From 17be5522f6de1d4920e7d9235bfb0e0c682c6f8f Mon Sep 17 00:00:00 2001 From: Jean Delvare Date: Wed, 15 Oct 2008 19:57:12 +0200 Subject: ALSA: ASoC: Convert wm8580 to a new-style i2c driver Convert the wm8580 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8580.c | 108 +++++++++++++++++++++++----------------------- sound/soc/codecs/wm8580.h | 1 + 2 files changed, 54 insertions(+), 55 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 627ebfb4209..cbcd7c324ab 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -900,85 +900,85 @@ static struct snd_soc_device *wm8580_socdev; * low = 0x1a * high = 0x1b */ -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8580_i2c_driver; -static struct i2c_client client_template; - -static int wm8580_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8580_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8580_socdev; - struct wm8580_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; - } i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - dev_err(&i2c->dev, "failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8580_init(socdev); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "failed to initialise WM8580\n"); - goto err; - } - - return ret; - -err: - kfree(codec); - kfree(i2c); return ret; } -static int wm8580_i2c_detach(struct i2c_client *client) +static int wm8580_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8580_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8580_codec_probe); -} +static const struct i2c_device_id wm8580_i2c_id[] = { + { "wm8580", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8580_i2c_driver = { .driver = { .name = "WM8580 I2C Codec", .owner = THIS_MODULE, }, - .attach_adapter = wm8580_i2c_attach, - .detach_client = wm8580_i2c_detach, - .command = NULL, + .probe = wm8580_i2c_probe, + .remove = wm8580_i2c_remove, + .id_table = wm8580_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8580", - .driver = &wm8580_i2c_driver, -}; +static int wm8580_add_i2c_device(struct platform_device *pdev, + const struct wm8580_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8580_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8580", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8580_i2c_driver); + return -ENODEV; +} #endif static int wm8580_probe(struct platform_device *pdev) @@ -1011,11 +1011,8 @@ static int wm8580_probe(struct platform_device *pdev) #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8580_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8580_add_i2c_device(pdev, setup); } #else /* Add other interfaces here */ @@ -1034,6 +1031,7 @@ static int wm8580_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8580_i2c_driver); #endif kfree(codec->private_data); diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 589ddaba21d..09e4422f6f2 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -29,6 +29,7 @@ #define WM8580_CLKSRC_NONE 5 struct wm8580_setup_data { + int i2c_bus; unsigned short i2c_address; }; -- cgit v1.2.3-70-g09d2 From 8ae6a5523f4188dbe2b98a9385f5860df6ee47a3 Mon Sep 17 00:00:00 2001 From: Jean Delvare Date: Wed, 15 Oct 2008 19:58:12 +0200 Subject: ALSA: ASoC: Convert wm8900 to a new-style i2c driver Convert the wm8900 codec driver to the new (standard) device driver binding model. Signed-off-by: Jean Delvare Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8900.c | 113 ++++++++++++++++++++++------------------------ sound/soc/codecs/wm8900.h | 1 + 2 files changed, 55 insertions(+), 59 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 3b326c9b558..de016f41e04 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1387,89 +1387,86 @@ static struct snd_soc_device *wm8900_socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) -static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END }; - -/* Magic definition of all other variables and things */ -I2C_CLIENT_INSMOD; - -static struct i2c_driver wm8900_i2c_driver; -static struct i2c_client client_template; - /* If the i2c layer weren't so broken, we could pass this kind of data around */ -static int wm8900_codec_probe(struct i2c_adapter *adap, int addr, int kind) +static int wm8900_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { struct snd_soc_device *socdev = wm8900_socdev; - struct wm8900_setup_data *setup = socdev->codec_data; struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c; int ret; - if (addr != setup->i2c_address) - return -ENODEV; - - dev_err(&adap->dev, "Probe on %x\n", addr); - - client_template.adapter = adap; - client_template.addr = addr; - - i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); - return -ENOMEM; - } i2c_set_clientdata(i2c, codec); codec->control_data = i2c; - ret = i2c_attach_client(i2c); - if (ret < 0) { - dev_err(&adap->dev, - "failed to attach codec at addr %x\n", addr); - goto err; - } - ret = wm8900_init(socdev); - if (ret < 0) { - dev_err(&adap->dev, "failed to initialise WM8900\n"); - goto err; - } - return ret; - -err: - kfree(codec); - kfree(i2c); + if (ret < 0) + dev_err(&i2c->dev, "failed to initialise WM8900\n"); return ret; } -static int wm8900_i2c_detach(struct i2c_client *client) +static int wm8900_i2c_remove(struct i2c_client *client) { struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_detach_client(client); kfree(codec->reg_cache); - kfree(client); return 0; } -static int wm8900_i2c_attach(struct i2c_adapter *adap) -{ - return i2c_probe(adap, &addr_data, wm8900_codec_probe); -} +static const struct i2c_device_id wm8900_i2c_id[] = { + { "wm8900", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); -/* corgi i2c codec control layer */ static struct i2c_driver wm8900_i2c_driver = { .driver = { .name = "WM8900 I2C codec", .owner = THIS_MODULE, }, - .attach_adapter = wm8900_i2c_attach, - .detach_client = wm8900_i2c_detach, - .command = NULL, + .probe = wm8900_i2c_probe, + .remove = wm8900_i2c_remove, + .id_table = wm8900_i2c_id, }; -static struct i2c_client client_template = { - .name = "WM8900", - .driver = &wm8900_i2c_driver, -}; +static int wm8900_add_i2c_device(struct platform_device *pdev, + const struct wm8900_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8900_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8900", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8900_i2c_driver); + return -ENODEV; +} #endif static int wm8900_probe(struct platform_device *pdev) @@ -1497,11 +1494,8 @@ static int wm8900_probe(struct platform_device *pdev) wm8900_socdev = socdev; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { - normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8900_i2c_driver); - if (ret != 0) - printk(KERN_ERR "can't add i2c driver"); + ret = wm8900_add_i2c_device(pdev, setup); } #else #error Non-I2C interfaces not yet supported @@ -1521,6 +1515,7 @@ static int wm8900_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); i2c_del_driver(&wm8900_i2c_driver); #endif kfree(codec); diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h index ba450d99e90..2249a446ad3 100644 --- a/sound/soc/codecs/wm8900.h +++ b/sound/soc/codecs/wm8900.h @@ -55,6 +55,7 @@ #define WM8900_ struct wm8900_setup_data { + int i2c_bus; unsigned short i2c_address; }; -- cgit v1.2.3-70-g09d2 From e775f6c0fb6ac25ab8845d4ad1e17b4b015487f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 28 Oct 2008 15:04:35 +0000 Subject: ASoC: Do a warm reset after cold when resetting the WM9713 The WM9713 comes out of cold reset in low power mode so always requires a warm reset to bring up the AC97 link after a cold reset. Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba402b3c99..3214aa503ea 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1097,6 +1097,8 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) } soc_ac97_ops.reset(codec->ac97); + if (soc_ac97_ops.warm_reset) + soc_ac97_ops.warm_reset(codec->ac97); if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; -- cgit v1.2.3-70-g09d2 From ca53fb24dd21bff32c4b41b2be1035a1adfc0135 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 22 Oct 2008 22:41:11 +0100 Subject: ASoC: Use finer grained dependencies in SND_SOC_ALL_CODECS Move the bus dependencies in SND_SOC_ALL_CODECS into the individual codec options rather than have them centrally. This allows the inclusion of AC97 codecs when testing on platforms with AC97 support and will also handle codecs on multi-function devices more gracefully. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 44 ++++++++++++++++++++++++-------------------- 1 file changed, 24 insertions(+), 20 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 38a0e3b620a..3c76cae68b4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,31 +1,35 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" - depends on I2C - select SPI - select SPI_MASTER - select SND_SOC_AD73311 - select SND_SOC_AK4535 - select SND_SOC_CS4270 - select SND_SOC_SSM2602 - select SND_SOC_TLV320AIC23 - select SND_SOC_TLV320AIC26 - select SND_SOC_TLV320AIC3X - select SND_SOC_UDA1380 - select SND_SOC_WM8510 - select SND_SOC_WM8580 - select SND_SOC_WM8731 - select SND_SOC_WM8750 - select SND_SOC_WM8753 - select SND_SOC_WM8900 - select SND_SOC_WM8903 - select SND_SOC_WM8971 - select SND_SOC_WM8990 + select SND_SOC_AC97 if SND_SOC_AC97_BUS + select SND_SOC_AD1980 if SND_SOC_AC97_BUS + select SND_SOC_AD73311 if I2C + select SND_SOC_AK4535 if I2C + select SND_SOC_CS4270 if I2C + select SND_SOC_SSM2602 if I2C + select SND_SOC_TLV320AIC23 if I2C + select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_UDA1380 if I2C + select SND_SOC_WM8510 if (I2C || SPI_MASTER) + select SND_SOC_WM8580 if I2C + select SND_SOC_WM8731 if (I2C || SPI_MASTER) + select SND_SOC_WM8750 if (I2C || SPI_MASTER) + select SND_SOC_WM8753 if (I2C || SPI_MASTER) + select SND_SOC_WM8900 if I2C + select SND_SOC_WM8903 if I2C + select SND_SOC_WM8971 if I2C + select SND_SOC_WM8990 if I2C + select SND_SOC_WM9712 if SND_SOC_AC97_BUS + select SND_SOC_WM9713 if SND_SOC_AC97_BUS help Normally ASoC codec drivers are only built if a machine driver which uses them is also built since they are only usable with a machine driver. Selecting this option will allow these drivers to be built without an explicit machine driver for test and development purposes. + Support for the bus types used to access the codecs to be built must + be selected separately. + If unsure select "N". -- cgit v1.2.3-70-g09d2 From 0c235d1e837c142b7565814318b6ba5917d5ac32 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Thu, 7 Aug 2008 11:22:32 -0500 Subject: ASoC: Disable automatic volume control in the CS4270 sound driver Disable the automatic volume control feature of the CS4270 audio codec. This feature, which is enabled by default, causes volume change commands to be delayed. Sometimes the volume change happens after playback is started. Signed-off-by: Timur Tabi Signed-off-by: Mark Brown --- sound/soc/codecs/cs4270.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0bbd94501d7..0ff476d7057 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -450,6 +450,19 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, return ret; } + /* Disable automatic volume control. It's enabled by default, and + * it causes volume change commands to be delayed, sometimes until + * after playback has started. + */ + + reg = cs4270_read_reg_cache(codec, CS4270_TRANS); + reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO); + ret = cs4270_i2c_write(codec, CS4270_TRANS, reg); + if (ret < 0) { + printk(KERN_ERR "I2C write failed\n"); + return ret; + } + /* Thaw and power-up the codec */ ret = snd_soc_write(codec, CS4270_PWRCTL, 0); -- cgit v1.2.3-70-g09d2 From 0763722d28b7b58fa1f9b83d3378efcde855b18a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Oct 2008 17:53:19 +0100 Subject: ALSA: ASoC - Fix a typo in Kconfig The last change to Kconfig ca53fb24dd21bff32c4b41b2be1035a1adfc0135 added a wrong item SND_SOC_AC97, which must be SND_SOC_AC97_CODEC. Signed-off-by: Takashi Iwai --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3c76cae68b4..0fd3341e6e3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1,6 +1,6 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" - select SND_SOC_AC97 if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 if I2C select SND_SOC_AK4535 if I2C -- cgit v1.2.3-70-g09d2 From cc17557e7876a92e11d4b406a367d28e103e42e6 Mon Sep 17 00:00:00 2001 From: Steve Sakoman Date: Thu, 30 Oct 2008 21:35:26 -0700 Subject: ASoC: Add support for TWL4030 audio codec Signed-off-by: Steve Sakoman Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/twl4030.c | 653 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 197 ++++++++++++++ 4 files changed, 857 insertions(+) create mode 100644 sound/soc/codecs/twl4030.c create mode 100644 sound/soc/codecs/twl4030.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0fd3341e6e3..b73c36aad67 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -9,6 +9,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C + select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA1380 if I2C select SND_SOC_WM8510 if (I2C || SPI_MASTER) select SND_SOC_WM8580 if I2C @@ -79,6 +80,10 @@ config SND_SOC_TLV320AIC3X tristate depends on I2C +config SND_SOC_TWL4030 + tristate + depends on TWL4030_CORE + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 90f0a585fc7..3b9b58a0ea7 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-twl4030-objs := twl4030.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o @@ -29,6 +30,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c new file mode 100644 index 00000000000..ee2f0d37765 --- /dev/null +++ b/sound/soc/codecs/twl4030.c @@ -0,0 +1,653 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman, + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "twl4030.h" + +/* + * twl4030 register cache & default register settings + */ +static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { + 0x00, /* this register not used */ + 0x93, /* REG_CODEC_MODE (0x1) */ + 0xc3, /* REG_OPTION (0x2) */ + 0x00, /* REG_UNKNOWN (0x3) */ + 0x00, /* REG_MICBIAS_CTL (0x4) */ + 0x24, /* REG_ANAMICL (0x5) */ + 0x04, /* REG_ANAMICR (0x6) */ + 0x0a, /* REG_AVADC_CTL (0x7) */ + 0x00, /* REG_ADCMICSEL (0x8) */ + 0x00, /* REG_DIGMIXING (0x9) */ + 0x0c, /* REG_ATXL1PGA (0xA) */ + 0x0c, /* REG_ATXR1PGA (0xB) */ + 0x00, /* REG_AVTXL2PGA (0xC) */ + 0x00, /* REG_AVTXR2PGA (0xD) */ + 0x01, /* REG_AUDIO_IF (0xE) */ + 0x00, /* REG_VOICE_IF (0xF) */ + 0x00, /* REG_ARXR1PGA (0x10) */ + 0x00, /* REG_ARXL1PGA (0x11) */ + 0x6c, /* REG_ARXR2PGA (0x12) */ + 0x6c, /* REG_ARXL2PGA (0x13) */ + 0x00, /* REG_VRXPGA (0x14) */ + 0x00, /* REG_VSTPGA (0x15) */ + 0x00, /* REG_VRX2ARXPGA (0x16) */ + 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_ARX2VTXPGA (0x18) */ + 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ + 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ + 0x4b, /* REG_ARXL2_APGA_CTL (0x1B) */ + 0x4b, /* REG_ARXR2_APGA_CTL (0x1C) */ + 0x00, /* REG_ATX2ARXPGA (0x1D) */ + 0x00, /* REG_BT_IF (0x1E) */ + 0x00, /* REG_BTPGA (0x1F) */ + 0x00, /* REG_BTSTPGA (0x20) */ + 0x00, /* REG_EAR_CTL (0x21) */ + 0x24, /* REG_HS_SEL (0x22) */ + 0x0a, /* REG_HS_GAIN_SET (0x23) */ + 0x00, /* REG_HS_POPN_SET (0x24) */ + 0x00, /* REG_PREDL_CTL (0x25) */ + 0x00, /* REG_PREDR_CTL (0x26) */ + 0x00, /* REG_PRECKL_CTL (0x27) */ + 0x00, /* REG_PRECKR_CTL (0x28) */ + 0x00, /* REG_HFL_CTL (0x29) */ + 0x00, /* REG_HFR_CTL (0x2A) */ + 0x00, /* REG_ALC_CTL (0x2B) */ + 0x00, /* REG_ALC_SET1 (0x2C) */ + 0x00, /* REG_ALC_SET2 (0x2D) */ + 0x00, /* REG_BOOST_CTL (0x2E) */ + 0x01, /* REG_SOFTVOL_CTL (0x2F) */ + 0x00, /* REG_DTMF_FREQSEL (0x30) */ + 0x00, /* REG_DTMF_TONEXT1H (0x31) */ + 0x00, /* REG_DTMF_TONEXT1L (0x32) */ + 0x00, /* REG_DTMF_TONEXT2H (0x33) */ + 0x00, /* REG_DTMF_TONEXT2L (0x34) */ + 0x00, /* REG_DTMF_TONOFF (0x35) */ + 0x00, /* REG_DTMF_WANONOFF (0x36) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_H (0x37) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_M (0x38) */ + 0x00, /* REG_I2S_RX_SCRAMBLE_L (0x39) */ + 0x16, /* REG_APLL_CTL (0x3A) */ + 0x00, /* REG_DTMF_CTL (0x3B) */ + 0x00, /* REG_DTMF_PGA_CTL2 (0x3C) */ + 0x00, /* REG_DTMF_PGA_CTL1 (0x3D) */ + 0x00, /* REG_MISC_SET_1 (0x3E) */ + 0x00, /* REG_PCMBTMUX (0x3F) */ + 0x00, /* not used (0x40) */ + 0x00, /* not used (0x41) */ + 0x00, /* not used (0x42) */ + 0x00, /* REG_RX_PATH_SEL (0x43) */ + 0x00, /* REG_VDL_APGA_CTL (0x44) */ + 0x00, /* REG_VIBRA_CTL (0x45) */ + 0x00, /* REG_VIBRA_SET (0x46) */ + 0x00, /* REG_VIBRA_PWM_SET (0x47) */ + 0x00, /* REG_ANAMIC_GAIN (0x48) */ + 0x00, /* REG_MISC_SET_2 (0x49) */ +}; + +/* + * read twl4030 register cache + */ +static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + return cache[reg]; +} + +/* + * write twl4030 register cache + */ +static inline void twl4030_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, u8 value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= TWL4030_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the twl4030 register space + */ +static int twl4030_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + twl4030_write_reg_cache(codec, reg, value); + return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); +} + +static void twl4030_clear_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode & ~TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_set_codecpdz(struct snd_soc_codec *codec) +{ + u8 mode; + + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, + mode | TWL4030_CODECPDZ); + + /* REVISIT: this delay is present in TI sample drivers */ + /* but there seems to be no TRM requirement for it */ + udelay(10); +} + +static void twl4030_init_chip(struct snd_soc_codec *codec) +{ + int i; + + /* clear CODECPDZ prior to setting register defaults */ + twl4030_clear_codecpdz(codec); + + /* set all audio section registers to reasonable defaults */ + for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++) + twl4030_write(codec, i, twl4030_reg[i]); + +} + +static const struct snd_kcontrol_new twl4030_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 127, 0), + SOC_DOUBLE_R("Capture Volume", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 127, 0), +}; + +/* add non dapm controls */ +static int twl4030_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&twl4030_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("INL"), + SND_SOC_DAPM_INPUT("INR"), + + SND_SOC_DAPM_OUTPUT("OUTL"), + SND_SOC_DAPM_OUTPUT("OUTR"), + + SND_SOC_DAPM_DAC("DACL", "Left Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DACR", "Right Playback", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* outputs */ + {"OUTL", NULL, "DACL"}, + {"OUTR", NULL, "DACR"}, + + /* inputs */ + {"ADCL", NULL, "INL"}, + {"ADCR", NULL, "INR"}, +}; + +static int twl4030_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, twl4030_dapm_widgets, + ARRAY_SIZE(twl4030_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static void twl4030_power_up(struct snd_soc_codec *codec) +{ + u8 anamicl, regmisc1, byte, popn, hsgain; + int i = 0; + + /* set CODECPDZ to turn on codec */ + twl4030_set_codecpdz(codec); + + /* initiate offset cancellation */ + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + twl4030_write(codec, TWL4030_REG_ANAMICL, + anamicl | TWL4030_CNCL_OFFSET_START); + + /* wait for offset cancellation to complete */ + do { + /* this takes a little while, so don't slam i2c */ + udelay(2000); + twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + TWL4030_REG_ANAMICL); + } while ((i++ < 100) && + ((byte & TWL4030_CNCL_OFFSET_START) == + TWL4030_CNCL_OFFSET_START)); + + /* anti-pop when changing analog gain */ + regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1); + twl4030_write(codec, TWL4030_REG_MISC_SET_1, + regmisc1 | TWL4030_SMOOTH_ANAVOL_EN); + + /* toggle CODECPDZ as per TRM */ + twl4030_clear_codecpdz(codec); + twl4030_set_codecpdz(codec); + + /* program anti-pop with bias ramp delay */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= TWL4030_RAMP_DELAY; + popn |= TWL4030_RAMP_DELAY_645MS; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + popn |= TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* enable output stage and gain setting */ + hsgain = TWL4030_HSR_GAIN_0DB | TWL4030_HSL_GAIN_0DB; + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); + + /* enable anti-pop ramp */ + popn |= TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); +} + +static void twl4030_power_down(struct snd_soc_codec *codec) +{ + u8 popn, hsgain; + + /* disable anti-pop ramp */ + popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); + popn &= ~TWL4030_RAMP_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* disable output stage and gain setting */ + hsgain = TWL4030_HSR_GAIN_PWR_DOWN | TWL4030_HSL_GAIN_PWR_DOWN; + twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); + + /* disable bias out */ + popn &= ~TWL4030_VMID_EN; + twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); + + /* power down */ + twl4030_clear_codecpdz(codec); +} + +static int twl4030_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + twl4030_power_up(codec); + break; + case SND_SOC_BIAS_PREPARE: + /* TODO: develop a twl4030_prepare function */ + break; + case SND_SOC_BIAS_STANDBY: + /* TODO: develop a twl4030_standby function */ + twl4030_power_down(codec); + break; + case SND_SOC_BIAS_OFF: + twl4030_power_down(codec); + break; + } + codec->bias_level = level; + + return 0; +} + +static int twl4030_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u8 mode, old_mode, format, old_format; + + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode & ~TWL4030_APLL_RATE; + + switch (params_rate(params)) { + case 8000: + mode |= TWL4030_APLL_RATE_8000; + break; + case 11025: + mode |= TWL4030_APLL_RATE_11025; + break; + case 12000: + mode |= TWL4030_APLL_RATE_12000; + break; + case 16000: + mode |= TWL4030_APLL_RATE_16000; + break; + case 22050: + mode |= TWL4030_APLL_RATE_22050; + break; + case 24000: + mode |= TWL4030_APLL_RATE_24000; + break; + case 32000: + mode |= TWL4030_APLL_RATE_32000; + break; + case 44100: + mode |= TWL4030_APLL_RATE_44100; + break; + case 48000: + mode |= TWL4030_APLL_RATE_48000; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_set_codecpdz(codec); + } + + /* sample size */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + format &= ~TWL4030_DATA_WIDTH; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + format |= TWL4030_DATA_WIDTH_16S_16W; + break; + case SNDRV_PCM_FORMAT_S24_LE: + format |= TWL4030_DATA_WIDTH_32S_24W; + break; + default: + printk(KERN_ERR "TWL4030 hw params: unknown format %d\n", + params_format(params)); + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + return 0; +} + +static int twl4030_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 infreq; + + switch (freq) { + case 19200000: + infreq = TWL4030_APLL_INFREQ_19200KHZ; + break; + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + case 38400000: + infreq = TWL4030_APLL_INFREQ_38400KHZ; + break; + default: + printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + format &= ~(TWL4030_AIF_SLAVE_EN); + format |= TWL4030_CLK256FS_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + format &= ~(TWL4030_CLK256FS_EN); + format |= TWL4030_AIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + format &= ~TWL4030_AIF_FORMAT; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= TWL4030_AIF_FORMAT_CODEC; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + + /* clear CODECPDZ before changing format (codec requirement) */ + twl4030_clear_codecpdz(codec); + + /* change format */ + twl4030_write(codec, TWL4030_REG_AUDIO_IF, format); + + /* set CODECPDZ afterwards */ + twl4030_set_codecpdz(codec); + } + + return 0; +} + +#define TWL4030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_dai twl4030_dai = { + .name = "twl4030", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = TWL4030_RATES, + .formats = TWL4030_FORMATS,}, + .ops = { + .hw_params = twl4030_hw_params, + }, + .dai_ops = { + .set_sysclk = twl4030_set_dai_sysclk, + .set_fmt = twl4030_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(twl4030_dai); + +static int twl4030_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int twl4030_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl4030_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +/* + * initialize the driver + * register the mixer and dsp interfaces with the kernel + */ + +static int twl4030_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + printk(KERN_INFO "TWL4030 Audio Codec init \n"); + + codec->name = "twl4030"; + codec->owner = THIS_MODULE; + codec->read = twl4030_read_reg_cache; + codec->write = twl4030_write; + codec->set_bias_level = twl4030_set_bias_level; + codec->dai = &twl4030_dai; + codec->num_dai = 1; + codec->reg_cache_size = sizeof(twl4030_reg); + codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to create pcms\n"); + goto pcm_err; + } + + twl4030_init_chip(codec); + + /* power on device */ + twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + twl4030_add_controls(codec); + twl4030_add_widgets(codec); + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "twl4030: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *twl4030_socdev; + +static int twl4030_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + twl4030_socdev = socdev; + twl4030_init(socdev); + + return 0; +} + +static int twl4030_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + printk(KERN_INFO "TWL4030 Audio Codec remove\n"); + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_twl4030 = { + .probe = twl4030_probe, + .remove = twl4030_remove, + .suspend = twl4030_suspend, + .resume = twl4030_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); + +MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); +MODULE_AUTHOR("Steve Sakoman"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h new file mode 100644 index 00000000000..09865d9f520 --- /dev/null +++ b/sound/soc/codecs/twl4030.h @@ -0,0 +1,197 @@ +/* + * ALSA SoC TWL4030 codec driver + * + * Author: Steve Sakoman + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __TWL4030_AUDIO_H__ +#define __TWL4030_AUDIO_H__ + +#define TWL4030_REG_CODEC_MODE 0x1 +#define TWL4030_REG_OPTION 0x2 +#define TWL4030_REG_UNKNOWN 0x3 +#define TWL4030_REG_MICBIAS_CTL 0x4 +#define TWL4030_REG_ANAMICL 0x5 +#define TWL4030_REG_ANAMICR 0x6 +#define TWL4030_REG_AVADC_CTL 0x7 +#define TWL4030_REG_ADCMICSEL 0x8 +#define TWL4030_REG_DIGMIXING 0x9 +#define TWL4030_REG_ATXL1PGA 0xA +#define TWL4030_REG_ATXR1PGA 0xB +#define TWL4030_REG_AVTXL2PGA 0xC +#define TWL4030_REG_AVTXR2PGA 0xD +#define TWL4030_REG_AUDIO_IF 0xE +#define TWL4030_REG_VOICE_IF 0xF +#define TWL4030_REG_ARXR1PGA 0x10 +#define TWL4030_REG_ARXL1PGA 0x11 +#define TWL4030_REG_ARXR2PGA 0x12 +#define TWL4030_REG_ARXL2PGA 0x13 +#define TWL4030_REG_VRXPGA 0x14 +#define TWL4030_REG_VSTPGA 0x15 +#define TWL4030_REG_VRX2ARXPGA 0x16 +#define TWL4030_REG_AVDAC_CTL 0x17 +#define TWL4030_REG_ARX2VTXPGA 0x18 +#define TWL4030_REG_ARXL1_APGA_CTL 0x19 +#define TWL4030_REG_ARXR1_APGA_CTL 0x1A +#define TWL4030_REG_ARXL2_APGA_CTL 0x1B +#define TWL4030_REG_ARXR2_APGA_CTL 0x1C +#define TWL4030_REG_ATX2ARXPGA 0x1D +#define TWL4030_REG_BT_IF 0x1E +#define TWL4030_REG_BTPGA 0x1F +#define TWL4030_REG_BTSTPGA 0x20 +#define TWL4030_REG_EAR_CTL 0x21 +#define TWL4030_REG_HS_SEL 0x22 +#define TWL4030_REG_HS_GAIN_SET 0x23 +#define TWL4030_REG_HS_POPN_SET 0x24 +#define TWL4030_REG_PREDL_CTL 0x25 +#define TWL4030_REG_PREDR_CTL 0x26 +#define TWL4030_REG_PRECKL_CTL 0x27 +#define TWL4030_REG_PRECKR_CTL 0x28 +#define TWL4030_REG_HFL_CTL 0x29 +#define TWL4030_REG_HFR_CTL 0x2A +#define TWL4030_REG_ALC_CTL 0x2B +#define TWL4030_REG_ALC_SET1 0x2C +#define TWL4030_REG_ALC_SET2 0x2D +#define TWL4030_REG_BOOST_CTL 0x2E +#define TWL4030_REG_SOFTVOL_CTL 0x2F +#define TWL4030_REG_DTMF_FREQSEL 0x30 +#define TWL4030_REG_DTMF_TONEXT1H 0x31 +#define TWL4030_REG_DTMF_TONEXT1L 0x32 +#define TWL4030_REG_DTMF_TONEXT2H 0x33 +#define TWL4030_REG_DTMF_TONEXT2L 0x34 +#define TWL4030_REG_DTMF_TONOFF 0x35 +#define TWL4030_REG_DTMF_WANONOFF 0x36 +#define TWL4030_REG_I2S_RX_SCRAMBLE_H 0x37 +#define TWL4030_REG_I2S_RX_SCRAMBLE_M 0x38 +#define TWL4030_REG_I2S_RX_SCRAMBLE_L 0x39 +#define TWL4030_REG_APLL_CTL 0x3A +#define TWL4030_REG_DTMF_CTL 0x3B +#define TWL4030_REG_DTMF_PGA_CTL2 0x3C +#define TWL4030_REG_DTMF_PGA_CTL1 0x3D +#define TWL4030_REG_MISC_SET_1 0x3E +#define TWL4030_REG_PCMBTMUX 0x3F +#define TWL4030_REG_RX_PATH_SEL 0x43 +#define TWL4030_REG_VDL_APGA_CTL 0x44 +#define TWL4030_REG_VIBRA_CTL 0x45 +#define TWL4030_REG_VIBRA_SET 0x46 +#define TWL4030_REG_VIBRA_PWM_SET 0x47 +#define TWL4030_REG_ANAMIC_GAIN 0x48 +#define TWL4030_REG_MISC_SET_2 0x49 + +#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) + +/* Bitfield Definitions */ + +/* TWL4030_CODEC_MODE (0x01) Fields */ + +#define TWL4030_APLL_RATE 0xF0 +#define TWL4030_APLL_RATE_8000 0x00 +#define TWL4030_APLL_RATE_11025 0x10 +#define TWL4030_APLL_RATE_12000 0x20 +#define TWL4030_APLL_RATE_16000 0x40 +#define TWL4030_APLL_RATE_22050 0x50 +#define TWL4030_APLL_RATE_24000 0x60 +#define TWL4030_APLL_RATE_32000 0x80 +#define TWL4030_APLL_RATE_44100 0x90 +#define TWL4030_APLL_RATE_48000 0xA0 +#define TWL4030_SEL_16K 0x04 +#define TWL4030_CODECPDZ 0x02 +#define TWL4030_OPT_MODE 0x01 + +/* ANAMICL (0x05) Fields */ +#define TWL4030_CNCL_OFFSET_START 0x80 +#define TWL4030_OFFSET_CNCL_SEL 0x60 +#define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 +#define TWL4030_OFFSET_CNCL_SEL_ARX2 0x20 +#define TWL4030_OFFSET_CNCL_SEL_VRX 0x40 +#define TWL4030_OFFSET_CNCL_SEL_ALL 0x60 +#define TWL4030_MICAMPL_EN 0x10 +#define TWL4030_CKMIC_EN 0x08 +#define TWL4030_AUXL_EN 0x04 +#define TWL4030_HSMIC_EN 0x02 +#define TWL4030_MAINMIC_EN 0x01 + +/* ANAMICR (0x06) Fields */ +#define TWL4030_MICAMPR_EN 0x10 +#define TWL4030_AUXR_EN 0x04 +#define TWL4030_SUBMIC_EN 0x01 + +/* AUDIO_IF (0x0E) Fields */ + +#define TWL4030_AIF_SLAVE_EN 0x80 +#define TWL4030_DATA_WIDTH 0x60 +#define TWL4030_DATA_WIDTH_16S_16W 0x00 +#define TWL4030_DATA_WIDTH_32S_16W 0x40 +#define TWL4030_DATA_WIDTH_32S_24W 0x60 +#define TWL4030_AIF_FORMAT 0x18 +#define TWL4030_AIF_FORMAT_CODEC 0x00 +#define TWL4030_AIF_FORMAT_LEFT 0x08 +#define TWL4030_AIF_FORMAT_RIGHT 0x10 +#define TWL4030_AIF_FORMAT_TDM 0x18 +#define TWL4030_AIF_TRI_EN 0x04 +#define TWL4030_CLK256FS_EN 0x02 +#define TWL4030_AIF_EN 0x01 + +/* HS_GAIN_SET (0x23) Fields */ + +#define TWL4030_HSR_GAIN 0x0C +#define TWL4030_HSR_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSR_GAIN_PLUS_6DB 0x04 +#define TWL4030_HSR_GAIN_0DB 0x08 +#define TWL4030_HSR_GAIN_MINUS_6DB 0x0C +#define TWL4030_HSL_GAIN 0x03 +#define TWL4030_HSL_GAIN_PWR_DOWN 0x00 +#define TWL4030_HSL_GAIN_PLUS_6DB 0x01 +#define TWL4030_HSL_GAIN_0DB 0x02 +#define TWL4030_HSL_GAIN_MINUS_6DB 0x03 + +/* HS_POPN_SET (0x24) Fields */ + +#define TWL4030_VMID_EN 0x40 +#define TWL4030_EXTMUTE 0x20 +#define TWL4030_RAMP_DELAY 0x1C +#define TWL4030_RAMP_DELAY_20MS 0x00 +#define TWL4030_RAMP_DELAY_40MS 0x04 +#define TWL4030_RAMP_DELAY_81MS 0x08 +#define TWL4030_RAMP_DELAY_161MS 0x0C +#define TWL4030_RAMP_DELAY_323MS 0x10 +#define TWL4030_RAMP_DELAY_645MS 0x14 +#define TWL4030_RAMP_DELAY_1291MS 0x18 +#define TWL4030_RAMP_DELAY_2581MS 0x1C +#define TWL4030_RAMP_EN 0x02 + +/* APLL_CTL (0x3A) Fields */ + +#define TWL4030_APLL_EN 0x10 +#define TWL4030_APLL_INFREQ 0x0F +#define TWL4030_APLL_INFREQ_19200KHZ 0x05 +#define TWL4030_APLL_INFREQ_26000KHZ 0x06 +#define TWL4030_APLL_INFREQ_38400KHZ 0x0F + +/* REG_MISC_SET_1 (0x3E) Fields */ + +#define TWL4030_CLK64_EN 0x80 +#define TWL4030_SCRAMBLE_EN 0x40 +#define TWL4030_FMLOOP_EN 0x20 +#define TWL4030_SMOOTH_ANAVOL_EN 0x02 +#define TWL4030_DIGMIC_LR_SWAP_EN 0x01 + +extern struct snd_soc_dai twl4030_dai; +extern struct snd_soc_codec_device soc_codec_dev_twl4030; + +#endif /* End of __TWL4030_AUDIO_H__ */ -- cgit v1.2.3-70-g09d2 From 3865675c60aec3e81d72d484680b544afc6fc51d Mon Sep 17 00:00:00 2001 From: Huang Weiyi Date: Fri, 31 Oct 2008 22:50:00 +0800 Subject: ALSA: ASoC codec: remove unused #include The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION. sound/soc/codecs/ad73311.c This patch removes the said #include . Signed-off-by: Huang Weiyi Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/ad73311.