +37,17 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, + }, + }; static int hdmi_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From e94a093c1c7a1b06fa574c99f69ed594e0c52ff2 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Thu, 18 Jul 2013 14:38:20 +0800 Subject: ASoC: wm8904: fix the typo error for LINER Mux fix the typo error, from "LINEL Mux" to "LINER Mux" Signed-off-by: Bo Shen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4c9fb142cb2d..91dfbfeda6f8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1012,7 +1012,7 @@ static const struct soc_enum liner_enum = SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); static const struct snd_kcontrol_new liner_mux = - SOC_DAPM_ENUM("LINEL Mux", liner_enum); + SOC_DAPM_ENUM("LINER Mux", liner_enum); static const char *sidetone_text[] = { "None", "Left", "Right" -- cgit v1.2.3 From 2df7c6aad63f432befe51ac3144a96b37fa5b4ba Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:43:00 +0100 Subject: ASoC: max98090: Use power efficient workqueue None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/max98090.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ad5313f98f28..0569a4c3ae00 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2084,8 +2084,9 @@ static irqreturn_t max98090_interrupt(int irq, void *data) pm_wakeup_event(codec->dev, 100); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); } if (active & M98090_DRCACT_MASK) @@ -2132,8 +2133,9 @@ int max98090_mic_detect(struct snd_soc_codec *codec, snd_soc_jack_report(max98090->jack, 0, SND_JACK_HEADSET | SND_JACK_BTN_0); - schedule_delayed_work(&max98090->jack_work, - msecs_to_jiffies(100)); + queue_delayed_work(system_power_efficient_wq, + &max98090->jack_work, + msecs_to_jiffies(100)); return 0; } -- cgit v1.2.3 From a14d982962c1c3caee99ddeea632d97fc851ea60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:43:19 +0100 Subject: ASoC: sta32x: Use power efficient workqueue None of the delayed work the driver schedules has particularly short delays and it is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/sta32x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index cfb55fe35e98..06edb396e733 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -363,16 +363,18 @@ static void sta32x_watchdog(struct work_struct *work) } if (!sta32x->shutdown) - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } static void sta32x_watchdog_start(struct sta32x_priv *sta32x) { if (sta32x->pdata->needs_esd_watchdog) { sta32x->shutdown = 0; - schedule_delayed_work(&sta32x->watchdog_work, - round_jiffies_relative(HZ)); + queue_delayed_work(system_power_efficient_wq, + &sta32x->watchdog_work, + round_jiffies_relative(HZ)); } } -- cgit v1.2.3 From 76394509f579cd4292b076f708da49404be6af37 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:44:03 +0100 Subject: ASoC: twl6040: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/twl6040.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621ddc332d..caf8784e7716 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.2.3 From 2c5920a787e68e0a61886d24b05cdbde344c4b0c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:45:40 +0100 Subject: ASoC: wm8350: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8350.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 0e8b3aaf6c8d..af1318ddb062 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1301,7 +1301,8 @@ static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpl.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpl.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } @@ -1318,7 +1319,8 @@ static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data) if (device_may_wakeup(wm8350->dev)) pm_wakeup_event(wm8350->dev, 250); - schedule_delayed_work(&priv->hpr.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hpr.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.2.3 From ec1d648d6c6589986072913a7d45b1cef49eb4b0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:46:24 +0100 Subject: ASoC: wm8753: Use power efficient workqueue The work used to allow the capcitors to ramp is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8753.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 0a4ab4c423d1..d96ebf52d953 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1456,8 +1456,9 @@ static int wm8753_resume(struct snd_soc_codec *codec) if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE); codec->dapm.bias_level = SND_SOC_BIAS_ON; - schedule_delayed_work(&codec->dapm.delayed_work, - msecs_to_jiffies(caps_charge)); + queue_delayed_work(system_power_efficient_wq, + &codec->dapm.delayed_work, + msecs_to_jiffies(caps_charge)); } return 0; -- cgit v1.2.3 From 68defe585f333223f0f3733340136d1b02003062 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:47:02 +0100 Subject: ASoC: wm8994: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8994.c | 23 ++++++++++++++--------- 1 file changed, 14 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 02c320f71cdf..24131a7f9390 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -819,8 +819,9 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w, * don't want false reports. */ if (wm8994->jackdet && !wm8994->clk_has_run) { - schedule_delayed_work(&wm8994->jackdet_bootstrap, - msecs_to_jiffies(1000)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->jackdet_bootstrap, + msecs_to_jiffies(1000)); wm8994->clk_has_run = true; } break; @@ -3487,7 +3488,8 @@ static irqreturn_t wm8994_mic_irq(int irq, void *data) pm_wakeup_event(codec->dev, 300); - schedule_delayed_work(&priv->mic_work, msecs_to_jiffies(250)); + queue_delayed_work(system_power_efficient_wq, + &priv->mic_work, msecs_to_jiffies(250)); return IRQ_HANDLED; } @@ -3575,8 +3577,9 @@ static void wm8958_mic_id(void *data, u16 status) /* If nothing present then clear our statuses */ dev_dbg(codec->dev, "Detected open circuit\n"); - schedule_delayed_work(&wm8994->open_circuit_work, - msecs_to_jiffies(2500)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->open_circuit_work, + msecs_to_jiffies(2500)); return; } @@ -3690,8 +3693,9 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) WM1811_JACKDET_DB, 0); delay = control->pdata.micdet_delay; - schedule_delayed_work(&wm8994->mic_work, - msecs_to_jiffies(delay)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->mic_work, + msecs_to_jiffies(delay)); } else { dev_dbg(codec->dev, "Jack not detected\n"); @@ -3940,8 +3944,9 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) id_delay = wm8994->wm8994->pdata.mic_id_delay; if (wm8994->mic_detecting) - schedule_delayed_work(&wm8994->mic_complete_work, - msecs_to_jiffies(id_delay)); + queue_delayed_work(system_power_efficient_wq, + &wm8994->mic_complete_work, + msecs_to_jiffies(id_delay)); else wm8958_button_det(codec, reg); -- cgit v1.2.3 From 444fc4b369f9341d2cbcffe2d1ffde4cad5b4945 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 19 Jul 2013 10:22:21 -0300 Subject: ASoC: wm8962: Do not call configure_bclk() inside wm8962_set_dai_sysclk() Currently after playing any audio file, we get the following error message: $ aplay clarinet.wav Playing WAVE 'clarinet.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ wm8962 0-001a: Unsupported sysclk ratio 544 This error message appears about 5 seconds after the audio playback has finished. Quoting Mark Brown [1]: "The issue here is triggered by the machine switching from the FLL to direct MCLK usage where the MCLK isn't generating a useful ratio. I suspect we should just kill the configure_bclk() in set_sysclk(), that one isn't safe as we can't reconfigure a live SYSCLK and it's probably the one that generates your warnings." Confirmed that the "Unsupported sysclk ratio" error message comes from wm8962_set_dai_sysclk(), so get rid of wm8962_configure_bclk() inside this function. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2013-July/064241.html Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e2de9ecfd641..8b8905dfc510 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2621,8 +2621,6 @@ static int wm8962_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, wm8962->sysclk_rate = freq; - wm8962_configure_bclk(codec); - return 0; } -- cgit v1.2.3 From a06e427d088d8a9b81defd42e6bae5f1cd69fc3f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:44:03 +0100 Subject: ASoC: twl6040: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar Acked-by: Peter Ujfalusi --- sound/soc/codecs/twl6040.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621ddc332d..caf8784e7716 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -429,7 +429,8 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) struct snd_soc_codec *codec = data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - schedule_delayed_work(&priv->hs_jack.work, msecs_to_jiffies(200)); + queue_delayed_work(system_power_efficient_wq, + &priv->hs_jack.work, msecs_to_jiffies(200)); return IRQ_HANDLED; } -- cgit v1.2.3 From da72c9619f4df033d431a0a4cee715cf14c78433 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Jul 2013 22:46:46 +0100 Subject: ASoC: wm8962: Use power efficient workqueue The accessory detect debounce work is not performance sensitive so let the scheduler run it wherever is most efficient rather than in a per CPU workqueue by using the system power efficient workqueue. Signed-off-by: Mark Brown Acked-by: Viresh Kumar --- sound/soc/codecs/wm8962.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8b8905dfc510..36782f067cc5 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3044,8 +3044,9 @@ static irqreturn_t wm8962_irq(int irq, void *data) pm_wakeup_event(dev, 300); - schedule_delayed_work(&wm8962->mic_work, - msecs_to_jiffies(250)); + queue_delayed_work(system_power_efficient_wq, + &wm8962->mic_work, + msecs_to_jiffies(250)); } return IRQ_HANDLED; -- cgit v1.2.3 From af2d8b5d95f12c36bfabe90c0879923efedefd2c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 22 Jul 2013 18:49:52 +0200 Subject: ASoC: Build adau1701 when SND_SOC_ALL_CODECS is selected The adau1701 driver was removed from SND_SOC_ALL_CODECS in commit commit ee8c7e9 ("ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency") due to the dependency on the SigmaDSP firmware loader which was only available on a limited set of platforms. This was fixed quite some time ago in commit 40216ce7 ("ASoC: Move SigmaDSP firmware loader to ASoC") though, so we can add the driver back again to SND_SOC_ALL_CODECS. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..adddb39f3c33 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -20,6 +20,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI + select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C -- cgit v1.2.3 From 96b9bc6174a030691a4a60b0117ba7718d2bb27a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 22 Jul 2013 18:49:53 +0200 Subject: ASoC: adau1701: Add adau1702 and adau1401(a) device ids Both the adau1702 and the adau1401(a) are register compatible to the adau1701, so add them to adau1701 driver's device id table. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index d1124a5b3471..44d8a95d9ec3 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -734,7 +734,10 @@ static int adau1701_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id adau1701_i2c_id[] = { + { "adau1401", 0 }, + { "adau1401a", 0 }, { "adau1701", 0 }, + { "adau1702", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, adau1701_i2c_id); -- cgit v1.2.3 From 5d99d778495cb02eafd38292f462c4466fc7189f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 24 Jul 2013 15:27:39 +0200 Subject: ASoC: tlv320aic3x: Use snd_soc_dapm_mixer_update_power Use snd_soc_dapm_mixer_update_power() instead of reimplementing its functionality. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 49 ++++++++++++++---------------------------- 1 file changed, 16 insertions(+), 33 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e5b926883131..1325c0c0df50 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -147,10 +147,9 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - unsigned short val, val_mask; - int ret; - struct snd_soc_dapm_path *path; - int found = 0; + unsigned short val; + struct snd_soc_dapm_update update; + int connect, change; val = (ucontrol->value.integer.value[0] & mask); @@ -158,42 +157,26 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, if (val) val = mask; + connect = !!val; + if (invert) val = mask - val; - val_mask = mask << shift; - val = val << shift; - - mutex_lock(&widget->codec->mutex); - if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) { - /* find dapm widget path assoc with kcontrol */ - list_for_each_entry(path, &widget->dapm->card->paths, list) { - if (path->kcontrol != kcontrol) - continue; + mask <<= shift; + val <<= shift; - /* found, now check type */ - found = 1; - if (val) - /* new connection */ - path->connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - path->connect = invert ? 1 : 0; + change = snd_soc_test_bits(widget->codec, val, mask, reg); + if (change) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - dapm_mark_dirty(path->source, "tlv320aic3x source"); - dapm_mark_dirty(path->sink, "tlv320aic3x sink"); - - break; - } + snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect, + &update); } - mutex_unlock(&widget->codec->mutex); - - if (found) - snd_soc_dapm_sync(widget->dapm); - - ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); - return ret; + return change; } /* -- cgit v1.2.3 From ba51cbb8206cdba789a1f65b06526bb20f51d594 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 25 Jul 2013 19:40:17 +0300 Subject: ASoC: adau1701: type bug with ADAU1707_CLKDIV_UNSET ADAU1707_CLKDIV_UNSET is always compared against an unsigned int and not an unsigned long. The current tests are always false. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/codecs/adau1701.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 44d8a95d9ec3..2c102522bbbc 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -91,7 +91,7 @@ #define ADAU1701_OSCIPOW_OPD 0x04 #define ADAU1701_DACSET_DACINIT 1 -#define ADAU1707_CLKDIV_UNSET (-1UL) +#define ADAU1707_CLKDIV_UNSET (-1U) #define ADAU1701_FIRMWARE "adau1701.bin" -- cgit v1.2.3 From 07ccc0f4f190070aaba8fb587307f7fefad97981 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 28 Jul 2013 18:45:28 +0200 Subject: ASoC: lm4857: Use table based setup for DAPM and controls Let the ASoC core take care of registering the DAPM widget and routes as well as the controls. This makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 26 +++++++------------------- 1 file changed, 7 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 9f9f59573f72..5ea2ed053eb7 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -174,28 +174,9 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { static int lm4857_probe(struct snd_soc_codec *codec) { struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; codec->control_data = lm4857->i2c; - ret = snd_soc_add_codec_controls(codec, lm4857_controls, - ARRAY_SIZE(lm4857_controls)); - if (ret) - return ret; - - ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets, - ARRAY_SIZE(lm4857_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, lm4857_routes, - ARRAY_SIZE(lm4857_routes)); - if (ret) - return ret; - - snd_soc_dapm_new_widgets(dapm); - return 0; } @@ -207,6 +188,13 @@ static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { .reg_word_size = sizeof(uint8_t), .reg_cache_default = lm4857_default_regs, .set_bias_level = lm4857_set_bias_level, + + .controls = lm4857_controls, + .num_controls = ARRAY_SIZE(lm4857_controls), + .dapm_widgets = lm4857_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(lm4857_dapm_widgets), + .dapm_routes = lm4857_routes, + .num_dapm_routes = ARRAY_SIZE(lm4857_routes), }; static int lm4857_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3 From 9b2709687a81297bca53f98100b6f13cd5e05b44 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sun, 28 Jul 2013 18:45:29 +0200 Subject: ASoC: lm4857: Convert to regmap Use regmap for IO for the lm4857 driver instead of open-coding the IO read/write functions. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 85 +++++++++++++++-------------------------------- 1 file changed, 26 insertions(+), 59 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 5ea2ed053eb7..0e5743ea79df 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include @@ -23,12 +24,15 @@ #include struct lm4857 { - struct i2c_client *i2c; + struct regmap *regmap; uint8_t mode; }; -static const uint8_t lm4857_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, +static const struct reg_default lm4857_default_regs[] = { + { 0x0, 0x00 }, + { 0x1, 0x00 }, + { 0x2, 0x00 }, + { 0x3, 0x00 }, }; /* The register offsets in the cache array */ @@ -42,39 +46,6 @@ static const uint8_t lm4857_default_regs[] = { #define LM4857_WAKEUP 5 #define LM4857_EPGAIN 4 -static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - uint8_t data; - int ret; - - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return ret; - - data = (reg << 6) | value; - ret = i2c_master_send(codec->control_data, &data, 1); - if (ret != 1) { - dev_err(codec->dev, "Failed to write register: %d\n", ret); - return ret; - } - - return 0; -} - -static unsigned int lm4857_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - unsigned int val; - int ret; - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret) - return -1; - - return val; -} - static int lm4857_get_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -96,7 +67,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, lm4857->mode = value; if (codec->dapm.bias_level == SND_SOC_BIAS_ON) - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, value + 6); return 1; } @@ -108,10 +79,11 @@ static int lm4857_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, + lm4857->mode + 6); break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0); + regmap_update_bits(lm4857->regmap, LM4857_CTRL, 0x0F, 0); break; default: break; @@ -171,22 +143,7 @@ static const struct snd_soc_dapm_route lm4857_routes[] = { {"EP", NULL, "IN"}, }; -static int lm4857_probe(struct snd_soc_codec *codec) -{ - struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec); - - codec->control_data = lm4857->i2c; - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { - .write = lm4857_write, - .read = lm4857_read, - .probe = lm4857_probe, - .reg_cache_size = ARRAY_SIZE(lm4857_default_regs), - .reg_word_size = sizeof(uint8_t), - .reg_cache_default = lm4857_default_regs, .set_bias_level = lm4857_set_bias_level, .controls = lm4857_controls, @@ -197,11 +154,21 @@ static struct snd_soc_codec_driver soc_codec_dev_lm4857 = { .num_dapm_routes = ARRAY_SIZE(lm4857_routes), }; +static const struct regmap_config lm4857_regmap_config = { + .val_bits = 6, + .reg_bits = 2, + + .max_register = LM4857_CTRL, + + .cache_type = REGCACHE_FLAT, + .reg_defaults = lm4857_default_regs, + .num_reg_defaults = ARRAY_SIZE(lm4857_default_regs), +}; + static int lm4857_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct lm4857 *lm4857; - int ret; lm4857 = devm_kzalloc(&i2c->dev, sizeof(*lm4857), GFP_KERNEL); if (!lm4857) @@ -209,11 +176,11 @@ static int lm4857_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, lm4857); - lm4857->i2c = i2c; - - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); + lm4857->regmap = devm_regmap_init_i2c(i2c, &lm4857_regmap_config); + if (IS_ERR(lm4857->regmap)) + return PTR_ERR(lm4857->regmap); - return ret; + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0); } static int lm4857_i2c_remove(struct i2c_client *i2c) -- cgit v1.2.3 From 81ad93ecfda64cb37129d29adb384affd0d0fa5b Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Mon, 29 Jul 2013 13:51:59 +0100 Subject: ASoC: wm_adsp: Simplify kcontrol handling Get rid off the wm_coeff struct and the wm_coeff_add_kcontrol() function. We are now using the snd_soc_card_kcontrol() function to get the kcontrol pointers. No need to call into ALSA code to register the kcontrols. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 103 ++++++++++++++------------------------------- sound/soc/codecs/wm_adsp.h | 2 +- 2 files changed, 32 insertions(+), 73 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 05252ac936a3..3168224bc104 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -225,15 +225,9 @@ struct wm_coeff_ctl_ops { struct snd_ctl_elem_info *uinfo); }; -struct wm_coeff { - struct device *dev; - struct list_head ctl_list; - struct regmap *regmap; -}; - struct wm_coeff_ctl { const char *name; - struct snd_card *card; + struct snd_soc_card *card; struct wm_adsp_alg_region region; struct wm_coeff_ctl_ops ops; struct wm_adsp *adsp; @@ -378,7 +372,6 @@ static int wm_coeff_info(struct snd_kcontrol *kcontrol, static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, const void *buf, size_t len) { - struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; struct wm_adsp_alg_region *region = &ctl->region; const struct wm_adsp_region *mem; @@ -401,7 +394,7 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, if (!scratch) return -ENOMEM; - ret = regmap_raw_write(wm_coeff->regmap, reg, scratch, + ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { adsp_err(adsp, "Failed to write %zu bytes to %x\n", @@ -434,7 +427,6 @@ static int wm_coeff_put(struct snd_kcontrol *kcontrol, static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, void *buf, size_t len) { - struct wm_coeff *wm_coeff= snd_kcontrol_chip(kcontrol); struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kcontrol->private_value; struct wm_adsp_alg_region *region = &ctl->region; const struct wm_adsp_region *mem; @@ -457,7 +449,7 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, if (!scratch) return -ENOMEM; - ret = regmap_raw_read(wm_coeff->regmap, reg, scratch, ctl->len); + ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); if (ret) { adsp_err(adsp, "Failed to read %zu bytes from %x\n", ctl->len, reg); @@ -481,37 +473,18 @@ static int wm_coeff_get(struct snd_kcontrol *kcontrol, return 0; } -static int wm_coeff_add_kcontrol(struct wm_coeff *wm_coeff, - struct wm_coeff_ctl *ctl, - const struct snd_kcontrol_new *kctl) -{ - int ret; - struct snd_kcontrol *kcontrol; - - kcontrol = snd_ctl_new1(kctl, wm_coeff); - ret = snd_ctl_add(ctl->card, kcontrol); - if (ret < 0) { - dev_err(wm_coeff->dev, "Failed to add %s: %d\n", - kctl->name, ret); - return ret; - } - ctl->kcontrol = kcontrol; - return 0; -} - struct wmfw_ctl_work { - struct wm_coeff *wm_coeff; + struct wm_adsp *adsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; -static int wmfw_add_ctl(struct wm_coeff *wm_coeff, - struct wm_coeff_ctl *ctl) +static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; int ret; - if (!wm_coeff || !ctl || !ctl->name || !ctl->card) + if (!ctl || !ctl->name || !ctl->card) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); @@ -525,14 +498,17 @@ static int wmfw_add_ctl(struct wm_coeff *wm_coeff, kcontrol->put = wm_coeff_put; kcontrol->private_value = (unsigned long)ctl; - ret = wm_coeff_add_kcontrol(wm_coeff, - ctl, kcontrol); + ret = snd_soc_add_card_controls(ctl->card, + kcontrol, 1); if (ret < 0) goto err_kcontrol; kfree(kcontrol); - list_add(&ctl->list, &wm_coeff->ctl_list); + ctl->kcontrol = snd_soc_card_get_kcontrol(ctl->card, + ctl->name); + + list_add(&ctl->list, &adsp->ctl_list); return 0; err_kcontrol: @@ -753,13 +729,12 @@ out: return ret; } -static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) +static int wm_coeff_init_control_caches(struct wm_adsp *adsp) { struct wm_coeff_ctl *ctl; int ret; - list_for_each_entry(ctl, &wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &adsp->ctl_list, list) { if (!ctl->enabled || ctl->set) continue; ret = wm_coeff_read_control(ctl->kcontrol, @@ -772,13 +747,12 @@ static int wm_coeff_init_control_caches(struct wm_coeff *wm_coeff) return 0; } -static int wm_coeff_sync_controls(struct wm_coeff *wm_coeff) +static int wm_coeff_sync_controls(struct wm_adsp *adsp) { struct wm_coeff_ctl *ctl; int ret; - list_for_each_entry(ctl, &wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &adsp->ctl_list, list) { if (!ctl->enabled) continue; if (ctl->set) { @@ -799,7 +773,7 @@ static void wm_adsp_ctl_work(struct work_struct *work) struct wmfw_ctl_work, work); - wmfw_add_ctl(ctl_work->wm_coeff, ctl_work->ctl); + wmfw_add_ctl(ctl_work->adsp, ctl_work->ctl); kfree(ctl_work); } @@ -842,7 +816,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, snprintf(name, PAGE_SIZE, "DSP%d %s %x", dsp->num, region_name, region->alg); - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, + list_for_each_entry(ctl, &dsp->ctl_list, list) { if (!strcmp(ctl->name, name)) { if (!ctl->enabled) @@ -866,7 +840,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; - ctl->card = codec->card->snd_card; + ctl->card = codec->card; ctl->adsp = dsp; ctl->len = region->len; @@ -882,7 +856,7 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, goto err_ctl_cache; } - ctl_work->wm_coeff = dsp->wm_coeff; + ctl_work->adsp = dsp; ctl_work->ctl = ctl; INIT_WORK(&ctl_work->work, wm_adsp_ctl_work); schedule_work(&ctl_work->work); @@ -1434,12 +1408,12 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, goto err; /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp->wm_coeff); + ret = wm_coeff_init_control_caches(dsp); if (ret != 0) goto err; /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp->wm_coeff); + ret = wm_coeff_sync_controls(dsp); if (ret != 0) goto err; @@ -1460,10 +1434,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, ADSP1_SYS_ENA, 0); - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - } break; default: @@ -1591,12 +1563,12 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, goto err; /* Initialize caches for enabled and unset controls */ - ret = wm_coeff_init_control_caches(dsp->wm_coeff); + ret = wm_coeff_init_control_caches(dsp); if (ret != 0) goto err; /* Sync set controls */ - ret = wm_coeff_sync_controls(dsp->wm_coeff); + ret = wm_coeff_sync_controls(dsp); if (ret != 0) goto err; @@ -1637,10 +1609,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, ret); } - list_for_each_entry(ctl, &dsp->wm_coeff->ctl_list, - list) { + list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; - } while (!list_empty(&dsp->alg_regions)) { alg_region = list_first_entry(&dsp->alg_regions, @@ -1679,49 +1649,38 @@ int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs) } INIT_LIST_HEAD(&adsp->alg_regions); - - adsp->wm_coeff = kzalloc(sizeof(*adsp->wm_coeff), - GFP_KERNEL); - if (!adsp->wm_coeff) - return -ENOMEM; - adsp->wm_coeff->regmap = adsp->regmap; - adsp->wm_coeff->dev = adsp->dev; - INIT_LIST_HEAD(&adsp->wm_coeff->ctl_list); + INIT_LIST_HEAD(&adsp->ctl_list); if (dvfs) { adsp->dvfs = devm_regulator_get(adsp->dev, "DCVDD"); if (IS_ERR(adsp->dvfs)) { ret = PTR_ERR(adsp->dvfs); dev_err(adsp->dev, "Failed to get DCVDD: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_enable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to enable DCVDD: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_set_voltage(adsp->dvfs, 1200000, 1800000); if (ret != 0) { dev_err(adsp->dev, "Failed to initialise DVFS: %d\n", ret); - goto out_coeff; + return ret; } ret = regulator_disable(adsp->dvfs); if (ret != 0) { dev_err(adsp->dev, "Failed to disable DCVDD: %d\n", ret); - goto out_coeff; + return ret; } } return 0; - -out_coeff: - kfree(adsp->wm_coeff); - return ret; } EXPORT_SYMBOL_GPL(wm_adsp2_init); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 9f922c82536c..64087fb1cdec 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -57,7 +57,7 @@ struct wm_adsp { struct regulator *dvfs; - struct wm_coeff *wm_coeff; + struct list_head ctl_list; }; #define WM_ADSP1(wname, num) \ -- cgit v1.2.3 From eee5d7f99ae95059e1a3d1cfa2dea3ed8dbd94ac Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 29 Jul 2013 17:13:57 +0200 Subject: ASoC: dapm: Add a helper to get the CODEC for DAPM kcontrol We use the same 3 lines to get the CODEC for a kcontrol in a quite a few places. This patch puts them into a common helper function. Having this encapsulated in a helper function will also make it more easier to eventually change the data layout of the kcontrol's private data. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 7 +++---- sound/soc/codecs/twl6040.c | 4 +--- sound/soc/codecs/wm8903.c | 4 +--- sound/soc/codecs/wm8994.c | 4 +--- sound/soc/codecs/wm8995.c | 5 +---- sound/soc/codecs/wm_hubs.c | 8 ++------ 6 files changed, 9 insertions(+), 23 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 1325c0c0df50..fec0db04262d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -138,8 +138,7 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; @@ -165,14 +164,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, mask <<= shift; val <<= shift; - change = snd_soc_test_bits(widget->codec, val, mask, reg); + change = snd_soc_test_bits(codec, val, mask, reg); if (change) { update.kcontrol = kcontrol; update.reg = reg; update.mask = mask; update.val = val; - snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, connect, + snd_soc_dapm_mixer_update_power(&codec->dapm, kcontrol, connect, &update); } diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 44621ddc332d..d6c5bf14179a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -437,9 +437,7 @@ static irqreturn_t twl6040_audio_handler(int irq, void *data) static int twl6040_soc_dapm_put_vibra_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int val; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index fa24cedee687..eebcb1da3b7b 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -364,9 +364,7 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, static int wm8903_class_w_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); u16 reg; int ret; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index ba832b77c543..eee2a01f2691 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1437,9 +1437,7 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec = w->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 90a65c427541..da2899e6c401 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -549,12 +549,9 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source, static int wm8995_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *w = wlist->widgets[0]; - struct snd_soc_codec *codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; - codec = w->codec; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); wm8995_update_class_w(codec); return ret; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2d9e099415a5..8b50e5958de5 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -699,9 +699,7 @@ EXPORT_SYMBOL_GPL(wm_hubs_update_class_w); static int class_w_put_volsw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol); @@ -721,9 +719,7 @@ static int class_w_put_volsw(struct snd_kcontrol *kcontrol, static int class_w_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol); - struct snd_soc_dapm_widget *widget = wlist->widgets[0]; - struct snd_soc_codec *codec = widget->codec; + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); int ret; ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); -- cgit v1.2.3 From 113591e477acb6b6dbc186ad2ee29a2502e68c33 Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 30 Jul 2013 11:18:52 +0100 Subject: ASoC: uda134x: fix codec driver by converting to DAPM For some reason, the DAC/ADCs are not being powered up when I try and use the UDA1341 driver; this used to work. Looking back in the git history, I don't see anything obvious which would cause this regression. However, from dumping the register writes, it seems that the codec is powered down, and nothing calls set_bias_level to wake the codec up. Moreover, this driver hasn't had DAPM support added to it, which prevents platform drivers from taking advantage of DAPMs facilities. So, let's add DAPM support to the driver. As we move the power control for the DAC/ADC into DAPM, we no longer need it in set_bias_level() - this function just becomes a way to manipulate the power control and sync the register cache with the hardware at the appropriate point. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/uda134x.c | 88 ++++++++++++++++++++++++++++------------------ 1 file changed, 53 insertions(+), 35 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 6d0aa44c3757..c94d4c1e3dac 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -325,7 +325,6 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u8 reg; struct uda134x_platform_data *pd = codec->control_data; int i; u8 *cache = codec->reg_cache; @@ -334,23 +333,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - /* ADC, DAC on */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg | 0x03); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_PREPARE: /* power on */ @@ -362,23 +344,6 @@ static int uda134x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_STANDBY: - /* ADC, DAC power off */ - switch (pd->model) { - case UDA134X_UDA1340: - case UDA134X_UDA1344: - case UDA134X_UDA1345: - reg = uda134x_read_reg_cache(codec, UDA134X_DATA011); - uda134x_write(codec, UDA134X_DATA011, reg & ~(0x03)); - break; - case UDA134X_UDA1341: - reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); - uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); - break; - default: - printk(KERN_ERR "UDA134X SoC codec: " - "unsupported model %d\n", pd->model); - return -EINVAL; - } break; case SND_SOC_BIAS_OFF: /* power off */ @@ -450,6 +415,37 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; +/* UDA1341 has the DAC/ADC power down in STATUS1 */ +static const struct snd_soc_dapm_widget uda1341_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_STATUS1, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_STATUS1, 1, 0), +}; + +/* UDA1340/4/5 has the DAC/ADC pwoer down in DATA0 11 */ +static const struct snd_soc_dapm_widget uda1340_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", UDA134X_DATA011, 0, 0), + SND_SOC_DAPM_ADC("ADC", "Capture", UDA134X_DATA011, 1, 0), +}; + +/* Common DAPM widgets */ +static const struct snd_soc_dapm_widget uda134x_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("VINL1"), + SND_SOC_DAPM_INPUT("VINR1"), + SND_SOC_DAPM_INPUT("VINL2"), + SND_SOC_DAPM_INPUT("VINR2"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route uda134x_dapm_routes[] = { + { "ADC", NULL, "VINL1" }, + { "ADC", NULL, "VINR1" }, + { "ADC", NULL, "VINL2" }, + { "ADC", NULL, "VINR2" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, +}; + static const struct snd_soc_dai_ops uda134x_dai_ops = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, @@ -485,6 +481,8 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) { struct uda134x_priv *uda134x; struct uda134x_platform_data *pd = codec->card->dev->platform_data; + const struct snd_soc_dapm_widget *widgets; + unsigned num_widgets; int ret; @@ -526,6 +524,22 @@ static int uda134x_soc_probe(struct snd_soc_codec *codec) else uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + if (pd->model == UDA134X_UDA1341) { + widgets = uda1341_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1341_dapm_widgets); + } else { + widgets = uda1340_dapm_widgets; + num_widgets = ARRAY_SIZE(uda1340_dapm_widgets); + } + + ret = snd_soc_dapm_new_controls(&codec->dapm, widgets, num_widgets); + if (ret) { + printk(KERN_ERR "%s failed to register dapm controls: %d", + __func__, ret); + kfree(uda134x); + return ret; + } + switch (pd->model) { case UDA134X_UDA1340: case UDA134X_UDA1344: @@ -599,6 +613,10 @@ static struct snd_soc_codec_driver soc_codec_dev_uda134x = { .read = uda134x_read_reg_cache, .write = uda134x_write, .set_bias_level = uda134x_set_bias_level, + .dapm_widgets = uda134x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda134x_dapm_widgets), + .dapm_routes = uda134x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(uda134x_dapm_routes), }; static int uda134x_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From 0890c2b7be08b0928f7f507e371918205a0312f7 Mon Sep 17 00:00:00 2001 From: Richard Genoud Date: Tue, 30 Jul 2013 11:59:45 +0200 Subject: ASoC: wm8731: add rates constraints Depending on the mclk (or crystal) selected, the wm8731 codec have some constraints on its data sampling rates: e.g. with a 12.288MHz or 18.432MHz crystal, the authorized rates are 8KHz, 32KHz, 48KHz and 96KHz. Signed-off-by: Richard Genoud Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 60 ++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 57 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5276062d6c79..456bb8c6d759 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -45,6 +45,7 @@ static const char *wm8731_supply_names[WM8731_NUM_SUPPLIES] = { struct wm8731_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; + const struct snd_pcm_hw_constraint_list *constraints; unsigned int sysclk; int sysclk_type; int playback_fs; @@ -290,6 +291,36 @@ static const struct _coeff_div coeff_div[] = { {12000000, 88200, 136, 0xf, 0x1, 0x1}, }; +/* rates constraints */ +static const unsigned int wm8731_rates_12000000[] = { + 8000, 32000, 44100, 48000, 96000, 88200, +}; + +static const unsigned int wm8731_rates_12288000_18432000[] = { + 8000, 32000, 48000, 96000, +}; + +static const unsigned int wm8731_rates_11289600_16934400[] = { + 8000, 44100, 88200, +}; + +static const struct snd_pcm_hw_constraint_list wm8731_constraints_12000000 = { + .list = wm8731_rates_12000000, + .count = ARRAY_SIZE(wm8731_rates_12000000), +}; + +static const +struct snd_pcm_hw_constraint_list wm8731_constraints_12288000_18432000 = { + .list = wm8731_rates_12288000_18432000, + .count = ARRAY_SIZE(wm8731_rates_12288000_18432000), +}; + +static const +struct snd_pcm_hw_constraint_list wm8731_constraints_11289600_16934400 = { + .list = wm8731_rates_11289600_16934400, + .count = ARRAY_SIZE(wm8731_rates_11289600_16934400), +}; + static inline int get_coeff(int mclk, int rate) { int i; @@ -362,17 +393,26 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, } switch (freq) { - case 11289600: + case 0: + wm8731->constraints = NULL; + break; case 12000000: + wm8731->constraints = &wm8731_constraints_12000000; + break; case 12288000: - case 16934400: case 18432000: - wm8731->sysclk = freq; + wm8731->constraints = &wm8731_constraints_12288000_18432000; + break; + case 16934400: + case 11289600: + wm8731->constraints = &wm8731_constraints_11289600_16934400; break; default: return -EINVAL; } + wm8731->sysclk = freq; + snd_soc_dapm_sync(&codec->dapm); return 0; @@ -475,12 +515,26 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, return 0; } +static int wm8731_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(dai->codec); + + if (wm8731->constraints) + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + wm8731->constraints); + + return 0; +} + #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { + .startup = wm8731_startup, .hw_params = wm8731_hw_params, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, -- cgit v1.2.3 From db5ff9541b61ef8394bad0fb05508921b8c5b17b Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 31 Jul 2013 20:07:05 +0800 Subject: ASoC: spdif: Add S20_3LE and S24_LE support for dummy codec drivers Generally, S/PDIF supports 20bit and optional 24bit samples. Thus add these two formats for the dummy codec drivers. If one S/PDIF controller has its own limitation, its CPU DAI driver should set the supported format by its own circumstance, since the soc-pcm driver will use the intersection of cpu_dai's formats and codec_dai's formats. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_receiver.c | 2 ++ sound/soc/codecs/spdif_transmitter.c | 5 +++-- 2 files changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index e9d7881ed2c8..26d34744c677 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -25,6 +25,8 @@ #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) static struct snd_soc_codec_driver soc_codec_spdif_dir; diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 18280499fd55..efc3d88d7f8c 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -25,8 +25,9 @@ #define DRV_NAME "spdif-dit" #define STUB_RATES SNDRV_PCM_RATE_8000_96000 -#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE - +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_codec_driver soc_codec_spdif_dit; -- cgit v1.2.3 From d4780eec779c4e6d2fe5963dd2aee0a85d956122 Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 1 Aug 2013 09:53:45 +0100 Subject: ASoC: wm0010: Use DMA-safe memory for SPI transfers We should be allocating our buffers for the SPI transfers from the DMA zone. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 10adc4145d46..d5ebcb00019b 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -420,7 +420,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) xfer->codec = codec; list_add_tail(&xfer->list, &xfer_list); - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); @@ -429,7 +429,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) } xfer->t.rx_buf = out; - img = kzalloc(len, GFP_KERNEL); + img = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); @@ -523,14 +523,14 @@ static int wm0010_stage2_load(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Downloading %zu byte stage 2 loader\n", fw->size); /* Copy to local buffer first as vmalloc causes problems for dma */ - img = kzalloc(fw->size, GFP_KERNEL); + img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!img) { dev_err(codec->dev, "Failed to allocate image buffer\n"); ret = -ENOMEM; goto abort2; } - out = kzalloc(fw->size, GFP_KERNEL); + out = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate output buffer\n"); ret = -ENOMEM; @@ -670,14 +670,14 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = -ENOMEM; len = pll_rec.length + 8; - out = kzalloc(len, GFP_KERNEL); + out = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!out) { dev_err(codec->dev, "Failed to allocate RX buffer\n"); goto abort; } - img_swap = kzalloc(len, GFP_KERNEL); + img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); if (!img_swap) { dev_err(codec->dev, "Failed to allocate image buffer\n"); -- cgit v1.2.3 From 95169d080fcaad6c990ce3602d9b3d38753b1fa4 Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Thu, 1 Aug 2013 11:14:58 +0200 Subject: ASoC: Add PCM1681 codec driver. PCM1681 can be controlled via I2C, SPI or in bootstrap mode (no control mode). This code add support only for I2C mode. Signed-off-by: Marek Belisko Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1681.c | 313 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 319 insertions(+) create mode 100644 sound/soc/codecs/pcm1681.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..e2daf820179b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1681 if I2C select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1681 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..4a068d20444d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c new file mode 100644 index 000000000000..27da41b2dfcd --- /dev/null +++ b/sound/soc/codecs/pcm1681.c @@ -0,0 +1,313 @@ +/* + * PCM1681 ASoC codec driver + * + * Copyright (c) StreamUnlimited GmbH 2013 + * Marek Belisko + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define PCM1681_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +#define PCM1681_PCM_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1681_SOFT_MUTE_ALL 0xff +#define PCM1681_DEEMPH_RATE_MASK 0x18 +#define PCM1681_DEEMPH_MASK 0x01 + +#define PCM1681_ATT_CONTROL(X) (X <= 6 ? X : X + 9) /* Attenuation level */ +#define PCM1681_SOFT_MUTE 0x07 /* Soft mute control register */ +#define PCM1681_DAC_CONTROL 0x08 /* DAC operation control */ +#define PCM1681_FMT_CONTROL 0x09 /* Audio interface data format */ +#define PCM1681_DEEMPH_CONTROL 0x0a /* De-emphasis control */ +#define PCM1681_ZERO_DETECT_STATUS 0x0e /* Zero detect status reg */ + +static const struct reg_default pcm1681_reg_defaults[] = { + { 0x01, 0xff }, + { 0x02, 0xff }, + { 0x03, 0xff }, + { 0x04, 0xff }, + { 0x05, 0xff }, + { 0x06, 0xff }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x06 }, + { 0x0A, 0x00 }, + { 0x0B, 0xff }, + { 0x0C, 0x0f }, + { 0x0D, 0x00 }, + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, +}; + +static bool pcm1681_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x00) || (reg == 0x0f)); +} + +static bool pcm1681_writeable_reg(struct device *dev, unsigned register reg) +{ + return pcm1681_accessible_reg(dev, reg) && + (reg != PCM1681_ZERO_DETECT_STATUS); +} + +struct pcm1681_private { + struct regmap *regmap; + unsigned int format; + /* Current deemphasis status */ + unsigned int deemph; + /* Current rate for deemphasis control */ + unsigned int rate; +}; + +static const int pcm1681_deemph[] = { 44100, 48000, 32000 }; + +static int pcm1681_set_deemph(struct snd_soc_codec *codec) +{ + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int i = 0, val = -1, enable = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(pcm1681_deemph); i++) + if (pcm1681_deemph[i] == priv->rate) + val = i; + + if (val != -1) { + regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_RATE_MASK, val); + enable = 1; + } else + enable = 0; + + /* enable/disable deemphasis functionality */ + return regmap_update_bits(priv->regmap, PCM1681_DEEMPH_CONTROL, + PCM1681_DEEMPH_MASK, enable); +} + +static int pcm1681_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int pcm1681_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return pcm1681_set_deemph(codec); +} + +static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The PCM1681 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + priv->format = format; + + return 0; +} + +static int pcm1681_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val; + + if (mute) + val = PCM1681_SOFT_MUTE_ALL; + else + val = 0; + + return regmap_write(priv->regmap, PCM1681_SOFT_MUTE, val); +} + +static int pcm1681_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) + val = 0x00; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x03; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x04; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x05; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, PCM1681_FMT_CONTROL, 0x0f, val); + if (ret < 0) + return ret; + + return pcm1681_set_deemph(codec); +} + +static const struct snd_soc_dai_ops pcm1681_dai_ops = { + .set_fmt = pcm1681_set_dai_fmt, + .hw_params = pcm1681_hw_params, + .digital_mute = pcm1681_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); + +static const struct snd_kcontrol_new pcm1681_controls[] = { + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + PCM1681_ATT_CONTROL(1), PCM1681_ATT_CONTROL(2), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + PCM1681_ATT_CONTROL(3), PCM1681_ATT_CONTROL(4), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + PCM1681_ATT_CONTROL(5), PCM1681_ATT_CONTROL(6), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 7/8 Playback Volume", + PCM1681_ATT_CONTROL(7), PCM1681_ATT_CONTROL(8), 0, + 0x7f, 0, pcm1681_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + pcm1681_get_deemph, pcm1681_put_deemph), +}; + +struct snd_soc_dai_driver pcm1681_dai = { + .name = "pcm1681-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = PCM1681_PCM_RATES, + .formats = PCM1681_PCM_FORMATS, + }, + .ops = &pcm1681_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1681_dt_ids[] = { + { .compatible = "ti,pcm1681", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); +#endif + +static const struct regmap_config pcm1681_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .reg_defaults = pcm1681_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), + .writeable_reg = pcm1681_writeable_reg, + .readable_reg = pcm1681_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { + .controls = pcm1681_controls, + .num_controls = ARRAY_SIZE(pcm1681_controls), +}; + +static const struct i2c_device_id pcm1681_i2c_id[] = { + {"pcm1681", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, pcm1681_i2c_id); + +static int pcm1681_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + int ret; + struct pcm1681_private *priv; + + priv = devm_kzalloc(&client->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(client, &pcm1681_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(client, priv); + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_pcm1681, + &pcm1681_dai, 1); +} + +static int pcm1681_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver pcm1681_i2c_driver = { + .driver = { + .name = "pcm1681", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(pcm1681_dt_ids), + }, + .id_table = pcm1681_i2c_id, + .probe = pcm1681_i2c_probe, + .remove = pcm1681_i2c_remove, +}; + +module_i2c_driver(pcm1681_i2c_driver); + +MODULE_DESCRIPTION("Texas Instruments PCM1681 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Marek Belisko "); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 92bb4c32708ee3e1d6eb0e185d678dab35152daf Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Thu, 1 Aug 2013 11:11:28 +0100 Subject: ASoC: wm_adsp: Sanitize parameter passing No need to hold on to the `codec' pointer. We can use the `dsp' pointer and grab all the information we need from there. This makes the parameters for the functions a bit more sane and idiomatic. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 31 ++++++++++++++++--------------- sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 17 insertions(+), 15 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3168224bc104..b38f3506418f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -227,7 +227,6 @@ struct wm_coeff_ctl_ops { struct wm_coeff_ctl { const char *name; - struct snd_soc_card *card; struct wm_adsp_alg_region region; struct wm_coeff_ctl_ops ops; struct wm_adsp *adsp; @@ -484,7 +483,7 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) struct snd_kcontrol_new *kcontrol; int ret; - if (!ctl || !ctl->name || !ctl->card) + if (!ctl || !ctl->name) return -EINVAL; kcontrol = kzalloc(sizeof(*kcontrol), GFP_KERNEL); @@ -498,14 +497,14 @@ static int wmfw_add_ctl(struct wm_adsp *adsp, struct wm_coeff_ctl *ctl) kcontrol->put = wm_coeff_put; kcontrol->private_value = (unsigned long)ctl; - ret = snd_soc_add_card_controls(ctl->card, + ret = snd_soc_add_card_controls(adsp->card, kcontrol, 1); if (ret < 0) goto err_kcontrol; kfree(kcontrol); - ctl->kcontrol = snd_soc_card_get_kcontrol(ctl->card, + ctl->kcontrol = snd_soc_card_get_kcontrol(adsp->card, ctl->name); list_add(&ctl->list, &adsp->ctl_list); @@ -777,11 +776,10 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } -static int wm_adsp_create_control(struct snd_soc_codec *codec, +static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *region) { - struct wm_adsp *dsp = snd_soc_codec_get_drvdata(codec); struct wm_coeff_ctl *ctl; struct wmfw_ctl_work *ctl_work; char *name; @@ -840,7 +838,6 @@ static int wm_adsp_create_control(struct snd_soc_codec *codec, ctl->set = 0; ctl->ops.xget = wm_coeff_get; ctl->ops.xput = wm_coeff_put; - ctl->card = codec->card; ctl->adsp = dsp; ctl->len = region->len; @@ -877,7 +874,7 @@ err_name: return ret; } -static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) +static int wm_adsp_setup_algs(struct wm_adsp *dsp) { struct regmap *regmap = dsp->regmap; struct wmfw_adsp1_id_hdr adsp1_id; @@ -1065,7 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1082,7 +1079,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp1_alg[i].alg.id)); @@ -1111,7 +1108,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1128,7 +1125,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1145,7 +1142,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp, struct snd_soc_codec *codec) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); - wm_adsp_create_control(codec, region); + wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", be32_to_cpu(adsp2_alg[i].alg.id)); @@ -1365,6 +1362,8 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, int ret; int val; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: regmap_update_bits(dsp->regmap, dsp->base + ADSP1_CONTROL_30, @@ -1399,7 +1398,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; @@ -1492,6 +1491,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, unsigned int val; int ret; + dsp->card = codec->card; + switch (event) { case SND_SOC_DAPM_POST_PMU: /* @@ -1554,7 +1555,7 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, if (ret != 0) goto err; - ret = wm_adsp_setup_algs(dsp, codec); + ret = wm_adsp_setup_algs(dsp); if (ret != 0) goto err; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 64087fb1cdec..d018dea6254d 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -39,6 +39,7 @@ struct wm_adsp { int type; struct device *dev; struct regmap *regmap; + struct snd_soc_card *card; int base; int sysclk_reg; -- cgit v1.2.3 From 868ead653e7f65a9ac05777d0736a181a3c1c150 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Thu, 1 Aug 2013 19:29:22 +0800 Subject: ASoC: rt5640: remove unused mux Remove unused "INL Mux" and "INR Mux". Signed-off-by: Bard Liao Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 26 -------------------------- 1 file changed, 26 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index ce585e37e38a..4db7314baabc 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -737,29 +737,6 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; -/* INL/R source */ -static const char * const rt5640_inl_src[] = { - "IN2P", "MONOP" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inl_enum, RT5640_INL_INR_VOL, - RT5640_INL_SEL_SFT, rt5640_inl_src); - -static const struct snd_kcontrol_new rt5640_inl_mux = - SOC_DAPM_ENUM("INL source", rt5640_inl_enum); - -static const char * const rt5640_inr_src[] = { - "IN2N", "MONON" -}; - -static const SOC_ENUM_SINGLE_DECL( - rt5640_inr_enum, RT5640_INL_INR_VOL, - RT5640_INR_SEL_SFT, rt5640_inr_src); - -static const struct snd_kcontrol_new rt5640_inr_mux = - SOC_DAPM_ENUM("INR source", rt5640_inr_enum); - /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -1005,9 +982,6 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_IN_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR VOL", RT5640_PWR_VOL, RT5640_PWR_IN_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL Mux", SND_SOC_NOPM, 0, 0, &rt5640_inl_mux), - SND_SOC_DAPM_MUX("INR Mux", SND_SOC_NOPM, 0, 0, &rt5640_inr_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5640_PWR_MIXER, RT5640_PWR_RM_L_BIT, 0, rt5640_rec_l_mix, ARRAY_SIZE(rt5640_rec_l_mix)), -- cgit v1.2.3 From 6ad709482e151068b7197f4572edeeae5eeaff93 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 4 Aug 2013 20:32:04 +0100 Subject: ASoC: spdif_transceiver: add output pin widget CODECs without any outputs now remain powered down, which means any paths to these codecs also remain powered down. Add an always-enabled output pin widget to the spdif transceiver codec. This enables DAPM to correctly identify that the spdif transceiver is in use when playback is enabled, which will then allow DAPM to power up any links from the CPU DAI to the S/PDIF transceiver. Signed-off-by: Russell King Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transmitter.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index efc3d88d7f8c..4e96d10d61a2 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -29,7 +29,20 @@ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dit; +static const struct snd_soc_dapm_widget dit_widgets[] = { + SND_SOC_DAPM_OUTPUT("spdif-out"), +}; + +static const const struct snd_soc_dapm_route dit_routes[] = { + { "spdif-out", NULL, "Playback" }, +}; + +static struct snd_soc_codec_driver soc_codec_spdif_dit = { + .dapm_widgets = dit_widgets, + .num_dapm_widgets = ARRAY_SIZE(dit_widgets), + .dapm_routes = dit_routes, + .num_dapm_routes = ARRAY_SIZE(dit_routes), +}; static struct snd_soc_dai_driver dit_stub_dai = { .name = "dit-hifi", -- cgit v1.2.3 From e06e4c2d530fd4995c41083009647263ccd77d3b Mon Sep 17 00:00:00 2001 From: Oskar Schirmer Date: Mon, 5 Aug 2013 07:36:02 +0000 Subject: ASoC: sgtl5000: fix codec clock source transition to avoid clockless moment Powering down PLL before switching to a mode that does not use it is a bad idea. It would cause the SGTL5000 be without internal clock supply, especially on the I2C interface, which would make subsequent access to it fail. Thus, in case of not using PLL any longer, first set the mode control, then power down PLL. Signed-off-by: Oskar Schirmer Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 7c99f3ccb1c6..54ca169ec27e 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -644,16 +644,19 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP); + + /* if using pll, clk_ctrl must be set after pll power up */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); } else { + /* otherwise, clk_ctrl must be set before pll power down */ + snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); + /* power down pll */ snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP, 0); } - /* if using pll, clk_ctrl must be set after pll power up */ - snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl); - return 0; } -- cgit v1.2.3 From 13b02fa0dbb1311d08dfacd897a6ff41232d7cfb Mon Sep 17 00:00:00 2001 From: Michael Trimarchi Date: Sat, 3 Aug 2013 16:20:43 +0200 Subject: ASoC: Add PCM1792A spi mode codec support Add PCM1792A spi mode codec support. This version implements only a subset of functionalities. Tested connect to a pandaboard ES device and based on recently pcm1681 codec. Signed-off-by: Michael Trimarchi Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm1792a.c | 245 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/pcm1792a.h | 26 +++++ 4 files changed, 277 insertions(+) create mode 100644 sound/soc/codecs/pcm1792a.c create mode 100644 sound/soc/codecs/pcm1792a.