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 37af8607b00..59c4c8f18cb 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -15,7 +15,6 @@ #include #include -#include #include #include #include -- cgit v1.2.3-70-g09d2 From e18c94d20224f3df584531a48d944d8cccfda46d Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Wed, 5 Nov 2008 23:51:05 +0200 Subject: ALSA: ASoC: TWL4030 codec - fix 256*Fs clock According to TRM, 256*Fs clock output should be enabled when TWL4030 is in slave mode, not master. This allows sound to work on OMAP3 Pandora, which uses 256*Fs clock. Signed-off-by: Grazvydas Ignotas Acked-by: Steve Sakoman Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ee2f0d37765..90f3b4decba 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -469,11 +469,11 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_AIF_SLAVE_EN); - format |= TWL4030_CLK256FS_EN; + format &= ~(TWL4030_CLK256FS_EN); break; case SND_SOC_DAIFMT_CBS_CFS: - format &= ~(TWL4030_CLK256FS_EN); format |= TWL4030_AIF_SLAVE_EN; + format |= TWL4030_CLK256FS_EN; break; default: return -EINVAL; -- cgit v1.2.3-70-g09d2 From 26df91c36fb976af9d08c20028b5cb1317eedcb3 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Wed, 5 Nov 2008 18:53:28 +0000 Subject: ASoC: TLV320AIC23B Support more sample rates Add support for more sample rates, different crystals and split playback/capture rates. Signed-off-by: Troy Kisky Acked-by: Arun KS Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 222 ++++++++++++++++++++++++++++++++--------- 1 file changed, 177 insertions(+), 45 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308dac9e1..a95b538b8fe 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -37,12 +37,6 @@ #define AIC23_VERSION "0.1" -struct tlv320aic23_srate_reg_info { - u32 sample_rate; - u8 control; /* SR3, SR2, SR1, SR0 and BOSR */ - u8 divider; /* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ -}; - /* * AIC23 register cache */ @@ -261,20 +255,151 @@ static const struct snd_soc_dapm_route intercon[] = { }; -/* tlv320aic23 related */ -static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { - {4000, 0x06, 1}, /* 4000 */ - {8000, 0x06, 0}, /* 8000 */ - {16000, 0x0C, 1}, /* 16000 */ - {22050, 0x11, 1}, /* 22050 */ - {24000, 0x00, 1}, /* 24000 */ - {32000, 0x0C, 0}, /* 32000 */ - {44100, 0x11, 0}, /* 44100 */ - {48000, 0x00, 0}, /* 48000 */ - {88200, 0x1F, 0}, /* 88200 */ - {96000, 0x0E, 0}, /* 96000 */ +/* AIC23 driver data */ +struct aic23 { + struct snd_soc_codec codec; + int mclk; + int requested_adc; + int requested_dac; +}; + +/* + * Common Crystals used + * 11.2896 Mhz /128 = *88.2k /192 = 58.8k + * 12.0000 Mhz /125 = *96k /136 = 88.235K + * 12.2880 Mhz /128 = *96k /192 = 64k + * 16.9344 Mhz /128 = 132.3k /192 = *88.2k + * 18.4320 Mhz /128 = 144k /192 = *96k + */ + +/* + * Normal BOSR 0-256/2 = 128, 1-384/2 = 192 + * USB BOSR 0-250/2 = 125, 1-272/2 = 136 + */ +static const int bosr_usb_divisor_table[] = { + 128, 125, 192, 136 +}; +#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7)) +#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) +static const unsigned short sr_valid_mask[] = { + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/ + LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/ + LOWER_GROUP, /* Usb, bosr - 0*/ + UPPER_GROUP, /* Usb, bosr - 1*/ +}; +/* + * Every divisor is a factor of 11*12 + */ +#define SR_MULT (11*12) +#define A(x) (x) ? (SR_MULT/x) : 0 +static const unsigned char sr_adc_mult_table[] = { + A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1), + A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1) +}; +static const unsigned char sr_dac_mult_table[] = { + A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1), + A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1) }; +static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc, + int dac, int dac_l, int dac_h, int need_dac) +{ + if ((adc >= adc_l) && (adc <= adc_h) && + (dac >= dac_l) && (dac <= dac_h)) { + int diff_adc = need_adc - adc; + int diff_dac = need_dac - dac; + return abs(diff_adc) + abs(diff_dac); + } + return UINT_MAX; +} + +static int find_rate(int mclk, u32 need_adc, u32 need_dac) +{ + int i, j; + int best_i = -1; + int best_j = -1; + int best_div = 0; + unsigned best_score = UINT_MAX; + int adc_l, adc_h, dac_l, dac_h; + + need_adc *= SR_MULT; + need_dac *= SR_MULT; + /* + * rates given are +/- 1/32 + */ + adc_l = need_adc - (need_adc >> 5); + adc_h = need_adc + (need_adc >> 5); + dac_l = need_dac - (need_dac >> 5); + dac_h = need_dac + (need_dac >> 5); + for (i = 0; i < 4; i++) { + int base = mclk / bosr_usb_divisor_table[i]; + int mask = sr_valid_mask[i]; + for (j = 0; j < 16; j++, mask >>= 1) { + int adc; + int dac; + int score; + if ((mask & 1) == 0) + continue; + adc = base * sr_adc_mult_table[j]; + dac = base * sr_dac_mult_table[j]; + score = get_score(adc, adc_l, adc_h, need_adc, + dac, dac_l, dac_h, need_dac); + if (best_score > score) { + best_score = score; + best_i = i; + best_j = j; + best_div = 0; + } + score = get_score((adc >> 1), adc_l, adc_h, need_adc, + (dac >> 1), dac_l, dac_h, need_dac); + /* prefer to have a /2 */ + if ((score != UINT_MAX) && (best_score >= score)) { + best_score = score; + best_i = i; + best_j = j; + best_div = 1; + } + } + } + return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); +} + +static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, + u32 *sample_rate_adc, u32 *sample_rate_dac) +{ + int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE); + int sr = (src >> 2) & 0x0f; + int val = (mclk / bosr_usb_divisor_table[src & 3]); + int adc = (val * sr_adc_mult_table[sr]) / SR_MULT; + int dac = (val * sr_dac_mult_table[sr]) / SR_MULT; + if (src & TLV320AIC23_CLKIN_HALF) { + adc >>= 1; + dac >>= 1; + } + *sample_rate_adc = adc; + *sample_rate_dac = dac; +} + +static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, + u32 sample_rate_adc, u32 sample_rate_dac) +{ + /* Search for the right sample rate */ + int data = find_rate(mclk, sample_rate_adc, sample_rate_dac); + if (data < 0) { + printk(KERN_ERR "%s:Invalid rate %u,%u requested\n", + __func__, sample_rate_adc, sample_rate_dac); + return -EINVAL; + } + tlv320aic23_write(codec, TLV320AIC23_SRATE, data); + if (1) { + int adc, dac; + get_current_sample_rates(codec, mclk, &adc, &dac); + printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", + adc, dac, data); + } + return 0; +} + static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, @@ -293,27 +418,30 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; - u16 iface_reg, data; - u8 count = 0; + u16 iface_reg; + int ret; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + u32 sample_rate_adc = aic23->requested_adc; + u32 sample_rate_dac = aic23->requested_dac; + u32 sample_rate = params_rate(params); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + aic23->requested_dac = sample_rate_dac = sample_rate; + if (!sample_rate_adc) + sample_rate_adc = sample_rate; + } else { + aic23->requested_adc = sample_rate_adc = sample_rate; + if (!sample_rate_dac) + sample_rate_dac = sample_rate; + } + ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc, + sample_rate_dac); + if (ret < 0) + return ret; iface_reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2); - - /* Search for the right sample rate */ - /* Verify what happens if the rate is not supported - * now it goes to 96Khz */ - while ((srate_reg_info[count].sample_rate != params_rate(params)) && - (count < ARRAY_SIZE(srate_reg_info))) { - count++; - } - - data = (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) | - (srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) | - TLV320AIC23_USB_CLK_ON; - - tlv320aic23_write(codec, TLV320AIC23_SRATE, data); - switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: break; @@ -349,12 +477,17 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); /* deactivate */ if (!codec->active) { udelay(50); tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0); } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + aic23->requested_dac = 0; + else + aic23->requested_adc = 0; } static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute) @@ -422,12 +555,9 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - - switch (freq) { - case 12000000: - return 0; - } - return -EINVAL; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + aic23->mclk = freq; + return 0; } static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, @@ -659,14 +789,15 @@ static int tlv320aic23_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct aic23 *aic23; int ret = 0; printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION); - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) + aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL); + if (aic23 == NULL) return -ENOMEM; - + codec = &aic23->codec; socdev->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -687,6 +818,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); if (codec->control_data) tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -697,7 +829,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) i2c_del_driver(&tlv320aic23_i2c_driver); #endif kfree(codec->reg_cache); - kfree(codec); + kfree(aic23); return 0; } -- cgit v1.2.3-70-g09d2 From bbba944410310181de14a5c60a7c161ff2447dd9 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 12 Nov 2008 17:05:41 +0200 Subject: ASoC: Fix supported sample rates of TWL4030 audio codec TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec mode register accordingly in twl4030_hw_params. Expose this info so that ASoC can match other rates than 44.1 kHz or 48 kHz as well. Signed-off-by: Jarkko Nikula Acked-by: Steve Sakoman Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 90f3b4decba..c1893d23703 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -504,7 +504,7 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -#define TWL4030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) struct snd_soc_dai twl4030_dai = { -- cgit v1.2.3-70-g09d2 From 71cfc9028d762419ce4dea62b4afc9c32c4b4820 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 13 Nov 2008 14:33:14 +0000 Subject: ASoC: Add WM8728 codec driver The WM8728 is a high performance stereo DAC designed for applications such as DVD, home theatre and digital TV. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8728.c | 574 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8728.h | 30 +++ 4 files changed, 610 insertions(+) create mode 100644 sound/soc/codecs/wm8728.c create mode 100644 sound/soc/codecs/wm8728.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b73c36aad67..8a84460a6f7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -13,6 +13,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA1380 if I2C select SND_SOC_WM8510 if (I2C || SPI_MASTER) select SND_SOC_WM8580 if I2C + select SND_SOC_WM8728 if (I2C || SPI_MASTER) select SND_SOC_WM8731 if (I2C || SPI_MASTER) select SND_SOC_WM8750 if (I2C || SPI_MASTER) select SND_SOC_WM8753 if (I2C || SPI_MASTER) @@ -93,6 +94,9 @@ config SND_SOC_WM8510 config SND_SOC_WM8580 tristate +config SND_SOC_WM8728 + tristate + config SND_SOC_WM8731 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3b9b58a0ea7..7ae17a6ea27 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -11,6 +11,7 @@ snd-soc-twl4030-objs := twl4030.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o +snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o @@ -34,6 +35,7 @@ obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o +obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c new file mode 100644 index 00000000000..3e39dea6124 --- /dev/null +++ b/sound/soc/codecs/wm8728.c @@ -0,0 +1,574 @@ +/* + * wm8728.c -- WM8728 ALSA SoC Audio driver + * + * Copyright 2008 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. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "wm8728.h" + +struct snd_soc_codec_device soc_codec_dev_wm8728; + +/* + * We can't read the WM8728 register space so we cache them instead. + * Note that the defaults here aren't the physical defaults, we latch + * the volume update bits, mute the output and enable infinite zero + * detect. + */ +static const u16 wm8728_reg_defaults[] = { + 0x1ff, + 0x1ff, + 0x001, + 0x100, +}; + +static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + return cache[reg]; +} + +static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults)); + cache[reg] = value; +} + +/* + * write to the WM8728 register space + */ +static int wm8728_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8728 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8728_write_reg_cache(codec, reg, value); + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static const DECLARE_TLV_DB_SCALE(wm8728_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new wm8728_snd_controls[] = { + +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL, + 0, 255, 0, wm8728_tlv), + +SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0), +}; + +static int wm8728_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8728_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +/* + * DAPM controls. + */ +static const struct snd_soc_dapm_widget wm8728_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("VOUTL"), +SND_SOC_DAPM_OUTPUT("VOUTR"), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"VOUTL", NULL, "DAC"}, + {"VOUTR", NULL, "DAC"}, +}; + +static int wm8728_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8728_dapm_widgets, + ARRAY_SIZE(wm8728_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + + return 0; +} + +static int wm8728_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + + if (mute) + wm8728_write(codec, WM8728_DACCTL, mute_reg | 1); + else + wm8728_write(codec, WM8728_DACCTL, mute_reg & ~1); + + return 0; +} + +static int wm8728_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL); + + dac &= ~0x18; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + dac |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + dac |= 0x08; + break; + default: + return -EINVAL; + } + + wm8728_write(codec, WM8728_DACCTL, dac); + + return 0; +} + +static int wm8728_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8728_read_reg_cache(codec, WM8728_IFCTL); + + /* Currently only I2S is supported by the driver, though the + * hardware is more flexible. + */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 1; + break; + default: + return -EINVAL; + } + + /* The hardware only support full slave mode */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface &= ~0x22; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x20; + iface &= ~0x02; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x02; + iface &= ~0x20; + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x22; + break; + default: + return -EINVAL; + } + + wm8728_write(codec, WM8728_IFCTL, iface); + return 0; +} + +static int wm8728_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 reg; + int i; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Power everything up... */ + reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + wm8728_write(codec, WM8728_DACCTL, reg & ~0x4); + + /* ..then sync in the register cache. */ + for (i = 0; i < ARRAY_SIZE(wm8728_reg_defaults); i++) + wm8728_write(codec, i, + wm8728_read_reg_cache(codec, i)); + } + break; + + case SND_SOC_BIAS_OFF: + reg = wm8728_read_reg_cache(codec, WM8728_DACCTL); + wm8728_write(codec, WM8728_DACCTL, reg | 0x4); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8728_RATES (SNDRV_PCM_RATE_8000_192000) + +#define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai wm8728_dai = { + .name = "WM8728", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8728_RATES, + .formats = WM8728_FORMATS, + }, + .ops = { + .hw_params = wm8728_hw_params, + }, + .dai_ops = { + .digital_mute = wm8728_mute, + .set_fmt = wm8728_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(wm8728_dai); + +static int wm8728_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8728_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8728_set_bias_level(codec, codec->suspend_bias_level); + + return 0; +} + +/* + * initialise the WM8728 driver + * register the mixer and dsp interfaces with the kernel + */ +static int wm8728_init(struct snd_soc_device *socdev) +{ + struct snd_soc_codec *codec = socdev->codec; + int ret = 0; + + codec->name = "WM8728"; + codec->owner = THIS_MODULE; + codec->read = wm8728_read_reg_cache; + codec->write = wm8728_write; + codec->set_bias_level = wm8728_set_bias_level; + codec->dai = &wm8728_dai; + codec->num_dai = 1; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->reg_cache_size = ARRAY_SIZE(wm8728_reg_defaults); + codec->reg_cache = kmemdup(wm8728_reg_defaults, + sizeof(wm8728_reg_defaults), + GFP_KERNEL); + if (codec->reg_cache == NULL) + return -ENOMEM; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to create pcms\n"); + goto pcm_err; + } + + /* power on device */ + wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + wm8728_add_controls(codec); + wm8728_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "wm8728: failed to register card\n"); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); + return ret; +} + +static struct snd_soc_device *wm8728_socdev; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + +/* + * WM8728 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ + +static int wm8728_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct snd_soc_device *socdev = wm8728_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + ret = wm8728_init(socdev); + if (ret < 0) + pr_err("failed to initialise WM8728\n"); + + return ret; +} + +static int wm8728_i2c_remove(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + kfree(codec->reg_cache); + return 0; +} + +static const struct i2c_device_id wm8728_i2c_id[] = { + { "wm8728", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8728_i2c_id); + +static struct i2c_driver wm8728_i2c_driver = { + .driver = { + .name = "WM8728 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8728_i2c_probe, + .remove = wm8728_i2c_remove, + .id_table = wm8728_i2c_id, +}; + +static int wm8728_add_i2c_device(struct platform_device *pdev, + const struct wm8728_setup_data *setup) +{ + struct i2c_board_info info; + struct i2c_adapter *adapter; + struct i2c_client *client; + int ret; + + ret = i2c_add_driver(&wm8728_i2c_driver); + if (ret != 0) { + dev_err(&pdev->dev, "can't add i2c driver\n"); + return ret; + } + + memset(&info, 0, sizeof(struct i2c_board_info)); + info.addr = setup->i2c_address; + strlcpy(info.type, "wm8728", I2C_NAME_SIZE); + + adapter = i2c_get_adapter(setup->i2c_bus); + if (!adapter) { + dev_err(&pdev->dev, "can't get i2c adapter %d\n", + setup->i2c_bus); + goto err_driver; + } + + client = i2c_new_device(adapter, &info); + i2c_put_adapter(adapter); + if (!client) { + dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", + (unsigned int)info.addr); + goto err_driver; + } + + return 0; + +err_driver: + i2c_del_driver(&wm8728_i2c_driver); + return -ENODEV; +} +#endif + +#if defined(CONFIG_SPI_MASTER) +static int __devinit wm8728_spi_probe(struct spi_device *spi) +{ + struct snd_soc_device *socdev = wm8728_socdev; + struct snd_soc_codec *codec = socdev->codec; + int ret; + + codec->control_data = spi; + + ret = wm8728_init(socdev); + if (ret < 0) + dev_err(&spi->dev, "failed to initialise WM8728\n"); + + return ret; +} + +static int __devexit wm8728_spi_remove(struct spi_device *spi) +{ + return 0; +} + +static struct spi_driver wm8728_spi_driver = { + .driver = { + .name = "wm8728", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8728_spi_probe, + .remove = __devexit_p(wm8728_spi_remove), +}; + +static int wm8728_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} +#endif /* CONFIG_SPI_MASTER */ + +static int wm8728_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct wm8728_setup_data *setup; + struct snd_soc_codec *codec; + int ret = 0; + + setup = socdev->codec_data; + codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (codec == NULL) + return -ENOMEM; + + socdev->codec = codec; + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + wm8728_socdev = socdev; + ret = -ENODEV; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + if (setup->i2c_address) { + codec->hw_write = (hw_write_t)i2c_master_send; + ret = wm8728_add_i2c_device(pdev, setup); + } +#endif +#if defined(CONFIG_SPI_MASTER) + if (setup->spi) { + codec->hw_write = (hw_write_t)wm8728_spi_write; + ret = spi_register_driver(&wm8728_spi_driver); + if (ret != 0) + printk(KERN_ERR "can't add spi driver"); + } +#endif + + if (ret != 0) + kfree(codec); + + return ret; +} + +/* power down chip */ +static int wm8728_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec->control_data) + wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_unregister_device(codec->control_data); + i2c_del_driver(&wm8728_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8728_spi_driver); +#endif + kfree(codec); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8728 = { + .probe = wm8728_probe, + .remove = wm8728_remove, + .suspend = wm8728_suspend, + .resume = wm8728_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728); + +MODULE_DESCRIPTION("ASoC WM8728 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8728.h b/sound/soc/codecs/wm8728.h new file mode 100644 index 00000000000..d269c132474 --- /dev/null +++ b/sound/soc/codecs/wm8728.h @@ -0,0 +1,30 @@ +/* + * wm8728.h -- WM8728 ASoC codec driver + * + * Copyright 2008 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 _WM8728_H +#define _WM8728_H + +#define WM8728_DACLVOL 0x00 +#define WM8728_DACRVOL 0x01 +#define WM8728_DACCTL 0x02 +#define WM8728_IFCTL 0x03 + +struct wm8728_setup_data { + int spi; + int i2c_bus; + unsigned short i2c_address; +}; + +extern struct snd_soc_dai wm8728_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8728; + +#endif -- cgit v1.2.3-70-g09d2 From 6e5d9db271ab57789b09bcc61083ab71b7eabea9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 14 Nov 2008 08:57:28 +0200 Subject: ASoC: Fix for master playback/capture volume range for TWL4030 codec FGAIN for playback is in range of 0-0x3f, while for capture GAIN it is in the range of 0-0x1f. The original value of 128 (0x7f) would modify the CGAIN also for playback. Signed-off-by: Peter Ujfalusi Acked-by: Steve Sakoman Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c1893d23703..c778eb446a5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,10 +192,10 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 127, 0), + 0, 0x3f, 0), SOC_DOUBLE_R("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 127, 0), + 0, 0x1f, 0), }; /* add non dapm controls */ -- cgit v1.2.3-70-g09d2 From 1cad1de1b216b355a60d907c103b2daf1a285345 Mon Sep 17 00:00:00 2001 From: Christian Pellegrin Date: Sat, 15 Nov 2008 08:58:16 +0100 Subject: ASoC: UDA134x codec driver Signed-off-by: Christian Pellegrin Signed-off-by: Mark Brown --- include/sound/l3.h | 18 ++ include/sound/uda134x.h | 26 ++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/l3.c | 91 ++++++ sound/soc/codecs/uda134x.c | 656 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda134x_codec.h | 36 +++ 7 files changed, 839 insertions(+) create mode 100644 include/sound/l3.h create mode 100644 include/sound/uda134x.h create mode 100644 sound/soc/codecs/l3.c create mode 100644 sound/soc/codecs/uda134x.c create mode 100644 sound/soc/codecs/uda134x_codec.h (limited to 'sound/soc/codecs') diff --git a/include/sound/l3.h b/include/sound/l3.h new file mode 100644 index 00000000000..423a08f0f1b --- /dev/null +++ b/include/sound/l3.h @@ -0,0 +1,18 @@ +#ifndef _L3_H_ +#define _L3_H_ 1 + +struct l3_pins { + void (*setdat)(int); + void (*setclk)(int); + void (*setmode)(int); + int data_hold; + int data_setup; + int clock_high; + int mode_hold; + int mode; + int mode_setup; +}; + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len); + +#endif diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h new file mode 100644 index 00000000000..475ef8bb7dc --- /dev/null +++ b/include/sound/uda134x.h @@ -0,0 +1,26 @@ +/* + * uda134x.h -- UDA134x ALSA SoC Codec driver + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * 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 _UDA134X_H +#define _UDA134X_H + +#include + +struct uda134x_platform_data { + struct l3_pins l3; + void (*power) (int); + int model; +#define UDA134X_UDA1340 1 +#define UDA134X_UDA1341 2 +#define UDA134X_UDA1344 3 +}; + +#endif /* _UDA134X_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8a84460a6f7..04f49f5c3c3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TWL4030 if TWL4030_CORE + select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WM8510 if (I2C || SPI_MASTER) select SND_SOC_WM8580 if I2C @@ -66,6 +67,9 @@ config SND_SOC_CS4270_VD33_ERRATA bool depends on SND_SOC_CS4270 +config SND_SOC_L3 + tristate + config SND_SOC_SSM2602 tristate @@ -85,6 +89,10 @@ config SND_SOC_TWL4030 tristate depends on TWL4030_CORE +config SND_SOC_UDA134X + tristate + select SND_SOC_L3 + config SND_SOC_UDA1380 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7ae17a6ea27..de6572356d1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,11 +3,13 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o +snd-soc-l3-objs := l3.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o +snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o @@ -27,11 +29,13 @@ obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o +obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o +obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o diff --git a/sound/soc/codecs/l3.c b/sound/soc/codecs/l3.c new file mode 100644 index 00000000000..5353af58862 --- /dev/null +++ b/sound/soc/codecs/l3.c @@ -0,0 +1,91 @@ +/* + * L3 code + * + * Copyright (C) 2008, Christian Pellegrin + * + * 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. + * + * + * based on: + * + * L3 bus algorithm module. + * + * Copyright (C) 2001 Russell King, All Rights Reserved. + * + * + */ + +#include +#include +#include + +#include + +/* + * Send one byte of data to the chip. Data is latched into the chip on + * the rising edge of the clock. + */ +static void sendbyte(struct l3_pins *adap, unsigned int byte) +{ + int i; + + for (i = 0; i < 8; i++) { + adap->setclk(0); + udelay(adap->data_hold); + adap->setdat(byte & 1); + udelay(adap->data_setup); + adap->setclk(1); + udelay(adap->clock_high); + byte >>= 1; + } +} + +/* + * Send a set of bytes to the chip. We need to pulse the MODE line + * between each byte, but never at the start nor at the end of the + * transfer. + */ +static void sendbytes(struct l3_pins *adap, const u8 *buf, + int len) +{ + int i; + + for (i = 0; i < len; i++) { + if (i) { + udelay(adap->mode_hold); + adap->setmode(0); + udelay(adap->mode); + } + adap->setmode(1); + udelay(adap->mode_setup); + sendbyte(adap, buf[i]); + } +} + +int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len) +{ + adap->setclk(1); + adap->setdat(1); + adap->setmode(1); + udelay(adap->mode); + + adap->setmode(0); + udelay(adap->mode_setup); + sendbyte(adap, addr); + udelay(adap->mode_hold); + + sendbytes(adap, data, len); + + adap->setclk(1); + adap->setdat(1); + adap->setmode(0); + + return len; +} +EXPORT_SYMBOL_GPL(l3_write); + +MODULE_DESCRIPTION("L3 bit-banging driver"); +MODULE_AUTHOR("Christian Pellegrin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c new file mode 100644 index 00000000000..04b30da1022 --- /dev/null +++ b/sound/soc/codecs/uda134x.c @@ -0,0 +1,656 @@ +/* + * uda134x.c -- UDA134X ALSA SoC Codec driver + * + * Modifications by Christian Pellegrin + * + * Copyright 2007 Dension Audio Systems Ltd. + * Author: Zoltan Devai + * + * Based on the WM87xx drivers by Liam Girdwood and Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "uda134x_codec.h" + + +#define POWER_OFF_ON_STANDBY 1 +/* + ALSA SOC usually puts the device in standby mode when it's not used + for sometime. If you define POWER_OFF_ON_STANDBY the driver will + turn off the ADC/DAC when this callback is invoked and turn it back + on when needed. Unfortunately this will result in a very light bump + (it can be audible only with good earphones). If this bothers you + just comment this line, you will have slightly higher power + consumption . Please note that sending the L3 command for ADC is + enough to make the bump, so it doesn't make difference if you + completely take off power from the codec. + */ + +#define UDA134X_RATES SNDRV_PCM_RATE_8000_48000 +#define UDA134X_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S20_3LE) + +struct uda134x_priv { + int sysclk; + int dai_fmt; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + +/* In-data addresses are hard-coded into the reg-cache values */ +static const char uda134x_reg[UDA134X_REGS_NUM] = { + /* Extended address registers */ + 0x04, 0x04, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, + /* Status, data regs */ + 0x00, 0x83, 0x00, 0x40, 0x80, 0x00, +}; + +/* + * The codec has no support for reading its registers except for peak level... + */ +static inline unsigned int uda134x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return -1; + return cache[reg]; +} + +/* + * Write the register cache + */ +static inline void uda134x_write_reg_cache(struct snd_soc_codec *codec, + u8 reg, unsigned int value) +{ + u8 *cache = codec->reg_cache; + + if (reg >= UDA134X_REGS_NUM) + return; + cache[reg] = value; +} + +/* + * Write to the uda134x registers + * + */ +static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + int ret; + u8 addr; + u8 data = value; + struct uda134x_platform_data *pd = codec->control_data; + + pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); + + if (reg >= UDA134X_REGS_NUM) { + printk(KERN_ERR "%s unkown register: reg: %d", + __func__, reg); + return -EINVAL; + } + + uda134x_write_reg_cache(codec, reg, value); + + switch (reg) { + case UDA134X_STATUS0: + case UDA134X_STATUS1: + addr = UDA134X_STATUS_ADDR; + break; + case UDA134X_DATA000: + case UDA134X_DATA001: + case UDA134X_DATA010: + addr = UDA134X_DATA0_ADDR; + break; + case UDA134X_DATA1: + addr = UDA134X_DATA1_ADDR; + break; + default: + /* It's an extended address register */ + addr = (reg | UDA134X_EXTADDR_PREFIX); + + ret = l3_write(&pd->l3, + UDA134X_DATA0_ADDR, &addr, 1); + if (ret != 1) + return -EIO; + + addr = UDA134X_DATA0_ADDR; + data = (value | UDA134X_EXTDATA_PREFIX); + break; + } + + ret = l3_write(&pd->l3, + addr, &data, 1); + if (ret != 1) + return -EIO; + + return 0; +} + +static inline void uda134x_reset(struct snd_soc_codec *codec) +{ + u8 reset_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + uda134x_write(codec, UDA134X_STATUS0, reset_reg | (1<<6)); + msleep(1); + uda134x_write(codec, UDA134X_STATUS0, reset_reg & ~(1<<6)); +} + +static int uda134x_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mute_reg = uda134x_read_reg_cache(codec, UDA134X_DATA010); + + pr_debug("%s mute: %d\n", __func__, mute); + + if (mute) + mute_reg |= (1<<2); + else + mute_reg &= ~(1<<2); + + uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + + return 0; +} + +static int uda134x_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + struct snd_pcm_runtime *master_runtime; + + if (uda134x->master_substream) { + master_runtime = uda134x->master_substream->runtime; + + pr_debug("%s constraining to %d bits at %d\n", __func__, + master_runtime->sample_bits, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + master_runtime->rate, + master_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + master_runtime->sample_bits, + master_runtime->sample_bits); + + uda134x->slave_substream = substream; + } else + uda134x->master_substream = substream; + + return 0; +} + +static void uda134x_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + + if (uda134x->master_substream == substream) + uda134x->master_substream = uda134x->slave_substream; + + uda134x->slave_substream = NULL; +} + +static int uda134x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct uda134x_priv *uda134x = codec->private_data; + u8 hw_params; + + if (substream == uda134x->slave_substream) { + pr_debug("%s ignoring hw_params for slave substream\n", + __func__); + return 0; + } + + hw_params = uda134x_read_reg_cache(codec, UDA134X_STATUS0); + hw_params &= STATUS0_SYSCLK_MASK; + hw_params &= STATUS0_DAIFMT_MASK; + + pr_debug("%s sysclk: %d, rate:%d\n", __func__, + uda134x->sysclk, params_rate(params)); + + /* set SYSCLK / fs ratio */ + switch (uda134x->sysclk / params_rate(params)) { + case 512: + break; + case 384: + hw_params |= (1<<4); + break; + case 256: + hw_params |= (1<<5); + break; + default: + printk(KERN_ERR "%s unsupported fs\n", __func__); + return -EINVAL; + } + + pr_debug("%s dai_fmt: %d, params_format:%d\n", __func__, + uda134x->dai_fmt, params_format(params)); + + /* set DAI format and word length */ + switch (uda134x->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + hw_params |= (1<<1); + break; + case SNDRV_PCM_FORMAT_S18_3LE: + hw_params |= (1<<2); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + hw_params |= ((1<<2) | (1<<1)); + break; + default: + printk(KERN_ERR "%s unsupported format (right)\n", + __func__); + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_LEFT_J: + hw_params |= (1<<3); + break; + default: + printk(KERN_ERR "%s unsupported format\n", __func__); + return -EINVAL; + } + + uda134x_write(codec, UDA134X_STATUS0, hw_params); + + return 0; +} + +static int uda134x_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__, + clk_id, freq, dir); + + /* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable + because the codec is slave. Of course limitations of the clock + master (the IIS controller) apply. + We'll error out on set_hw_params if it's not OK */ + if ((freq >= (256 * 8000)) && (freq <= (512 * 48000))) { + uda134x->sysclk = freq; + return 0; + } + + printk(KERN_ERR "%s unsupported sysclk\n", __func__); + return -EINVAL; +} + +static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct uda134x_priv *uda134x = codec->private_data; + + pr_debug("%s fmt: %08X\n", __func__, fmt); + + /* codec supports only full slave mode */ + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + printk(KERN_ERR "%s unsupported slave mode\n", __func__); + return -EINVAL; + } + + /* no support for clock inversion */ + if ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) { + printk(KERN_ERR "%s unsupported clock inversion\n", __func__); + return -EINVAL; + } + + /* We can't setup DAI format here as it depends on the word bit num */ + /* so let's just store the value for later */ + uda134x->dai_fmt = fmt; + + return 0; +} + +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; + + pr_debug("%s bias level %d\n", __func__, level); + + switch (level) { + case SND_SOC_BIAS_ON: + /* ADC, DAC on */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg | 0x03); + break; + case SND_SOC_BIAS_PREPARE: + /* power on */ + if (pd->power) { + pd->power(1); + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(uda134x_reg); i++) + codec->write(codec, i, *cache++); + } + break; + case SND_SOC_BIAS_STANDBY: + /* ADC, DAC power off */ + reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + uda134x_write(codec, UDA134X_STATUS1, reg & ~(0x03)); + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (pd->power) + pd->power(0); + break; + } + codec->bias_level = level; + return 0; +} + +static const char *uda134x_dsp_setting[] = {"Flat", "Minimum1", + "Minimum2", "Maximum"}; +static const char *uda134x_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *uda134x_mixmode[] = {"Differential", "Analog1", + "Analog2", "Both"}; + +static const struct soc_enum uda134x_mixer_enum[] = { +SOC_ENUM_SINGLE(UDA134X_DATA010, 0, 0x04, uda134x_dsp_setting), +SOC_ENUM_SINGLE(UDA134X_DATA010, 3, 0x04, uda134x_deemph), +SOC_ENUM_SINGLE(UDA134X_EA010, 0, 0x04, uda134x_mixmode), +}; + +static const struct snd_kcontrol_new uda1341_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), +SOC_SINGLE("Capture Volume", UDA134X_EA010, 2, 0x07, 0), +SOC_SINGLE("Analog1 Volume", UDA134X_EA000, 0, 0x1F, 1), +SOC_SINGLE("Analog2 Volume", UDA134X_EA001, 0, 0x1F, 1), + +SOC_SINGLE("Mic Sensitivity", UDA134X_EA010, 2, 7, 0), +SOC_SINGLE("Mic Volume", UDA134X_EA101, 0, 0x1F, 0), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), +SOC_ENUM("Input Mux", uda134x_mixer_enum[2]), + +SOC_SINGLE("AGC Switch", UDA134X_EA100, 4, 1, 0), +SOC_SINGLE("AGC Target Volume", UDA134X_EA110, 0, 0x03, 1), +SOC_SINGLE("AGC Timing", UDA134X_EA110, 2, 0x07, 0), + +SOC_SINGLE("DAC +6dB Switch", UDA134X_STATUS1, 6, 1, 0), +SOC_SINGLE("ADC +6dB Switch", UDA134X_STATUS1, 5, 1, 0), +SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), +SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), +SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static const struct snd_kcontrol_new uda1340_snd_controls[] = { +SOC_SINGLE("Master Playback Volume", UDA134X_DATA000, 0, 0x3F, 1), + +SOC_SINGLE("Tone Control - Bass", UDA134X_DATA001, 2, 0xF, 0), +SOC_SINGLE("Tone Control - Treble", UDA134X_DATA001, 0, 3, 0), + +SOC_ENUM("Sound Processing Filter", uda134x_mixer_enum[0]), +SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]), + +SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), +}; + +static int uda134x_add_controls(struct snd_soc_codec *codec) +{ + int err, i, n; + const struct snd_kcontrol_new *ctrls; + struct uda134x_platform_data *pd = codec->control_data; + + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1344: + n = ARRAY_SIZE(uda1340_snd_controls); + ctrls = uda1340_snd_controls; + break; + case UDA134X_UDA1341: + n = ARRAY_SIZE(uda1341_snd_controls); + ctrls = uda1341_snd_controls; + break; + default: + printk(KERN_ERR "%s unkown codec type: %d", + __func__, pd->model); + return -EINVAL; + } + + for (i = 0; i < n; i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&ctrls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +struct snd_soc_dai uda134x_dai = { + .name = "UDA134X", + /* playback capabilities */ + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* capture capabilities */ + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = UDA134X_RATES, + .formats = UDA134X_FORMATS, + }, + /* pcm operations */ + .ops = { + .startup = uda134x_startup, + .shutdown = uda134x_shutdown, + .hw_params = uda134x_hw_params, + }, + /* DAI operations */ + .dai_ops = { + .digital_mute = uda134x_mute, + .set_sysclk = uda134x_set_dai_sysclk, + .set_fmt = uda134x_set_dai_fmt, + } +}; +EXPORT_SYMBOL(uda134x_dai); + + +static int uda134x_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct uda134x_priv *uda134x; + void *codec_setup_data = socdev->codec_data; + int ret = -ENOMEM; + struct uda134x_platform_data *pd; + + printk(KERN_INFO "UDA134X SoC Audio Codec\n"); + + if (!codec_setup_data) { + printk(KERN_ERR "UDA134X SoC codec: " + "missing L3 bitbang function\n"); + return -ENODEV; + } + + pd = codec_setup_data; + switch (pd->model) { + case UDA134X_UDA1340: + case UDA134X_UDA1341: + case UDA134X_UDA1344: + break; + default: + printk(KERN_ERR "UDA134X SoC codec: " + "unsupported model %d\n", + pd->model); + return -EINVAL; + } + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return ret; + + codec = socdev->codec; + + uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL); + if (uda134x == NULL) + goto priv_err; + codec->private_data = uda134x; + + codec->reg_cache = kmemdup(uda134x_reg, sizeof(uda134x_reg), + GFP_KERNEL); + if (codec->reg_cache == NULL) + goto reg_err; + + mutex_init(&codec->mutex); + + codec->reg_cache_size = sizeof(uda134x_reg); + codec->reg_cache_step = 1; + + codec->name = "UDA134X"; + codec->owner = THIS_MODULE; + codec->dai = &uda134x_dai; + codec->num_dai = 1; + codec->read = uda134x_read_reg_cache; + codec->write = uda134x_write; +#ifdef POWER_OFF_ON_STANDBY + codec->set_bias_level = uda134x_set_bias_level; +#endif + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->control_data = codec_setup_data; + + if (pd->power) + pd->power(1); + + uda134x_reset(codec); + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register pcms\n"); + goto pcm_err; + } + + ret = uda134x_add_controls(codec); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register controls\n"); + goto pcm_err; + } + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "UDA134X: failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + kfree(codec->reg_cache); +reg_err: + kfree(codec->private_data); +priv_err: + kfree(codec); + return ret; +} + +/* power down chip */ +static int uda134x_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec); + + return 0; +} + +#if defined(CONFIG_PM) +static int uda134x_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int uda134x_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE); + uda134x_set_bias_level(codec, SND_SOC_BIAS_ON); + return 0; +} +#else +#define uda134x_soc_suspend NULL +#define uda134x_soc_resume NULL +#endif /* CONFIG_PM */ + +struct snd_soc_codec_device soc_codec_dev_uda134x = { + .probe = uda134x_soc_probe, + .remove = uda134x_soc_remove, + .suspend = uda134x_soc_suspend, + .resume = uda134x_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x); + +MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); +MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x_codec.h b/sound/soc/codecs/uda134x_codec.h new file mode 100644 index 00000000000..94f440490b3 --- /dev/null +++ b/sound/soc/codecs/uda134x_codec.h @@ -0,0 +1,36 @@ +#ifndef _UDA134X_CODEC_H +#define _UDA134X_CODEC_H + +#define UDA134X_L3ADDR 5 +#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) +#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) +#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) + +#define UDA134X_EXTADDR_PREFIX 0xC0 +#define UDA134X_EXTDATA_PREFIX 0xE0 + +/* UDA134X registers */ +#define UDA134X_EA000 0 +#define UDA134X_EA001 1 +#define UDA134X_EA010 2 +#define UDA134X_EA011 3 +#define UDA134X_EA100 4 +#define UDA134X_EA101 5 +#define UDA134X_EA110 6 +#define UDA134X_EA111 7 +#define UDA134X_STATUS0 8 +#define UDA134X_STATUS1 9 +#define UDA134X_DATA000 10 +#define UDA134X_DATA001 11 +#define UDA134X_DATA010 12 +#define UDA134X_DATA1 13 + +#define UDA134X_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda134x_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda134x; + +#endif -- cgit v1.2.3-70-g09d2 From ba533e95b929c577d69237692ee588001347be8a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Nov 2008 16:59:24 +0000 Subject: ASoC: Allow writes to uncached registers in WM8990 Only fully documented registers are cached in the WM8990 but additional registers exist. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 572d22b0880..5c84f02c457 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -106,6 +106,7 @@ static const u16 wm8990_reg[] = { 0x0008, /* R60 - PLL1 */ 0x0031, /* R61 - PLL2 */ 0x0026, /* R62 - PLL3 */ + 0x0000, /* R63 - Driver internal */ }; /* @@ -126,10 +127,9 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { u16 *cache = codec->reg_cache; - BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1); - /* Reset register is uncached */ - if (reg == 0) + /* Reset register and reserved registers are uncached */ + if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1) return; cache[reg] = value; -- cgit v1.2.3-70-g09d2 From be1b87c70af69acfadb8a27a7a76dfb61de92643 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Nov 2008 17:09:34 +0000 Subject: ASoC: Enable WM8990 ADC clocking workaround Enable a hardware workaround which avoids problems with the clocking of the ADCs in certain configurations. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 6 ++++-- sound/soc/codecs/wm8990.h | 4 ++-- 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5c84f02c457..938e1542920 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1272,9 +1272,11 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ wm8990_write(codec, WM8990_ANTIPOP2, WM8990_BUFIOEN); - } else { - /* ON -> standby */ + /* Enable workaround for ADC clocking issue. */ + wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0x2); + wm8990_write(codec, WM8990_EXT_CTL1, 0xa003); + wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0); } break; diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index 0e192f3b078..7114ddc88b4 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -80,8 +80,8 @@ #define WM8990_PLL3 0x3E #define WM8990_INTDRIVBITS 0x3F -#define WM8990_REGISTER_COUNT 60 -#define WM8990_MAX_REGISTER 0x3F +#define WM8990_EXT_ACCESS_ENA 0x75 +#define WM8990_EXT_CTL1 0x7a /* * Field Definitions. -- cgit v1.2.3-70-g09d2 From 2adb9833d1782262c20b21457d645163928cf2a2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Nov 2008 17:11:14 +0000 Subject: ASoC: Manage VMID mode for WM8990 A small additional power saving can be achieved for the WM8990 by maintaining VMID using a 2*250k divider when in standby mode. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 938e1542920..2d7b0096d92 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1222,8 +1222,14 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: break; + case SND_SOC_BIAS_PREPARE: + /* VMID=2*50k */ + val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + ~WM8990_VMID_MODE_MASK; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2); break; + case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { /* Enable all output discharge bits */ @@ -1278,6 +1284,11 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, wm8990_write(codec, WM8990_EXT_CTL1, 0xa003); wm8990_write(codec, WM8990_EXT_ACCESS_ENA, 0); } + + /* VMID=2*250k */ + val = wm8990_read_reg_cache(codec, WM8990_POWER_MANAGEMENT_1) & + ~WM8990_VMID_MODE_MASK; + wm8990_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4); break; case SND_SOC_BIAS_OFF: -- cgit v1.2.3-70-g09d2 From 8d702f2376d25ab277c38b57015f4aa990bc7f16 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 17 Nov 2008 21:42:01 +0000 Subject: ASoC: Build tlv320aic23 cleanly Also merge down a couple of last minute style changes that got lost in the shuffle. Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a95b538b8fe..c903e4f48dc 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -331,10 +331,11 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) adc_h = need_adc + (need_adc >> 5); dac_l = need_dac - (need_dac >> 5); dac_h = need_dac + (need_dac >> 5); - for (i = 0; i < 4; i++) { + for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) { int base = mclk / bosr_usb_divisor_table[i]; int mask = sr_valid_mask[i]; - for (j = 0; j < 16; j++, mask >>= 1) { + for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table); + j++, mask >>= 1) { int adc; int dac; int score; @@ -364,6 +365,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac) return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT); } +#ifdef DEBUG static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, u32 *sample_rate_adc, u32 *sample_rate_dac) { @@ -379,6 +381,7 @@ static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk, *sample_rate_adc = adc; *sample_rate_dac = dac; } +#endif static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, u32 sample_rate_adc, u32 sample_rate_dac) @@ -391,12 +394,14 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk, return -EINVAL; } tlv320aic23_write(codec, TLV320AIC23_SRATE, data); - if (1) { - int adc, dac; +#ifdef DEBUG + { + u32 adc, dac; get_current_sample_rates(codec, mclk, &adc, &dac); printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n", adc, dac, data); } +#endif return 0; } -- cgit v1.2.3-70-g09d2 From 9905ed35fdec0ebb3be8a724021ff3b104571667 Mon Sep 17 00:00:00 2001 From: Cliff Cai Date: Tue, 18 Nov 2008 16:18:16 +0800 Subject: ASoC: AD1980 codec: add multi-channel function support We added multi-channel function to this codec driver and Blackfin ASoC driver as well. It was tested on Blackfin hardware. Signed-off-by: Cliff Cai Signed-off-by: Bryan Wu Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 1397b8e06c0..410fed953c5 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -85,6 +85,9 @@ SOC_DOUBLE("Line HP Swap Switch", AC97_AD_MISC, 10, 5, 1, 0), SOC_DOUBLE("Surround Playback Volume", AC97_SURROUND_MASTER, 8, 0, 31, 1), SOC_DOUBLE("Surround Playback Switch", AC97_SURROUND_MASTER, 15, 7, 1, 1), +SOC_DOUBLE("Center/LFE Playback Volume", AC97_CENTER_LFE_MASTER, 8, 0, 31, 1), +SOC_DOUBLE("Center/LFE Playback Switch", AC97_CENTER_LFE_MASTER, 15, 7, 1, 1), + SOC_ENUM("Capture Source", ad1980_cap_src), SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0), @@ -145,7 +148,7 @@ struct snd_soc_dai ad1980_dai = { .playback = { .stream_name = "Playback", .channels_min = 2, - .channels_max = 2, + .channels_max = 6, .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { @@ -192,6 +195,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0; u16 vendor_id2; + u16 ext_status; printk(KERN_INFO "AD1980 SoC Audio Codec\n"); @@ -253,9 +257,16 @@ static int ad1980_soc_probe(struct platform_device *pdev) "supported\n"); } - ac97_write(codec, AC97_MASTER, 0x0000); /* unmute line out volume */ - ac97_write(codec, AC97_PCM, 0x0000); /* unmute PCM out volume */ - ac97_write(codec, AC97_REC_GAIN, 0x0000);/* unmute record volume */ + /* unmute captures and playbacks volume */ + ac97_write(codec, AC97_MASTER, 0x0000); + ac97_write(codec, AC97_PCM, 0x0000); + ac97_write(codec, AC97_REC_GAIN, 0x0000); + ac97_write(codec, AC97_CENTER_LFE_MASTER, 0x0000); + ac97_write(codec, AC97_SURROUND_MASTER, 0x0000); + + /*power on LFE/CENTER/Surround DACs*/ + ext_status = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); ad1980_add_controls(codec); ret = snd_soc_register_card(socdev); -- cgit v1.2.3-70-g09d2 From 72f2b894455775b980a5ac7ae70ab560b3d3d247 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Nov 2008 12:25:46 +0000 Subject: ASoC: Move uda134x_codec.h to uda134x.h For consistency with other ASoC codec drivers. Signed-off-by: Mark Brown --- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda134x.h | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/uda134x_codec.h | 36 ------------------------------------ sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- 4 files changed, 38 insertions(+), 38 deletions(-) create mode 100644 sound/soc/codecs/uda134x.h delete mode 100644 sound/soc/codecs/uda134x_codec.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 04b30da1022..69ef521a2ed 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -24,7 +24,7 @@ #include #include -#include "uda134x_codec.h" +#include "uda134x.h" #define POWER_OFF_ON_STANDBY 1 diff --git a/sound/soc/codecs/uda134x.h b/sound/soc/codecs/uda134x.h new file mode 100644 index 00000000000..94f440490b3 --- /dev/null +++ b/sound/soc/codecs/uda134x.h @@ -0,0 +1,36 @@ +#ifndef _UDA134X_CODEC_H +#define _UDA134X_CODEC_H + +#define UDA134X_L3ADDR 5 +#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) +#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) +#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) + +#define UDA134X_EXTADDR_PREFIX 0xC0 +#define UDA134X_EXTDATA_PREFIX 0xE0 + +/* UDA134X registers */ +#define UDA134X_EA000 0 +#define UDA134X_EA001 1 +#define UDA134X_EA010 2 +#define UDA134X_EA011 3 +#define UDA134X_EA100 4 +#define UDA134X_EA101 5 +#define UDA134X_EA110 6 +#define UDA134X_EA111 7 +#define UDA134X_STATUS0 8 +#define UDA134X_STATUS1 9 +#define UDA134X_DATA000 10 +#define UDA134X_DATA001 11 +#define UDA134X_DATA010 12 +#define UDA134X_DATA1 13 + +#define UDA134X_REGS_NUM 14 + +#define STATUS0_DAIFMT_MASK (~(7<<1)) +#define STATUS0_SYSCLK_MASK (~(3<<4)) + +extern struct snd_soc_dai uda134x_dai; +extern struct snd_soc_codec_device soc_codec_dev_uda134x; + +#endif diff --git a/sound/soc/codecs/uda134x_codec.h b/sound/soc/codecs/uda134x_codec.h deleted file mode 100644 index 94f440490b3..00000000000 --- a/sound/soc/codecs/uda134x_codec.h +++ /dev/null @@ -1,36 +0,0 @@ -#ifndef _UDA134X_CODEC_H -#define _UDA134X_CODEC_H - -#define UDA134X_L3ADDR 5 -#define UDA134X_DATA0_ADDR ((UDA134X_L3ADDR << 2) | 0) -#define UDA134X_DATA1_ADDR ((UDA134X_L3ADDR << 2) | 1) -#define UDA134X_STATUS_ADDR ((UDA134X_L3ADDR << 2) | 2) - -#define UDA134X_EXTADDR_PREFIX 0xC0 -#define UDA134X_EXTDATA_PREFIX 0xE0 - -/* UDA134X registers */ -#define UDA134X_EA000 0 -#define UDA134X_EA001 1 -#define UDA134X_EA010 2 -#define UDA134X_EA011 3 -#define UDA134X_EA100 4 -#define UDA134X_EA101 5 -#define UDA134X_EA110 6 -#define UDA134X_EA111 7 -#define UDA134X_STATUS0 8 -#define UDA134X_STATUS1 9 -#define UDA134X_DATA000 10 -#define UDA134X_DATA001 11 -#define UDA134X_DATA010 12 -#define UDA134X_DATA1 13 - -#define UDA134X_REGS_NUM 14 - -#define STATUS0_DAIFMT_MASK (~(7<<1)) -#define STATUS0_SYSCLK_MASK (~(3<<4)) - -extern struct snd_soc_dai uda134x_dai; -extern struct snd_soc_codec_device soc_codec_dev_uda134x; - -#endif diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c index 29a68132f16..487b010730b 100644 --- a/sound/soc/s3c24xx/s3c24xx_uda134x.c +++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c @@ -26,7 +26,7 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -#include "../codecs/uda134x_codec.h" +#include "../codecs/uda134x.h" /* #define ENFORCE_RATES 1 */ -- cgit v1.2.3-70-g09d2 From 1c0090c280da18f79e3e94168b5f3bfe4eb5f1c8 Mon Sep 17 00:00:00 2001 From: Hugo Villeneuve Date: Wed, 19 Nov 2008 01:37:31 -0500 Subject: ASoC: Add PCM3008 ALSA SoC driver The PCM3008 is a 16-bit stereo audio codec. It accepts left-justified format for ADC, and right-justified format for DAC. Independent power-down modes for ADC and DAC are provided, as well as a digital de-emphasis filter (4 modes). [Merged Makefile & Kconfig, changed asm/gpio.h to linux/gpio.h -- broonie] Signed-off-by: Hugo Villeneuve Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm3008.c | 201 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/pcm3008.h | 25 ++++++ 4 files changed, 232 insertions(+) create mode 100644 sound/soc/codecs/pcm3008.c create mode 100644 sound/soc/codecs/pcm3008.h (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 04f49f5c3c3..bf68052d692 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -5,6 +5,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD73311 if I2C select SND_SOC_AK4535 if I2C select SND_SOC_CS4270 if I2C + select SND_SOC_PCM3008 select SND_SOC_SSM2602 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -70,6 +71,9 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_L3 tristate +config SND_SOC_PCM3008 + tristate + config SND_SOC_SSM2602 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de6572356d1..9a20fddd09c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -4,6 +4,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4535-objs := ak4535.o snd-soc-cs4270-objs := cs4270.o snd-soc-l3-objs := l3.o +snd-soc-pcm3008-objs := pcm3008.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o +obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c new file mode 100644 index 00000000000..2b26e1d80c8 --- /dev/null +++ b/sound/soc/codecs/pcm3008.c @@ -0,0 +1,201 @@ +/* + * ALSA Soc PCM3008 codec support + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * Based on AC97 Soc codec, original copyright follow: + * Copyright 2005 Wolfson Microelectronics PLC. + * + * 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. + * + * Generic PCM3008 support. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "pcm3008.h" + +#define PCM3008_VERSION "0.2" + +#define PCM3008_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +struct snd_soc_dai pcm3008_dai = { + .name = "PCM3008 HiFi", + .type = SND_SOC_DAI_I2S, + .playback = { + .stream_name = "PCM3008 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "PCM3008 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = PCM3008_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; +EXPORT_SYMBOL_GPL(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); +} + +static int pcm3008_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + int ret = 0; + + printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (!socdev->codec) + return -ENOMEM; + + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->name = "PCM3008"; + codec->owner = THIS_MODULE; + codec->dai = &pcm3008_dai; + codec->num_dai = 1; + codec->write = NULL; + codec->read = NULL; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + /* Register PCMs. */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to create pcms\n"); + goto pcm_err; + } + + /* Register Card. */ + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "pcm3008: failed to register card\n"); + goto card_err; + } + + /* DEM1 DEM0 DE-EMPHASIS_MODE + * Low Low De-emphasis 44.1 kHz ON + * Low High De-emphasis OFF + * High Low De-emphasis 48 kHz ON + * High High De-emphasis 32 kHz ON + */ + + /* 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); + if (ret != 0) + goto gpio_err; + + /* 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); + if (ret != 0) + goto gpio_err; + + /* Configure PDAD GPIO. */ + ret = gpio_request(setup->pdad_pin, "codec_pdad"); + if (ret == 0) + ret = gpio_direction_output(setup->pdad_pin, 1); + if (ret != 0) + goto gpio_err; + + /* Configure PDDA GPIO. */ + ret = gpio_request(setup->pdda_pin, "codec_pdda"); + if (ret == 0) + ret = gpio_direction_output(setup->pdda_pin, 1); + if (ret != 0) + goto gpio_err; + + return ret; + +gpio_err: + pcm3008_gpio_free(setup); +card_err: + snd_soc_free_pcms(socdev); +pcm_err: + kfree(socdev->codec); + + return ret; +} + +static int pcm3008_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct pcm3008_setup_data *setup = socdev->codec_data; + + if (!codec) + return 0; + + pcm3008_gpio_free(setup); + snd_soc_free_pcms(socdev); + kfree(socdev->codec); + + return 0; +} + +#ifdef CONFIG_PM +static int pcm3008_soc_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_data; + + gpio_set_value(setup->pdad_pin, 0); + gpio_set_value(setup->pdda_pin, 0); + + return 0; +} + +static int pcm3008_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct pcm3008_setup_data *setup = socdev->codec_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 + +struct snd_soc_codec_device soc_codec_dev_pcm3008 = { + .probe = pcm3008_soc_probe, + .remove = pcm3008_soc_remove, + .suspend = pcm3008_soc_suspend, + .resume = pcm3008_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008); + +MODULE_DESCRIPTION("Soc PCM3008 driver"); +MODULE_AUTHOR("Hugo Villeneuve"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h new file mode 100644 index 00000000000..d04e87d3c06 --- /dev/null +++ b/sound/soc/codecs/pcm3008.h @@ -0,0 +1,25 @@ +/* + * PCM3008 ALSA SoC Layer + * + * Author: Hugo Villeneuve + * Copyright (C) 2008 Lyrtech inc + * + * 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 __LINUX_SND_SOC_PCM3008_H +#define __LINUX_SND_SOC_PCM3008_H + +struct pcm3008_setup_data { + unsigned dem0_pin; + unsigned dem1_pin; + unsigned pdad_pin; + unsigned pdda_pin; +}; + +extern struct snd_soc_codec_device soc_codec_dev_pcm3008; +extern struct snd_soc_dai pcm3008_dai; + +#endif -- cgit v1.2.3-70-g09d2 From faab5a32f4d0784d6bde57963267be0453be3546 Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Thu, 20 Nov 2008 15:39:27 +0100 Subject: ASoC: ssm2602: Fix priv substreams refs Clean up our record of the active streams in shutdown(), fixing subsequent failures of snd_pcm_hw_constraints_complete after closure of a stream. NOTE: - The ssm2602 allows pairs of non-matching PB/REC rates. - This is a fix for less evil: The logic is flawed (e.g. the slave might startup before the master's rate and sample_bits are set). Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 44ef0dacd56..0e522e718df 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -292,9 +292,15 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3; int i = get_coeff(ssm2602->sysclk, params_rate(params)); + if (substream == ssm2602->slave_substream) { + dev_dbg(&i2c->dev, "Ignoring hw_params for slave substream\n"); + return 0; + } + /*no match is found*/ if (i == ARRAY_SIZE(coeff_div)) return -EINVAL; @@ -330,13 +336,19 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct ssm2602_priv *ssm2602 = codec->private_data; + struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or * capture going then constrain this substream to match it. + * TODO: the ssm2602 allows pairs of non-matching PB/REC rates */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; + dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + master_runtime->sample_bits, + master_runtime->rate); + snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, master_runtime->rate, @@ -370,9 +382,15 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; + struct ssm2602_priv *ssm2602 = codec->private_data; /* deactivate */ if (!codec->active) ssm2602_write(codec, SSM2602_ACTIVE, 0); + + if (ssm2602->master_substream == substream) + ssm2602->master_substream = ssm2602->slave_substream; + + ssm2602->slave_substream = NULL; } static int ssm2602_mute(struct snd_soc_dai *dai, int mute) -- cgit v1.2.3-70-g09d2 From 5de27b6cc0a8a1d27158ec9047cb5981745edfc0 Mon Sep 17 00:00:00 2001 From: Karl Beldan Date: Thu, 20 Nov 2008 15:39:31 +0100 Subject: ASoC: ssm2602: Update supported stream formats Signed-off-by: Karl Beldan Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 0e522e718df..56dc1c9c7c5 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -514,6 +514,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ SNDRV_PCM_RATE_96000) +#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + struct snd_soc_dai ssm2602_dai = { .name = "SSM2602", .playback = { @@ -521,13 +524,13 @@ struct snd_soc_dai ssm2602_dai = { .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SSM2602_RATES, - .formats = SNDRV_PCM_FMTBIT_S32_LE,}, + .formats = SSM2602_FORMATS,}, .ops = { .startup = ssm2602_startup, .prepare = ssm2602_pcm_prepare, -- cgit v1.2.3-70-g09d2 From dee89c4d94433520e4e3977ae203d4cfbfe385fb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 18 Nov 2008 22:11:38 +0000 Subject: ASoC: Merge snd_soc_ops into snd_soc_dai_ops Liam Girdwood's ASoC v2 work avoids having two different ops structures for DAIs by merging the members of struct snd_soc_ops into struct snd_soc_dai_ops, allowing per DAI configuration for everything. Backport this change. This paves the way for future work allowing any combination of DAIs to be connected rather than having fixed purpose CODEC and CPU DAIs and only allowing CODEC<->CPU interconnections. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 20 +++++++++-- sound/soc/atmel/atmel_ssc_dai.c | 21 +++++------ sound/soc/au1x/psc-ac97.c | 5 +-- sound/soc/au1x/psc-i2s.c | 8 ++--- sound/soc/blackfin/bf5xx-i2s.c | 12 ++++--- sound/soc/codecs/ac97.c | 3 +- sound/soc/codecs/ak4535.c | 5 ++- sound/soc/codecs/cs4270.c | 11 +++--- sound/soc/codecs/ssm2602.c | 14 ++++---- sound/soc/codecs/tlv320aic23.c | 19 +++++----- sound/soc/codecs/tlv320aic26.c | 5 ++- sound/soc/codecs/tlv320aic3x.c | 5 ++- sound/soc/codecs/twl4030.c | 5 ++- sound/soc/codecs/uda134x.c | 12 +++---- sound/soc/codecs/uda1380.c | 15 ++++---- sound/soc/codecs/wm8510.c | 5 ++- sound/soc/codecs/wm8580.c | 12 +++---- sound/soc/codecs/wm8728.c | 5 ++- sound/soc/codecs/wm8731.c | 11 +++--- sound/soc/codecs/wm8750.c | 5 ++- sound/soc/codecs/wm8753.c | 25 ++++++------- sound/soc/codecs/wm8900.c | 5 ++- sound/soc/codecs/wm8903.c | 11 +++--- sound/soc/codecs/wm8971.c | 5 ++- sound/soc/codecs/wm8990.c | 6 ++-- sound/soc/codecs/wm9712.c | 6 ++-- sound/soc/codecs/wm9713.c | 21 +++++------ sound/soc/davinci/davinci-i2s.c | 12 ++++--- sound/soc/davinci/davinci-sffsdr.c | 3 +- sound/soc/fsl/fsl_ssi.c | 14 ++++---- sound/soc/fsl/mpc5200_psc_i2s.c | 17 +++++---- sound/soc/omap/omap-mcbsp.c | 14 ++++---- sound/soc/omap/omap2evm.c | 3 +- sound/soc/pxa/pxa-ssp.c | 20 +++++------ sound/soc/pxa/pxa2xx-ac97.c | 9 +++-- sound/soc/pxa/pxa2xx-i2s.c | 15 ++++---- sound/soc/s3c24xx/s3c2412-i2s.c | 8 ++--- sound/soc/s3c24xx/s3c2443-ac97.c | 8 +++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 9 ++--- sound/soc/sh/hac.c | 3 +- sound/soc/sh/ssi.c | 16 ++++----- sound/soc/soc-core.c | 74 +++++++++++++++++++------------------- 42 files changed, 265 insertions(+), 237 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 08b8f7025c6..f51cb55902f 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -156,6 +156,23 @@ struct snd_soc_dai_ops { * Called by soc-core to minimise any pops. */ int (*digital_mute)(struct snd_soc_dai *dai, int mute); + + /* + * ALSA PCM audio operations - all optional. + * Called by soc-core during audio PCM operations. + */ + int (*startup)(struct snd_pcm_substream *, + struct snd_soc_dai *); + void (*shutdown)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*hw_params)(struct snd_pcm_substream *, + struct snd_pcm_hw_params *, struct snd_soc_dai *); + int (*hw_free)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*prepare)(struct snd_pcm_substream *, + struct snd_soc_dai *); + int (*trigger)(struct snd_pcm_substream *, int, + struct snd_soc_dai *); }; /* @@ -180,8 +197,7 @@ struct snd_soc_dai { struct snd_soc_dai *dai); /* ops */ - struct snd_soc_ops ops; - struct snd_soc_dai_ops dai_ops; + struct snd_soc_dai_ops ops; /* DAI capabilities */ struct snd_soc_pcm_stream capture; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index d290b789491..916f73b9a18 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -202,7 +202,8 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) /* * Startup. Only that one substream allowed in each direction. */ -static int atmel_ssc_startup(struct snd_pcm_substream *substream) +static int atmel_ssc_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; @@ -231,7 +232,8 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream) * Shutdown. Clear DMA parameters and shutdown the SSC if there * are no other substreams open. */ -static void atmel_ssc_shutdown(struct snd_pcm_substream *substream) +static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; @@ -332,7 +334,8 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, * Configure the SSC. */ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); int id = rtd->dai->cpu_dai->id; @@ -600,7 +603,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, } -static int atmel_ssc_prepare(struct snd_pcm_substream *substream) +static int atmel_ssc_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct atmel_ssc_info *ssc_p = &ssc_info[rtd->dai->cpu_dai->id]; @@ -715,8 +719,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .startup = atmel_ssc_startup, .shutdown = atmel_ssc_shutdown, .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params,}, - .dai_ops = { + .hw_params = atmel_ssc_hw_params, .set_fmt = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, .private_data = &ssc_info[0], @@ -741,8 +744,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .startup = atmel_ssc_startup, .shutdown = atmel_ssc_shutdown, .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params,}, - .dai_ops = { + .hw_params = atmel_ssc_hw_params, .set_fmt = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, .private_data = &ssc_info[1], @@ -766,8 +768,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { .startup = atmel_ssc_startup, .shutdown = atmel_ssc_shutdown, .prepare = atmel_ssc_prepare, - .hw_params = atmel_ssc_hw_params,}, - .dai_ops = { + .hw_params = atmel_ssc_hw_params, .set_fmt = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv,}, .private_data = &ssc_info[2], diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 57facbad682..ad60a6042ca 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -160,7 +160,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = { EXPORT_SYMBOL_GPL(soc_ac97_ops); static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; @@ -210,7 +211,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, } static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, - int cmd) + int cmd, struct snd_soc_dai *dai) { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 9384702c7eb..05a5acbb16a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -116,7 +116,8 @@ out: } static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; @@ -240,7 +241,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype) return 0; } -static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata; int ret, stype = SUBSTREAM_TYPE(substream); @@ -389,8 +391,6 @@ struct snd_soc_dai au1xpsc_i2s_dai = { .ops = { .trigger = au1xpsc_i2s_trigger, .hw_params = au1xpsc_i2s_hw_params, - }, - .dai_ops = { .set_fmt = au1xpsc_i2s_set_fmt, }, }; diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index e020c160ee4..4e675b55b18 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -132,7 +132,8 @@ static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return ret; } -static int bf5xx_i2s_startup(struct snd_pcm_substream *substream) +static int bf5xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); @@ -142,7 +143,8 @@ static int bf5xx_i2s_startup(struct snd_pcm_substream *substream) } static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { int ret = 0; @@ -193,7 +195,8 @@ static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream) +static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { pr_debug("%s enter\n", __func__); bf5xx_i2s.counter--; @@ -307,8 +310,7 @@ struct snd_soc_dai bf5xx_i2s_dai = { .ops = { .startup = bf5xx_i2s_startup, .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = bf5xx_i2s_hw_params, .set_fmt = bf5xx_i2s_set_dai_fmt, }, }; diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index bd1ebdc6c86..8a93aff359d 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -24,7 +24,8 @@ #define AC97_VERSION "0.6" -static int ac97_prepare(struct snd_pcm_substream *substream) +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 2a89b5888e1..c742290e553 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -339,7 +339,8 @@ static int ak4535_set_dai_sysclk(struct snd_soc_dai *codec_dai, } static int ak4535_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -451,8 +452,6 @@ struct snd_soc_dai ak4535_dai = { .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { .hw_params = ak4535_hw_params, - }, - .dai_ops = { .set_fmt = ak4535_set_dai_fmt, .digital_mute = ak4535_mute, .set_sysclk = ak4535_set_dai_sysclk, diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 0ff476d7057..7507d468b20 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -360,13 +360,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, /* * Program the CS4270 with the given hardware parameters. * - * The .dai_ops functions are used to provide board-specific data, like + * The .ops functions are used to provide board-specific data, like * input frequencies, to this driver. This function takes that information, * combines it with the hardware parameters provided, and programs the * hardware accordingly. */ static int cs4270_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -710,10 +711,10 @@ static int cs4270_probe(struct platform_device *pdev) if (codec->control_data) { /* Initialize codec ops */ cs4270_dai.ops.hw_params = cs4270_hw_params; - cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk; - cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt; + cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt; #ifdef CONFIG_SND_SOC_CS4270_HWMUTE - cs4270_dai.dai_ops.digital_mute = cs4270_mute; + cs4270_dai.ops.digital_mute = cs4270_mute; #endif } else printk(KERN_INFO "cs4270: no I2C device found, " diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 56dc1c9c7c5..0c5884ea1b0 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -285,7 +285,8 @@ static inline int get_coeff(int mclk, int rate) } static int ssm2602_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { u16 srate; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -330,7 +331,8 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream, return 0; } -static int ssm2602_startup(struct snd_pcm_substream *substream) +static int ssm2602_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -366,7 +368,8 @@ static int ssm2602_startup(struct snd_pcm_substream *substream) return 0; } -static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) +static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -377,7 +380,8 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void ssm2602_shutdown(struct snd_pcm_substream *substream) +static void ssm2602_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -536,8 +540,6 @@ struct snd_soc_dai ssm2602_dai = { .prepare = ssm2602_pcm_prepare, .hw_params = ssm2602_hw_params, .shutdown = ssm2602_shutdown, - }, - .dai_ops = { .digital_mute = ssm2602_mute, .set_sysclk = ssm2602_set_dai_sysclk, .set_fmt = ssm2602_set_dai_fmt, diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c903e4f48dc..a4e13d0688c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -418,7 +418,8 @@ static int tlv320aic23_add_widgets(struct snd_soc_codec *codec) } static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -465,7 +466,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream, return 0; } -static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) +static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -477,7 +479,8 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void tlv320aic23_shutdown(struct snd_pcm_substream *substream) +static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -613,12 +616,10 @@ struct snd_soc_dai tlv320aic23_dai = { .prepare = tlv320aic23_pcm_prepare, .hw_params = tlv320aic23_hw_params, .shutdown = tlv320aic23_shutdown, - }, - .dai_ops = { - .digital_mute = tlv320aic23_mute, - .set_fmt = tlv320aic23_set_dai_fmt, - .set_sysclk = tlv320aic23_set_dai_sysclk, - } + .digital_mute = tlv320aic23_mute, + .set_fmt = tlv320aic23_set_dai_fmt, + .set_sysclk = tlv320aic23_set_dai_sysclk, + } }; EXPORT_SYMBOL_GPL(tlv320aic23_dai); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index bed8a9e63dd..6b7ddfc9257 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -125,7 +125,8 @@ static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, * Digital Audio Interface Operations */ static int aic26_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -287,8 +288,6 @@ struct snd_soc_dai aic26_dai = { }, .ops = { .hw_params = aic26_hw_params, - }, - .dai_ops = { .digital_mute = aic26_mute, .set_sysclk = aic26_set_sysclk, .set_fmt = aic26_set_fmt, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index cff276ee261..b76bcc3c411 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -694,7 +694,8 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) } static int aic3x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1009,8 +1010,6 @@ struct snd_soc_dai aic3x_dai = { .formats = AIC3X_FORMATS,}, .ops = { .hw_params = aic3x_hw_params, - }, - .dai_ops = { .digital_mute = aic3x_mute, .set_sysclk = aic3x_set_dai_sysclk, .set_fmt = aic3x_set_dai_fmt, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c778eb446a5..33489515b92 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -343,7 +343,8 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, } static int twl4030_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -523,8 +524,6 @@ struct snd_soc_dai twl4030_dai = { .formats = TWL4030_FORMATS,}, .ops = { .hw_params = twl4030_hw_params, - }, - .dai_ops = { .set_sysclk = twl4030_set_dai_sysclk, .set_fmt = twl4030_set_dai_fmt, } diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 69ef521a2ed..91f333cdc7c 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -168,7 +168,8 @@ static int uda134x_mute(struct snd_soc_dai *dai, int mute) return 0; } -static int uda134x_startup(struct snd_pcm_substream *substream) +static int uda134x_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -200,7 +201,8 @@ static int uda134x_startup(struct snd_pcm_substream *substream) return 0; } -static void uda134x_shutdown(struct snd_pcm_substream *substream) +static void uda134x_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -214,7 +216,8 @@ static void uda134x_shutdown(struct snd_pcm_substream *substream) } static int uda134x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -484,9 +487,6 @@ struct snd_soc_dai uda134x_dai = { .startup = uda134x_startup, .shutdown = uda134x_shutdown, .hw_params = uda134x_hw_params, - }, - /* DAI operations */ - .dai_ops = { .digital_mute = uda134x_mute, .set_sysclk = uda134x_set_dai_sysclk, .set_fmt = uda134x_set_dai_fmt, diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a69ee72a7af..330877c7069 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -407,7 +407,8 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai, * when the DAI is being clocked by the CPU DAI. It's up to the * machine and cpu DAI driver to do this before we are called. */ -static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) +static int uda1380_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -439,7 +440,8 @@ static int uda1380_pcm_prepare(struct snd_pcm_substream *substream) } static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -477,7 +479,8 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream) +static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -560,8 +563,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, @@ -579,8 +580,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .digital_mute = uda1380_mute, .set_fmt = uda1380_set_dai_fmt, }, @@ -598,8 +597,6 @@ struct snd_soc_dai uda1380_dai[] = { .hw_params = uda1380_pcm_hw_params, .shutdown = uda1380_pcm_shutdown, .prepare = uda1380_pcm_prepare, - }, - .dai_ops = { .set_fmt = uda1380_set_dai_fmt, }, }, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index d8ca2da8d63..173b66c0c76 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -463,7 +463,8 @@ static int wm8510_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -585,8 +586,6 @@ struct snd_soc_dai wm8510_dai = { .formats = WM8510_FORMATS,}, .ops = { .hw_params = wm8510_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8510_mute, .set_fmt = wm8510_set_dai_fmt, .set_clkdiv = wm8510_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index cbcd7c324ab..220d4b68904 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -548,13 +548,13 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai_link *dai = rtd->dai; struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; - u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->codec_dai->id); + u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id); paifb &= ~WM8580_AIF_LENGTH_MASK; /* bit size */ @@ -574,7 +574,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - wm8580_write(codec, WM8580_PAIF3 + dai->codec_dai->id, paifb); + wm8580_write(codec, WM8580_PAIF3 + dai->id, paifb); return 0; } @@ -798,8 +798,6 @@ struct snd_soc_dai wm8580_dai[] = { }, .ops = { .hw_params = wm8580_paif_hw_params, - }, - .dai_ops = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, @@ -818,8 +816,6 @@ struct snd_soc_dai wm8580_dai[] = { }, .ops = { .hw_params = wm8580_paif_hw_params, - }, - .dai_ops = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 3e39dea6124..71949bd320d 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -147,7 +147,8 @@ static int wm8728_mute(struct snd_soc_dai *dai, int mute) } static int wm8728_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -269,8 +270,6 @@ struct snd_soc_dai wm8728_dai = { }, .ops = { .hw_params = wm8728_hw_params, - }, - .dai_ops = { .digital_mute = wm8728_mute, .set_fmt = wm8728_set_dai_fmt, } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7f8a7e36b33..c0f277053bb 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -264,7 +264,8 @@ static inline int get_coeff(int mclk, int rate) } static int wm8731_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -293,7 +294,8 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) +static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -305,7 +307,8 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static void wm8731_shutdown(struct snd_pcm_substream *substream) +static void wm8731_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -461,8 +464,6 @@ struct snd_soc_dai wm8731_dai = { .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, .shutdown = wm8731_shutdown, - }, - .dai_ops = { .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 9b7296ee5b0..860a1d56830 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -614,7 +614,8 @@ static int wm8750_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -709,8 +710,6 @@ struct snd_soc_dai wm8750_dai = { .formats = WM8750_FORMATS,}, .ops = { .hw_params = wm8750_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8750_mute, .set_fmt = wm8750_set_dai_fmt, .set_sysclk = wm8750_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d426eaa2218..5e4cd3bb824 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -922,7 +922,8 @@ static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1155,7 +1156,8 @@ static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1323,16 +1325,15 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS}, .capture = { /* dummy for fast DAI switching */ .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = WM8753_RATES, - .formats = WM8753_FORMATS,}, + .formats = WM8753_FORMATS}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode1h_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1356,8 +1357,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_pcm_hw_params,}, - .dai_ops = { + .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode1v_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1385,8 +1385,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_pcm_hw_params,}, - .dai_ops = { + .hw_params = wm8753_pcm_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode2_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1410,8 +1409,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode3_4_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, @@ -1439,8 +1437,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = { .rates = WM8753_RATES, .formats = WM8753_FORMATS,}, .ops = { - .hw_params = wm8753_i2s_hw_params,}, - .dai_ops = { + .hw_params = wm8753_i2s_hw_params, .digital_mute = wm8753_mute, .set_fmt = wm8753_mode3_4_set_dai_fmt, .set_clkdiv = wm8753_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index de016f41e04..d1326be91c8 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -727,7 +727,8 @@ static int wm8900_add_widgets(struct snd_soc_codec *codec) } static int wm8900_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1117,8 +1118,6 @@ struct snd_soc_dai wm8900_dai = { }, .ops = { .hw_params = wm8900_hw_params, - }, - .dai_ops = { .set_clkdiv = wm8900_set_dai_clkdiv, .set_pll = wm8900_set_dai_pll, .set_fmt = wm8900_set_dai_fmt, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index ce40d787760..efbe8927b7d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1257,7 +1257,8 @@ static struct { { 0, 0 }, }; -static int wm8903_startup(struct snd_pcm_substream *substream) +static int wm8903_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1298,7 +1299,8 @@ static int wm8903_startup(struct snd_pcm_substream *substream) return 0; } -static void wm8903_shutdown(struct snd_pcm_substream *substream) +static void wm8903_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1317,7 +1319,8 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream) } static int wm8903_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1515,8 +1518,6 @@ struct snd_soc_dai wm8903_dai = { .startup = wm8903_startup, .shutdown = wm8903_shutdown, .hw_params = wm8903_hw_params, - }, - .dai_ops = { .digital_mute = wm8903_digital_mute, .set_fmt = wm8903_set_dai_fmt, .set_sysclk = wm8903_set_dai_sysclk diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index f41a578ddd4..26edcc9d6e8 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -541,7 +541,8 @@ static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -634,8 +635,6 @@ struct snd_soc_dai wm8971_dai = { .formats = WM8971_FORMATS,}, .ops = { .hw_params = wm8971_pcm_hw_params, - }, - .dai_ops = { .digital_mute = wm8971_mute, .set_fmt = wm8971_set_dai_fmt, .set_sysclk = wm8971_set_dai_sysclk, diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 2d7b0096d92..13926516d16 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1172,7 +1172,8 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai, * Set PCM DAI bit size and sample rate. */ static int wm8990_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -1362,8 +1363,7 @@ struct snd_soc_dai wm8990_dai = { .rates = WM8990_RATES, .formats = WM8990_FORMATS,}, .ops = { - .hw_params = wm8990_hw_params,}, - .dai_ops = { + .hw_params = wm8990_hw_params, .digital_mute = wm8990_mute, .set_fmt = wm8990_set_dai_fmt, .set_clkdiv = wm8990_set_dai_clkdiv, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index ffb471e420e..81c38e7ad34 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -487,7 +487,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } -static int ac97_prepare(struct snd_pcm_substream *substream) +static int ac97_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -507,7 +508,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, reg, runtime->rate); } -static int ac97_aux_prepare(struct snd_pcm_substream *substream) +static int ac97_aux_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 740bf3cde18..a0cc5ac969a 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -928,7 +928,8 @@ static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai, } static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -954,7 +955,8 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) +static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_device *socdev = rtd->socdev; @@ -969,7 +971,8 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) ac97_write(codec, AC97_EXTENDED_MID, status); } -static int ac97_hifi_prepare(struct snd_pcm_substream *substream) +static int ac97_hifi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -989,7 +992,8 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream) return ac97_write(codec, reg, runtime->rate); } -static int ac97_aux_prepare(struct snd_pcm_substream *substream) +static int ac97_aux_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -1042,8 +1046,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { - .prepare = ac97_hifi_prepare,}, - .dai_ops = { + .prepare = ac97_hifi_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll,}, }, @@ -1056,8 +1059,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { - .prepare = ac97_aux_prepare,}, - .dai_ops = { + .prepare = ac97_aux_prepare, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll,}, }, @@ -1077,8 +1079,7 @@ struct snd_soc_dai wm9713_dai[] = { .formats = WM9713_PCM_FORMATS,}, .ops = { .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown,}, - .dai_ops = { + .shutdown = wm9713_voiceshutdown, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, .set_fmt = wm9713_set_dai_fmt, diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 95df51e803b..7a17cd0ecf6 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -188,7 +188,8 @@ static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); } -static int davinci_i2s_startup(struct snd_pcm_substream *substream) +static int davinci_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -285,7 +286,8 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, } static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; @@ -349,7 +351,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -473,8 +476,7 @@ struct snd_soc_dai davinci_i2s_dai = { .ops = { .startup = davinci_i2s_startup, .trigger = davinci_i2s_trigger, - .hw_params = davinci_i2s_hw_params,}, - .dai_ops = { + .hw_params = davinci_i2s_hw_params, .set_fmt = davinci_i2s_set_dai_fmt, }, }; diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index fa38f9cd350..e95fde1766b 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -34,7 +34,8 @@ #include "davinci-i2s.h" static int sffsdr_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 157a7895ffa..52c290bb47b 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -266,7 +266,8 @@ static irqreturn_t fsl_ssi_isr(int irq, void *dev_id) * If this is the first stream open, then grab the IRQ and program most of * the SSI registers. */ -static int fsl_ssi_startup(struct snd_pcm_substream *substream) +static int fsl_ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -411,7 +412,8 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the * clock master. */ -static int fsl_ssi_prepare(struct snd_pcm_substream *substream) +static int fsl_ssi_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -441,7 +443,8 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) * The DMA channel is in external master start and pause mode, which * means the SSI completely controls the flow of data. */ -static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) +static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -490,7 +493,8 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) * * Shutdown the SSI if there are no other substreams open. */ -static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) +static void fsl_ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; @@ -578,8 +582,6 @@ static struct snd_soc_dai fsl_ssi_dai_template = { .prepare = fsl_ssi_prepare, .shutdown = fsl_ssi_shutdown, .trigger = fsl_ssi_trigger, - }, - .dai_ops = { .set_sysclk = fsl_ssi_set_sysclk, .set_fmt = fsl_ssi_set_fmt, }, diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 94a02eaa482..e2c172f3897 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -187,7 +187,8 @@ static irqreturn_t psc_i2s_bcom_irq(int irq, void *_psc_i2s_stream) * If this is the first stream open, then grab the IRQ and program most of * the PSC registers. */ -static int psc_i2s_startup(struct snd_pcm_substream *substream) +static int psc_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -220,7 +221,8 @@ static int psc_i2s_startup(struct snd_pcm_substream *substream) } static int psc_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -256,7 +258,8 @@ static int psc_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int psc_i2s_hw_free(struct snd_pcm_substream *substream) +static int psc_i2s_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { snd_pcm_set_runtime_buffer(substream, NULL); return 0; @@ -268,7 +271,8 @@ static int psc_i2s_hw_free(struct snd_pcm_substream *substream) * This function is called by ALSA to start, stop, pause, and resume the DMA * transfer of data. */ -static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -383,7 +387,8 @@ static int psc_i2s_trigger(struct snd_pcm_substream *substream, int cmd) * * Shutdown the PSC if there are no other substreams open. */ -static void psc_i2s_shutdown(struct snd_pcm_substream *substream) +static void psc_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct psc_i2s *psc_i2s = rtd->dai->cpu_dai->private_data; @@ -483,8 +488,6 @@ static struct snd_soc_dai psc_i2s_dai_template = { .hw_free = psc_i2s_hw_free, .shutdown = psc_i2s_shutdown, .trigger = psc_i2s_trigger, - }, - .dai_ops = { .set_sysclk = psc_i2s_set_sysclk, .set_fmt = psc_i2s_set_fmt, }, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 3d4060b00eb..6013898f49b 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -138,7 +138,8 @@ static const unsigned long omap34xx_mcbsp_port[][2] = { static const unsigned long omap34xx_mcbsp_port[][2] = {}; #endif -static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) +static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -151,7 +152,8 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream) return err; } -static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) +static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -163,7 +165,8 @@ static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream) } } -static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) +static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -192,7 +195,8 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd) } static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -470,8 +474,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, .shutdown = omap_mcbsp_dai_shutdown, \ .trigger = omap_mcbsp_dai_trigger, \ .hw_params = omap_mcbsp_dai_hw_params, \ - }, \ - .dai_ops = { \ .set_fmt = omap_mcbsp_dai_set_dai_fmt, \ .set_clkdiv = omap_mcbsp_dai_set_clkdiv, \ .set_sysclk = omap_mcbsp_dai_set_dai_sysclk, \ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 5bea31157a9..7b160f9d83f 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -38,7 +38,8 @@ #include "../codecs/twl4030.h" static int omap2evm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e2b54b88c38..d0dd6245a20 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -212,7 +212,8 @@ static struct pxa2xx_pcm_dma_params *ssp_dma_params[4][4] = { }, }; -static int pxa_ssp_startup(struct snd_pcm_substream *substream) +static int pxa_ssp_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -228,7 +229,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream) return ret; } -static void pxa_ssp_shutdown(struct snd_pcm_substream *substream) +static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -604,7 +606,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, * Can be called multiple times by oss emulation. */ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -678,7 +681,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd) +static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -806,8 +810,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { .shutdown = pxa_ssp_shutdown, .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, - }, - .dai_ops = { .set_sysclk = pxa_ssp_set_dai_sysclk, .set_clkdiv = pxa_ssp_set_dai_clkdiv, .set_pll = pxa_ssp_set_dai_pll, @@ -840,8 +842,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { .shutdown = pxa_ssp_shutdown, .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, - }, - .dai_ops = { .set_sysclk = pxa_ssp_set_dai_sysclk, .set_clkdiv = pxa_ssp_set_dai_clkdiv, .set_pll = pxa_ssp_set_dai_pll, @@ -875,8 +875,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { .shutdown = pxa_ssp_shutdown, .