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..4afe94330820 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -54,6 +54,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C select SND_SOC_HDMI_CODEC + select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C @@ -292,6 +293,9 @@ config SND_SOC_MAX9850 config SND_SOC_HDMI_CODEC tristate +config SND_SOC_PCM1792A + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..811ca12febcb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -42,6 +42,7 @@ snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-hdmi-codec-objs := hdmi.o +snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -171,6 +172,7 @@ obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o +obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c new file mode 100644 index 000000000000..3f83bf9895d4 --- /dev/null +++ b/sound/soc/codecs/pcm1792a.c @@ -0,0 +1,245 @@ +/* + * PCM1792A ASoC codec driver + * + * Copyright (c) Amarula Solutions B.V. 2013 + * + * Michael Trimarchi + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include + +#include "pcm1792a.h" + +#define PCM1792A_DAC_VOL_LEFT 0x10 +#define PCM1792A_DAC_VOL_RIGHT 0x11 +#define PCM1792A_FMT_CONTROL 0x12 +#define PCM1792A_SOFT_MUTE PCM1792A_FMT_CONTROL + +#define PCM1792A_FMT_MASK 0x70 +#define PCM1792A_FMT_SHIFT 4 +#define PCM1792A_MUTE_MASK 0x01 +#define PCM1792A_MUTE_SHIFT 0 +#define PCM1792A_ATLD_ENABLE (1 << 7) + +static const struct reg_default pcm1792a_reg_defaults[] = { + { 0x10, 0xff }, + { 0x11, 0xff }, + { 0x12, 0x50 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x01 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, +}; + +static bool pcm1792a_accessible_reg(struct device *dev, unsigned int reg) +{ + return reg >= 0x10 && reg <= 0x17; +} + +static bool pcm1792a_writeable_reg(struct device *dev, unsigned register reg) +{ + bool accessible; + + accessible = pcm1792a_accessible_reg(dev, reg); + + return accessible && reg != 0x16 && reg != 0x17; +} + +struct pcm1792a_private { + struct regmap *regmap; + unsigned int format; + unsigned int rate; +}; + +static int pcm1792a_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->format = format; + + return 0; +} + +static int pcm1792a_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regmap_update_bits(priv->regmap, PCM1792A_SOFT_MUTE, + PCM1792A_MUTE_MASK, !!mute); + if (ret < 0) + return ret; + + return 0; +} + +static int pcm1792a_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); + int val = 0, ret; + int pcm_format = params_format(params); + + priv->rate = params_rate(params); + + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x02; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || + pcm_format == SNDRV_PCM_FORMAT_S32_LE) + val = 0x05; + else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) + val = 0x04; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + val = val << PCM1792A_FMT_SHIFT | PCM1792A_ATLD_ENABLE; + + ret = regmap_update_bits(priv->regmap, PCM1792A_FMT_CONTROL, + PCM1792A_FMT_MASK | PCM1792A_ATLD_ENABLE, val); + if (ret < 0) + return ret; + + return 0; +} + +static const struct snd_soc_dai_ops pcm1792a_dai_ops = { + .set_fmt = pcm1792a_set_dai_fmt, + .hw_params = pcm1792a_hw_params, + .digital_mute = pcm1792a_digital_mute, +}; + +static const DECLARE_TLV_DB_SCALE(pcm1792a_dac_tlv, -12000, 50, 1); + +static const struct snd_kcontrol_new pcm1792a_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("DAC Playback Volume", PCM1792A_DAC_VOL_LEFT, + PCM1792A_DAC_VOL_RIGHT, 0, 0xf, 0xff, 0, + pcm1792a_dac_tlv), +}; + +static struct snd_soc_dai_driver pcm1792a_dai = { + .name = "pcm1792a-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PCM1792A_RATES, + .formats = PCM1792A_FORMATS, }, + .capture = { + .channels_min = 0, + .channels_max = 0, + }, + .ops = &pcm1792a_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id pcm1792a_of_match[] = { + { .compatible = "ti,pcm1792a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm1792a_of_match); +#endif + +static const struct regmap_config pcm1792a_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 24, + .reg_defaults = pcm1792a_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), + .writeable_reg = pcm1792a_writeable_reg, + .readable_reg = pcm1792a_accessible_reg, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { + .controls = pcm1792a_controls, + .num_controls = ARRAY_SIZE(pcm1792a_controls), +}; + +static int pcm1792a_spi_probe(struct spi_device *spi) +{ + struct pcm1792a_private *pcm1792a; + int ret; + + pcm1792a = devm_kzalloc(&spi->dev, sizeof(struct pcm1792a_private), + GFP_KERNEL); + if (!pcm1792a) + return -ENOMEM; + + spi_set_drvdata(spi, pcm1792a); + + pcm1792a->regmap = devm_regmap_init_spi(spi, &pcm1792a_regmap); + if (IS_ERR(pcm1792a->regmap)) { + ret = PTR_ERR(pcm1792a->regmap); + dev_err(&spi->dev, "Failed to register regmap: %d\n", ret); + return ret; + } + + return snd_soc_register_codec(&spi->dev, + &soc_codec_dev_pcm1792a, &pcm1792a_dai, 1); +} + +static int pcm1792a_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm1792a_spi_ids[] = { + { "pcm1792a", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm1792a_spi_ids); + +static struct spi_driver pcm1792a_codec_driver = { + .driver = { + .name = "pcm1792a", + .owner = THIS_MODULE, + .of_match_table = pcm1792a_of_match, + }, + .id_table = pcm1792a_spi_ids, + .probe = pcm1792a_spi_probe, + .remove = pcm1792a_spi_remove, +}; + +module_spi_driver(pcm1792a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM1792A driver"); +MODULE_AUTHOR("Michael Trimarchi "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm1792a.h b/sound/soc/codecs/pcm1792a.h new file mode 100644 index 000000000000..7a83d1fc102a --- /dev/null +++ b/sound/soc/codecs/pcm1792a.h @@ -0,0 +1,26 @@ +/* + * definitions for PCM1792A + * + * Copyright 2013 Amarula Solutions + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#ifndef __PCM1792A_H__ +#define __PCM1792A_H__ + +#define PCM1792A_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) + +#define PCM1792A_FORMATS (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S16_LE) + +#endif -- cgit v1.2.3 From 55af2d23c6c9caf7da6c9a55bbea83dccbc1af2b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Aug 2013 18:20:53 +0100 Subject: ASoC: pcm1792a: Fix build with !OF Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 3f83bf9895d4..72cf8353e812 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -169,13 +169,11 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .ops = &pcm1792a_dai_ops, }; -#ifdef CONFIG_OF static const struct of_device_id pcm1792a_of_match[] = { { .compatible = "ti,pcm1792a", }, { } }; MODULE_DEVICE_TABLE(of, pcm1792a_of_match); -#endif static const struct regmap_config pcm1792a_regmap = { .reg_bits = 8, @@ -231,7 +229,7 @@ static struct spi_driver pcm1792a_codec_driver = { .driver = { .name = "pcm1792a", .owner = THIS_MODULE, - .of_match_table = pcm1792a_of_match, + .of_match_table = of_match_ptr(pcm1792a_of_match), }, .id_table = pcm1792a_spi_ids, .probe = pcm1792a_spi_probe, -- cgit v1.2.3 From ed6a27723979cfffab62c450baba4f75ebcbda78 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 5 Aug 2013 23:34:17 +0100 Subject: ASoC: wm8994: Fix class W controls Commit 6e0650 (ASoC: wm8994: Use SOC_SINGLE_EXT() instead of open-coding it) went too far and converted a DAPM control to use SOC_SINGLE_EXT() which crashes. Revert that portion of the patch. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 24131a7f9390..c99b6da79efd 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1433,7 +1433,7 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING, #define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \ SOC_SINGLE_EXT(xname, reg, shift, max, invert, \ - snd_soc_get_volsw, wm8994_put_class_w) + snd_soc_dapm_get_volsw, wm8994_put_class_w) static int wm8994_put_class_w(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3 From f8f11795b96a3632edb25a8924c61bfb74581cb0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Aug 2013 13:39:29 +0200 Subject: ASoC: tlv320aic26: Fix keyclick feature The tlv320aic26 contains a embedded snd_soc_codec struct which is referenced in the keyclick code. That struct is never initialized though, replace the embedded struct with a pointer and use that in the keyclick code. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b1f6982c7c9c..b192cd4705a0 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,7 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; - struct snd_soc_codec codec; + struct snd_soc_codec *codec; int master; int datfm; int mclk; @@ -330,7 +330,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -346,9 +346,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val; - val = aic26_reg_read_cache(&aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(&aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -360,8 +360,11 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); */ static int aic26_probe(struct snd_soc_codec *codec) { + struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, err, i, reg; + aic26->codec = codec; + dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ -- cgit v1.2.3 From 95ad868289a24dbc072412ce2fb0d40cb34c5794 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Aug 2013 13:39:31 +0200 Subject: ASoC: mc13783: Remove embedded snd_soc_codec structs from private data structs It is unused and a leftover of the pre multi-component era. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 5402dfbbb716..4d3c8fd8c5db 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -94,7 +94,6 @@ #define AUDIO_DAC_CFS_DLY_B (1 << 10) struct mc13783_priv { - struct snd_soc_codec codec; struct mc13xxx *mc13xxx; enum mc13783_ssi_port adc_ssi_port; -- cgit v1.2.3 From 0d59ff3a24ad099f741da5efd9e3e02bfd64496e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 6 Aug 2013 13:39:30 +0200 Subject: ASoC: twl4030: Remove embedded snd_soc_codec structs from private data structs It is unused and a leftover of the pre multi-component era. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 8e6e5b016021..1e3884d6b3fb 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -137,8 +137,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { /* codec private data */ struct twl4030_priv { - struct snd_soc_codec codec; - unsigned int codec_powered; /* reference counts of AIF/APLL users */ -- cgit v1.2.3 From c7f3843575eac1eea1fbda2f6b61d36627fa8f7c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 6 Aug 2013 17:03:55 +0100 Subject: ASoC: wm5110: Correct input OSR bits for wm5110 The input OSR bits are specified differently for wm5110 than for current revs of wm5102. This patch corrects support for this on wm5110. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 20 ++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 12 ++++-------- 3 files changed, 25 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8dc6881496de..779a0eeac67c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -553,6 +553,26 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static const char * const arizona_in_dmic_osr_text[] = { + "1.536MHz", "3.072MHz", "6.144MHz", +}; + +const struct soc_enum arizona_in_dmic_osr[] = { + SOC_ENUM_SINGLE(ARIZONA_IN1L_CONTROL, ARIZONA_IN1_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN2L_CONTROL, ARIZONA_IN2_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN3L_CONTROL, ARIZONA_IN3_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), + SOC_ENUM_SINGLE(ARIZONA_IN4L_CONTROL, ARIZONA_IN4_OSR_SHIFT, + ARRAY_SIZE(arizona_in_dmic_osr_text), + arizona_in_dmic_osr_text), +}; +EXPORT_SYMBOL_GPL(arizona_in_dmic_osr); + static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) { struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index fe1b794bd5f0..b6b6d7036ea0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -198,6 +198,7 @@ extern const struct soc_enum arizona_lhpf3_mode; extern const struct soc_enum arizona_lhpf4_mode; extern const struct soc_enum arizona_ng_hold; +extern const struct soc_enum arizona_in_dmic_osr[]; extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index fc410377277f..77fd531bf3cc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -58,14 +58,10 @@ static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); SOC_SINGLE(name " NG SPKDAT2R Switch", base, 11, 1, 0) static const struct snd_kcontrol_new wm5110_snd_controls[] = { -SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3_OSR_SHIFT, 1, 0), -SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, - ARIZONA_IN4_OSR_SHIFT, 1, 0), +SOC_ENUM("IN1 OSR", arizona_in_dmic_osr[0]), +SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), +SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), +SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -- cgit v1.2.3 From 9e7e474c0963dfd1f60b200160ff9e7cefb32b06 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Aug 2013 23:51:12 +0100 Subject: ASoC: ad1980: Provide stub DAPM support Since non-DAPM devices are not going to be supported provide DAPM input and output widgets and hook them up to the DAIs. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ad1980.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 89fcf7d6e7b8..7257a8885f42 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -96,6 +96,44 @@ SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), }; +static const struct snd_soc_dapm_widget ad1980_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2"), +SND_SOC_DAPM_INPUT("CD_L"), +SND_SOC_DAPM_INPUT("CD_R"), +SND_SOC_DAPM_INPUT("AUX_L"), +SND_SOC_DAPM_INPUT("AUX_R"), +SND_SOC_DAPM_INPUT("LINE_IN_L"), +SND_SOC_DAPM_INPUT("LINE_IN_R"), + +SND_SOC_DAPM_OUTPUT("LFE_OUT"), +SND_SOC_DAPM_OUTPUT("CENTER_OUT"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_L"), +SND_SOC_DAPM_OUTPUT("LINE_OUT_R"), +SND_SOC_DAPM_OUTPUT("MONO_OUT"), +SND_SOC_DAPM_OUTPUT("HP_OUT_L"), +SND_SOC_DAPM_OUTPUT("HP_OUT_R"), +}; + +static const struct snd_soc_dapm_route ad1980_dapm_routes[] = { + { "Capture", NULL, "MIC1" }, + { "Capture", NULL, "MIC2" }, + { "Capture", NULL, "CD_L" }, + { "Capture", NULL, "CD_R" }, + { "Capture", NULL, "AUX_L" }, + { "Capture", NULL, "AUX_R" }, + { "Capture", NULL, "LINE_IN_L" }, + { "Capture", NULL, "LINE_IN_R" }, + + { "LFE_OUT", NULL, "Playback" }, + { "CENTER_OUT", NULL, "Playback" }, + { "LINE_OUT_L", NULL, "Playback" }, + { "LINE_OUT_R", NULL, "Playback" }, + { "MONO_OUT", NULL, "Playback" }, + { "HP_OUT_L", NULL, "Playback" }, + { "HP_OUT_R", NULL, "Playback" }, +}; + static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg) { @@ -253,6 +291,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ad1980 = { .reg_cache_step = 2, .write = ac97_write, .read = ac97_read, + + .dapm_widgets = ad1980_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad1980_dapm_widgets), + .dapm_routes = ad1980_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad1980_dapm_routes), }; static int ad1980_probe(struct platform_device *pdev) -- cgit v1.2.3 From 45a14a8b50465a6ce61005f7fe9f3fd5c06823d5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 18:24:09 +0100 Subject: ASoC: ads711x: Add DAPM support This makes it easier to hook into boards and ensures the driver continues to work with support for non-DAPM CODECs removed. Signed-off-by: Mark Brown --- sound/soc/codecs/ads117x.c | 29 ++++++++++++++++++++++++++++- 1 file changed, 28 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index 506d474c4d22..8f388edff586 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -23,6 +23,28 @@ #define ADS117X_RATES (SNDRV_PCM_RATE_8000_48000) #define ADS117X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE) +static const struct snd_soc_dapm_widget ads117x_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("Input1"), +SND_SOC_DAPM_INPUT("Input2"), +SND_SOC_DAPM_INPUT("Input3"), +SND_SOC_DAPM_INPUT("Input4"), +SND_SOC_DAPM_INPUT("Input5"), +SND_SOC_DAPM_INPUT("Input6"), +SND_SOC_DAPM_INPUT("Input7"), +SND_SOC_DAPM_INPUT("Input8"), +}; + +static const struct snd_soc_dapm_route ads117x_dapm_routes[] = { + { "Capture", NULL, "Input1" }, + { "Capture", NULL, "Input2" }, + { "Capture", NULL, "Input3" }, + { "Capture", NULL, "Input4" }, + { "Capture", NULL, "Input5" }, + { "Capture", NULL, "Input6" }, + { "Capture", NULL, "Input7" }, + { "Capture", NULL, "Input8" }, +}; + static struct snd_soc_dai_driver ads117x_dai = { /* ADC */ .name = "ads117x-hifi", @@ -34,7 +56,12 @@ static struct snd_soc_dai_driver ads117x_dai = { .formats = ADS117X_FORMATS,}, }; -static struct snd_soc_codec_driver soc_codec_dev_ads117x; +static struct snd_soc_codec_driver soc_codec_dev_ads117x = { + .dapm_widgets = ads117x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ads117x_dapm_widgets), + .dapm_routes = ads117x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ads117x_dapm_routes), +}; static int ads117x_probe(struct platform_device *pdev) { -- cgit v1.2.3 From 16695971bec3b8b2398f7ab8dfa4c5a22bfcf95d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 8 Aug 2013 12:25:57 +0100 Subject: ASoC: pcm1681: Staticise DAI driver It is not exported so doesn't need to be in the global namespace and sparse warns on this. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 27da41b2dfcd..51b18662e6aa 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -225,7 +225,7 @@ static const struct snd_kcontrol_new pcm1681_controls[] = { pcm1681_get_deemph, pcm1681_put_deemph), }; -struct snd_soc_dai_driver pcm1681_dai = { +static struct snd_soc_dai_driver pcm1681_dai = { .name = "pcm1681-hifi", .playback = { .stream_name = "Playback", -- cgit v1.2.3 From 827d22f13618557bd35f938b020c954d83a82977 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 17:59:51 +0100 Subject: ASoC: ad73311: Add DAPM support This makes it possible to hook up other devices in boards and is required by removal of support for non-DAPM CODECs in the core. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/ad73311.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index b1f2baf42b48..5fac8adbc136 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -23,6 +23,21 @@ #include "ad73311.h" +static const struct snd_soc_dapm_widget ad73311_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINP"), +SND_SOC_DAPM_INPUT("VINN"), +SND_SOC_DAPM_OUTPUT("VOUTN"), +SND_SOC_DAPM_OUTPUT("VOUTP"), +}; + +static const struct snd_soc_dapm_route ad73311_dapm_routes[] = { + { "Capture", NULL, "VINP" }, + { "Capture", NULL, "VINN" }, + + { "VOUTN", NULL, "Playback" }, + { "VOUTP", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ad73311_dai = { .name = "ad73311-hifi", .playback = { @@ -39,7 +54,12 @@ static struct snd_soc_dai_driver ad73311_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE, }, }; -static struct snd_soc_codec_driver soc_codec_dev_ad73311; +static struct snd_soc_codec_driver soc_codec_dev_ad73311 = { + .dapm_widgets = ad73311_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ad73311_dapm_widgets), + .dapm_routes = ad73311_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ad73311_dapm_routes), +}; static int ad73311_probe(struct platform_device *pdev) { -- cgit v1.2.3 From 439fe8a7bb07f8394fef03d7aa4f207166a32b88 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 13:26:43 +0100 Subject: ASoC: max9768: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/max9768.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index a6ac2313047d..31f91560e9f6 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -118,6 +118,18 @@ static const struct snd_kcontrol_new max9768_mute[] = { SOC_SINGLE_BOOL_EXT("Playback Switch", 0, max9768_get_gpio, max9768_set_gpio), }; +static const struct snd_soc_dapm_widget max9768_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("IN"), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +}; + +static const struct snd_soc_dapm_route max9768_dapm_routes[] = { + { "OUT+", NULL, "IN" }, + { "OUT-", NULL, "IN" }, +}; + static int max9768_probe(struct snd_soc_codec *codec) { struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); @@ -148,6 +160,10 @@ static struct snd_soc_codec_driver max9768_codec_driver = { .probe = max9768_probe, .controls = max9768_volume, .num_controls = ARRAY_SIZE(max9768_volume), + .dapm_widgets = max9768_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9768_dapm_widgets), + .dapm_routes = max9768_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9768_dapm_routes), }; static const struct regmap_config max9768_i2c_regmap_config = { -- cgit v1.2.3 From bad268f3504e2a58e406c3f0e282c1de629bd42f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 13:12:13 +0100 Subject: ASoC: cs4271: Convert to table based control init Signed-off-by: Mark Brown Acked-by: Alexander Sverdlin --- sound/soc/codecs/cs4271.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 03036b326732..65ad56c43c13 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -576,8 +576,7 @@ static int cs4271_probe(struct snd_soc_codec *codec) CS4271_MODE2_MUTECAEQUB, CS4271_MODE2_MUTECAEQUB); - return snd_soc_add_codec_controls(codec, cs4271_snd_controls, - ARRAY_SIZE(cs4271_snd_controls)); + return 0; } static int cs4271_remove(struct snd_soc_codec *codec) @@ -596,6 +595,9 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .remove = cs4271_remove, .suspend = cs4271_soc_suspend, .resume = cs4271_soc_resume, + + .controls = cs4271_snd_controls, + .num_controls = ARRAY_SIZE(cs4271_snd_controls), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From 2e7fb942a30563125d6aac497fa0dcddbb7d731d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 13:15:10 +0100 Subject: ASoC: cs4271: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown Acked-by: Alexander Sverdlin --- sound/soc/codecs/cs4271.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 65ad56c43c13..a20f1bb8f071 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -173,6 +173,26 @@ struct cs4271_private { bool enable_soft_reset; }; +static const struct snd_soc_dapm_widget cs4271_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINA"), +SND_SOC_DAPM_INPUT("AINB"), + +SND_SOC_DAPM_OUTPUT("AOUTA+"), +SND_SOC_DAPM_OUTPUT("AOUTA-"), +SND_SOC_DAPM_OUTPUT("AOUTB+"), +SND_SOC_DAPM_OUTPUT("AOUTB-"), +}; + +static const struct snd_soc_dapm_route cs4271_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA+", NULL, "Playback" }, + { "AOUTA-", NULL, "Playback" }, + { "AOUTB+", NULL, "Playback" }, + { "AOUTB-", NULL, "Playback" }, +}; + /* * @freq is the desired MCLK rate * MCLK rate should (c) be the sample rate, multiplied by one of the @@ -598,6 +618,10 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .controls = cs4271_snd_controls, .num_controls = ARRAY_SIZE(cs4271_snd_controls), + .dapm_widgets = cs4271_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4271_dapm_widgets), + .dapm_routes = cs4271_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From 9190aeb4ecbdcab7d66d186c207f76d09b41d082 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 15:07:36 +0100 Subject: ASoC: adau1701: Use gpio_set_value_cansleep() The GPIO manipulation done by this driver is never in atomic context so we can use gpio_set_value_cansleep() and support GPIOs that can't be set from atomic context. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/codecs/adau1701.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2c102522bbbc..ebff1128be59 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -247,21 +247,21 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) gpio_is_valid(adau1701->gpio_pll_mode[1])) { switch (clkdiv) { case 64: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 256: - gpio_set_value(adau1701->gpio_pll_mode[0], 0); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; case 384: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 0); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 0); break; case 0: /* fallback */ case 512: - gpio_set_value(adau1701->gpio_pll_mode[0], 1); - gpio_set_value(adau1701->gpio_pll_mode[1], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[0], 1); + gpio_set_value_cansleep(adau1701->gpio_pll_mode[1], 1); break; } } @@ -269,10 +269,10 @@ static int adau1701_reset(struct snd_soc_codec *codec, unsigned int clkdiv) adau1701->pll_clkdiv = clkdiv; if (gpio_is_valid(adau1701->gpio_nreset)) { - gpio_set_value(adau1701->gpio_nreset, 0); + gpio_set_value_cansleep(adau1701->gpio_nreset, 0); /* minimum reset time is 20ns */ udelay(1); - gpio_set_value(adau1701->gpio_nreset, 1); + gpio_set_value_cansleep(adau1701->gpio_nreset, 1); /* power-up time may be as long as 85ms */ mdelay(85); } -- cgit v1.2.3 From 2e61926cb4a42d79a406aa64f04869d1227ca42c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 19:01:40 +0100 Subject: ASoC: ak4104: Add stub DAPM support This makes it easer to integrate the device with other on-board components and ensures correct operation following removal of support for non-DAPM CODECs. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index c7cfdf957e4d..9a7c89b9cb35 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -51,6 +51,14 @@ struct ak4104_private { struct regmap *regmap; }; +static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { + { "TX", NULL, "Playback" }, +}; + static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { @@ -214,6 +222,11 @@ static int ak4104_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_ak4104 = { .probe = ak4104_probe, .remove = ak4104_remove, + + .dapm_widgets = ak4104_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4104_dapm_widgets), + .dapm_routes = ak4104_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4104_dapm_routes), }; static const struct regmap_config ak4104_regmap = { -- cgit v1.2.3 From a5db4d50fa578936275c1e26d5d2fda25c0d2bf6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 7 Aug 2013 19:05:47 +0100 Subject: ASoC: ak4104: Manage TXE using DAPM Saves some code. We should also be able to manage the power up and reset registers using DAPM but it's probably more trouble than it's worth in mains powered systems. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 23 ++++------------------- 1 file changed, 4 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 9a7c89b9cb35..71059c07ae7b 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -52,11 +52,14 @@ struct ak4104_private { }; static const struct snd_soc_dapm_widget ak4104_dapm_widgets[] = { +SND_SOC_DAPM_PGA("TXE", AK4104_REG_TX, AK4104_TX_TXE, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("TX"), }; static const struct snd_soc_dapm_route ak4104_dapm_routes[] = { - { "TX", NULL, "Playback" }, + { "TXE", NULL, "Playback" }, + { "TX", NULL, "TXE" }, }; static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, @@ -146,29 +149,11 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* enable transmitter */ - ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); - if (ret < 0) - return ret; - return 0; } -static int ak4104_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); - - /* disable transmitter */ - return regmap_update_bits(ak4104->regmap, AK4104_REG_TX, - AK4104_TX_TXE, 0); -} - static const struct snd_soc_dai_ops ak4101_dai_ops = { .hw_params = ak4104_hw_params, - .hw_free = ak4104_hw_free, .set_fmt = ak4104_set_dai_fmt, }; -- cgit v1.2.3 From dcf1439a493f75336f7e9d272d01b04bc1c4ca8e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 12:28:56 +0100 Subject: ASoC: ak5386: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/ak5386.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index 1f303983ae02..72e953b2cb41 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -22,7 +22,22 @@ struct ak5386_priv { int reset_gpio; }; -static struct snd_soc_codec_driver soc_codec_ak5386; +static const struct snd_soc_dapm_widget ak5386_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route ak5386_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + +static struct snd_soc_codec_driver soc_codec_ak5386 = { + .dapm_widgets = ak5386_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak5386_dapm_widgets), + .dapm_routes = ak5386_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak5386_dapm_routes), +}; static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) -- cgit v1.2.3 From e7edb2731bf8e00aaeb7d20800ae108068618f63 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 12 Aug 2013 11:33:32 +0100 Subject: ASoC: arizona: Add widget<->mux route into mux route macro The routes linking the widget and the input mux were being added manually, rather than by the ARIZONA_MUX_ROUTES macro. This patchs adds the routes to the macro. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.h | 3 ++- sound/soc/codecs/wm5102.c | 41 ++++++++++++----------------------------- sound/soc/codecs/wm5110.c | 8 ++++---- 3 files changed, 18 insertions(+), 34 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b6b6d7036ea0..9e81b6392692 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -150,7 +150,8 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MUX(name_str " Aux 5", &name##_aux5_mux), \ ARIZONA_MUX(name_str " Aux 6", &name##_aux6_mux) -#define ARIZONA_MUX_ROUTES(name) \ +#define ARIZONA_MUX_ROUTES(widget, name) \ + { widget, NULL, name " Input" }, \ ARIZONA_MIXER_INPUT_ROUTES(name " Input") #define ARIZONA_MIXER_ROUTES(widget, name) \ diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a6cbdb4b5c0f..f38c52d43b8d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1501,23 +1501,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L PGA", NULL, "IN3L" }, { "IN3R PGA", NULL, "IN3R" }, - { "ASRC1L", NULL, "ASRC1L Input" }, - { "ASRC1R", NULL, "ASRC1R Input" }, - { "ASRC2L", NULL, "ASRC2L Input" }, - { "ASRC2R", NULL, "ASRC2R Input" }, - - { "ISRC1DEC1", NULL, "ISRC1DEC1 Input" }, - { "ISRC1DEC2", NULL, "ISRC1DEC2 Input" }, - - { "ISRC1INT1", NULL, "ISRC1INT1 Input" }, - { "ISRC1INT2", NULL, "ISRC1INT2 Input" }, - - { "ISRC2DEC1", NULL, "ISRC2DEC1 Input" }, - { "ISRC2DEC2", NULL, "ISRC2DEC2 Input" }, - - { "ISRC2INT1", NULL, "ISRC2INT1 Input" }, - { "ISRC2INT2", NULL, "ISRC2INT2 Input" }, - ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), @@ -1569,22 +1552,22 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), - ARIZONA_MUX_ROUTES("ISRC1INT1"), - ARIZONA_MUX_ROUTES("ISRC1INT2"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC1INT2"), - ARIZONA_MUX_ROUTES("ISRC1DEC1"), - ARIZONA_MUX_ROUTES("ISRC1DEC2"), + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), - ARIZONA_MUX_ROUTES("ISRC2INT1"), - ARIZONA_MUX_ROUTES("ISRC2INT2"), + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), - ARIZONA_MUX_ROUTES("ISRC2DEC1"), - ARIZONA_MUX_ROUTES("ISRC2DEC2"), + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), ARIZONA_DSP_ROUTES("DSP1"), diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 77fd531bf3cc..38e50c81a953 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -978,10 +978,10 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), - ARIZONA_MUX_ROUTES("ASRC1L"), - ARIZONA_MUX_ROUTES("ASRC1R"), - ARIZONA_MUX_ROUTES("ASRC2L"), - ARIZONA_MUX_ROUTES("ASRC2R"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, -- cgit v1.2.3 From 40843aea5a9bd2c3d7917d086e6d23cb02cc4b39 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 12 Aug 2013 23:46:55 +0100 Subject: ASoC: wm8997: Initial CODEC driver The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus interfacing, for smartphones, tablets and other portable audio devices based on the Arizona platform. This patch adds the wm8997 CODEC driver. [Fixed some interface churn from bitrot due to the patch not going via the MFD tree as expected -- broonie] Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 13 +- sound/soc/codecs/wm8997.c | 1175 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8997.h | 23 + 5 files changed, 1216 insertions(+), 3 deletions(-) create mode 100644 sound/soc/codecs/wm8997.c create mode 100644 sound/soc/codecs/wm8997.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index badb6fbacaa6..bb34c8a4bf0e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -122,6 +122,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8994 if MFD_WM8994 select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8996 if I2C + select SND_SOC_WM8997 if MFD_WM8997 select SND_SOC_WM9081 if I2C select SND_SOC_WM9090 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS @@ -145,8 +146,10 @@ config SND_SOC_ARIZONA tristate default y if SND_SOC_WM5102=y default y if SND_SOC_WM5110=y + default y if SND_SOC_WM8997=y default m if SND_SOC_WM5102=m default m if SND_SOC_WM5110=m + default m if SND_SOC_WM8997=m config SND_SOC_WM_HUBS tristate @@ -500,6 +503,9 @@ config SND_SOC_WM8995 config SND_SOC_WM8996 tristate +config SND_SOC_WM8997 + tristate + config SND_SOC_WM9081 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70fd8066f546..68ea0a2c169c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -114,6 +114,7 @@ snd-soc-wm8991-objs := wm8991.o snd-soc-wm8993-objs := wm8993.o snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o snd-soc-wm8995-objs := wm8995.o +snd-soc-wm8997-objs := wm8997.o snd-soc-wm9081-objs := wm9081.o snd-soc-wm9090-objs := wm9090.o snd-soc-wm9705-objs := wm9705.o @@ -239,6 +240,7 @@ obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o +obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 779a0eeac67c..657808ba1418 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -200,9 +200,16 @@ int arizona_init_spk(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); - if (ret != 0) - return ret; + switch (arizona->type) { + case WM8997: + break; + default: + ret = snd_soc_dapm_new_controls(&codec->dapm, + &arizona_spkr, 1); + if (ret != 0) + return ret; + break; + } ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, "Thermal warning", arizona_thermal_warn, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c new file mode 100644 index 000000000000..0a43bac2f4e3 --- /dev/null +++ b/sound/soc/codecs/wm8997.c @@ -0,0 +1,1175 @@ +/* + * wm8997.c -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Charles Keepax + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "arizona.h" +#include "wm8997.h" + +struct wm8997_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0); + +static const struct reg_default wm8997_sysclk_reva_patch[] = { + { 0x301D, 0x7B15 }, + { 0x301B, 0x0050 }, + { 0x305D, 0x7B17 }, + { 0x305B, 0x0050 }, + { 0x3001, 0x08FE }, + { 0x3003, 0x00F4 }, + { 0x3041, 0x08FF }, + { 0x3043, 0x0005 }, + { 0x3020, 0x0225 }, + { 0x3021, 0x0A00 }, + { 0x3022, 0xE24D }, + { 0x3023, 0x0800 }, + { 0x3024, 0xE24D }, + { 0x3025, 0xF000 }, + { 0x3060, 0x0226 }, + { 0x3061, 0x0A00 }, + { 0x3062, 0xE252 }, + { 0x3063, 0x0800 }, + { 0x3064, 0xE252 }, + { 0x3065, 0xF000 }, + { 0x3116, 0x022B }, + { 0x3117, 0xFA00 }, + { 0x3110, 0x246C }, + { 0x3111, 0x0A03 }, + { 0x3112, 0x246E }, + { 0x3113, 0x0A03 }, + { 0x3114, 0x2470 }, + { 0x3115, 0x0A03 }, + { 0x3126, 0x246C }, + { 0x3127, 0x0A02 }, + { 0x3128, 0x246E }, + { 0x3129, 0x0A02 }, + { 0x312A, 0x2470 }, + { 0x312B, 0xFA02 }, + { 0x3125, 0x0800 }, +}; + +static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 0: + patch = wm8997_sysclk_reva_patch; + patch_size = ARRAY_SIZE(wm8997_sysclk_reva_patch); + break; + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + default: + break; + } + + return 0; +} + +static const char *wm8997_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm8997_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm8997_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm8997_osr_text, wm8997_osr_val), +}; + +#define WM8997_NG_SRC(name, base) \ + SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ + SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ + SOC_SINGLE(name " NG EPOUT Switch", base, 4, 1, 0), \ + SOC_SINGLE(name " NG SPKOUT Switch", base, 6, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1L Switch", base, 8, 1, 0), \ + SOC_SINGLE(name " NG SPKDAT1R Switch", base, 9, 1, 0) + +static const struct snd_kcontrol_new wm8997_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), + +SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_SINGLE_TLV("IN1L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN1R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_IN1R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2L Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("IN2R Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_IN2R_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), + +SOC_ENUM("Input Ramp Up", arizona_in_vi_ramp), +SOC_ENUM("Input Ramp Down", arizona_in_vd_ramp), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("EQ1 Coefficeints", ARIZONA_EQ1_1, 21, + ARIZONA_EQ1_ENA_MASK), +SND_SOC_BYTES_MASK("EQ2 Coefficeints", ARIZONA_EQ2_1, 21, + ARIZONA_EQ2_ENA_MASK), +SND_SOC_BYTES_MASK("EQ3 Coefficeints", ARIZONA_EQ3_1, 21, + ARIZONA_EQ3_ENA_MASK), +SND_SOC_BYTES_MASK("EQ4 Coefficeints", ARIZONA_EQ4_1, 21, + ARIZONA_EQ4_ENA_MASK), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +SND_SOC_BYTES("LHPF1 Coefficients", ARIZONA_HPLPF1_2, 1), +SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), +SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), +SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), + +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUT", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_OUT4L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_VALUE_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_VALUE_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), + +SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), +SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +SOC_SINGLE("Noise Gate Switch", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_ENA_SHIFT, 1, 0), +SOC_SINGLE_TLV("Noise Gate Threshold Volume", ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_THR_SHIFT, 7, 1, ng_tlv), +SOC_ENUM("Noise Gate Hold", arizona_ng_hold), + +WM8997_NG_SRC("HPOUT1L", ARIZONA_NOISE_GATE_SELECT_1L), +WM8997_NG_SRC("HPOUT1R", ARIZONA_NOISE_GATE_SELECT_1R), +WM8997_NG_SRC("EPOUT", ARIZONA_NOISE_GATE_SELECT_3L), +WM8997_NG_SRC("SPKOUT", ARIZONA_NOISE_GATE_SELECT_4L), +WM8997_NG_SRC("SPKDAT1L", ARIZONA_NOISE_GATE_SELECT_5L), +WM8997_NG_SRC("SPKDAT1R", ARIZONA_NOISE_GATE_SELECT_5R), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("SLIMTX1", ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX2", ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX3", ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX4", ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX5", ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX6", ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX7", ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SLIMTX8", ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUT, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(SLIMTX1, ARIZONA_SLIMTX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX2, ARIZONA_SLIMTX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX3, ARIZONA_SLIMTX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX4, ARIZONA_SLIMTX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX5, ARIZONA_SLIMTX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX6, ARIZONA_SLIMTX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX7, ARIZONA_SLIMTX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SLIMTX8, ARIZONA_SLIMTX8MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1INT1, ARIZONA_ISRC1INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1INT2, ARIZONA_ISRC1INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC1DEC1, ARIZONA_ISRC1DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC1DEC2, ARIZONA_ISRC1DEC2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2INT1, ARIZONA_ISRC2INT1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2INT2, ARIZONA_ISRC2INT2MIX_INPUT_1_SOURCE); + +ARIZONA_MUX_ENUMS(ISRC2DEC1, ARIZONA_ISRC2DEC1MIX_INPUT_1_SOURCE); +ARIZONA_MUX_ENUMS(ISRC2DEC2, ARIZONA_ISRC2DEC2MIX_INPUT_1_SOURCE); + +static const char *wm8997_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "EPOUT", "SPKOUT", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm8997_aec_loopback_values[] = { + 0, 1, 4, 6, 8, 9, +}; + +static const struct soc_enum wm8997_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, 