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, - }, - .dai_ops = { .set_sysclk = pxa_ssp_set_dai_sysclk, .set_clkdiv = pxa_ssp_set_dai_clkdiv, .set_pll = pxa_ssp_set_dai_pll, @@ -910,8 +908,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { .shutdown = pxa_ssp_shutdown, .trigger = pxa_ssp_trigger, .hw_params = pxa_ssp_hw_params, - }, - .dai_ops = { .set_sysclk = pxa_ssp_set_dai_sysclk, .set_clkdiv = pxa_ssp_set_dai_clkdiv, .set_pll = pxa_ssp_set_dai_pll, diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index a7a3a9c5c6f..86667d2f1b7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -117,7 +117,8 @@ static void pxa2xx_ac97_remove(struct platform_device *pdev, } static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -131,7 +132,8 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, } static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -145,7 +147,8 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, } static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index e758034db5c..9a3e55b4812 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -121,7 +121,8 @@ static struct pxa2xx_gpio gpio_bus[] = { }, }; -static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) +static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -187,7 +188,8 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -248,7 +250,8 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -269,7 +272,8 @@ static int pxa2xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream) +static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { SACR1 |= SACR1_DRPL; @@ -353,8 +357,7 @@ struct snd_soc_dai pxa_i2s_dai = { .startup = pxa2xx_i2s_startup, .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, - .hw_params = pxa2xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = pxa2xx_i2s_hw_params, .set_fmt = pxa2xx_i2s_set_dai_fmt, .set_sysclk = pxa2xx_i2s_set_dai_sysclk, }, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ded7d995a92..360cc2a49d9 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -343,7 +343,8 @@ static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; @@ -373,7 +374,8 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; @@ -730,8 +732,6 @@ struct snd_soc_dai s3c2412_i2s_dai = { .ops = { .trigger = s3c2412_i2s_trigger, .hw_params = s3c2412_i2s_hw_params, - }, - .dai_ops = { .set_fmt = s3c2412_i2s_set_fmt, .set_clkdiv = s3c2412_i2s_set_clkdiv, .set_sysclk = s3c2412_i2s_set_sysclk, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 19c5c3cf5d8..31377821b2c 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -271,7 +271,8 @@ static void s3c2443_ac97_remove(struct platform_device *pdev, } static int s3c2443_ac97_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -313,7 +314,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd) } static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; @@ -327,7 +329,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, } static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, - int cmd) + int cmd, struct snd_soc_dai *dai) { u32 ac_glbctrl; diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index c18977bceaf..1bac9dd3dbd 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -243,7 +243,8 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; @@ -279,7 +280,8 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } -static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { int ret = 0; @@ -475,8 +477,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,}, .ops = { .trigger = s3c24xx_i2s_trigger, - .hw_params = s3c24xx_i2s_hw_params,}, - .dai_ops = { + .hw_params = s3c24xx_i2s_hw_params, .set_fmt = s3c24xx_i2s_set_fmt, .set_clkdiv = s3c24xx_i2s_set_clkdiv, .set_sysclk = s3c24xx_i2s_set_sysclk, diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index df7bc345c32..3318071dc80 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -236,7 +236,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = { EXPORT_SYMBOL_GPL(soc_ac97_ops); static int hac_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 55c3464163a..52a233840d2 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -89,7 +89,8 @@ struct ssi_priv { * track usage of the SSI; it is simplex-only so prevent attempts of * concurrent playback + capture. FIXME: any locking required? */ -static int ssi_startup(struct snd_pcm_substream *substream) +static int ssi_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -101,7 +102,8 @@ static int ssi_startup(struct snd_pcm_substream *substream) return 0; } -static void ssi_shutdown(struct snd_pcm_substream *substream) +static void ssi_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -109,7 +111,8 @@ static void ssi_shutdown(struct snd_pcm_substream *substream) ssi->inuse = 0; } -static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -129,7 +132,8 @@ static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) } static int ssi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; @@ -354,8 +358,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { .shutdown = ssi_shutdown, .trigger = ssi_trigger, .hw_params = ssi_hw_params, - }, - .dai_ops = { .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, .set_fmt = ssi_set_fmt, @@ -383,8 +385,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { .shutdown = ssi_shutdown, .trigger = ssi_trigger, .hw_params = ssi_hw_params, - }, - .dai_ops = { .set_sysclk = ssi_set_sysclk, .set_clkdiv = ssi_set_clkdiv, .set_fmt = ssi_set_fmt, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c5cb9516fea..43f4060dbe7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -134,7 +134,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* startup the audio subsystem */ if (cpu_dai->ops.startup) { - ret = cpu_dai->ops.startup(substream); + ret = cpu_dai->ops.startup(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open interface %s\n", cpu_dai->name); @@ -151,7 +151,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } if (codec_dai->ops.startup) { - ret = codec_dai->ops.startup(substream); + ret = codec_dai->ops.startup(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't open codec %s\n", codec_dai->name); @@ -248,7 +248,7 @@ codec_dai_err: platform_err: if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream); + cpu_dai->ops.shutdown(substream, cpu_dai); out: mutex_unlock(&pcm_mutex); return ret; @@ -339,10 +339,10 @@ static int soc_codec_close(struct snd_pcm_substream *substream) snd_soc_dai_digital_mute(codec_dai, 1); if (cpu_dai->ops.shutdown) - cpu_dai->ops.shutdown(substream); + cpu_dai->ops.shutdown(substream, cpu_dai); if (codec_dai->ops.shutdown) - codec_dai->ops.shutdown(substream); + codec_dai->ops.shutdown(substream, codec_dai); if (machine->ops && machine->ops->shutdown) machine->ops->shutdown(substream); @@ -406,7 +406,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } if (codec_dai->ops.prepare) { - ret = codec_dai->ops.prepare(substream); + ret = codec_dai->ops.prepare(substream, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: codec DAI prepare error\n"); goto out; @@ -414,7 +414,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } if (cpu_dai->ops.prepare) { - ret = cpu_dai->ops.prepare(substream); + ret = cpu_dai->ops.prepare(substream, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: cpu DAI prepare error\n"); goto out; @@ -491,7 +491,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } if (codec_dai->ops.hw_params) { - ret = codec_dai->ops.hw_params(substream, params); + ret = codec_dai->ops.hw_params(substream, params, codec_dai); if (ret < 0) { printk(KERN_ERR "asoc: can't set codec %s hw params\n", codec_dai->name); @@ -500,7 +500,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } if (cpu_dai->ops.hw_params) { - ret = cpu_dai->ops.hw_params(substream, params); + ret = cpu_dai->ops.hw_params(substream, params, cpu_dai); if (ret < 0) { printk(KERN_ERR "asoc: interface %s hw params failed\n", cpu_dai->name); @@ -523,11 +523,11 @@ out: platform_err: if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream); + cpu_dai->ops.hw_free(substream, cpu_dai); interface_err: if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream); + codec_dai->ops.hw_free(substream, codec_dai); codec_err: if (machine->ops && machine->ops->hw_free) @@ -566,10 +566,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* now free hw params for the DAI's */ if (codec_dai->ops.hw_free) - codec_dai->ops.hw_free(substream); + codec_dai->ops.hw_free(substream, codec_dai); if (cpu_dai->ops.hw_free) - cpu_dai->ops.hw_free(substream); + cpu_dai->ops.hw_free(substream, cpu_dai); mutex_unlock(&pcm_mutex); return 0; @@ -586,7 +586,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) int ret; if (codec_dai->ops.trigger) { - ret = codec_dai->ops.trigger(substream, cmd); + ret = codec_dai->ops.trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } @@ -598,7 +598,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) } if (cpu_dai->ops.trigger) { - ret = cpu_dai->ops.trigger(substream, cmd); + ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; } @@ -637,10 +637,10 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) snd_power_change_state(codec->card, SNDRV_CTL_POWER_D3hot); /* mute any active DAC's */ - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; - if (dai->dai_ops.digital_mute && dai->playback.active) - dai->dai_ops.digital_mute(dai, 1); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + if (dai->ops.digital_mute && dai->playback.active) + dai->ops.digital_mute(dai, 1); } /* suspend all pcms */ @@ -733,10 +733,10 @@ static void soc_resume_deferred(struct work_struct *work) } /* unmute any active DACs */ - for (i = 0; i < machine->num_links; i++) { - struct snd_soc_dai *dai = machine->dai_link[i].codec_dai; - if (dai->dai_ops.digital_mute && dai->playback.active) - dai->dai_ops.digital_mute(dai, 0); + for (i = 0; i < card->num_links; i++) { + struct snd_soc_dai *dai = card->dai_link[i].codec_dai; + if (dai->ops.digital_mute && dai->playback.active) + dai->ops.digital_mute(dai, 0); } for (i = 0; i < card->num_links; i++) { @@ -1849,8 +1849,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_sysclk(dai, clk_id, freq, dir); + if (dai->ops.set_sysclk) + return dai->ops.set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; } @@ -1869,8 +1869,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->dai_ops.set_clkdiv) - return dai->dai_ops.set_clkdiv(dai, div_id, div); + if (dai->ops.set_clkdiv) + return dai->ops.set_clkdiv(dai, div_id, div); else return -EINVAL; } @@ -1888,8 +1888,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->dai_ops.set_pll) - return dai->dai_ops.set_pll(dai, pll_id, freq_in, freq_out); + if (dai->ops.set_pll) + return dai->ops.set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; } @@ -1905,8 +1905,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->dai_ops.set_fmt) - return dai->dai_ops.set_fmt(dai, fmt); + if (dai->ops.set_fmt) + return dai->ops.set_fmt(dai, fmt); else return -EINVAL; } @@ -1924,8 +1924,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_tdm_slot(dai, mask, slots); + if (dai->ops.set_sysclk) + return dai->ops.set_tdm_slot(dai, mask, slots); else return -EINVAL; } @@ -1940,8 +1940,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->dai_ops.set_sysclk) - return dai->dai_ops.set_tristate(dai, tristate); + if (dai->ops.set_sysclk) + return dai->ops.set_tristate(dai, tristate); else return -EINVAL; } @@ -1956,8 +1956,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->dai_ops.digital_mute) - return dai->dai_ops.digital_mute(dai, mute); + if (dai->ops.digital_mute) + return dai->ops.digital_mute(dai, mute); else return -EINVAL; } -- cgit v1.2.3-70-g09d2 From 39639faba98eafeb327a30bc10b7d921c398a59a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 21 Nov 2008 14:28:49 +0000 Subject: ASoC: Improve error reporting for AC97 reset failures Print something a bit more verbose to help make errors a little more obvious. Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 410fed953c5..1a3ea059a6d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -238,7 +238,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) ret = ad1980_reset(codec, 0); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset AD1980: AC97 link error\n"); goto reset_err; } diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 81c38e7ad34..6e3e0f34017 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -690,7 +690,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) ret = wm9712_reset(codec, 0); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n"); goto reset_err; } diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a0cc5ac969a..7be811a3b02 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1243,7 +1243,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) wm9713_reset(codec, 0); ret = wm9713_reset(codec, 1); if (ret < 0) { - printk(KERN_ERR "AC97 link error\n"); + printk(KERN_ERR "Failed to reset WM9713: AC97 link error\n"); goto reset_err; } -- cgit v1.2.3-70-g09d2 From 55b8bac50a494871594e81a05b37c15e7283f868 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Nov 2008 14:05:29 +0000 Subject: ASoC: Use supplied DAI for WM9713 rather than substream Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 16 ++++------------ 1 file changed, 4 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 7be811a3b02..a502667fca7 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -931,9 +931,7 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = dai->codec; u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; switch (params_format(params)) { @@ -958,9 +956,7 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; + struct snd_soc_codec *codec = dai->codec; u16 status; /* Gracefully shut down the voice interface. */ @@ -974,10 +970,8 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, static int ac97_hifi_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; int reg; u16 vra; @@ -995,10 +989,8 @@ static int ac97_hifi_prepare(struct snd_pcm_substream *substream, static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_device *socdev = rtd->socdev; - struct snd_soc_codec *codec = socdev->codec; u16 vra, xsle; vra = ac97_read(codec, AC97_EXTENDED_STATUS); -- cgit v1.2.3-70-g09d2 From f8d05bdbb07458e5f2c6a8281bde08056836fea6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 24 Nov 2008 08:25:45 +0200 Subject: ASoC: TWL4030: Disable soft-volume Keep Soft-volume disabled for now, since if it is enabled the FGAIN volume controls are not working in the current configuration: CODEC_MODE:OPT_MODE = 1 OPTION:ARXR2_EN = 1 OPTION:ARXL2_EN = 1 OPTION:ARXR1_EN = 0 OPTION:ARXL1_VRX_EN = 0 RX_PATH_SEL:RXL1_SEL = 0x0 (or 0x1) RX_PATH_SEL:RXR1_SEL = 0x0 (or 0x1) After the patch, FGAIN volume control works. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 33489515b92..49c7da39497 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -87,7 +87,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_ALC_SET1 (0x2C) */ 0x00, /* REG_ALC_SET2 (0x2D) */ 0x00, /* REG_BOOST_CTL (0x2E) */ - 0x01, /* REG_SOFTVOL_CTL (0x2F) */ + 0x00, /* REG_SOFTVOL_CTL (0x2F) */ 0x00, /* REG_DTMF_FREQSEL (0x30) */ 0x00, /* REG_DTMF_TONEXT1H (0x31) */ 0x00, /* REG_DTMF_TONEXT1L (0x32) */ -- cgit v1.2.3-70-g09d2 From c10b82cf085c38f2568609ffb10a6d725130f389 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 24 Nov 2008 13:49:35 +0200 Subject: ASoC: TWL4030: Change the Master volume control to TLV TWL4030 FGAIN volume control has a range: -62 to 0 dB in 1 dB steps, 0 in the FGAIN means mute. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 49c7da39497..498c42f7c6e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -33,6 +33,7 @@ #include #include #include +#include #include "twl4030.h" @@ -189,10 +190,16 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } +/* + * FGAIN volume control: + * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) + */ +static DECLARE_TLV_DB_SCALE(master_tlv, -6300, 100, 1); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { - SOC_DOUBLE_R("Master Playback Volume", + SOC_DOUBLE_R_TLV("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 0x3f, 0), + 0, 0x3f, 0, master_tlv), SOC_DOUBLE_R("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0), -- cgit v1.2.3-70-g09d2 From 0d33ea0b0f954dddd3996597c663c111249d4df9 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 24 Nov 2008 13:49:36 +0200 Subject: ASoC: TWL4030: Add CGAIN volume control Add CGAIN (Coarse gain control) to TWL4030 codec. The range of the CGAIN is: 0 dB to 12 dB in 6 dB steps. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 498c42f7c6e..91effd341c0 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -196,10 +196,20 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) */ static DECLARE_TLV_DB_SCALE(master_tlv, -6300, 100, 1); +/* + * CGAIN volume control: + * 0 dB to 12 dB in 6 dB steps + * value 2 and 3 means 12 dB + */ +static DECLARE_TLV_DB_SCALE(master_coarse_tlv, 0, 600, 0); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, 0, 0x3f, 0, master_tlv), + SOC_DOUBLE_R_TLV("Master PCM Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 0x2, 0, master_coarse_tlv), SOC_DOUBLE_R("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0), -- cgit v1.2.3-70-g09d2 From b0bd53a7399f65e2d1b37cd44c5003e55b886c1e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 24 Nov 2008 13:49:38 +0200 Subject: ASoC: TWL4030: Add helper function for output gain controls Some of the gain controls in TWL (mostly those which are associated with the outputs) are implemented in an interesting way: 0x0 : Power down (mute) 0x1 : 6dB 0x2 : 0 dB 0x3 : -6 dB Inverting not going to help with these. Custom volsw and volsw_2r get/put functions to handle these gains. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 157 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 157 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 91effd341c0..41362314789 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -190,6 +190,163 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } +/* + * Some of the gain controls in TWL (mostly those which are associated with + * the outputs) are implemented in an interesting way: + * 0x0 : Power down (mute) + * 0x1 : 6dB + * 0x2 : 0 dB + * 0x3 : -6 dB + * Inverting not going to help with these. + * Custom volsw and volsw_2r get/put functions to handle these gain bits. + */ +#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw, \ + .get = snd_soc_get_volsw_twl4030, \ + .put = snd_soc_put_volsw_twl4030, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = shift_left, .rshift = shift_right,\ + .max = xmax, .invert = xinvert} } +#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = snd_soc_get_volsw_r2_twl4030,\ + .put = snd_soc_put_volsw_r2_twl4030, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .max = xmax, .invert = xinvert} } +#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \ + SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \ + xinvert, tlv_array) + +static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + if (ucontrol->value.integer.value[0]) + ucontrol->value.integer.value[0] = + max + 1 - ucontrol->value.integer.value[0]; + + if (shift != rshift) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg) >> rshift) & mask; + if (ucontrol->value.integer.value[1]) + ucontrol->value.integer.value[1] = + max + 1 - ucontrol->value.integer.value[1]; + } + + return 0; +} + +static int snd_soc_put_volsw_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + unsigned short val, val2, val_mask; + + val = (ucontrol->value.integer.value[0] & mask); + + val_mask = mask << shift; + if (val) + val = max + 1 - val; + val = val << shift; + if (shift != rshift) { + val2 = (ucontrol->value.integer.value[1] & mask); + val_mask |= mask << rshift; + if (val2) + val2 = max + 1 - val2; + val |= val2 << rshift; + } + return snd_soc_update_bits(codec, reg, val_mask, val); +} + +static int snd_soc_get_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + int mask = (1<value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, reg2) >> shift) & mask; + + if (ucontrol->value.integer.value[0]) + ucontrol->value.integer.value[0] = + max + 1 - ucontrol->value.integer.value[0]; + if (ucontrol->value.integer.value[1]) + ucontrol->value.integer.value[1] = + max + 1 - ucontrol->value.integer.value[1]; + + return 0; +} + +static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + int mask = (1 << fls(max)) - 1; + int err; + unsigned short val, val2, val_mask; + + val_mask = mask << shift; + val = (ucontrol->value.integer.value[0] & mask); + val2 = (ucontrol->value.integer.value[1] & mask); + + if (val) + val = max + 1 - val; + if (val2) + val2 = max + 1 - val2; + + val = val << shift; + val2 = val2 << shift; + + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) + return err; + + err = snd_soc_update_bits(codec, reg2, val_mask, val2); + return err; +} + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) -- cgit v1.2.3-70-g09d2 From 3ba9e10a6d3b6abf5f5952572cff8f8d5a35ae54 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Nov 2008 18:01:05 +0000 Subject: ASoC: Remove DAI type information DAI type information is only ever used within ASoC in order to special case AC97 and for diagnostic purposes. Since modern CPUs and codecs support multi function DAIs which can be configured for several modes it is more trouble than it's worth to maintain anything other than a flag identifying AC97 DAIs so remove the type field and replace it with an ac97_control flag. Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 2 +- include/sound/soc.h | 8 -------- sound/soc/atmel/atmel_ssc_dai.c | 3 --- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 1 - sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/pcm3008.c | 1 - sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/davinci/davinci-i2s.c | 1 - sound/soc/fsl/mpc5200_psc_i2s.c | 1 - sound/soc/omap/omap-mcbsp.c | 1 - sound/soc/pxa/pxa-ssp.c | 4 ---- sound/soc/pxa/pxa2xx-ac97.c | 6 +++--- sound/soc/pxa/pxa2xx-i2s.c | 1 - sound/soc/s3c24xx/neo1973_wm8753.c | 1 - sound/soc/s3c24xx/s3c2412-i2s.c | 1 - sound/soc/s3c24xx/s3c2443-ac97.c | 4 ++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 1 - sound/soc/sh/hac.c | 4 ++-- sound/soc/sh/ssi.c | 2 -- sound/soc/soc-core.c | 31 ++++++++----------------------- 24 files changed, 21 insertions(+), 63 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index f51cb55902f..a01a24b1019 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -184,7 +184,7 @@ struct snd_soc_dai { /* DAI description */ char *name; unsigned int id; - unsigned char type; + int ac97_control; /* DAI callbacks */ int (*probe)(struct platform_device *pdev, diff --git a/include/sound/soc.h b/include/sound/soc.h index e4465f73aa4..444f9c21137 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -143,14 +143,6 @@ enum snd_soc_bias_level { SND_SOC_BIAS_OFF, }; -/* - * Digital Audio Interface (DAI) types - */ -#define SND_SOC_DAI_AC97 0x1 -#define SND_SOC_DAI_I2S 0x2 -#define SND_SOC_DAI_PCM 0x4 -#define SND_SOC_DAI_AC97_BUS 0x8 /* for custom i.e. non ac97_codec.c */ - struct snd_soc_device; struct snd_soc_pcm_stream; struct snd_soc_ops; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 916f73b9a18..0bb18dfa949 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -702,7 +702,6 @@ static int atmel_ssc_resume(struct platform_device *pdev, struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { { .name = "atmel-ssc0", .id = 0, - .type = SND_SOC_DAI_PCM, .suspend = atmel_ssc_suspend, .resume = atmel_ssc_resume, .playback = { @@ -727,7 +726,6 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { #if NUM_SSC_DEVICES == 3 { .name = "atmel-ssc1", .id = 1, - .type = SND_SOC_DAI_PCM, .suspend = atmel_ssc_suspend, .resume = atmel_ssc_resume, .playback = { @@ -751,7 +749,6 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { }, { .name = "atmel-ssc2", .id = 2, - .type = SND_SOC_DAI_PCM, .suspend = atmel_ssc_suspend, .resume = atmel_ssc_resume, .playback = { diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index ad60a6042ca..a0bcfeaf5f8 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -346,7 +346,7 @@ static int au1xpsc_ac97_resume(struct platform_device *pdev, struct snd_soc_dai au1xpsc_ac97_dai = { .name = "au1xpsc_ac97", - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = au1xpsc_ac97_probe, .remove = au1xpsc_ac97_remove, .suspend = au1xpsc_ac97_suspend, diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 05a5acbb16a..f4217e70a78 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -371,7 +371,6 @@ static int au1xpsc_i2s_resume(struct platform_device *pdev, struct snd_soc_dai au1xpsc_i2s_dai = { .name = "au1xpsc_i2s", - .type = SND_SOC_DAI_I2S, .probe = au1xpsc_i2s_probe, .remove = au1xpsc_i2s_remove, .suspend = au1xpsc_i2s_suspend, diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index 5dcd3f665ab..709bdf08e39 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -409,7 +409,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev, struct snd_soc_dai bfin_ac97_dai = { .name = "bf5xx-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = bf5xx_ac97_probe, .remove = bf5xx_ac97_remove, .suspend = bf5xx_ac97_suspend, diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 4e675b55b18..6e5036bf924 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -292,7 +292,6 @@ static int bf5xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai bf5xx_i2s_dai = { .name = "bf5xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = bf5xx_i2s_probe, .remove = bf5xx_i2s_remove, .suspend = bf5xx_i2s_suspend, diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 8a93aff359d..c4208c8210c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -43,7 +43,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai ac97_dai = { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .stream_name = "AC97 Playback", .channels_min = 1, diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 2b26e1d80c8..651a15eb8c2 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -33,7 +33,6 @@ struct snd_soc_dai pcm3008_dai = { .name = "PCM3008 HiFi", - .type = SND_SOC_DAI_I2S, .playback = { .stream_name = "PCM3008 Playback", .channels_min = 1, diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 6e3e0f34017..40f14061fb7 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -535,7 +535,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai wm9712_dai[] = { { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97_BUS, + .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index a502667fca7..9dad0bffcb0 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1024,7 +1024,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai wm9713_dai[] = { { .name = "AC97 HiFi", - .type = SND_SOC_DAI_AC97_BUS, + .ac97_control = 1, .playback = { .stream_name = "HiFi Playback", .channels_min = 1, diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 7a17cd0ecf6..cf31b3bb96c 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -460,7 +460,6 @@ static void davinci_i2s_remove(struct platform_device *pdev, struct snd_soc_dai davinci_i2s_dai = { .name = "davinci-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = davinci_i2s_probe, .remove = davinci_i2s_remove, .playback = { diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index e2c172f3897..9ad8f9a2d8e 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -469,7 +469,6 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format) * psc_i2s_dai_template: template CPU Digital Audio Interface */ static struct snd_soc_dai psc_i2s_dai_template = { - .type = SND_SOC_DAI_I2S, .playback = { .channels_min = 2, .channels_max = 2, diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2eeb135c1e4..252bc7ebb19 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -456,7 +456,6 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, { \ .name = "omap-mcbsp-dai-"#link_id, \ .id = (link_id), \ - .type = SND_SOC_DAI_I2S, \ .playback = { \ .channels_min = 2, \ .channels_max = 2, \ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d0dd6245a20..402fc5ba65e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -788,7 +788,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { { .name = "pxa2xx-ssp1", .id = 0, - .type = SND_SOC_DAI_PCM, .probe = pxa_ssp_probe, .remove = pxa_ssp_remove, .suspend = pxa_ssp_suspend, @@ -820,7 +819,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { }, { .name = "pxa2xx-ssp2", .id = 1, - .type = SND_SOC_DAI_PCM, .probe = pxa_ssp_probe, .remove = pxa_ssp_remove, .suspend = pxa_ssp_suspend, @@ -853,7 +851,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { { .name = "pxa2xx-ssp3", .id = 2, - .type = SND_SOC_DAI_PCM, .probe = pxa_ssp_probe, .remove = pxa_ssp_remove, .suspend = pxa_ssp_suspend, @@ -886,7 +883,6 @@ struct snd_soc_dai pxa_ssp_dai[] = { { .name = "pxa2xx-ssp4", .id = 3, - .type = SND_SOC_DAI_PCM, .probe = pxa_ssp_probe, .remove = pxa_ssp_remove, .suspend = pxa_ssp_suspend, diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 86667d2f1b7..bffbe288634 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -173,7 +173,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = pxa2xx_ac97_probe, .remove = pxa2xx_ac97_remove, .suspend = pxa2xx_ac97_suspend, @@ -196,7 +196,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97-aux", .id = 1, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .stream_name = "AC97 Aux Playback", .channels_min = 1, @@ -215,7 +215,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { { .name = "pxa2xx-ac97-mic", .id = 2, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 9a3e55b4812..f9a9e2ebafa 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -340,7 +340,6 @@ static int pxa2xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai pxa_i2s_dai = { .name = "pxa2xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .suspend = pxa2xx_i2s_suspend, .resume = pxa2xx_i2s_resume, .playback = { diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 528fc3f1b45..3df2224a672 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -548,7 +548,6 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, - .type = SND_SOC_DAI_PCM, .playback = { .channels_min = 1, .channels_max = 1, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 360cc2a49d9..1c741047ae3 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -713,7 +713,6 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = s3c2412_i2s_probe, .suspend = s3c2412_i2s_suspend, .resume = s3c2412_i2s_resume, diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 31377821b2c..41bde6a3883 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -358,7 +358,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .probe = s3c2443_ac97_probe, .remove = s3c2443_ac97_remove, .playback = { @@ -380,7 +380,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "pxa2xx-ac97-mic", .id = 1, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .capture = { .stream_name = "AC97 Mic Capture", .channels_min = 1, diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 1bac9dd3dbd..8d9135f41bc 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -461,7 +461,6 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, - .type = SND_SOC_DAI_I2S, .probe = s3c24xx_i2s_probe, .suspend = s3c24xx_i2s_suspend, .resume = s3c24xx_i2s_resume, diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 3318071dc80..c435913c60e 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -271,7 +271,7 @@ struct snd_soc_dai sh4_hac_dai[] = { { .name = "HAC0", .id = 0, - .type = SND_SOC_DAI_AC97, + .ac97_control = 1, .playback = { .rates = AC97_RATES, .formats = AC97_FMTS, @@ -291,8 +291,8 @@ struct snd_soc_dai sh4_hac_dai[] = { #ifdef CONFIG_CPU_SUBTYPE_SH7760 { .name = "HAC1", + .ac97_control = 1, .id = 1, - .type = SND_SOC_DAI_AC97, .playback = { .rates = AC97_RATES, .formats = AC97_FMTS, diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 52a233840d2..fed544a3def 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -340,7 +340,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI0", .id = 0, - .type = SND_SOC_DAI_I2S, .playback = { .rates = SSI_RATES, .formats = SSI_FMTS, @@ -367,7 +366,6 @@ struct snd_soc_dai sh4_ssi_dai[] = { { .name = "SSI1", .id = 1, - .type = SND_SOC_DAI_I2S, .playback = { .rates = SSI_RATES, .formats = SSI_FMTS, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 43f4060dbe7..0d47696ccd0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -100,20 +100,6 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif -static inline const char *get_dai_name(int type) -{ - switch (type) { - case SND_SOC_DAI_AC97_BUS: - case SND_SOC_DAI_AC97: - return "AC97"; - case SND_SOC_DAI_I2S: - return "I2S"; - case SND_SOC_DAI_PCM: - return "PCM"; - } - return NULL; -} - /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -652,7 +638,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->suspend && cpu_dai->type != SND_SOC_DAI_AC97) + if (cpu_dai->suspend && !cpu_dai->ac97_control) cpu_dai->suspend(pdev, cpu_dai); if (platform->suspend) platform->suspend(pdev, cpu_dai); @@ -678,7 +664,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->suspend && cpu_dai->type == SND_SOC_DAI_AC97) + if (cpu_dai->suspend && cpu_dai->ac97_control) cpu_dai->suspend(pdev, cpu_dai); } @@ -714,7 +700,7 @@ static void soc_resume_deferred(struct work_struct *work) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->resume && cpu_dai->type == SND_SOC_DAI_AC97) + if (cpu_dai->resume && cpu_dai->ac97_control) cpu_dai->resume(pdev, cpu_dai); } @@ -741,7 +727,7 @@ static void soc_resume_deferred(struct work_struct *work) for (i = 0; i < card->num_links; i++) { struct snd_soc_dai *cpu_dai = card->dai_link[i].cpu_dai; - if (cpu_dai->resume && cpu_dai->type != SND_SOC_DAI_AC97) + if (cpu_dai->resume && !cpu_dai->ac97_control) cpu_dai->resume(pdev, cpu_dai); if (platform->resume) platform->resume(pdev, cpu_dai); @@ -898,8 +884,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev, codec_dai->codec = socdev->codec; /* check client and interface hw capabilities */ - sprintf(new_name, "%s %s-%s-%d", dai_link->stream_name, codec_dai->name, - get_dai_name(cpu_dai->type), num); + sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name, + num); if (codec_dai->playback.channels_min) playback = 1; @@ -1270,8 +1256,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) continue; } } - if (card->dai_link[i].codec_dai->type == - SND_SOC_DAI_AC97_BUS) + if (card->dai_link[i].codec_dai->ac97_control) ac97 = 1; } snprintf(codec->card->shortname, sizeof(codec->card->shortname), @@ -1335,7 +1320,7 @@ void snd_soc_free_pcms(struct snd_soc_device *socdev) #ifdef CONFIG_SND_SOC_AC97_BUS for (i = 0; i < codec->num_dai; i++) { codec_dai = &codec->dai[i]; - if (codec_dai->type == SND_SOC_DAI_AC97_BUS && codec->ac97) { + if (codec_dai->ac97_control && codec->ac97) { soc_ac97_dev_unregister(codec); goto free_card; } -- cgit v1.2.3-70-g09d2 From 67c91513b81a101800f113013234d2ab06bc5e52 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Nov 2008 17:45:26 +0000 Subject: ASoC: Flag AD1980 as an AC97 interface Special handling is required for suspend and resume of AC97 codecs due to the control path going over the data bus. Signed-off-by: Mark Brown --- sound/soc/codecs/ad1980.