0xf, + ARRAY_SIZE(wm8997_aec_loopback_texts), + wm8997_aec_loopback_texts, + wm8997_aec_loopback_values); + +static const struct snd_kcontrol_new wm8997_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm8997_aec_loopback); + +static const struct snd_soc_dapm_widget wm8997_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, wm8997_sysclk_ev, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, + ARIZONA_OPCLK_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCOPCLK", ARIZONA_OUTPUT_ASYNC_CLOCK, + ARIZONA_OPCLK_ASYNC_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0, SND_SOC_DAPM_REGULATOR_BYPASS), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDD", 0, 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), +SND_SOC_DAPM_SIGGEN("HAPTICS"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB3_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1INT1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1INT2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC1DEC1", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC1DEC2", ARIZONA_ISRC_1_CTRL_3, + ARIZONA_ISRC1_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2INT1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2INT2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_INT1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("ISRC2DEC1", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC0_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ISRC2DEC2", ARIZONA_ISRC_2_CTRL_3, + ARIZONA_ISRC2_DEC1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("SLIMTX1", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX2", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX3", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX4", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX5", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX6", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX7", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("SLIMTX8", NULL, 0, + ARIZONA_SLIMBUS_TX_CHANNEL_ENABLE, + ARIZONA_SLIMTX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("SLIMRX1", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX2", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX3", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX4", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX5", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX6", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX7", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("SLIMRX8", NULL, 0, + ARIZONA_SLIMBUS_RX_CHANNEL_ENABLE, + ARIZONA_SLIMRX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm8997_aec_loopback_mux), + +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUT, "SPKOUT"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(SLIMTX1, "SLIMTX1"), +ARIZONA_MIXER_WIDGETS(SLIMTX2, "SLIMTX2"), +ARIZONA_MIXER_WIDGETS(SLIMTX3, "SLIMTX3"), +ARIZONA_MIXER_WIDGETS(SLIMTX4, "SLIMTX4"), +ARIZONA_MIXER_WIDGETS(SLIMTX5, "SLIMTX5"), +ARIZONA_MIXER_WIDGETS(SLIMTX6, "SLIMTX6"), +ARIZONA_MIXER_WIDGETS(SLIMTX7, "SLIMTX7"), +ARIZONA_MIXER_WIDGETS(SLIMTX8, "SLIMTX8"), + +ARIZONA_MUX_WIDGETS(ISRC1DEC1, "ISRC1DEC1"), +ARIZONA_MUX_WIDGETS(ISRC1DEC2, "ISRC1DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC1INT1, "ISRC1INT1"), +ARIZONA_MUX_WIDGETS(ISRC1INT2, "ISRC1INT2"), + +ARIZONA_MUX_WIDGETS(ISRC2DEC1, "ISRC2DEC1"), +ARIZONA_MUX_WIDGETS(ISRC2DEC2, "ISRC2DEC2"), + +ARIZONA_MUX_WIDGETS(ISRC2INT1, "ISRC2INT1"), +ARIZONA_MUX_WIDGETS(ISRC2INT2, "ISRC2INT2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTN"), +SND_SOC_DAPM_OUTPUT("SPKOUTP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), + +SND_SOC_DAPM_OUTPUT("MICSUPP"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "Haptics", "HAPTICS" }, \ + { name, "AEC", "AEC Loopback" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "SLIMRX1", "SLIMRX1" }, \ + { name, "SLIMRX2", "SLIMRX2" }, \ + { name, "SLIMRX3", "SLIMRX3" }, \ + { name, "SLIMRX4", "SLIMRX4" }, \ + { name, "SLIMRX5", "SLIMRX5" }, \ + { name, "SLIMRX6", "SLIMRX6" }, \ + { name, "SLIMRX7", "SLIMRX7" }, \ + { name, "SLIMRX8", "SLIMRX8" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ISRC1DEC1", "ISRC1DEC1" }, \ + { name, "ISRC1DEC2", "ISRC1DEC2" }, \ + { name, "ISRC1INT1", "ISRC1INT1" }, \ + { name, "ISRC1INT2", "ISRC1INT2" }, \ + { name, "ISRC2DEC1", "ISRC2DEC1" }, \ + { name, "ISRC2DEC2", "ISRC2DEC2" }, \ + { name, "ISRC2INT1", "ISRC2INT1" }, \ + { name, "ISRC2INT2", "ISRC2INT2" } + +static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDD" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + + { "IN1L", NULL, "SYSCLK" }, + { "IN1R", NULL, "SYSCLK" }, + { "IN2L", NULL, "SYSCLK" }, + { "IN2R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "SYSCLK" }, + { "Tone Generator 1", NULL, "SYSCLK" }, + { "Tone Generator 2", NULL, "SYSCLK" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "Slim1 Capture", NULL, "SLIMTX1" }, + { "Slim1 Capture", NULL, "SLIMTX2" }, + { "Slim1 Capture", NULL, "SLIMTX3" }, + { "Slim1 Capture", NULL, "SLIMTX4" }, + + { "SLIMRX1", NULL, "Slim1 Playback" }, + { "SLIMRX2", NULL, "Slim1 Playback" }, + { "SLIMRX3", NULL, "Slim1 Playback" }, + { "SLIMRX4", NULL, "Slim1 Playback" }, + + { "Slim2 Capture", NULL, "SLIMTX5" }, + { "Slim2 Capture", NULL, "SLIMTX6" }, + + { "SLIMRX5", NULL, "Slim2 Playback" }, + { "SLIMRX6", NULL, "Slim2 Playback" }, + + { "Slim3 Capture", NULL, "SLIMTX7" }, + { "Slim3 Capture", NULL, "SLIMTX8" }, + + { "SLIMRX7", NULL, "Slim3 Playback" }, + { "SLIMRX8", NULL, "Slim3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "Slim1 Playback", NULL, "SYSCLK" }, + { "Slim2 Playback", NULL, "SYSCLK" }, + { "Slim3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "Slim1 Capture", NULL, "SYSCLK" }, + { "Slim2 Capture", NULL, "SYSCLK" }, + { "Slim3 Capture", NULL, "SYSCLK" }, + + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUT"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("SLIMTX1", "SLIMTX1"), + ARIZONA_MIXER_ROUTES("SLIMTX2", "SLIMTX2"), + ARIZONA_MIXER_ROUTES("SLIMTX3", "SLIMTX3"), + ARIZONA_MIXER_ROUTES("SLIMTX4", "SLIMTX4"), + ARIZONA_MIXER_ROUTES("SLIMTX5", "SLIMTX5"), + ARIZONA_MIXER_ROUTES("SLIMTX6", "SLIMTX6"), + ARIZONA_MIXER_ROUTES("SLIMTX7", "SLIMTX7"), + ARIZONA_MIXER_ROUTES("SLIMTX8", "SLIMTX8"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), + ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC1DEC1", "ISRC1DEC1"), + ARIZONA_MUX_ROUTES("ISRC1DEC2", "ISRC1DEC2"), + + ARIZONA_MUX_ROUTES("ISRC2INT1", "ISRC2INT1"), + ARIZONA_MUX_ROUTES("ISRC2INT2", "ISRC2INT2"), + + ARIZONA_MUX_ROUTES("ISRC2DEC1", "ISRC2DEC1"), + ARIZONA_MUX_ROUTES("ISRC2DEC2", "ISRC2DEC2"), + + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "AEC Loopback", "EPOUT", "OUT3L" }, + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "AEC Loopback", "SPKOUT", "OUT4L" }, + { "SPKOUTN", NULL, "OUT4L" }, + { "SPKOUTP", NULL, "OUT4L" }, + + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "MICSUPP", NULL, "SYSCLK" }, +}; + +static int wm8997_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm8997_priv *wm8997 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM8997_FLL1: + return arizona_set_fll(&wm8997->fll[0], source, Fref, Fout); + case WM8997_FLL2: + return arizona_set_fll(&wm8997->fll[1], source, Fref, Fout); + case WM8997_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[0], source, Fref, + Fout); + case WM8997_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm8997->fll[1], source, Fref, + Fout); + default: + return -EINVAL; + } +} + +#define WM8997_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM8997_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm8997_dai[] = { + { + .name = "wm8997-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm8997-slim1", + .id = 3, + .playback = { + .stream_name = "Slim1 Playback", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim1 Capture", + .channels_min = 1, + .channels_max = 4, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim2", + .id = 4, + .playback = { + .stream_name = "Slim2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, + { + .name = "wm8997-slim3", + .id = 5, + .playback = { + .stream_name = "Slim3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .capture = { + .stream_name = "Slim3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8997_RATES, + .formats = WM8997_FORMATS, + }, + .ops = &arizona_simple_dai_ops, + }, +}; + +static int wm8997_codec_probe(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = priv->core.arizona->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + if (ret != 0) + return ret; + + arizona_init_spk(codec); + + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); + + priv->core.arizona->dapm = &codec->dapm; + + return 0; +} + +static int wm8997_codec_remove(struct snd_soc_codec *codec) +{ + struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); + + priv->core.arizona->dapm = NULL; + + return 0; +} + +#define WM8997_DIG_VU 0x0200 + +static unsigned int wm8997_digital_vu[] = { + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm8997 = { + .probe = wm8997_codec_probe, + .remove = wm8997_codec_remove, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm8997_set_fll, + + .controls = wm8997_snd_controls, + .num_controls = ARRAY_SIZE(wm8997_snd_controls), + .dapm_widgets = wm8997_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8997_dapm_widgets), + .dapm_routes = wm8997_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8997_dapm_routes), +}; + +static int wm8997_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm8997_priv *wm8997; + int i; + + wm8997 = devm_kzalloc(&pdev->dev, sizeof(struct wm8997_priv), + GFP_KERNEL); + if (wm8997 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm8997); + + wm8997->core.arizona = arizona; + wm8997->core.num_inputs = 4; + + for (i = 0; i < ARRAY_SIZE(wm8997->fll); i++) + wm8997->fll[i].vco_mult = 1; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm8997->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm8997->fll[1]); + + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + + for (i = 0; i < ARRAY_SIZE(wm8997_dai); i++) + arizona_init_dai(&wm8997->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm8997_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm8997_digital_vu[i], + WM8997_DIG_VU, WM8997_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8997, + wm8997_dai, ARRAY_SIZE(wm8997_dai)); +} + +static int wm8997_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm8997_codec_driver = { + .driver = { + .name = "wm8997-codec", + .owner = THIS_MODULE, + }, + .probe = wm8997_probe, + .remove = wm8997_remove, +}; + +module_platform_driver(wm8997_codec_driver); + +MODULE_DESCRIPTION("ASoC WM8997 driver"); +MODULE_AUTHOR("Charles Keepax "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8997-codec"); diff --git a/sound/soc/codecs/wm8997.h b/sound/soc/codecs/wm8997.h new file mode 100644 index 000000000000..5e91c6a7d567 --- /dev/null +++ b/sound/soc/codecs/wm8997.h @@ -0,0 +1,23 @@ +/* + * wm8997.h -- WM8997 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8997_H +#define _WM8997_H + +#include "arizona.h" + +#define WM8997_FLL1 1 +#define WM8997_FLL2 2 +#define WM8997_FLL1_REFCLK 3 +#define WM8997_FLL2_REFCLK 4 + +#endif -- cgit v1.2.3 From 4601736a6f8e7ae09f1010df02e1ced605043cad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 12:28:42 +0100 Subject: ASoC: ak4554: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4554.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4554.c b/sound/soc/codecs/ak4554.c index 6aed9c4d06d6..79e9555766c0 100644 --- a/sound/soc/codecs/ak4554.c +++ b/sound/soc/codecs/ak4554.c @@ -29,6 +29,22 @@ * CPU-DAI2 (capture only fmt = LEFT_J) ---+ */ +static const struct snd_soc_dapm_widget ak4554_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route ak4554_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, + + { "AOUTL", NULL, "Playback" }, + { "AOUTR", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver ak4554_dai = { .name = "ak4554-hifi", .playback = { @@ -49,6 +65,10 @@ static struct snd_soc_dai_driver ak4554_dai = { }; static struct snd_soc_codec_driver soc_codec_dev_ak4554 = { + .dapm_widgets = ak4554_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4554_dapm_widgets), + .dapm_routes = ak4554_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ak4554_dapm_routes), }; static int ak4554_soc_probe(struct platform_device *pdev) -- cgit v1.2.3 From 997288e3824e4c6a7ab4ca4f580fe35e138d62e8 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 13:10:19 +0100 Subject: ASoC: max9877: Convert to use regmap API Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 187 ++++++++++++++++++++++++++++----------------- 1 file changed, 118 insertions(+), 69 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 6b6c74cd83e2..7e2fe5023fb2 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -14,27 +14,21 @@ #include #include #include +#include #include #include #include "max9877.h" -static struct i2c_client *i2c; +static struct regmap *regmap; -static u8 max9877_regs[5] = { 0x40, 0x00, 0x00, 0x00, 0x49 }; - -static void max9877_write_regs(void) -{ - unsigned int i; - u8 data[6]; - - data[0] = MAX9877_INPUT_MODE; - for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) - data[i + 1] = max9877_regs[i]; - - if (i2c_master_send(i2c, data, 6) != 6) - dev_err(&i2c->dev, "i2c write failed\n"); -} +static struct reg_default max9877_regs[] = { + { 0, 0x40 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x49 }, +}; static int max9877_get_reg(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -45,8 +39,14 @@ static int max9877_get_reg(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int mask = mc->max; unsigned int invert = mc->invert; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; + ucontrol->value.integer.value[0] = (val >> shift) & mask; if (invert) ucontrol->value.integer.value[0] = @@ -65,18 +65,21 @@ static int max9877_set_reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int invert = mc->invert; unsigned int val = (ucontrol->value.integer.value[0] & mask); + bool change; + int ret; if (invert) val = mask - val; - if (((max9877_regs[reg] >> shift) & mask) == val) - return 0; - - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_write_regs(); + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change); + if (ret != 0) + return ret; - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_2reg(struct snd_kcontrol *kcontrol, @@ -88,9 +91,18 @@ static int max9877_get_2reg(struct snd_kcontrol *kcontrol, unsigned int reg2 = mc->rreg; unsigned int shift = mc->shift; unsigned int mask = mc->max; + unsigned int val; + int ret; + + ret = regmap_read(regmap, reg, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[0] = (val >> shift) & mask; - ucontrol->value.integer.value[0] = (max9877_regs[reg] >> shift) & mask; - ucontrol->value.integer.value[1] = (max9877_regs[reg2] >> shift) & mask; + ret = regmap_read(regmap, reg2, &val); + if (ret != 0) + return ret; + ucontrol->value.integer.value[1] = (val >> shift) & mask; return 0; } @@ -106,77 +118,99 @@ static int max9877_set_2reg(struct snd_kcontrol *kcontrol, unsigned int mask = mc->max; unsigned int val = (ucontrol->value.integer.value[0] & mask); unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - unsigned int change = 0; - - if (((max9877_regs[reg] >> shift) & mask) != val) - change = 1; - - if (((max9877_regs[reg2] >> shift) & mask) != val2) - change = 1; - - if (change) { - max9877_regs[reg] &= ~(mask << shift); - max9877_regs[reg] |= val << shift; - max9877_regs[reg2] &= ~(mask << shift); - max9877_regs[reg2] |= val2 << shift; - max9877_write_regs(); - } - - return change; + bool change1, change2; + int ret; + + ret = regmap_update_bits_check(regmap, reg, mask << shift, + val << shift, &change1); + if (ret != 0) + return ret; + + ret = regmap_update_bits_check(regmap, reg2, mask << shift, + val2 << shift, &change2); + if (ret != 0) + return ret; + + if (change1 || change2) + return 1; + else + return 0; } static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK; + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OUTMODE_MASK; + if (val) + val--; - if (value) - value -= 1; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; + unsigned int val; + bool change; + int ret; - value += 1; + val = ucontrol->value.integer.value[0] + 1; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OUTMODE_MASK) == value) - return 0; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OUTMODE_MASK, val, &change); + if (ret != 0) + return ret; - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OUTMODE_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; + if (change) + return 1; + else + return 0; } static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = (max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK); + unsigned int val; + int ret; + + ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); + if (ret != 0) + return ret; + + val &= MAX9877_OSC_MASK; + val >>= MAX9877_OSC_OFFSET; - value = value >> MAX9877_OSC_OFFSET; + ucontrol->value.integer.value[0] = val; - ucontrol->value.integer.value[0] = value; return 0; } static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - u8 value = ucontrol->value.integer.value[0]; - - value = value << MAX9877_OSC_OFFSET; - if ((max9877_regs[MAX9877_OUTPUT_MODE] & MAX9877_OSC_MASK) == value) + unsigned int val; + bool change; + int ret; + + val = ucontrol->value.integer.value[0] << MAX9877_OSC_OFFSET; + ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, + MAX9877_OSC_MASK, val, &change); + if (ret != 0) + return ret; + + if (change) + return 1; + else return 0; - - max9877_regs[MAX9877_OUTPUT_MODE] &= ~MAX9877_OSC_MASK; - max9877_regs[MAX9877_OUTPUT_MODE] |= value; - max9877_write_regs(); - return 1; } static const unsigned int max9877_pgain_tlv[] = { @@ -258,19 +292,34 @@ int max9877_add_controls(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct regmap_config max9877_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .reg_defaults = max9877_regs, + .num_reg_defaults = ARRAY_SIZE(max9877_regs), + .cache_type = REGCACHE_RBTREE, +}; + static int max9877_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - i2c = client; + int i; + + regmap = devm_regmap_init_i2c(client, &max9877_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - max9877_write_regs(); + /* Ensure the device is in reset state */ + for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) + regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); return 0; } static int max9877_i2c_remove(struct i2c_client *client) { - i2c = NULL; + regmap = NULL; return 0; } -- cgit v1.2.3 From d76a96174b31bd916c1dfaa81a3db82fc8c54b91 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 13:20:15 +0100 Subject: ASoC: max9877: Convert to standard CODEC driver Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 256 ++++++--------------------------------------- 1 file changed, 31 insertions(+), 225 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 7e2fe5023fb2..8505b401d3c4 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -30,189 +30,6 @@ static struct reg_default max9877_regs[] = { { 4, 0x49 }, }; -static int max9877_get_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int invert = mc->invert; - unsigned int val; - int ret; - - ret = regmap_read(regmap, reg, &val); - if (ret != 0) - return ret; - - ucontrol->value.integer.value[0] = (val >> shift) & mask; - - if (invert) - ucontrol->value.integer.value[0] = - mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int max9877_set_reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int invert = mc->invert; - unsigned int val = (ucontrol->value.integer.value[0] & mask); - bool change; - int ret; - - if (invert) - val = mask - val; - - ret = regmap_update_bits_check(regmap, reg, mask << shift, - val << shift, &change); - if (ret != 0) - return ret; - - if (change) - return 1; - else - return 0; -} - -static int max9877_get_2reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int val; - int ret; - - ret = regmap_read(regmap, reg, &val); - if (ret != 0) - return ret; - ucontrol->value.integer.value[0] = (val >> shift) & mask; - - ret = regmap_read(regmap, reg2, &val); - if (ret != 0) - return ret; - ucontrol->value.integer.value[1] = (val >> shift) & mask; - - return 0; -} - -static int max9877_set_2reg(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct soc_mixer_control *mc = - (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; - unsigned int reg2 = mc->rreg; - unsigned int shift = mc->shift; - unsigned int mask = mc->max; - unsigned int val = (ucontrol->value.integer.value[0] & mask); - unsigned int val2 = (ucontrol->value.integer.value[1] & mask); - bool change1, change2; - int ret; - - ret = regmap_update_bits_check(regmap, reg, mask << shift, - val << shift, &change1); - if (ret != 0) - return ret; - - ret = regmap_update_bits_check(regmap, reg2, mask << shift, - val2 << shift, &change2); - if (ret != 0) - return ret; - - if (change1 || change2) - return 1; - else - return 0; -} - -static int max9877_get_out_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - int ret; - - ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); - if (ret != 0) - return ret; - - val &= MAX9877_OUTMODE_MASK; - if (val) - val--; - - ucontrol->value.integer.value[0] = val; - - return 0; -} - -static int max9877_set_out_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - bool change; - int ret; - - val = ucontrol->value.integer.value[0] + 1; - - ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, - MAX9877_OUTMODE_MASK, val, &change); - if (ret != 0) - return ret; - - if (change) - return 1; - else - return 0; -} - -static int max9877_get_osc_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - int ret; - - ret = regmap_read(regmap, MAX9877_OUTPUT_MODE, &val); - if (ret != 0) - return ret; - - val &= MAX9877_OSC_MASK; - val >>= MAX9877_OSC_OFFSET; - - ucontrol->value.integer.value[0] = val; - - return 0; -} - -static int max9877_set_osc_mode(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - unsigned int val; - bool change; - int ret; - - val = ucontrol->value.integer.value[0] << MAX9877_OSC_OFFSET; - ret = regmap_update_bits_check(regmap, MAX9877_OUTPUT_MODE, - MAX9877_OSC_MASK, val, &change); - if (ret != 0) - return ret; - - if (change) - return 1; - else - return 0; -} - static const unsigned int max9877_pgain_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 1, TLV_DB_SCALE_ITEM(0, 900, 0), @@ -246,51 +63,40 @@ static const char *max9877_osc_mode[] = { }; static const struct soc_enum max9877_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_out_mode), max9877_out_mode), - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), + SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, 0, ARRAY_SIZE(max9877_out_mode), + max9877_out_mode), + SOC_ENUM_SINGLE(MAX9877_OUTPUT_MODE, MAX9877_OSC_OFFSET, + ARRAY_SIZE(max9877_osc_mode), max9877_osc_mode), }; static const struct snd_kcontrol_new max9877_controls[] = { - SOC_SINGLE_EXT_TLV("MAX9877 PGAINA Playback Volume", - MAX9877_INPUT_MODE, 0, 2, 0, - max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), - SOC_SINGLE_EXT_TLV("MAX9877 PGAINB Playback Volume", - MAX9877_INPUT_MODE, 2, 2, 0, - max9877_get_reg, max9877_set_reg, max9877_pgain_tlv), - SOC_SINGLE_EXT_TLV("MAX9877 Amp Speaker Playback Volume", - MAX9877_SPK_VOLUME, 0, 31, 0, - max9877_get_reg, max9877_set_reg, max9877_output_tlv), - SOC_DOUBLE_R_EXT_TLV("MAX9877 Amp HP Playback Volume", - MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, - max9877_get_2reg, max9877_set_2reg, max9877_output_tlv), - SOC_SINGLE_EXT("MAX9877 INB Stereo Switch", - MAX9877_INPUT_MODE, 4, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 INA Stereo Switch", - MAX9877_INPUT_MODE, 5, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Zero-crossing detection Switch", - MAX9877_INPUT_MODE, 6, 1, 0, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Bypass Mode Switch", - MAX9877_OUTPUT_MODE, 6, 1, 0, - max9877_get_reg, max9877_set_reg), - SOC_SINGLE_EXT("MAX9877 Shutdown Mode Switch", - MAX9877_OUTPUT_MODE, 7, 1, 1, - max9877_get_reg, max9877_set_reg), - SOC_ENUM_EXT("MAX9877 Output Mode", max9877_enum[0], - max9877_get_out_mode, max9877_set_out_mode), - SOC_ENUM_EXT("MAX9877 Oscillator Mode", max9877_enum[1], - max9877_get_osc_mode, max9877_set_osc_mode), + SOC_SINGLE_TLV("MAX9877 PGAINA Playback Volume", + MAX9877_INPUT_MODE, 0, 2, 0, max9877_pgain_tlv), + SOC_SINGLE_TLV("MAX9877 PGAINB Playback Volume", + MAX9877_INPUT_MODE, 2, 2, 0, max9877_pgain_tlv), + SOC_SINGLE_TLV("MAX9877 Amp Speaker Playback Volume", + MAX9877_SPK_VOLUME, 0, 31, 0, max9877_output_tlv), + SOC_DOUBLE_R_TLV("MAX9877 Amp HP Playback Volume", + MAX9877_HPL_VOLUME, MAX9877_HPR_VOLUME, 0, 31, 0, + max9877_output_tlv), + SOC_SINGLE("MAX9877 INB Stereo Switch", + MAX9877_INPUT_MODE, 4, 1, 1), + SOC_SINGLE("MAX9877 INA Stereo Switch", + MAX9877_INPUT_MODE, 5, 1, 1), + SOC_SINGLE("MAX9877 Zero-crossing detection Switch", + MAX9877_INPUT_MODE, 6, 1, 0), + SOC_SINGLE("MAX9877 Bypass Mode Switch", + MAX9877_OUTPUT_MODE, 6, 1, 0), + SOC_SINGLE("MAX9877 Shutdown Mode Switch", + MAX9877_OUTPUT_MODE, 7, 1, 1), + SOC_ENUM("MAX9877 Output Mode", max9877_enum[0]), + SOC_ENUM("MAX9877 Oscillator Mode", max9877_enum[1]), }; -/* This function is called from ASoC machine driver */ -int max9877_add_controls(struct snd_soc_codec *codec) -{ - return snd_soc_add_codec_controls(codec, max9877_controls, - ARRAY_SIZE(max9877_controls)); -} -EXPORT_SYMBOL_GPL(max9877_add_controls); +static const struct snd_soc_codec_driver max9877_codec = { + .controls = max9877_controls, + .num_controls = ARRAY_SIZE(max9877_controls), +}; static const struct regmap_config max9877_regmap = { .reg_bits = 8, @@ -314,12 +120,12 @@ static int max9877_i2c_probe(struct i2c_client *client, for (i = 0; i < ARRAY_SIZE(max9877_regs); i++) regmap_write(regmap, max9877_regs[i].reg, max9877_regs[i].def); - return 0; + return snd_soc_register_codec(&client->dev, &max9877_codec, NULL, 0); } static int max9877_i2c_remove(struct i2c_client *client) { - regmap = NULL; + snd_soc_unregister_codec(&client->dev); return 0; } -- cgit v1.2.3 From 5cf9da8aacbfaed72ada8c195859f49d5d7f5f6c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 13:32:03 +0100 Subject: ASoC: max9877: Add basic DAPM support This does not fully map the power control available within the device but it provides the hooks for routing signals through the device and allows automatic management of the device low power mode. Signed-off-by: Mark Brown --- sound/soc/codecs/max9877.c | 39 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 37 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max9877.c b/sound/soc/codecs/max9877.c index 8505b401d3c4..29549cdbf4c1 100644 --- a/sound/soc/codecs/max9877.c +++ b/sound/soc/codecs/max9877.c @@ -87,15 +87,50 @@ static const struct snd_kcontrol_new max9877_controls[] = { MAX9877_INPUT_MODE, 6, 1, 0), SOC_SINGLE("MAX9877 Bypass Mode Switch", MAX9877_OUTPUT_MODE, 6, 1, 0), - SOC_SINGLE("MAX9877 Shutdown Mode Switch", - MAX9877_OUTPUT_MODE, 7, 1, 1), SOC_ENUM("MAX9877 Output Mode", max9877_enum[0]), SOC_ENUM("MAX9877 Oscillator Mode", max9877_enum[1]), }; +static const struct snd_soc_dapm_widget max9877_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("INA1"), +SND_SOC_DAPM_INPUT("INA2"), +SND_SOC_DAPM_INPUT("INB1"), +SND_SOC_DAPM_INPUT("INB2"), +SND_SOC_DAPM_INPUT("RXIN+"), +SND_SOC_DAPM_INPUT("RXIN-"), + +SND_SOC_DAPM_PGA("SHDN", MAX9877_OUTPUT_MODE, 7, 1, NULL, 0), + +SND_SOC_DAPM_OUTPUT("OUT+"), +SND_SOC_DAPM_OUTPUT("OUT-"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_route max9877_dapm_routes[] = { + { "SHDN", NULL, "INA1" }, + { "SHDN", NULL, "INA2" }, + { "SHDN", NULL, "INB1" }, + { "SHDN", NULL, "INB2" }, + + { "OUT+", NULL, "RXIN+" }, + { "OUT+", NULL, "SHDN" }, + + { "OUT-", NULL, "SHDN" }, + { "OUT-", NULL, "RXIN-" }, + + { "HPL", NULL, "SHDN" }, + { "HPR", NULL, "SHDN" }, +}; + static const struct snd_soc_codec_driver max9877_codec = { .controls = max9877_controls, .num_controls = ARRAY_SIZE(max9877_controls), + + .dapm_widgets = max9877_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9877_dapm_widgets), + .dapm_routes = max9877_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9877_dapm_routes), }; static const struct regmap_config max9877_regmap = { -- cgit v1.2.3 From c324aac01be55253488aba9481523cd6f546f4ca Mon Sep 17 00:00:00 2001 From: Ma Haijun Date: Wed, 14 Aug 2013 09:15:38 +0800 Subject: ASoC: wm8960: Fix ADC volume bits Signed-off-by: Ma Haijun Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd1d2a7..368d39f19cdb 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -263,8 +263,8 @@ SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), -SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, - 0, 127, 0), +SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC, + 0, 255, 0, adc_tlv), SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", WM8960_BYPASS1, 4, 7, 1, bypass_tlv), -- cgit v1.2.3 From 5332e1d26fd182444e15e6481029347ab032d7cb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 18:29:44 +0100 Subject: ASoC: pcm1792a: Remove empty capture DAI stub These intialisations are just what will be done for static data anyway so remove them. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 72cf8353e812..c57d3a5665c5 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -162,10 +162,6 @@ static struct snd_soc_dai_driver pcm1792a_dai = { .channels_max = 2, .rates = PCM1792A_RATES, .formats = PCM1792A_FORMATS, }, - .capture = { - .channels_min = 0, - .channels_max = 0, - }, .ops = &pcm1792a_dai_ops, }; -- cgit v1.2.3 From e7a5cb4223c86df522a97e21742aeef153db4ebb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 18:30:00 +0100 Subject: ASoC: pcm1792a: Add DAPM support Provide DAPM for the device, ensuring operation with DAPM required by the core and making it easier to hook up external hardware to it. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1792a.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index c57d3a5665c5..2a8eccf64c76 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -154,6 +154,20 @@ static const struct snd_kcontrol_new pcm1792a_controls[] = { pcm1792a_dac_tlv), }; +static const struct snd_soc_dapm_widget pcm1792a_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("IOUTL+"), +SND_SOC_DAPM_OUTPUT("IOUTL-"), +SND_SOC_DAPM_OUTPUT("IOUTR+"), +SND_SOC_DAPM_OUTPUT("IOUTR-"), +}; + +static const struct snd_soc_dapm_route pcm1792a_dapm_routes[] = { + { "IOUTL+", NULL, "Playback" }, + { "IOUTL-", NULL, "Playback" }, + { "IOUTR+", NULL, "Playback" }, + { "IOUTR-", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver pcm1792a_dai = { .name = "pcm1792a-hifi", .playback = { @@ -184,6 +198,10 @@ static const struct regmap_config pcm1792a_regmap = { static struct snd_soc_codec_driver soc_codec_dev_pcm1792a = { .controls = pcm1792a_controls, .num_controls = ARRAY_SIZE(pcm1792a_controls), + .dapm_widgets = pcm1792a_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1792a_dapm_widgets), + .dapm_routes = pcm1792a_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1792a_dapm_routes), }; static int pcm1792a_spi_probe(struct spi_device *spi) -- cgit v1.2.3 From b9281f99e30f795f28f6ea216289900b6e870d01 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 Aug 2013 18:25:41 +0100 Subject: ASoC: pcm1681: Add DAPM support Provide DAPM for the device, ensuring operation with DAPM required by the core and making it easier to hook up external hardware to it. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 51b18662e6aa..651ce0923675 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -206,6 +206,28 @@ static const struct snd_soc_dai_ops pcm1681_dai_ops = { .digital_mute = pcm1681_digital_mute, }; +static const struct snd_soc_dapm_widget pcm1681_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUT1"), +SND_SOC_DAPM_OUTPUT("VOUT2"), +SND_SOC_DAPM_OUTPUT("VOUT3"), +SND_SOC_DAPM_OUTPUT("VOUT4"), +SND_SOC_DAPM_OUTPUT("VOUT5"), +SND_SOC_DAPM_OUTPUT("VOUT6"), +SND_SOC_DAPM_OUTPUT("VOUT7"), +SND_SOC_DAPM_OUTPUT("VOUT8"), +}; + +static const struct snd_soc_dapm_route pcm1681_dapm_routes[] = { + { "VOUT1", NULL, "Playback" }, + { "VOUT2", NULL, "Playback" }, + { "VOUT3", NULL, "Playback" }, + { "VOUT4", NULL, "Playback" }, + { "VOUT5", NULL, "Playback" }, + { "VOUT6", NULL, "Playback" }, + { "VOUT7", NULL, "Playback" }, + { "VOUT8", NULL, "Playback" }, +}; + static const DECLARE_TLV_DB_SCALE(pcm1681_dac_tlv, -6350, 50, 1); static const struct snd_kcontrol_new pcm1681_controls[] = { @@ -258,6 +280,10 @@ static const struct regmap_config pcm1681_regmap = { static struct snd_soc_codec_driver soc_codec_dev_pcm1681 = { .controls = pcm1681_controls, .num_controls = ARRAY_SIZE(pcm1681_controls), + .dapm_widgets = pcm1681_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm1681_dapm_widgets), + .dapm_routes = pcm1681_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm1681_dapm_routes), }; static const struct i2c_device_id pcm1681_i2c_id[] = { -- cgit v1.2.3 From 12201398fc9ad25f5c6568527e70c9a4bcf5fcee Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 12:13:50 +0100 Subject: ASoC: tlv320aic26: Remove direct use of internal I/O functions Use the core to do I/O rather than directly calling the driver operations in order to support further refactoring. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index b192cd4705a0..a4f93608287e 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -174,9 +174,9 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG1, reg); reg = dval << 2; - aic26_reg_write(codec, AIC26_REG_PLL_PROG2, reg); + snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); @@ -185,13 +185,13 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, reg |= 0x0800; if (fsref == 48000) reg |= 0x2000; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Audio Control 1 (FSref divisor) */ reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); reg &= ~0x0fff; reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); return 0; } @@ -212,7 +212,7 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) reg |= 0x8080; else reg &= ~0x8080; - aic26_reg_write(codec, AIC26_REG_DAC_GAIN, reg); + snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); return 0; } @@ -348,7 +348,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); val |= 0x8000; - aic26_reg_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); return count; } @@ -368,20 +368,20 @@ static int aic26_probe(struct snd_soc_codec *codec) dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); /* Reset the codec to power on defaults */ - aic26_reg_write(codec, AIC26_REG_RESET, 0xBB00); + snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); /* Power up CODEC */ - aic26_reg_write(codec, AIC26_REG_POWER_CTRL, 0); + snd_soc_write(codec, AIC26_REG_POWER_CTRL, 0); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read(codec, AIC26_REG_AUDIO_CTRL3); + reg = snd_soc_read(codec, AIC26_REG_AUDIO_CTRL3); reg &= ~0xf800; reg |= 0x0800; /* set master mode */ - aic26_reg_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); /* Fill register cache */ for (i = 0; i < codec->driver->reg_cache_size; i++) - aic26_reg_read(codec, i); + snd_soc_read(codec, i); /* Register the sysfs files for debugging */ /* Create SysFS files */ -- cgit v1.2.3 From c21bb9b1b7de87ee33c8ebf94a155be2aa551849 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 12:14:42 +0100 Subject: ASoC: tlv320aic26: Remove noisy print Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index a4f93608287e..93cf692a8db7 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -365,8 +365,6 @@ static int aic26_probe(struct snd_soc_codec *codec) aic26->codec = codec; - dev_info(codec->dev, "Probing AIC26 SoC CODEC driver\n"); - /* Reset the codec to power on defaults */ snd_soc_write(codec, AIC26_REG_RESET, 0xBB00); -- cgit v1.2.3 From 4a11bc2fdd7f526c70e013366171d66f27656203 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 12:24:47 +0100 Subject: ASoC: tlv320aic26: Add basic DAPM support Provide external widgets for the CODEC to ensure the device continues to function with DAPM mandatory and to make it easier to hook the device up to other components. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 93cf692a8db7..7b8f3d965f43 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -119,6 +119,22 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } +static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICIN"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +}; + +static const struct snd_soc_dapm_route tlv320aic26_dapm_routes[] = { + { "Capture", NULL, "MICIN" }, + { "Capture", NULL, "AUX" }, + + { "HPL", NULL, "Playback" }, + { "HPR", NULL, "Playback" }, +}; + /* --------------------------------------------------------------------- * Digital Audio Interface Operations */ @@ -402,6 +418,10 @@ static struct snd_soc_codec_driver aic26_soc_codec_dev = { .write = aic26_reg_write, .reg_cache_size = AIC26_NUM_REGS, .reg_word_size = sizeof(u16), + .dapm_widgets = tlv320aic26_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), + .dapm_routes = tlv320aic26_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), }; /* --------------------------------------------------------------------- -- cgit v1.2.3 From ac0b82b17894120b21937d0031fd0080b3ee2d83 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 16 Aug 2013 11:54:51 +0100 Subject: ASoC: si476x: Add DAPM support This ensures the driver continues to work with DAPM mandatory and makes it easier to connect the device up to other components in the subsystem. Signed-off-by: Mark Brown Acked-by: Andrey Smirnov --- sound/soc/codecs/si476x.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 73e205c892a0..38f3b105c17d 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -102,6 +102,16 @@ static int si476x_codec_write(struct snd_soc_codec *codec, return err; } +static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("LOUT"), +SND_SOC_DAPM_OUTPUT("ROUT"), +}; + +static const struct snd_soc_dapm_route si476x_dapm_routes[] = { + { "Capture", NULL, "LOUT" }, + { "Capture", NULL, "ROUT" }, +}; + static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -260,6 +270,10 @@ static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, .read = si476x_codec_read, .write = si476x_codec_write, + .dapm_widgets = si476x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), + .dapm_routes = si476x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(si476x_dapm_routes), }; static int si476x_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From 85fa532b6ef920b32598df86b194571a7059a77c Mon Sep 17 00:00:00 2001 From: Mike Dyer Date: Fri, 16 Aug 2013 18:36:28 +0100 Subject: ASoC: wm8960: Fix PLL register writes Bit 9 of PLL2,3 and 4 is reserved as '0'. The 24bit fractional part should be split across each register in 8bit chunks. Signed-off-by: Mike Dyer Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8960.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 0a4ffdd1d2a7..5e5af898f7f8 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -857,9 +857,9 @@ static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, if (pll_div.k) { reg |= 0x20; - snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); - snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); - snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + snd_soc_write(codec, WM8960_PLL2, (pll_div.k >> 16) & 0xff); + snd_soc_write(codec, WM8960_PLL3, (pll_div.k >> 8) & 0xff); + snd_soc_write(codec, WM8960_PLL4, pll_div.k & 0xff); } snd_soc_write(codec, WM8960_PLL1, reg); -- cgit v1.2.3 From ea67afc3fdbe9196d76ee79503a3809a54300b5a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Aug 2013 11:53:28 +0100 Subject: ASoC: pcm3008: Use gpio_set_value_cansleep() We don't set the GPIO values from atomic context so support GPIOs that can't be controlled from atomic context. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index b883f99d6f9f..8b9b378bc0a7 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -54,8 +54,8 @@ static int pcm3008_soc_suspend(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; - gpio_set_value(setup->pdad_pin, 0); - gpio_set_value(setup->pdda_pin, 0); + gpio_set_value_cansleep(setup->pdad_pin, 0); + gpio_set_value_cansleep(setup->pdda_pin, 0); return 0; } @@ -64,8 +64,8 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) { struct pcm3008_setup_data *setup = codec->dev->platform_data; - gpio_set_value(setup->pdad_pin, 1); - gpio_set_value(setup->pdda_pin, 1); + gpio_set_value_cansleep(setup->pdad_pin, 1); + gpio_set_value_cansleep(setup->pdda_pin, 1); return 0; } -- cgit v1.2.3 From faaf36f21642a140715b7d6cf897ab4f4f5a924d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Aug 2013 12:01:40 +0100 Subject: ASoC: pcm3008: Add DAPM support Make it possible to connect external devices to the CODEC and ensure continued operation with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 8b9b378bc0a7..19f5028354f6 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,6 +28,22 @@ #include "pcm3008.h" +static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("VINL"), +SND_SOC_DAPM_INPUT("VINR"), + +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { + { "PCM3008 Capture", NULL, "VINL" }, + { "PCM3008 Capture", NULL, "VINR" }, + + { "VOUTL", NULL, "PCM3008 Playback" }, + { "VOUTR", NULL, "PCM3008 Playback" }, +}; + #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) @@ -77,6 +93,10 @@ static int pcm3008_soc_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { .suspend = pcm3008_soc_suspend, .resume = pcm3008_soc_resume, + .dapm_widgets = pcm3008_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), + .dapm_routes = pcm3008_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm3008_dapm_routes), }; static int pcm3008_codec_probe(struct platform_device *pdev) -- cgit v1.2.3 From 4fc932c6d8c0d2715bb7f2a2f657230ea360af87 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 15 Aug 2013 12:04:28 +0100 Subject: ASoC: pcm3008: Manage DAC and ADC power with DAPM Rather than leaving the DAC and ADC active whenever the system is running manage their power with DAPM. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm3008.c | 72 +++++++++++++++++++++++++--------------------- 1 file changed, 39 insertions(+), 33 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 19f5028354f6..b6618c4a7597 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -28,20 +28,53 @@ #include "pcm3008.h" +static int pcm3008_dac_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value_cansleep(setup->pdda_pin, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + +static int pcm3008_adc_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct pcm3008_setup_data *setup = codec->dev->platform_data; + + gpio_set_value_cansleep(setup->pdad_pin, + SND_SOC_DAPM_EVENT_ON(event)); + + return 0; +} + static const struct snd_soc_dapm_widget pcm3008_dapm_widgets[] = { SND_SOC_DAPM_INPUT("VINL"), SND_SOC_DAPM_INPUT("VINR"), +SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_dac_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, pcm3008_adc_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUTPUT("VOUTL"), SND_SOC_DAPM_OUTPUT("VOUTR"), }; static const struct snd_soc_dapm_route pcm3008_dapm_routes[] = { - { "PCM3008 Capture", NULL, "VINL" }, - { "PCM3008 Capture", NULL, "VINR" }, + { "PCM3008 Capture", NULL, "ADC" }, + { "ADC", NULL, "VINL" }, + { "ADC", NULL, "VINR" }, - { "VOUTL", NULL, "PCM3008 Playback" }, - { "VOUTR", NULL, "PCM3008 Playback" }, + { "DAC", NULL, "PCM3008 Playback" }, + { "VOUTL", NULL, "DAC" }, + { "VOUTR", NULL, "DAC" }, }; #define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -65,34 +98,7 @@ static struct snd_soc_dai_driver pcm3008_dai = { }, }; -#ifdef CONFIG_PM -static int pcm3008_soc_suspend(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value_cansleep(setup->pdad_pin, 0); - gpio_set_value_cansleep(setup->pdda_pin, 0); - - return 0; -} - -static int pcm3008_soc_resume(struct snd_soc_codec *codec) -{ - struct pcm3008_setup_data *setup = codec->dev->platform_data; - - gpio_set_value_cansleep(setup->pdad_pin, 1); - gpio_set_value_cansleep(setup->pdda_pin, 1); - - return 0; -} -#else -#define pcm3008_soc_suspend NULL -#define pcm3008_soc_resume NULL -#endif - static struct snd_soc_codec_driver soc_codec_dev_pcm3008 = { - .suspend = pcm3008_soc_suspend, - .resume = pcm3008_soc_resume, .dapm_widgets = pcm3008_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(pcm3008_dapm_widgets), .dapm_routes = pcm3008_dapm_routes, @@ -128,13 +134,13 @@ static int pcm3008_codec_probe(struct platform_device *pdev) /* Configure PDAD GPIO. */ ret = devm_gpio_request_one(&pdev->dev, setup->pdad_pin, - GPIOF_OUT_INIT_HIGH, "codec_pdad"); + GPIOF_OUT_INIT_LOW, "codec_pdad"); if (ret != 0) return ret; /* Configure PDDA GPIO. */ ret = devm_gpio_request_one(&pdev->dev, setup->pdda_pin, - GPIOF_OUT_INIT_HIGH, "codec_pdda"); + GPIOF_OUT_INIT_LOW, "codec_pdda"); if (ret != 0) return ret; -- cgit v1.2.3 From e29deb48189cea0be8b24f5912d8c21a18cb0244 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 18 Aug 2013 18:25:53 +0100 Subject: ASoC: wl1273: Add stub DAPM support In order to ensure that the device continues to work with DAPM support being mandatory provide stub DAPM widgets and routes. Note that the public information on the device appears to make no mention of the FM support the driver appears to have. Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index 54cd3da09abd..b7ab2ef567c8 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -290,6 +290,18 @@ static const struct snd_kcontrol_new wl1273_controls[] = { snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put), }; +static const struct snd_soc_dapm_widget wl1273_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route wl1273_dapm_routes[] = { + { "Capture", NULL, "RX" }, + + { "TX", NULL, "Playback" }, +}; + static int wl1273_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -483,6 +495,11 @@ static int wl1273_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wl1273 = { .probe = wl1273_probe, .remove = wl1273_remove, + + .dapm_widgets = wl1273_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wl1273_dapm_widgets), + .dapm_routes = wl1273_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wl1273_dapm_routes), }; static int wl1273_platform_probe(struct platform_device *pdev) -- cgit v1.2.3 From 782fbaba36731d46820f3a4f358a7b46a9cd795c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 11 Aug 2013 12:29:07 +0100 Subject: ASoC: cs4270: Add DAPM support This makes it possible to hook the device into a more complex board and ensures it will continue to work with non-DAPM support removed from the core. Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 8e4779812b96..83c835d9fd88 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -139,6 +139,22 @@ struct cs4270_private { struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; }; +static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), + +SND_SOC_DAPM_OUTPUT("AOUTL"), +SND_SOC_DAPM_OUTPUT("AOUTR"), +}; + +static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { + { "Capture", NULL, "AINA" }, + { "Capture", NULL, "AINB" }, + + { "AOUTA", NULL, "Playback" }, + { "AOUTB", NULL, "Playback" }, +}; + /** * struct cs4270_mode_ratios - clock ratio tables * @ratio: the ratio of MCLK to the sample rate @@ -612,6 +628,10 @@ static const struct snd_soc_codec_driver soc_codec_device_cs4270 = { .controls = cs4270_snd_controls, .num_controls = ARRAY_SIZE(cs4270_snd_controls), + .dapm_widgets = cs4270_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs4270_dapm_widgets), + .dapm_routes = cs4270_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs4270_dapm_routes), }; /* -- cgit v1.2.3 From 72a061f763c8af8ace650ccb1d01f484a6465608 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 18 Aug 2013 18:35:54 +0100 Subject: ASoC: wm8727: Add DAPM support In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8727.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 462f5e4d5c05..7b1a6d5c11c6 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -23,6 +23,16 @@ #include #include +static const struct snd_soc_dapm_widget wm8727_dapm_widgets[] = { +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route wm8727_dapm_routes[] = { + { "VOUTL", NULL, "Playback" }, + { "VOUTR", NULL, "Playback" }, +}; + /* * Note this is a simple chip with no configuration interface, sample rate is * determined automatically by examining the Master clock and Bit clock ratios @@ -43,7 +53,12 @@ static struct snd_soc_dai_driver wm8727_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8727; +static struct snd_soc_codec_driver soc_codec_dev_wm8727 = { + .dapm_widgets = wm8727_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8727_dapm_widgets), + .dapm_routes = wm8727_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8727_dapm_routes), +}; static int wm8727_probe(struct platform_device *pdev) { -- cgit v1.2.3 From 226059e1cdbb5d747bd008eba114af0b1a4a621e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 18 Aug 2013 18:36:06 +0100 Subject: ASoC: wm8782: Add DAPM support In order to make the device easier to hook up to external components in system designs and ensure operation when DAPM support becomes mandatory add DAPM support. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8782.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index f1fdbf63abb4..8092495605ce 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -26,6 +26,16 @@ #include #include +static const struct snd_soc_dapm_widget wm8782_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AINL"), +SND_SOC_DAPM_INPUT("AINR"), +}; + +static const struct snd_soc_dapm_route wm8782_dapm_routes[] = { + { "Capture", NULL, "AINL" }, + { "Capture", NULL, "AINR" }, +}; + static struct snd_soc_dai_driver wm8782_dai = { .name = "wm8782", .capture = { @@ -40,7 +50,12 @@ static struct snd_soc_dai_driver wm8782_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_wm8782; +static struct snd_soc_codec_driver soc_codec_dev_wm8782 = { + .dapm_widgets = wm8782_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8782_dapm_widgets), + .dapm_routes = wm8782_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8782_dapm_routes), +}; static int wm8782_probe(struct platform_device *pdev) { -- cgit v1.2.3 From 3efd8a6f1a74b4bbf54c992e1cf23381c64de216 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Aug 2013 23:58:58 +0100 Subject: ASoC: wm5102: Add inputs for noise and mic mixers The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index f38c52d43b8d..8bbddc151aa8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1423,9 +1423,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -1552,6 +1549,9 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), -- cgit v1.2.3 From 66e7aa22c751af82567f9af82fe7e1254f751870 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Aug 2013 23:59:08 +0100 Subject: ASoC: wm5110: Add inputs for noise and mic mixers The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 38e50c81a953..bbd64384ca1c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -841,9 +841,6 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -978,6 +975,9 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ASRC1L", "ASRC1L"), ARIZONA_MUX_ROUTES("ASRC1R", "ASRC1R"), ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), -- cgit v1.2.3 From c5efb38a1354890297aed2a7e197ec5b23ce966a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 12 Aug 2013 23:59:19 +0100 Subject: ASoC: wm8997: Add inputs for noise and mic mixers The noise and mic mixer inputs were not connected, do so. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 0a43bac2f4e3..6ec3de3efa4f 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -774,9 +774,6 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { { "Tone Generator 1", NULL, "TONE" }, { "Tone Generator 2", NULL, "TONE" }, - { "Mic Mute Mixer", NULL, "Noise Mixer" }, - { "Mic Mute Mixer", NULL, "Mic Mixer" }, - { "AIF1 Capture", NULL, "AIF1TX1" }, { "AIF1 Capture", NULL, "AIF1TX2" }, { "AIF1 Capture", NULL, "AIF1TX3" }, @@ -886,6 +883,9 @@ static const struct snd_soc_dapm_route wm8997_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Noise"), + ARIZONA_MIXER_ROUTES("Mic Mute Mixer", "Mic"), + ARIZONA_MUX_ROUTES("ISRC1INT1", "ISRC1INT1"), ARIZONA_MUX_ROUTES("ISRC1INT2", "ISRC2INT2"), -- cgit v1.2.3 From d2a369cb53a3f3733800d5160d60f9a5271fe44c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:18:07 +0100 Subject: ASoC: ac97: Provide stub DAPM integration Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index ec7351803c24..8d9ba4ba4bfe 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -23,6 +23,16 @@ #include #include +static const struct snd_soc_dapm_widget ac97_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route ac97_routes[] = { + { "AC97 Capture", NULL, "RX" }, + { "TX", NULL, "AC97 Playback" }, +}; + static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -117,6 +127,11 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, + + .dapm_widgets = ac97_widgets, + .num_dapm_widgets = ARRAY_SIZE(ac97_widgets), + .dapm_routes = ac97_routes, + .num_dapm_routes = ARRAY_SIZE(ac97_routes), }; static int ac97_probe(struct platform_device *pdev) -- cgit v1.2.3 From c34e51b12751c3e81c752b385f02a97bf3f862da Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:17:36 +0100 Subject: ASoC: hdmi: Provide stub DAPM integration Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown --- sound/soc/codecs/hdmi.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c index f0986b9f1939..68342b121c96 100644 --- a/sound/soc/codecs/hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -23,11 +23,20 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver hdmi_codec; +static const struct snd_soc_dapm_widget hdmi_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route hdmi_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; static struct snd_soc_dai_driver hdmi_codec_dai = { .name = "hdmi-hifi", .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 8, .rates = SNDRV_PCM_RATE_32000 | @@ -38,6 +47,7 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { SNDRV_PCM_FMTBIT_S24_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | @@ -50,6 +60,13 @@ static struct snd_soc_dai_driver hdmi_codec_dai = { }; +static struct snd_soc_codec_driver hdmi_codec = { + .dapm_widgets = hdmi_widgets, + .num_dapm_widgets = ARRAY_SIZE(hdmi_widgets), + .dapm_routes = hdmi_routes, + .num_dapm_routes = ARRAY_SIZE(hdmi_routes), +}; + static int hdmi_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, &hdmi_codec, -- cgit v1.2.3 From b9dff9c3d2c6c4a9cdb936f263ed293274e2f05a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:13:14 +0100 Subject: ASoC: bt-sco: Add generic compatible string Provide a common compatible string for device trees to list as a fallback for simplicity. We don't currently have a binding document but let's not fix that right now... Signed-off-by: Mark Brown --- sound/soc/codecs/bt-sco.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index a081d9fcb166..5c040ce33f1d 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -50,6 +50,9 @@ static struct platform_device_id bt_sco_driver_ids[] = { { .name = "dfbmcs320", }, + { + .name = "bt-sco", + }, {}, }; MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); -- cgit v1.2.3 From 5195ca4902fe0b2ae01eb43ce522b89163672804 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:16:19 +0100 Subject: ASoC: bt-sco: Provide stub DAPM integration Ensure continued operation with DAPM being mandatory. Signed-off-by: Mark Brown --- sound/soc/codecs/bt-sco.c | 19 ++++++++++++++++++- 1 file changed, 18 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index 5c040ce33f1d..c4cf0699e77f 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -15,15 +15,27 @@ #include +static const struct snd_soc_dapm_widget bt_sco_widgets[] = { + SND_SOC_DAPM_INPUT("RX"), + SND_SOC_DAPM_OUTPUT("TX"), +}; + +static const struct snd_soc_dapm_route bt_sco_routes[] = { + { "Capture", NULL, "RX" }, + { "TX", NULL, "Playback" }, +}; + static struct snd_soc_dai_driver bt_sco_dai = { .name = "bt-sco-pcm", .playback = { + .stream_name = "Playback", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_8000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 1, .channels_max = 1, .rates = SNDRV_PCM_RATE_8000, @@ -31,7 +43,12 @@ static struct snd_soc_dai_driver bt_sco_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_bt_sco; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco = { + .dapm_widgets = bt_sco_widgets, + .num_dapm_widgets = ARRAY_SIZE(bt_sco_widgets), + .dapm_routes = bt_sco_routes, + .num_dapm_routes = ARRAY_SIZE(bt_sco_routes), +}; static int bt_sco_probe(struct platform_device *pdev) { -- cgit v1.2.3 From 2f6e3ba0e0645011cbbd0289e9082d8007141498 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:09:39 +0100 Subject: ASoC: spdif: Add stub DAPM widgets for Rx Ensure that the driver continues to work with mandatory DAPM. Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_receiver.c | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index 26d34744c677..e3501f40c7b3 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -23,13 +23,26 @@ #include #include +static const struct snd_soc_dapm_widget dir_widgets[] = { + SND_SOC_DAPM_INPUT("spdif-in"), +}; + +static const struct snd_soc_dapm_route dir_routes[] = { + { "Capture", NULL, "spdif-in" }, +}; + #define STUB_RATES SNDRV_PCM_RATE_8000_192000 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) -static struct snd_soc_codec_driver soc_codec_spdif_dir; +static struct snd_soc_codec_driver soc_codec_spdif_dir = { + .dapm_widgets = dir_widgets, + .num_dapm_widgets = ARRAY_SIZE(dir_widgets), + .dapm_routes = dir_routes, + .num_dapm_routes = ARRAY_SIZE(dir_routes), +}; static struct snd_soc_dai_driver dir_stub_dai = { .name = "dir-hifi", -- cgit v1.2.3 From fc6061486534a8dfee02dd6b9dd523789abd9a3d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 Aug 2013 12:10:08 +0100 Subject: ASoC: spdif: Remove duplicate const Signed-off-by: Mark Brown --- sound/soc/codecs/spdif_transmitter.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 4e96d10d61a2..a078aa31052a 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -33,7 +33,7 @@ static const struct snd_soc_dapm_widget dit_widgets[] = { SND_SOC_DAPM_OUTPUT("spdif-out"), }; -static const const struct snd_soc_dapm_route dit_routes[] = { +static const struct snd_soc_dapm_route dit_routes[] = { { "spdif-out", NULL, "Playback" }, }; -- cgit v1.2.3