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 1a3ea059a6d..a9a268112d3 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -145,6 +145,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, struct snd_soc_dai ad1980_dai = { .name = "AC97", + .ac97_control = 1, .playback = { .stream_name = "Playback", .channels_min = 2, -- cgit v1.2.3-70-g09d2 From 54f01916297bafc18bd7df4e2300a0544a84fce3 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 26 Nov 2008 17:47:36 +0100 Subject: ASoC: Allow more routing features for tlv320aic3x This patch enables more routing functions for tlv320aic3x codecs. It is now possible to - control the volume of the PGA bypass path for the HPL, HPR, HPLCOM and HPRCOM outputs individually - route right line1 input to the left ADC channel - route left line1 input to the right ADC channel - route right mic3 input to left DAC channel - route left mic3 input to right DAC channel - route left line1 input to right line1 output - route right line1 input to left line1 output Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 96 +++++++++++++++++++++++++++++------------- sound/soc/codecs/tlv320aic3x.h | 12 ++++++ 2 files changed, 78 insertions(+), 30 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b76bcc3c411..255e784c805 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -272,8 +272,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACR1_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HP PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, - PGAR_2_HPROUT_VOL, 0, 0x7f, 1), + SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + 0, 0x7f, 1), + SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, + 0, 0x7f, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), @@ -281,8 +283,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), - SOC_DOUBLE_R("HPCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, - PGAR_2_HPRCOM_VOL, 0, 0x7f, 1), + SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL, + 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1), @@ -333,7 +337,8 @@ SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]); /* Left DAC_L1 Mixer */ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", DACL1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", DACL1_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HPCOM Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0), @@ -341,7 +346,8 @@ static const struct snd_kcontrol_new aic3x_left_dac_mixer_controls[] = { /* Right DAC_R1 Mixer */ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", DACR1_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", DACR1_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", DACR1_2_HPROUT_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HPCOM Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0), @@ -350,14 +356,18 @@ static const struct snd_kcontrol_new aic3x_right_dac_mixer_controls[] = { /* Left PGA Mixer */ static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1), }; /* Right PGA Mixer */ static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = { SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1), SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1), + SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1), SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1), }; @@ -379,34 +389,42 @@ SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]); /* Left PGA Bypass Mixer */ static const struct snd_kcontrol_new aic3x_left_pga_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", PGAL_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", PGAL_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HP Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPL Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPR Switch", PGAL_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0), }; /* Right PGA Bypass Mixer */ static const struct snd_kcontrol_new aic3x_right_pga_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", PGAR_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", PGAR_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HP Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPL Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPR Switch", PGAR_2_HPROUT_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0), }; /* Left Line2 Bypass Mixer */ static const struct snd_kcontrol_new aic3x_left_line2_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPLCOM Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0), }; /* Right Line2 Bypass Mixer */ static const struct snd_kcontrol_new aic3x_right_line2_bp_mixer_controls[] = { - SOC_DAPM_SINGLE("Line Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineL Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("LineR Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("Mono Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0), SOC_DAPM_SINGLE("HP Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0), - SOC_DAPM_SINGLE("HPCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), + SOC_DAPM_SINGLE("HPRCOM Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0), }; static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { @@ -439,22 +457,26 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { /* Mono Output */ SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0), - /* Left Inputs to Left ADC */ + /* Inputs to Left ADC */ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0, &aic3x_left_pga_mixer_controls[0], ARRAY_SIZE(aic3x_left_pga_mixer_controls)), SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line1_mux_controls), + SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0, + &aic3x_left_line1_mux_controls), SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0, &aic3x_left_line2_mux_controls), - /* Right Inputs to Right ADC */ + /* Inputs to Right ADC */ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", LINE1R_2_RADC_CTRL, 2, 0), SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0, &aic3x_right_pga_mixer_controls[0], ARRAY_SIZE(aic3x_right_pga_mixer_controls)), + SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0, + &aic3x_right_line1_mux_controls), SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0, &aic3x_right_line1_mux_controls), SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0, @@ -531,7 +553,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left DAC Mux", "DAC_L2", "Left DAC"}, {"Left DAC Mux", "DAC_L3", "Left DAC"}, - {"Left DAC_L1 Mixer", "Line Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "LineL Switch", "Left DAC Mux"}, + {"Left DAC_L1 Mixer", "LineR Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "Mono Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "HP Switch", "Left DAC Mux"}, {"Left DAC_L1 Mixer", "HPCOM Switch", "Left DAC Mux"}, @@ -557,7 +580,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right DAC Mux", "DAC_R2", "Right DAC"}, {"Right DAC Mux", "DAC_R3", "Right DAC"}, - {"Right DAC_R1 Mixer", "Line Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "LineL Switch", "Right DAC Mux"}, + {"Right DAC_R1 Mixer", "LineR Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "Mono Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "HP Switch", "Right DAC Mux"}, {"Right DAC_R1 Mixer", "HPCOM Switch", "Right DAC Mux"}, @@ -592,8 +616,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left Line2L Mux", "differential", "LINE2L"}, {"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"}, + {"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"}, {"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"}, {"Left PGA Mixer", "Mic3L Switch", "MIC3L"}, + {"Left PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Left ADC", NULL, "Left PGA Mixer"}, {"Left ADC", NULL, "GPIO1 dmic modclk"}, @@ -605,18 +631,23 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, + {"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"}, {"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"}, {"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"}, + {"Right PGA Mixer", "Mic3L Switch", "MIC3L"}, {"Right PGA Mixer", "Mic3R Switch", "MIC3R"}, {"Right ADC", NULL, "Right PGA Mixer"}, {"Right ADC", NULL, "GPIO1 dmic modclk"}, /* Left PGA Bypass */ - {"Left PGA Bypass Mixer", "Line Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "LineL Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "LineR Switch", "Left PGA Mixer"}, {"Left PGA Bypass Mixer", "Mono Switch", "Left PGA Mixer"}, - {"Left PGA Bypass Mixer", "HP Switch", "Left PGA Mixer"}, - {"Left PGA Bypass Mixer", "HPCOM Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPL Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPR Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPLCOM Switch", "Left PGA Mixer"}, + {"Left PGA Bypass Mixer", "HPRCOM Switch", "Left PGA Mixer"}, {"Left HPCOM Mux", "differential of HPLOUT", "Left PGA Bypass Mixer"}, {"Left HPCOM Mux", "constant VCM", "Left PGA Bypass Mixer"}, @@ -627,10 +658,13 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left HP Out", NULL, "Left PGA Bypass Mixer"}, /* Right PGA Bypass */ - {"Right PGA Bypass Mixer", "Line Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "LineL Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "LineR Switch", "Right PGA Mixer"}, {"Right PGA Bypass Mixer", "Mono Switch", "Right PGA Mixer"}, - {"Right PGA Bypass Mixer", "HP Switch", "Right PGA Mixer"}, - {"Right PGA Bypass Mixer", "HPCOM Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPL Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPR Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPLCOM Switch", "Right PGA Mixer"}, + {"Right PGA Bypass Mixer", "HPRCOM Switch", "Right PGA Mixer"}, {"Right HPCOM Mux", "differential of HPROUT", "Right PGA Bypass Mixer"}, {"Right HPCOM Mux", "constant VCM", "Right PGA Bypass Mixer"}, @@ -643,10 +677,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Right HP Out", NULL, "Right PGA Bypass Mixer"}, /* Left Line2 Bypass */ - {"Left Line2 Bypass Mixer", "Line Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "LineL Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "LineR Switch", "Left Line2L Mux"}, {"Left Line2 Bypass Mixer", "Mono Switch", "Left Line2L Mux"}, {"Left Line2 Bypass Mixer", "HP Switch", "Left Line2L Mux"}, - {"Left Line2 Bypass Mixer", "HPCOM Switch", "Left Line2L Mux"}, + {"Left Line2 Bypass Mixer", "HPLCOM Switch", "Left Line2L Mux"}, {"Left HPCOM Mux", "differential of HPLOUT", "Left Line2 Bypass Mixer"}, {"Left HPCOM Mux", "constant VCM", "Left Line2 Bypass Mixer"}, @@ -657,10 +692,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"Left HP Out", NULL, "Left Line2 Bypass Mixer"}, /* Right Line2 Bypass */ - {"Right Line2 Bypass Mixer", "Line Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "LineL Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "LineR Switch", "Right Line2R Mux"}, {"Right Line2 Bypass Mixer", "Mono Switch", "Right Line2R Mux"}, {"Right Line2 Bypass Mixer", "HP Switch", "Right Line2R Mux"}, - {"Right Line2 Bypass Mixer", "HPCOM Switch", "Right Line2R Mux"}, + {"Right Line2 Bypass Mixer", "HPRCOM Switch", "Right Line2R Mux"}, {"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"}, {"Right HPCOM Mux", "constant VCM", "Right Line2 Bypass Mixer"}, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 00a195aa02e..7e982acf399 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -48,7 +48,9 @@ #define MIC3LR_2_RADC_CTRL 18 /* Line1 Input control registers */ #define LINE1L_2_LADC_CTRL 19 +#define LINE1R_2_LADC_CTRL 21 #define LINE1R_2_RADC_CTRL 22 +#define LINE1L_2_RADC_CTRL 24 /* Line2 Input control registers */ #define LINE2L_2_LADC_CTRL 20 #define LINE2R_2_RADC_CTRL 23 @@ -79,6 +81,8 @@ #define LINE2L_2_HPLOUT_VOL 45 #define LINE2R_2_HPROUT_VOL 62 #define PGAL_2_HPLOUT_VOL 46 +#define PGAL_2_HPROUT_VOL 60 +#define PGAR_2_HPLOUT_VOL 49 #define PGAR_2_HPROUT_VOL 63 #define DACL1_2_HPLOUT_VOL 47 #define DACR1_2_HPROUT_VOL 64 @@ -88,6 +92,8 @@ #define LINE2L_2_HPLCOM_VOL 52 #define LINE2R_2_HPRCOM_VOL 69 #define PGAL_2_HPLCOM_VOL 53 +#define PGAR_2_HPLCOM_VOL 56 +#define PGAL_2_HPRCOM_VOL 67 #define PGAR_2_HPRCOM_VOL 70 #define DACL1_2_HPLCOM_VOL 54 #define DACR1_2_HPRCOM_VOL 71 @@ -103,11 +109,17 @@ #define MONOLOPM_CTRL 79 /* Line Output Plus/Minus control registers */ #define LINE2L_2_LLOPM_VOL 80 +#define LINE2L_2_RLOPM_VOL 87 +#define LINE2R_2_LLOPM_VOL 83 #define LINE2R_2_RLOPM_VOL 90 #define PGAL_2_LLOPM_VOL 81 +#define PGAL_2_RLOPM_VOL 88 +#define PGAR_2_LLOPM_VOL 84 #define PGAR_2_RLOPM_VOL 91 #define DACL1_2_LLOPM_VOL 82 +#define DACL1_2_RLOPM_VOL 89 #define DACR1_2_RLOPM_VOL 92 +#define DACR1_2_LLOPM_VOL 85 #define LLOPM_CTRL 86 #define RLOPM_CTRL 93 /* GPIO/IRQ registers */ -- cgit v1.2.3-70-g09d2 From 9171e5e6a20a9cd4992ff9c7cbee13c6fdf7b0b1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 1 Dec 2008 15:39:13 +0100 Subject: ALSA: soc - Fix compile warnings in wm8903.c MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Hide annoying uninitialized warnings: sound/soc/codecs/wm8903.c:382: warning: ‘reg’ may be used uninitialized in this function sound/soc/codecs/wm8903.c:383: warning: ‘shift’ may be used uninitialized in this function Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index efbe8927b7d..78070b2cd48 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -379,8 +379,8 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, struct wm8903_priv *wm8903 = codec->private_data; struct i2c_client *i2c = codec->control_data; u16 val; - u16 reg; - int shift; + u16 uninitialized_var(reg); + int uninitialized_var(shift); u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0); switch (w->reg) { -- cgit v1.2.3-70-g09d2 From 0bc286e2ac72e483d2b5a6dac0dafb05e9f047c8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 1 Dec 2008 19:59:35 +0100 Subject: Revert "ALSA: soc - Fix compile warnings in wm8903.c" This reverts commit 9171e5e6a20a9cd4992ff9c7cbee13c6fdf7b0b1. I can't reproduce the compile warnings any more. The warnings might be some weird cross-compiling set up. Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8903.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 78070b2cd48..efbe8927b7d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -379,8 +379,8 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, struct wm8903_priv *wm8903 = codec->private_data; struct i2c_client *i2c = codec->control_data; u16 val; - u16 uninitialized_var(reg); - int uninitialized_var(shift); + u16 reg; + int shift; u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0); switch (w->reg) { -- cgit v1.2.3-70-g09d2 From 968a6025aa9f909d487988efb542217a126023a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 28 Nov 2008 11:49:07 +0000 Subject: ASoC: Rename snd_soc_register_card() to snd_soc_init_card() Currently ASoC card initialisation is completed by a function called snd_soc_register_card(). As part of the work to allow independant registration of cards, codecs and machines in ASoC v2 a new function of the same name has been added so rename the existing function to facilitate the merge of v2. Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/soc-core.c | 6 +++--- 27 files changed, 29 insertions(+), 29 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc.h b/include/sound/soc.h index 444f9c21137..9356c1ce98c 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -162,7 +162,7 @@ extern struct snd_ac97_bus_ops soc_ac97_ops; /* pcm <-> DAI connect */ void snd_soc_free_pcms(struct snd_soc_device *socdev); int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid); -int snd_soc_register_card(struct snd_soc_device *socdev); +int snd_soc_init_card(struct snd_soc_device *socdev); /* set runtime hw params */ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index c4208c8210c..fb53e6511af 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -114,7 +114,7 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) goto bus_err; return 0; diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index a9a268112d3..73fdbb4d4a3 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -270,7 +270,7 @@ static int ad1980_soc_probe(struct platform_device *pdev) ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800); ad1980_add_controls(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad1980: failed to register card\n"); goto reset_err; diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 59c4c8f18cb..0f4110f4bce 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -67,7 +67,7 @@ static int ad73311_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ad73311: failed to register card\n"); goto register_err; diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index c742290e553..23062c952e8 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -512,7 +512,7 @@ static int ak4535_init(struct snd_soc_device *socdev) ak4535_add_controls(codec); ak4535_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "ak4535: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 7507d468b20..4667a07b566 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -723,7 +723,7 @@ static int cs4270_probe(struct platform_device *pdev) printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); #endif - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "cs4270: failed to register card\n"); goto error_del_driver; diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 651a15eb8c2..a5862555b44 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -91,7 +91,7 @@ static int pcm3008_soc_probe(struct platform_device *pdev) } /* Register Card. */ - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "pcm3008: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 0c5884ea1b0..973844973fe 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -624,7 +624,7 @@ static int ssm2602_init(struct snd_soc_device *socdev) ssm2602_add_controls(codec); ssm2602_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { pr_err("ssm2602: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a4e13d0688c..d209bec02a6 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -720,7 +720,7 @@ static int tlv320aic23_init(struct snd_soc_device *socdev) tlv320aic23_add_controls(codec); tlv320aic23_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "tlv320aic23: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 6b7ddfc9257..e33fb7e00d1 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -359,7 +359,7 @@ static int aic26_probe(struct platform_device *pdev) /* CODEC is setup, we can register the card now */ dev_dbg(&pdev->dev, "Registering card\n"); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { dev_err(&pdev->dev, "aic26: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 255e784c805..0f4067bdd4a 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1187,7 +1187,7 @@ static int aic3x_init(struct snd_soc_device *socdev) aic3x_add_controls(codec); aic3x_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "aic3x: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 41362314789..f3e9e591b52 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -764,7 +764,7 @@ static int twl4030_init(struct snd_soc_device *socdev) twl4030_add_controls(codec); twl4030_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "twl4030: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 91f333cdc7c..58de749185e 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -578,7 +578,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) goto pcm_err; } - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "UDA134X: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 330877c7069..42491650593 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -677,7 +677,7 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk) /* uda1380 init */ uda1380_add_controls(codec); uda1380_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { pr_err("uda1380: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 173b66c0c76..126c70f749d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -658,7 +658,7 @@ static int wm8510_init(struct snd_soc_device *socdev) wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm8510_add_controls(codec); wm8510_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8510: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 220d4b68904..572a31de321 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -869,7 +869,7 @@ static int wm8580_init(struct snd_soc_device *socdev) wm8580_add_controls(codec); wm8580_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8580: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 71949bd320d..28f12c6a6ac 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -332,7 +332,7 @@ static int wm8728_init(struct snd_soc_device *socdev) wm8728_add_controls(codec); wm8728_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8728: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index c0f277053bb..403dea13b5d 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -545,7 +545,7 @@ static int wm8731_init(struct snd_soc_device *socdev) wm8731_add_controls(codec); wm8731_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8731: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 860a1d56830..979446f5c98 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -818,7 +818,7 @@ static int wm8750_init(struct snd_soc_device *socdev) wm8750_add_controls(codec); wm8750_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8750: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 5e4cd3bb824..96c0453fffb 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1605,7 +1605,7 @@ static int wm8753_init(struct snd_soc_device *socdev) wm8753_add_controls(codec); wm8753_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8753: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index d1326be91c8..29cd83991c5 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1365,7 +1365,7 @@ static int wm8900_init(struct snd_soc_device *socdev) wm8900_add_controls(codec); wm8900_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { dev_err(&i2c_client->dev, "Failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index efbe8927b7d..393a4c19882 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1648,7 +1648,7 @@ static int wm8903_init(struct snd_soc_device *socdev) wm8903_add_controls(codec); wm8903_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { dev_err(&i2c->dev, "wm8903: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 26edcc9d6e8..53e6937e9ba 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -747,7 +747,7 @@ static int wm8971_init(struct snd_soc_device *socdev) wm8971_add_controls(codec); wm8971_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8971: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 13926516d16..5c5128b6b45 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1462,7 +1462,7 @@ static int wm8990_init(struct snd_soc_device *socdev) wm8990_add_controls(codec); wm8990_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm8990: failed to register card\n"); goto card_err; diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 40f14061fb7..af83d629078 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -700,7 +700,7 @@ static int wm9712_soc_probe(struct platform_device *pdev) wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY); wm9712_add_controls(codec); wm9712_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) { printk(KERN_ERR "wm9712: failed to register card\n"); goto reset_err; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 9dad0bffcb0..49962a88770 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1247,7 +1247,7 @@ static int wm9713_soc_probe(struct platform_device *pdev) wm9713_add_controls(codec); wm9713_add_widgets(codec); - ret = snd_soc_register_card(socdev); + ret = snd_soc_init_card(socdev); if (ret < 0) goto reset_err; return 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 13b4aaff0e9..0448708245e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1233,7 +1233,7 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) EXPORT_SYMBOL_GPL(snd_soc_new_pcms); /** - * snd_soc_register_card - register sound card + * snd_soc_init_card - register sound card * @socdev: the SoC audio device * * Register a SoC sound card. Also registers an AC97 device if the @@ -1241,7 +1241,7 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms); * * Returns 0 for success, else error. */ -int snd_soc_register_card(struct snd_soc_device *socdev) +int snd_soc_init_card(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->codec; struct snd_soc_card *card = socdev->card; @@ -1298,7 +1298,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) out: return ret; } -EXPORT_SYMBOL_GPL(snd_soc_register_card); +EXPORT_SYMBOL_GPL(snd_soc_init_card); /** * snd_soc_free_pcms - free sound card and pcms -- cgit v1.2.3-70-g09d2 From fa5c76978cee331b25e6d271482cf8e76f51e68b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 30 Nov 2008 22:55:46 +0000 Subject: ASoC: Remove in-code changelog from AD73311 driver Signed-off-by: Mark Brown --- sound/soc/codecs/ad73311.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 0f4110f4bce..500f9f3363d 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -8,9 +8,6 @@ * 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. - * - * Revision history - * 25th Sep 2008 Initial version. */ #include -- cgit v1.2.3-70-g09d2 From 381a22b564ff5a7ada09ad9a0831246da1dc5513 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 1 Dec 2008 10:03:45 +0200 Subject: ASoC: TWL4030: Change the capture volume control to TLV The digital Capture gain control has a range: 0 to 31 dB in 1 dB steps. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index f3e9e591b52..4b7a2d173a4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -360,6 +360,12 @@ static DECLARE_TLV_DB_SCALE(master_tlv, -6300, 100, 1); */ static DECLARE_TLV_DB_SCALE(master_coarse_tlv, 0, 600, 0); +/* + * Capture gain after the ADCs + * from 0 dB to 31 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Master Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, @@ -367,9 +373,11 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Master PCM Playback Volume", TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, 6, 0x2, 0, master_coarse_tlv), - SOC_DOUBLE_R("Capture Volume", - TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, - 0, 0x1f, 0), + + /* Common capture gain controls */ + SOC_DOUBLE_R_TLV("Capture Volume", + TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, + 0, 0x1f, 0, digital_capture_tlv), }; /* add non dapm controls */ -- cgit v1.2.3-70-g09d2 From d889a72c5c71161d6f934f9d7fca0e5b7e52bc08 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 1 Dec 2008 10:03:46 +0200 Subject: ASoC: TWL4030: Change the common playback volume controls Add Playback volume controls for all four DACs. All four paths has three levels of volume controls: Digital Fine gain, Digital Coarse gain, Analog gain. The controls are named to reflect their connection to the DACs. Per DAC volume can be performed, if needed: amixer sset 'DAC1 Analog' 5,10 DACL1 analog gain to 5 DACR1 analog gain to 10 Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++-------- 1 file changed, 29 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4b7a2d173a4..1dae73af527 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -351,14 +351,20 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) */ -static DECLARE_TLV_DB_SCALE(master_tlv, -6300, 100, 1); +static DECLARE_TLV_DB_SCALE(digital_fine_tlv, -6300, 100, 1); /* * CGAIN volume control: * 0 dB to 12 dB in 6 dB steps * value 2 and 3 means 12 dB */ -static DECLARE_TLV_DB_SCALE(master_coarse_tlv, 0, 600, 0); +static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0); + +/* + * Analog playback gain + * -24 dB to 12 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); /* * Capture gain after the ADCs @@ -367,12 +373,27 @@ static DECLARE_TLV_DB_SCALE(master_coarse_tlv, 0, 600, 0); static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); static const struct snd_kcontrol_new twl4030_snd_controls[] = { - SOC_DOUBLE_R_TLV("Master Playback Volume", - TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 0, 0x3f, 0, master_tlv), - SOC_DOUBLE_R_TLV("Master PCM Playback Volume", - TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, - 6, 0x2, 0, master_coarse_tlv), + /* Common playback gain controls */ + SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", + TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, + 0, 0x3f, 0, digital_fine_tlv), + SOC_DOUBLE_R_TLV("DAC2 Digital Fine Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 0, 0x3f, 0, digital_fine_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Digital Coarse Playback Volume", + TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, + 6, 0x2, 0, digital_coarse_tlv), + SOC_DOUBLE_R_TLV("DAC2 Digital Coarse Playback Volume", + TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, + 6, 0x2, 0, digital_coarse_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume", + TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL, + 3, 0x12, 1, analog_tlv), + SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 3, 0x12, 1, analog_tlv), /* Common capture gain controls */ SOC_DOUBLE_R_TLV("Capture Volume", -- cgit v1.2.3-70-g09d2 From 4290239cd05b6323da87b5e7e7db4c673bff5359 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 1 Dec 2008 10:03:47 +0200 Subject: ASoC: TWL4030: Add volume controls for outputs All outputs have dedicated gain controls except the HandsFree output. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1dae73af527..ffd5120697a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -366,6 +366,12 @@ static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0); */ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0); +/* + * Gain controls tied to outputs + * -6 dB to 6 dB in 6 dB steps (mute instead of -12) + */ +static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); + /* * Capture gain after the ADCs * from 0 dB to 31 dB in 1 dB steps @@ -395,6 +401,21 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, 3, 0x12, 1, analog_tlv), + /* Separate output gain controls */ + SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", + TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL, + 4, 3, 0, output_tvl), + + SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume", + TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl), + + SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume", + TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL, + 4, 3, 0, output_tvl), + + SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume", + TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl), + /* Common capture gain controls */ SOC_DOUBLE_R_TLV("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, -- cgit v1.2.3-70-g09d2 From 0ecfe7987855d21c2a89ffe003ddf0ee11b42d47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 1 Dec 2008 17:59:25 +0000 Subject: ASoC: Don't free static data in WM9713 Signed-off-by: Mark Brown --- sound/soc/codecs/wm9713.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 49962a88770..f3ca8aaf013 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1283,7 +1283,6 @@ static int wm9713_soc_remove(struct platform_device *pdev) snd_soc_free_ac97_codec(codec); kfree(codec->private_data); kfree(codec->reg_cache); - kfree(codec->dai); kfree(codec); return 0; } -- cgit v1.2.3-70-g09d2 From 5920b45303291057fef827f5bdafe04001c1bbae Mon Sep 17 00:00:00 2001 From: Grazvydas Ignotas Date: Tue, 2 Dec 2008 20:48:58 +0200 Subject: ASoC: TWL4030: Add input selection and gain controls The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 177 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 16 ++++ 2 files changed, 190 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ffd5120697a..3c9fdf2c6c7 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -46,9 +46,9 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ - 0x0a, /* REG_AVADC_CTL (0x7) */ + 0x20, /* REG_ANAMICL (0x5) */ + 0x00, /* REG_ANAMICR (0x6) */ + 0x00, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ 0x0c, /* REG_ATXL1PGA (0xA) */ @@ -347,6 +347,162 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +static int twl4030_get_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + int result = 0; + + /* one bit must be set a time */ + reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN; + if (reg != 0) { + result++; + while ((reg & 1) == 0) { + result++; + reg >>= 1; + } + } + + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int twl4030_put_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicl, micbias, avadc_ctl; + + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN); + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicl |= TWL4030_MAINMIC_EN; + micbias |= TWL4030_MICBIAS1_EN; + break; + case 2: + anamicl |= TWL4030_HSMIC_EN; + micbias |= TWL4030_HSMICBIAS_EN; + break; + case 3: + anamicl |= TWL4030_AUXL_EN; + break; + case 4: + anamicl |= TWL4030_CKMIC_EN; + break; + default: + break; + } + + /* If some input is selected, enable amp and ADC */ + if (value != 0) { + anamicl |= TWL4030_MICAMPL_EN; + avadc_ctl |= TWL4030_ADCL_EN; + } else { + anamicl &= ~TWL4030_MICAMPL_EN; + avadc_ctl &= ~TWL4030_ADCL_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static int twl4030_get_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + int value = 0; + + reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; + switch (reg) { + case TWL4030_SUBMIC_EN: + value = 1; + break; + case TWL4030_AUXR_EN: + value = 2; + break; + default: + break; + } + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int twl4030_put_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicr, micbias, avadc_ctl; + + anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~TWL4030_MICBIAS2_EN; + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicr |= TWL4030_SUBMIC_EN; + micbias |= TWL4030_MICBIAS2_EN; + break; + case 2: + anamicr |= TWL4030_AUXR_EN; + break; + default: + break; + } + + if (value != 0) { + anamicr |= TWL4030_MICAMPR_EN; + avadc_ctl |= TWL4030_ADCR_EN; + } else { + anamicr &= ~TWL4030_MICAMPR_EN; + avadc_ctl &= ~TWL4030_ADCR_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static const char *twl4030_left_in_sel[] = { + "None", + "Main Mic", + "Headset Mic", + "Line In", + "Carkit Mic", +}; + +static const char *twl4030_right_in_sel[] = { + "None", + "Sub Mic", + "Line In", +}; + +static const struct soc_enum twl4030_left_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel), + twl4030_left_in_sel); + +static const struct soc_enum twl4030_right_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel), + twl4030_right_in_sel); + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -378,6 +534,12 @@ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); */ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); +/* + * Gain control for input amplifiers + * 0 dB to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -420,6 +582,15 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0, digital_capture_tlv), + + SOC_DOUBLE_TLV("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN, + 0, 3, 5, 0, input_gain_tlv), + + /* Input source controls */ + SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux, + twl4030_get_left_input, twl4030_put_left_input), + SOC_ENUM_EXT("Right Input Source", twl4030_right_input_mux, + twl4030_get_right_input, twl4030_put_right_input), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 09865d9f520..a2065d417c2 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,7 +113,16 @@ #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ + +#define TWL4030_MICBIAS2_CTL 0x40 +#define TWL4030_MICBIAS1_CTL 0x20 +#define TWL4030_HSMICBIAS_EN 0x04 +#define TWL4030_MICBIAS2_EN 0x02 +#define TWL4030_MICBIAS1_EN 0x01 + /* ANAMICL (0x05) Fields */ + #define TWL4030_CNCL_OFFSET_START 0x80 #define TWL4030_OFFSET_CNCL_SEL 0x60 #define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 @@ -127,10 +136,17 @@ #define TWL4030_MAINMIC_EN 0x01 /* ANAMICR (0x06) Fields */ + #define TWL4030_MICAMPR_EN 0x10 #define TWL4030_AUXR_EN 0x04 #define TWL4030_SUBMIC_EN 0x01 +/* AVADC_CTL (0x07) Fields */ + +#define TWL4030_ADCL_EN 0x08 +#define TWL4030_AVADC_CLK_PRIORITY 0x04 +#define TWL4030_ADCR_EN 0x02 + /* AUDIO_IF (0x0E) Fields */ #define TWL4030_AIF_SLAVE_EN 0x80 -- cgit v1.2.3-70-g09d2 From 4b4fffdd9d179677cb030e97869286b62df25adc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 3 Dec 2008 11:21:08 +0000 Subject: ASoC: Fix WM8903 right mixer bypass path Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 393a4c19882..3c83b797307 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -773,14 +773,14 @@ static const struct snd_kcontrol_new left_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0), SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0), -SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 1, 1, 0), +SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_LEFT_MIX_0, 0, 1, 0), }; static const struct snd_kcontrol_new right_output_mixer[] = { SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 2, 1, 0), SOC_DAPM_SINGLE_W("Left Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0), -SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 1, 1, 0), +SOC_DAPM_SINGLE_W("Right Bypass Switch", WM8903_ANALOGUE_RIGHT_MIX_0, 0, 1, 0), }; static const struct snd_kcontrol_new left_speaker_mixer[] = { @@ -788,7 +788,7 @@ SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 3, 1, 0), SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 2, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, 1, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_LEFT_0, - 1, 1, 0), + 0, 1, 0), }; static const struct snd_kcontrol_new right_speaker_mixer[] = { @@ -797,7 +797,7 @@ SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 2, 1, 0), SOC_DAPM_SINGLE("Left Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, 1, 1, 0), SOC_DAPM_SINGLE("Right Bypass Switch", WM8903_ANALOGUE_SPK_MIX_RIGHT_0, - 1, 1, 0), + 0, 1, 0), }; static const struct snd_soc_dapm_widget wm8903_dapm_widgets[] = { -- cgit v1.2.3-70-g09d2 From 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 3 Dec 2008 11:44:17 +0100 Subject: ASoC: tlv320aic3x: headset/button press support - Add aic3x_set_headset_detection() function to define the headset detection mode for tlv32aic3x chips - added aic3x_button_pressed() - Read from the real-time registers in aic3x_headset_detected() to query headset presence without an occured interrupt Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 31 ++++++++++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.h | 46 +++++++++++++++++++++++++++++++++++++++++- 2 files changed, 74 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0f4067bdd4a..341e1adc916 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1018,14 +1018,41 @@ int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio) } EXPORT_SYMBOL_GPL(aic3x_get_gpio); +void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, + int headset_debounce, int button_debounce) +{ + u8 val; + + val = ((detect & AIC3X_HEADSET_DETECT_MASK) + << AIC3X_HEADSET_DETECT_SHIFT) | + ((headset_debounce & AIC3X_HEADSET_DEBOUNCE_MASK) + << AIC3X_HEADSET_DEBOUNCE_SHIFT) | + ((button_debounce & AIC3X_BUTTON_DEBOUNCE_MASK) + << AIC3X_BUTTON_DEBOUNCE_SHIFT); + + if (detect & AIC3X_HEADSET_DETECT_MASK) + val |= AIC3X_HEADSET_DETECT_ENABLED; + + aic3x_write(codec, AIC3X_HEADSET_DETECT_CTRL_A, val); +} +EXPORT_SYMBOL_GPL(aic3x_set_headset_detection); + int aic3x_headset_detected(struct snd_soc_codec *codec) { u8 val; - aic3x_read(codec, AIC3X_RT_IRQ_FLAGS_REG, &val); - return (val >> 2) & 1; + aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); + return (val >> 4) & 1; } EXPORT_SYMBOL_GPL(aic3x_headset_detected); +int aic3x_button_pressed(struct snd_soc_codec *codec) +{ + u8 val; + aic3x_read(codec, AIC3X_HEADSET_DETECT_CTRL_B, &val); + return (val >> 5) & 1; +} +EXPORT_SYMBOL_GPL(aic3x_button_pressed); + #define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 #define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 7e982acf399..73e35b6ec92 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -39,7 +39,9 @@ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 /* Audio codec digital filter control register */ #define AIC3X_CODEC_DFILT_CTRL 12 - +/* Headset/button press detection register */ +#define AIC3X_HEADSET_DETECT_CTRL_A 13 +#define AIC3X_HEADSET_DETECT_CTRL_B 14 /* ADC PGA Gain control registers */ #define LADC_VOL 15 #define RADC_VOL 16 @@ -233,7 +235,49 @@ enum { void aic3x_set_gpio(struct snd_soc_codec *codec, int gpio, int state); int aic3x_get_gpio(struct snd_soc_codec *codec, int gpio); + +/* headset detection / button API */ + +/* The AIC3x supports detection of stereo headsets (GND + left + right signal) + * and cellular headsets (GND + speaker output + microphone input). + * It is recommended to enable MIC bias for this function to work properly. + * For more information, please refer to the datasheet. */ +enum { + AIC3X_HEADSET_DETECT_OFF = 0, + AIC3X_HEADSET_DETECT_STEREO = 1, + AIC3X_HEADSET_DETECT_CELLULAR = 2, + AIC3X_HEADSET_DETECT_BOTH = 3 +}; + +enum { + AIC3X_HEADSET_DEBOUNCE_16MS = 0, + AIC3X_HEADSET_DEBOUNCE_32MS = 1, + AIC3X_HEADSET_DEBOUNCE_64MS = 2, + AIC3X_HEADSET_DEBOUNCE_128MS = 3, + AIC3X_HEADSET_DEBOUNCE_256MS = 4, + AIC3X_HEADSET_DEBOUNCE_512MS = 5 +}; + +enum { + AIC3X_BUTTON_DEBOUNCE_0MS = 0, + AIC3X_BUTTON_DEBOUNCE_8MS = 1, + AIC3X_BUTTON_DEBOUNCE_16MS = 2, + AIC3X_BUTTON_DEBOUNCE_32MS = 3 +}; + +#define AIC3X_HEADSET_DETECT_ENABLED 0x80 +#define AIC3X_HEADSET_DETECT_SHIFT 5 +#define AIC3X_HEADSET_DETECT_MASK 3 +#define AIC3X_HEADSET_DEBOUNCE_SHIFT 2 +#define AIC3X_HEADSET_DEBOUNCE_MASK 7 +#define AIC3X_BUTTON_DEBOUNCE_SHIFT 0 +#define AIC3X_BUTTON_DEBOUNCE_MASK 3 + +/* see the enums above for valid parameters to this function */ +void aic3x_set_headset_detection(struct snd_soc_codec *codec, int detect, + int headset_debounce, int button_debounce); int aic3x_headset_detected(struct snd_soc_codec *codec); +int aic3x_button_pressed(struct snd_soc_codec *codec); struct aic3x_setup_data { int i2c_bus; -- cgit v1.2.3-70-g09d2 From 28a1d869560a49d960ba2a3b450ec965712e5560 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 5 Dec 2008 17:31:00 +0100 Subject: ASoC: tlv320aic3x: control additions and cleanups - split "Line Playback Switch" into "LineL Playback Switch" and "LineR Playback Switch" - split "Line PGA Bypass Playback Volume" into "LineL Left PGA Bypass Playback Volume" and "LineR Right PGA Bypass Playback Volume" - split "Line Line2 Bypass Playback Volume" into "LineL Line2 Bypass Playback Volume" and "LineR Line2 Bypass Playback Volume" - Added "HP Right PGA Bypass Playback Volume" Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 341e1adc916..6a058298a3c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -253,11 +253,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, - 0x01, 0), - SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - PGAR_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, + DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, + 0, 0x7f, 1), + SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, @@ -272,6 +278,8 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACR1_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, + PGAR_2_HPROUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, -- cgit v1.2.3-70-g09d2 From 53b5047d994edfcafabc0e95bb681ae70d6e8604 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 08:45:43 +0200 Subject: ASoC: TWL4030: Correct DAPM_DAC with power control Add all four DACs to dapm_widgets with power switch. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 3c9fdf2c6c7..3543bf6e258 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -616,8 +616,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), - SND_SOC_DAPM_DAC("DACL", "Left Playback", SND_SOC_NOPM, 0, 0), - SND_SOC_DAPM_DAC("DACR", "Right Playback", SND_SOC_NOPM, 0, 0), + /* DACs */ + SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", + TWL4030_REG_AVDAC_CTL, 0, 0), + SND_SOC_DAPM_DAC("DACL1", "Left Front Playback", + TWL4030_REG_AVDAC_CTL, 1, 0), + SND_SOC_DAPM_DAC("DACR2", "Right Rear Playback", + TWL4030_REG_AVDAC_CTL, 2, 0), + SND_SOC_DAPM_DAC("DACL2", "Left Rear Playback", + TWL4030_REG_AVDAC_CTL, 3, 0), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), @@ -625,8 +632,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { static const struct snd_soc_dapm_route intercon[] = { /* outputs */ - {"OUTL", NULL, "DACL"}, - {"OUTR", NULL, "DACR"}, + {"OUTL", NULL, "DACL2"}, + {"OUTR", NULL, "DACR2"}, /* inputs */ {"ADCL", NULL, "INL"}, -- cgit v1.2.3-70-g09d2 From 44c5587035fbbdd368a3d5d8d11997d43758078a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 08:45:44 +0200 Subject: ASoC: TWL4030: Add Analog PGA control switch to DAPM Add all four APGA switch to DAPM routing and widgets. Add user control for DA enable for all APGA as normal control. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 25 +++++++++++++++++++++++-- 1 file changed, 23 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 3543bf6e258..4293ec7b502 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -562,6 +562,12 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC2 Analog Playback Volume", TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, 3, 0x12, 1, analog_tlv), + SOC_DOUBLE_R("DAC1 Analog Playback Switch", + TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL, + 1, 1, 0), + SOC_DOUBLE_R("DAC2 Analog Playback Switch", + TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL, + 1, 1, 0), /* Separate output gain controls */ SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume", @@ -626,14 +632,29 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_DAC("DACL2", "Left Rear Playback", TWL4030_REG_AVDAC_CTL, 3, 0), + /* Analog PGAs */ + SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, + 0, 0, NULL, 0), + SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_dapm_route intercon[] = { + {"ARXL1_APGA", NULL, "DACL1"}, + {"ARXR1_APGA", NULL, "DACR1"}, + {"ARXL2_APGA", NULL, "DACL2"}, + {"ARXR2_APGA", NULL, "DACR2"}, + /* outputs */ - {"OUTL", NULL, "DACL2"}, - {"OUTR", NULL, "DACR2"}, + {"OUTL", NULL, "ARXL2_APGA"}, + {"OUTR", NULL, "ARXR2_APGA"}, /* inputs */ {"ADCL", NULL, "INL"}, -- cgit v1.2.3-70-g09d2 From e8ff9c417ad6e8f7ef253e36f9d6e22dc2aa2512 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 12:35:46 +0200 Subject: ASoC: TWL4030: Add DAPM event handler for output MUX selection DAPM event handler is set to filter out invalid MUX settings for certain outputs. Earpiece: - 0 = Off - 1 = DACL1 - 2 = DACL2 - 3 = *** Invalid *** - 4 = DACR1 PreDriveL/R: - 0 = Off/Off - 1 = DACL1/DACR1 - 2 = DACL2/DACR2 - 3 = *** Invalid/Invalid *** - 4 = DACR2/DACL2 Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 4293ec7b502..9d1078325c3 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -190,6 +190,30 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } +static int outmixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + int ret = 0; + int val; + + switch (e->reg) { + case TWL4030_REG_PREDL_CTL: + case TWL4030_REG_PREDR_CTL: + case TWL4030_REG_EAR_CTL: + val = w->value >> e->shift_l; + if (val == 3) { + printk(KERN_WARNING + "Invalid MUX setting for register 0x%02x (%d)\n", + e->reg, val); + ret = -1; + } + break; + } + + return ret; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: -- cgit v1.2.3-70-g09d2 From 5e98a46449cd028b9b97a8ef2c2448c8f473d6c5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 12:35:47 +0200 Subject: ASoC: TWL4030: DAPM mapping of the Earpiece output Adds DAPM muxing, routing for the Earpiece output. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 9d1078325c3..1da46175519 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -190,6 +190,19 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) } +/* Earpiece */ +static const char *twl4030_earpiece_texts[] = + {"Off", "DACL1", "DACL2", "Invalid", + "DACR1"}; + +static const struct soc_enum twl4030_earpiece_enum = + SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, + ARRAY_SIZE(twl4030_earpiece_texts), + twl4030_earpiece_texts); + +static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = +SOC_DAPM_ENUM("Route", twl4030_earpiece_enum); + static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -645,6 +658,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), + SND_SOC_DAPM_OUTPUT("EARPIECE"), /* DACs */ SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", @@ -666,6 +680,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0), + /* Output MUX controls */ + /* Earpiece */ + SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_control, outmixer_event, + SND_SOC_DAPM_PRE_REG), + SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), }; @@ -676,9 +696,16 @@ static const struct snd_soc_dapm_route intercon[] = { {"ARXL2_APGA", NULL, "DACL2"}, {"ARXR2_APGA", NULL, "DACR2"}, + /* Internal playback routings */ + /* Earpiece */ + {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, + {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, + {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, {"OUTR", NULL, "ARXR2_APGA"}, + {"EARPIECE", NULL, "Earpiece Mux"}, /* inputs */ {"ADCL", NULL, "INL"}, -- cgit v1.2.3-70-g09d2 From 2a6f5c5892dcd17c81204fe5e26b92a37d2daafa Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 12:35:48 +0200 Subject: ASoC: TWL4030: DAPM mapping of the PreDriv outputs Adds DAPM muxing, routing for the PreDrive outputs. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1da46175519..c508344241e 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -203,6 +203,32 @@ static const struct soc_enum twl4030_earpiece_enum = static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = SOC_DAPM_ENUM("Route", twl4030_earpiece_enum); +/* PreDrive Left */ +static const char *twl4030_predrivel_texts[] = + {"Off", "DACL1", "DACL2", "Invalid", + "DACR2"}; + +static const struct soc_enum twl4030_predrivel_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, + ARRAY_SIZE(twl4030_predrivel_texts), + twl4030_predrivel_texts); + +static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = +SOC_DAPM_ENUM("Route", twl4030_predrivel_enum); + +/* PreDrive Right */ +static const char *twl4030_predriver_texts[] = + {"Off", "DACR1", "DACR2", "Invalid", + "DACL2"}; + +static const struct soc_enum twl4030_predriver_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, + ARRAY_SIZE(twl4030_predriver_texts), + twl4030_predriver_texts); + +static const struct snd_kcontrol_new twl4030_dapm_predriver_control = +SOC_DAPM_ENUM("Route", twl4030_predriver_enum); + static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -659,6 +685,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("OUTL"), SND_SOC_DAPM_OUTPUT("OUTR"), SND_SOC_DAPM_OUTPUT("EARPIECE"), + SND_SOC_DAPM_OUTPUT("PREDRIVEL"), + SND_SOC_DAPM_OUTPUT("PREDRIVER"), /* DACs */ SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", @@ -685,6 +713,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MUX_E("Earpiece Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_earpiece_control, outmixer_event, SND_SOC_DAPM_PRE_REG), + /* PreDrivL/R */ + SND_SOC_DAPM_MUX_E("PredriveL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_control, outmixer_event, + SND_SOC_DAPM_PRE_REG), + SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_control, outmixer_event, + SND_SOC_DAPM_PRE_REG), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), @@ -701,11 +736,21 @@ static const struct snd_soc_dapm_route intercon[] = { {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + /* PreDrivL */ + {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, + {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + /* PreDrivR */ + {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, + {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, {"OUTR", NULL, "ARXR2_APGA"}, {"EARPIECE", NULL, "Earpiece Mux"}, + {"PREDRIVEL", NULL, "PredriveL Mux"}, + {"PREDRIVER", NULL, "PredriveR Mux"}, /* inputs */ {"ADCL", NULL, "INL"}, -- cgit v1.2.3-70-g09d2 From dfad21a26f5b3cc379fbec9c5d12b5106dd1f9c5 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 12:35:49 +0200 Subject: ASoC: TWL4030: DAPM mapping of the Headset outputs Adds DAPM muxing, routing for the Headset outputs. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 39 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 39 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index c508344241e..86ff5a9ffa7 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -229,6 +229,30 @@ static const struct soc_enum twl4030_predriver_enum = static const struct snd_kcontrol_new twl4030_dapm_predriver_control = SOC_DAPM_ENUM("Route", twl4030_predriver_enum); +/* Headset Left */ +static const char *twl4030_hsol_texts[] = + {"Off", "DACL1", "DACL2"}; + +static const struct soc_enum twl4030_hsol_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, + ARRAY_SIZE(twl4030_hsol_texts), + twl4030_hsol_texts); + +static const struct snd_kcontrol_new twl4030_dapm_hsol_control = +SOC_DAPM_ENUM("Route", twl4030_hsol_enum); + +/* Headset Right */ +static const char *twl4030_hsor_texts[] = + {"Off", "DACR1", "DACR2"}; + +static const struct soc_enum twl4030_hsor_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, + ARRAY_SIZE(twl4030_hsor_texts), + twl4030_hsor_texts); + +static const struct snd_kcontrol_new twl4030_dapm_hsor_control = +SOC_DAPM_ENUM("Route", twl4030_hsor_enum); + static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -687,6 +711,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("EARPIECE"), SND_SOC_DAPM_OUTPUT("PREDRIVEL"), SND_SOC_DAPM_OUTPUT("PREDRIVER"), + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), /* DACs */ SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", @@ -720,6 +746,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MUX_E("PredriveR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_predriver_control, outmixer_event, SND_SOC_DAPM_PRE_REG), + /* HeadsetL/R */ + SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_control), + SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_control), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), @@ -744,6 +775,12 @@ static const struct snd_soc_dapm_route intercon[] = { {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + /* HeadsetL */ + {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, + {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + /* HeadsetR */ + {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, + {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, @@ -751,6 +788,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"EARPIECE", NULL, "Earpiece Mux"}, {"PREDRIVEL", NULL, "PredriveL Mux"}, {"PREDRIVER", NULL, "PredriveR Mux"}, + {"HSOL", NULL, "HeadsetL Mux"}, + {"HSOR", NULL, "HeadsetR Mux"}, /* inputs */ {"ADCL", NULL, "INL"}, -- cgit v1.2.3-70-g09d2 From 5152d8c28b95e421b91483ca0df76726e6e6c41e Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 12:35:50 +0200 Subject: ASoC: TWL4030: DAPM mapping of the Carkit outputs Adds DAPM muxing, routing for the Carkit outputs. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 37 +++++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 86ff5a9ffa7..08c33e9b96c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -253,6 +253,30 @@ static const struct soc_enum twl4030_hsor_enum = static const struct snd_kcontrol_new twl4030_dapm_hsor_control = SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +/* Carkit Left */ +static const char *twl4030_carkitl_texts[] = + {"Off", "DACL1", "DACL2"}; + +static const struct soc_enum twl4030_carkitl_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, + ARRAY_SIZE(twl4030_carkitl_texts), + twl4030_carkitl_texts); + +static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = +SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); + +/* Carkit Right */ +static const char *twl4030_carkitr_texts[] = + {"Off", "DACR1", "DACR2"}; + +static const struct soc_enum twl4030_carkitr_enum = + SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, + ARRAY_SIZE(twl4030_carkitr_texts), + twl4030_carkitr_texts); + +static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = +SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); + static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -751,6 +775,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_hsol_control), SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_hsor_control), + /* CarkitL/R */ + SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_control), + SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_control), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), @@ -781,6 +810,12 @@ static const struct snd_soc_dapm_route intercon[] = { /* HeadsetR */ {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + /* CarkitL */ + {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, + {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + /* CarkitR */ + {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, + {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, @@ -790,6 +825,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"PREDRIVER", NULL, "PredriveR Mux"}, {"HSOL", NULL, "HeadsetL Mux"}, {"HSOR", NULL, "HeadsetR Mux"}, + {"CARKITL", NULL, "CarkitL Mux"}, + {"CARKITR", NULL, "CarkitR Mux"}, /* inputs */ {"ADCL", NULL, "INL"}, -- cgit v1.2.3-70-g09d2 From df339804bbfc118eaca066b95488a2dbacc2e258 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 12:35:51 +0200 Subject: ASoC: TWL4030: DAPM mapping of the Handsfree outputs Adds DAPM muxing, routing for the Handsfree outputs. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 41 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 08c33e9b96c..d0612a4ca45 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -277,6 +277,30 @@ static const struct soc_enum twl4030_carkitr_enum = static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +/* Handsfree Left */ +static const char *twl4030_handsfreel_texts[] = + {"Voice", "DACL1", "DACL2", "DACR2"}; + +static const struct soc_enum twl4030_handsfreel_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, + ARRAY_SIZE(twl4030_handsfreel_texts), + twl4030_handsfreel_texts); + +static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control = +SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); + +/* Handsfree Right */ +static const char *twl4030_handsfreer_texts[] = + {"Voice", "DACR1", "DACR2", "DACL2"}; + +static const struct soc_enum twl4030_handsfreer_enum = + SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, + ARRAY_SIZE(twl4030_handsfreer_texts), + twl4030_handsfreer_texts); + +static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = +SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); + static int outmixer_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -737,6 +761,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("PREDRIVER"), SND_SOC_DAPM_OUTPUT("HSOL"), SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), /* DACs */ SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", @@ -780,6 +806,11 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_carkitl_control), SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitr_control), + /* HandsfreeL/R */ + SND_SOC_DAPM_MUX("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, + &twl4030_dapm_handsfreel_control), + SND_SOC_DAPM_MUX("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, + &twl4030_dapm_handsfreer_control), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), @@ -816,6 +847,14 @@ static const struct snd_soc_dapm_route intercon[] = { /* CarkitR */ {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + /* HandsfreeL */ + {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + /* HandsfreeR */ + {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, @@ -827,6 +866,8 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSOR", NULL, "HeadsetR Mux"}, {"CARKITL", NULL, "CarkitL Mux"}, {"CARKITR", NULL, "CarkitR Mux"}, + {"HFL", NULL, "HandsfreeL Mux"}, + {"HFR", NULL, "HandsfreeR Mux"}, /* inputs */ {"ADCL", NULL, "INL"}, -- cgit v1.2.3-70-g09d2 From ca4513fe06c483bf0111c990059d42f97288605d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 9 Dec 2008 12:35:52 +0200 Subject: ASoC: TWL4030: Do not alter the Headset output volume on power-up/down There is a separate gain control for the Headset output already. Do not reset the gain to 0 dB at power up. In power-down, there is no need to set the Headset output gain to power-down mode, since if the CODECPDZ is in powered off this setting has no effect. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index d0612a4ca45..358aa2b1aae 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -887,7 +887,7 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec) static void twl4030_power_up(struct snd_soc_codec *codec) { - u8 anamicl, regmisc1, byte, popn, hsgain; + u8 anamicl, regmisc1, byte, popn; int i = 0; /* set CODECPDZ to turn on codec */ @@ -925,10 +925,6 @@ static void twl4030_power_up(struct snd_soc_codec *codec) popn |= TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - /* enable output stage and gain setting */ - hsgain = TWL4030_HSR_GAIN_0DB | TWL4030_HSL_GAIN_0DB; - twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); - /* enable anti-pop ramp */ popn |= TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); @@ -936,17 +932,13 @@ static void twl4030_power_up(struct snd_soc_codec *codec) static void twl4030_power_down(struct snd_soc_codec *codec) { - u8 popn, hsgain; + u8 popn; /* disable anti-pop ramp */ popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET); popn &= ~TWL4030_RAMP_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); - /* disable output stage and gain setting */ - hsgain = TWL4030_HSR_GAIN_PWR_DOWN | TWL4030_HSL_GAIN_PWR_DOWN; - twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hsgain); - /* disable bias out */ popn &= ~TWL4030_VMID_EN; twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn); -- cgit v1.2.3-70-g09d2 From 64089b84abfe2f26a864ebd968429302dcb071de Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 8 Dec 2008 19:17:58 +0000 Subject: ASoC: Register non-AC97 codec DAIs Currently this is done at module probe time since ASoC ties in codec device probe to the instantiation of the entire ASoC device. Subsequent patches will refactor the codec drivers to handle probing separately. Note that the core does not yet use this information. AC97 is special since the codec is controlled over the AC97 link but we want to give the machine driver a chance to set up the system before trying to instantiate since it may need to do configuration before the AC97 link will operate Signed-off-by: Mark Brown --- sound/soc/codecs/ad73311.c | 12 ++++++++++++ sound/soc/codecs/ak4535.c | 12 ++++++++++++ sound/soc/codecs/cs4270.c | 12 ++++++++++++ sound/soc/codecs/pcm3008.c | 12 ++++++++++++ sound/soc/codecs/ssm2602.c | 12 ++++++++++++ sound/soc/codecs/tlv320aic23.c | 12 ++++++++++++ sound/soc/codecs/tlv320aic26.c | 15 ++++++++++++--- sound/soc/codecs/tlv320aic3x.c | 12 ++++++++++++ sound/soc/codecs/twl4030.c | 14 +++++++++++++- sound/soc/codecs/uda134x.c | 12 ++++++++++++ sound/soc/codecs/uda1380.c | 12 ++++++++++++ sound/soc/codecs/wm8510.c | 12 ++++++++++++ sound/soc/codecs/wm8580.c | 12 ++++++++++++ sound/soc/codecs/wm8728.c | 12 ++++++++++++ sound/soc/codecs/wm8731.c | 12 ++++++++++++ sound/soc/codecs/wm8750.c | 12 ++++++++++++ sound/soc/codecs/wm8753.c | 12 ++++++++++++ sound/soc/codecs/wm8900.c | 12 ++++++++++++ sound/soc/codecs/wm8903.c | 12 ++++++++++++ sound/soc/codecs/wm8971.c | 12 ++++++++++++ sound/soc/codecs/wm8990.c | 12 ++++++++++++ 21 files changed, 253 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index 500f9f3363d..e32f55034e6 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -98,6 +98,18 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); +static int __devinit ad73311_init(void) +{ + return snd_soc_register_dai(&ad73311_dai); +} +module_init(ad73311_init); + +static void __exit ad73311_exit(void) +{ + snd_soc_unregister_dai(&ad73311_dai); +} +module_exit(ad73311_exit); + MODULE_DESCRIPTION("ASoC ad73311 driver"); MODULE_AUTHOR("Cliff Cai "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 23062c952e8..94148fba911 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -688,6 +688,18 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); +static int __devinit ak4535_modinit(void) +{ + return snd_soc_register_dai(&ak4535_dai); +} +module_init(ak4535_modinit); + +static void __exit ak4535_exit(void) +{ + snd_soc_unregister_dai(&ak4535_dai); +} +module_exit(ak4535_exit); + MODULE_DESCRIPTION("Soc AK4535 driver"); MODULE_AUTHOR("Richard Purdie"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 4667a07b566..73aaf249d78 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -774,6 +774,18 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = { }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); +static int __devinit cs4270_init(void) +{ + return snd_soc_register_dai(&cs4270_dai); +} +module_init(cs4270_init); + +static void __exit cs4270_exit(void) +{ + snd_soc_unregister_dai(&cs4270_dai); +} +module_exit(cs4270_exit); + MODULE_AUTHOR("Timur Tabi "); MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index a5862555b44..f333e88ee25 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -195,6 +195,18 @@ struct snd_soc_codec_device soc_codec_dev_pcm3008 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008); +static int __devinit pcm3008_init(void) +{ + return snd_soc_register_dai(&pcm3008_dai); +} +module_init(pcm3008_init); + +static void __exit pcm3008_exit(void) +{ + snd_soc_unregister_dai(&pcm3008_dai); +} +module_exit(pcm3008_exit); + MODULE_DESCRIPTION("Soc PCM3008 driver"); MODULE_AUTHOR("Hugo Villeneuve"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 973844973fe..77fdcb4b9a1 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -793,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); +static int __devinit ssm2602_modinit(void) +{ + return snd_soc_register_dai(&ssm2602_dai); +} +module_init(ssm2602_modinit); + +static void __exit ssm2602_exit(void) +{ + snd_soc_unregister_dai(&ssm2602_dai); +} +module_exit(ssm2602_exit); + MODULE_DESCRIPTION("ASoC ssm2602 driver"); MODULE_AUTHOR("Cliff Cai"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index d209bec02a6..eac449b92bd 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -847,6 +847,18 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); +static int __devinit tlv320aic23_modinit(void) +{ + return snd_soc_register_dai(&tlv320aic23_dai); +} +module_init(tlv320aic23_modinit); + +static void __exit tlv320aic23_exit(void) +{ + snd_soc_unregister_dai(&tlv320aic23_dai); +} +module_exit(tlv320aic23_exit); + MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index e33fb7e00d1..29f2f1a017f 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -426,7 +426,7 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_spi_probe(struct spi_device *spi) { struct aic26 *aic26; - int rc, i, reg; + int ret, i, reg; dev_dbg(&spi->dev, "probing tlv320aic26 spi device\n"); @@ -456,6 +456,14 @@ static int aic26_spi_probe(struct spi_device *spi) aic26->codec.reg_cache_size = AIC26_NUM_REGS; aic26->codec.reg_cache = aic26->reg_cache; + aic26_dai.dev = &spi->dev; + ret = snd_soc_register_dai(&aic26_dai); + if (ret != 0) { + dev_err(&spi->dev, "Failed to register DAI: %d\n", ret); + kfree(aic26); + return ret; + } + /* Reset the codec to power on defaults */ aic26_reg_write(&aic26->codec, AIC26_REG_RESET, 0xBB00); @@ -474,8 +482,8 @@ static int aic26_spi_probe(struct spi_device *spi) /* Register the sysfs files for debugging */ /* Create SysFS files */ - rc = device_create_file(&spi->dev, &dev_attr_keyclick); - if (rc) + ret = device_create_file(&spi->dev, &dev_attr_keyclick); + if (ret) dev_info(&spi->dev, "error creating sysfs files\n"); #if defined(CONFIG_SND_SOC_OF_SIMPLE) @@ -492,6 +500,7 @@ static int aic26_spi_remove(struct spi_device *spi) { struct aic26 *aic26 = dev_get_drvdata(&spi->dev); + snd_soc_unregister_dai(&aic26_dai); kfree(aic26); return 0; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a058298a3c..ccd57596186 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1411,6 +1411,18 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x); +static int __devinit aic3x_modinit(void) +{ + return snd_soc_register_dai(&aic3x_dai); +} +module_init(aic3x_modinit); + +static void __exit aic3x_exit(void) +{ + snd_soc_unregister_dai(&aic3x_dai); +} +module_exit(aic3x_exit); + MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver"); MODULE_AUTHOR("Vladimir Barinov"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 358aa2b1aae..373daa486ce 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -358,7 +358,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, .put = snd_soc_put_volsw_r2_twl4030, \ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ - .max = xmax, .invert = xinvert} } + .rshift = xshift, .max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \ SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \ xinvert, tlv_array) @@ -1275,6 +1275,18 @@ struct snd_soc_codec_device soc_codec_dev_twl4030 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); +static int __devinit twl4030_init(void) +{ + return snd_soc_register_dai(&twl4030_dai); +} +module_init(twl4030_init); + +static void __exit twl4030_exit(void) +{ + snd_soc_unregister_dai(&twl4030_dai); +} +module_exit(twl4030_exit); + MODULE_DESCRIPTION("ASoC TWL4030 codec driver"); MODULE_AUTHOR("Steve Sakoman"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 58de749185e..8e035b5d733 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -651,6 +651,18 @@ struct snd_soc_codec_device soc_codec_dev_uda134x = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x); +static int __devinit uda134x_init(void) +{ + return snd_soc_register_dai(&uda134x_dai); +} +module_init(uda134x_init); + +static void __exit uda134x_exit(void) +{ + snd_soc_unregister_dai(&uda134x_dai); +} +module_exit(uda134x_exit); + MODULE_DESCRIPTION("UDA134X ALSA soc codec driver"); MODULE_AUTHOR("Zoltan Devai, Christian Pellegrin "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 42491650593..55a99b6a68a 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -841,6 +841,18 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); +static int __devinit uda1380_modinit(void) +{ + return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); +} +module_init(uda1380_modinit); + +static void __exit uda1380_exit(void) +{ + snd_soc_unregister_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); +} +module_exit(uda1380_exit); + MODULE_AUTHOR("Giorgio Padrin"); MODULE_DESCRIPTION("Audio support for codec Philips UDA1380"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 126c70f749d..a2af04bb4e9 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -889,6 +889,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); +static int __devinit wm8510_modinit(void) +{ + return snd_soc_register_dai(&wm8510_dai); +} +module_init(wm8510_modinit); + +static void __exit wm8510_exit(void) +{ + snd_soc_unregister_dai(&wm8510_dai); +} +module_exit(wm8510_exit); + MODULE_DESCRIPTION("ASoC WM8510 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 572a31de321..391ec2978ae 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1042,6 +1042,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); +static int __devinit wm8580_modinit(void) +{ + return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); +} +module_init(wm8580_modinit); + +static void __exit wm8580_exit(void) +{ + snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); +} +module_exit(wm8580_exit); + MODULE_DESCRIPTION("ASoC WM8580 driver"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 28f12c6a6ac..d905e25b1a9 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -568,6 +568,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8728 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728); +static int __devinit wm8728_modinit(void) +{ + return snd_soc_register_dai(&wm8728_dai); +} +module_init(wm8728_modinit); + +static void __exit wm8728_exit(void) +{ + snd_soc_unregister_dai(&wm8728_dai); +} +module_exit(wm8728_exit); + MODULE_DESCRIPTION("ASoC WM8728 driver"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 403dea13b5d..7b455a60d71 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -793,6 +793,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); +static int __devinit wm8731_modinit(void) +{ + return snd_soc_register_dai(&wm8731_dai); +} +module_init(wm8731_modinit); + +static void __exit wm8731_exit(void) +{ + snd_soc_unregister_dai(&wm8731_dai); +} +module_exit(wm8731_exit); + MODULE_DESCRIPTION("ASoC WM8731 driver"); MODULE_AUTHOR("Richard Purdie"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 979446f5c98..84a6307de90 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -1085,6 +1085,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); +static int __devinit wm8750_modinit(void) +{ + return snd_soc_register_dai(&wm8750_dai); +} +module_init(wm8750_modinit); + +static void __exit wm8750_exit(void) +{ + snd_soc_unregister_dai(&wm8750_dai); +} +module_exit(wm8750_exit); + MODULE_DESCRIPTION("ASoC WM8750 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 96c0453fffb..1caca30d081 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1874,6 +1874,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); +static int __devinit wm8753_modinit(void) +{ + return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); +} +module_init(wm8753_modinit); + +static void __exit wm8753_exit(void) +{ + snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); +} +module_exit(wm8753_exit); + MODULE_DESCRIPTION("ASoC WM8753 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 29cd83991c5..02bb1c999ba 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1530,6 +1530,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900); +static int __devinit wm8900_modinit(void) +{ + return snd_soc_register_dai(&wm8900_dai); +} +module_init(wm8900_modinit); + +static void __exit wm8900_exit(void) +{ + snd_soc_unregister_dai(&wm8900_dai); +} +module_exit(wm8900_exit); + MODULE_DESCRIPTION("ASoC WM8900 driver"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3c83b797307..5d8fe7e1571 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1809,6 +1809,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903); +static int __devinit wm8903_modinit(void) +{ + return snd_soc_register_dai(&wm8903_dai); +} +module_init(wm8903_modinit); + +static void __exit wm8903_exit(void) +{ + snd_soc_unregister_dai(&wm8903_dai); +} +module_exit(wm8903_exit); + MODULE_DESCRIPTION("ASoC WM8903 driver"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 53e6937e9ba..2979fc4f44f 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -935,6 +935,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = { EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971); +static int __devinit wm8971_modinit(void) +{ + return snd_soc_register_dai(&wm8971_dai); +} +module_init(wm8971_modinit); + +static void __exit wm8971_exit(void) +{ + snd_soc_unregister_dai(&wm8971_dai); +} +module_exit(wm8971_exit); + MODULE_DESCRIPTION("ASoC WM8971 driver"); MODULE_AUTHOR("Lab126"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 5c5128b6b45..53e71aafe6c 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1643,6 +1643,18 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990); +static int __devinit wm8990_modinit(void) +{ + return snd_soc_register_dai(&wm8990_dai); +} +module_init(wm8990_modinit); + +static void __exit wm8990_exit(void) +{ + snd_soc_unregister_dai(&wm8990_dai); +} +module_exit(wm8990_exit); + MODULE_DESCRIPTION("ASoC WM8990 driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); -- cgit v1.2.3-70-g09d2 From f0752331b89ce79063f765545dd7dd5f49d9a713 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 9 Dec 2008 12:51:56 +0000 Subject: ASoC: Convert WM8900 to allow registration by machine code This makes use of the support for delayed DAI registration to allow the WM8900 I2C device to be registered by general platform/architecture code rather than as part of the ASoC device probe. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 97 ++++++++++------------------------------------- 1 file changed, 20 insertions(+), 77 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 02bb1c999ba..34e58af0c65 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1382,32 +1382,21 @@ priv_err: return ret; } -static struct snd_soc_device *wm8900_socdev; +static struct i2c_client *wm8900_client; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - -/* If the i2c layer weren't so broken, we could pass this kind of data - around */ static int wm8900_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { - struct snd_soc_device *socdev = wm8900_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; - - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; - - ret = wm8900_init(socdev); - if (ret < 0) - dev_err(&i2c->dev, "failed to initialise WM8900\n"); - return ret; + wm8900_client = i2c; + wm8900_dai.dev = &i2c->dev; + return snd_soc_register_dai(&wm8900_dai); } static int wm8900_i2c_remove(struct i2c_client *client) { - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); + snd_soc_unregister_dai(&wm8900_dai); + wm8900_dai.dev = NULL; + wm8900_client = NULL; return 0; } @@ -1427,57 +1416,17 @@ static struct i2c_driver wm8900_i2c_driver = { .id_table = wm8900_i2c_id, }; -static int wm8900_add_i2c_device(struct platform_device *pdev, - const struct wm8900_setup_data *setup) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - int ret; - - ret = i2c_add_driver(&wm8900_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - return ret; - } - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = setup->i2c_address; - strlcpy(info.type, "wm8900", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "can't get i2c adapter %d\n", - setup->i2c_bus); - goto err_driver; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - dev_err(&pdev->dev, "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - goto err_driver; - } - - return 0; - -err_driver: - i2c_del_driver(&wm8900_i2c_driver); - return -ENODEV; -} -#endif - static int wm8900_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8900_setup_data *setup; struct snd_soc_codec *codec; int ret = 0; - dev_info(&pdev->dev, "WM8900 Audio Codec\n"); + if (!wm8900_client) { + dev_err(&pdev->dev, "I2C client not yet instantiated\n"); + return -ENODEV; + } - setup = socdev->codec_data; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; @@ -1490,15 +1439,13 @@ static int wm8900_probe(struct platform_device *pdev) codec->set_bias_level = wm8900_set_bias_level; - wm8900_socdev = socdev; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - if (setup->i2c_address) { - codec->hw_write = (hw_write_t)i2c_master_send; - ret = wm8900_add_i2c_device(pdev, setup); - } -#else -#error Non-I2C interfaces not yet supported -#endif + codec->hw_write = (hw_write_t)i2c_master_send; + codec->control_data = wm8900_client; + + ret = wm8900_init(socdev); + if (ret != 0) + kfree(codec); + return ret; } @@ -1513,10 +1460,6 @@ static int wm8900_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); - i2c_del_driver(&wm8900_i2c_driver); -#endif kfree(codec); return 0; @@ -1532,13 +1475,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900); static int __devinit wm8900_modinit(void) { - return snd_soc_register_dai(&wm8900_dai); + return i2c_add_driver(&wm8900_i2c_driver); } module_init(wm8900_modinit); static void __exit wm8900_exit(void) { - snd_soc_unregister_dai(&wm8900_dai); + i2c_del_driver(&wm8900_i2c_driver); } module_exit(wm8900_exit); -- cgit v1.2.3-70-g09d2 From 24e07db8cceb7dfe2d4005e4450a27f4bcda6499 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Dec 2008 07:40:24 +0100 Subject: ALSA: ASoC - Fix module init entry for twl4030.c Fixed the function name of module init entry for twl4030.c, which conflicted with the existing hardware init function: sound/soc/codecs/twl4030.c:1278: error: conflicting types for 'twl4030_init' sound/soc/codecs/twl4030.c:1187: error: previous definition of 'twl4030_init' was here Also fixed the section type of init function. Signed-off-by: Takashi Iwai --- sound/soc/codecs/twl4030.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 373daa486ce..71ab5342bea 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1275,11 +1275,11 @@ struct snd_soc_codec_device soc_codec_dev_twl4030 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); -static int __devinit twl4030_init(void) +static int __init twl4030_modinit(void) { return snd_soc_register_dai(&twl4030_dai); } -module_init(twl4030_init); +module_init(twl4030_modinit); static void __exit twl4030_exit(void) { -- cgit v1.2.3-70-g09d2 From c9b3a40ff2b3dea9914e36965a17c802650bb603 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 10 Dec 2008 07:47:22 +0100 Subject: ALSA: ASoC - Fix wrong section types The module init entries should be __init instead of __devinit. Signed-off-by: Takashi Iwai --- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 2 +- sound/soc/codecs/ad73311.c | 2 +- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-pcm.c | 2 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/pxa-ssp.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- sound/soc/sh/dma-sh7760.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/ssi.c | 2 +- sound/soc/soc-core.c | 2 +- 41 files changed, 41 insertions(+), 41 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 027eb13f9dd..1fac5efd285 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -477,7 +477,7 @@ struct snd_soc_platform atmel_soc_platform = { }; EXPORT_SYMBOL_GPL(atmel_soc_platform); -static int __devinit atmel_pcm_modinit(void) +static int __init atmel_pcm_modinit(void) { return snd_soc_register_platform(&atmel_soc_platform); } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 87904b6ab8c..c5d67900d66 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -772,7 +772,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = { }; EXPORT_SYMBOL_GPL(atmel_ssc_dai); -static int __devinit atmel_ssc_modinit(void) +static int __init atmel_ssc_modinit(void) { return snd_soc_register_dais(atmel_ssc_dai, ARRAY_SIZE(atmel_ssc_dai)); } diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 5b27e0d9d0e..8067cfafa3a 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -451,7 +451,7 @@ struct snd_soc_platform bf5xx_ac97_soc_platform = { }; EXPORT_SYMBOL_GPL(bf5xx_ac97_soc_platform); -static int __devinit bfin_ac97_init(void) +static int __init bfin_ac97_init(void) { return snd_soc_register_platform(&bf5xx_ac97_soc_platform); } diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index ad3efeeb6d4..3be2be60576 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -431,7 +431,7 @@ struct snd_soc_dai bfin_ac97_dai = { }; EXPORT_SYMBOL_GPL(bfin_ac97_dai); -static int __devinit bfin_ac97_init(void) +static int __init bfin_ac97_init(void) { return snd_soc_register_dai(&bfin_ac97_dai); } diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index c58b12a4487..53d290b3ea4 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -283,7 +283,7 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = { }; EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform); -static int __devinit bfin_i2s_init(void) +static int __init bfin_i2s_init(void) { return snd_soc_register_platform(&bf5xx_i2s_soc_platform); } diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c index 0d58d2b6db6..c17b131f662 100644 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ b/sound/soc/blackfin/bf5xx-i2s.c @@ -313,7 +313,7 @@ struct snd_soc_dai bf5xx_i2s_dai = { }; EXPORT_SYMBOL_GPL(bf5xx_i2s_dai); -static int __devinit bfin_i2s_init(void) +static int __init bfin_i2s_init(void) { return snd_soc_register_dai(&bfin_i2s_dai); } diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index e32f55034e6..b09289a1e55 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -98,7 +98,7 @@ struct snd_soc_codec_device soc_codec_dev_ad73311 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311); -static int __devinit ad73311_init(void) +static int __init ad73311_init(void) { return snd_soc_register_dai(&ad73311_dai); } diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 94148fba911..81300d8d42c 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -688,7 +688,7 @@ struct snd_soc_codec_device soc_codec_dev_ak4535 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ak4535); -static int __devinit ak4535_modinit(void) +static int __init ak4535_modinit(void) { return snd_soc_register_dai(&ak4535_dai); } diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 73aaf249d78..f1aa0c34421 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -774,7 +774,7 @@ struct snd_soc_codec_device soc_codec_device_cs4270 = { }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -static int __devinit cs4270_init(void) +static int __init cs4270_init(void) { return snd_soc_register_dai(&cs4270_dai); } diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index f333e88ee25..9a3e67e5319 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -195,7 +195,7 @@ struct snd_soc_codec_device soc_codec_dev_pcm3008 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_pcm3008); -static int __devinit pcm3008_init(void) +static int __init pcm3008_init(void) { return snd_soc_register_dai(&pcm3008_dai); } diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 77fdcb4b9a1..2325aefea41 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -793,7 +793,7 @@ struct snd_soc_codec_device soc_codec_dev_ssm2602 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_ssm2602); -static int __devinit ssm2602_modinit(void) +static int __init ssm2602_modinit(void) { return snd_soc_register_dai(&ssm2602_dai); } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index eac449b92bd..39f5b981d25 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -847,7 +847,7 @@ struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23); -static int __devinit tlv320aic23_modinit(void) +static int __init tlv320aic23_modinit(void) { return snd_soc_register_dai(&tlv320aic23_dai); } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index ccd57596186..8da9e5d2e2f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1411,7 +1411,7 @@ struct snd_soc_codec_device soc_codec_dev_aic3x = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_aic3x); -static int __devinit aic3x_modinit(void) +static int __init aic3x_modinit(void) { return snd_soc_register_dai(&aic3x_dai); } diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 8e035b5d733..a2c5064a774 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -651,7 +651,7 @@ struct snd_soc_codec_device soc_codec_dev_uda134x = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_uda134x); -static int __devinit uda134x_init(void) +static int __init uda134x_init(void) { return snd_soc_register_dai(&uda134x_dai); } diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 55a99b6a68a..e6bf0844fbf 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -841,7 +841,7 @@ struct snd_soc_codec_device soc_codec_dev_uda1380 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_uda1380); -static int __devinit uda1380_modinit(void) +static int __init uda1380_modinit(void) { return snd_soc_register_dais(uda1380_dai, ARRAY_SIZE(uda1380_dai)); } diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index a2af04bb4e9..40f8238df71 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -889,7 +889,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8510 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8510); -static int __devinit wm8510_modinit(void) +static int __init wm8510_modinit(void) { return snd_soc_register_dai(&wm8510_dai); } diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 391ec2978ae..d004e584529 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1042,7 +1042,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8580 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580); -static int __devinit wm8580_modinit(void) +static int __init wm8580_modinit(void) { return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai)); } diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index d905e25b1a9..80b11983e13 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -568,7 +568,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8728 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8728); -static int __devinit wm8728_modinit(void) +static int __init wm8728_modinit(void) { return snd_soc_register_dai(&wm8728_dai); } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7b455a60d71..c444b9f2701 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -793,7 +793,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); -static int __devinit wm8731_modinit(void) +static int __init wm8731_modinit(void) { return snd_soc_register_dai(&wm8731_dai); } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 84a6307de90..5997fa68e0d 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -1085,7 +1085,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); -static int __devinit wm8750_modinit(void) +static int __init wm8750_modinit(void) { return snd_soc_register_dai(&wm8750_dai); } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 1caca30d081..6c21b50c937 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1874,7 +1874,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8753 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753); -static int __devinit wm8753_modinit(void) +static int __init wm8753_modinit(void) { return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai)); } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index 34e58af0c65..ebf58fba1be 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1473,7 +1473,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8900 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8900); -static int __devinit wm8900_modinit(void) +static int __init wm8900_modinit(void) { return i2c_add_driver(&wm8900_i2c_driver); } diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 5d8fe7e1571..0b5bea37e3d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1809,7 +1809,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8903 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903); -static int __devinit wm8903_modinit(void) +static int __init wm8903_modinit(void) { return snd_soc_register_dai(&wm8903_dai); } diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 2979fc4f44f..88ead7f8dd9 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -935,7 +935,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8971 = { EXPORT_SYMBOL_GPL(soc_codec_dev_wm8971); -static int __devinit wm8971_modinit(void) +static int __init wm8971_modinit(void) { return snd_soc_register_dai(&wm8971_dai); } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 53e71aafe6c..5b5afc14447 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1643,7 +1643,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8990 = { }; EXPORT_SYMBOL_GPL(soc_codec_dev_wm8990); -static int __devinit wm8990_modinit(void) +static int __init wm8990_modinit(void) { return snd_soc_register_dai(&wm8990_dai); } diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d89fc2f009a..81ff5c37ab5 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -481,7 +481,7 @@ struct snd_soc_dai davinci_i2s_dai = { }; EXPORT_SYMBOL_GPL(davinci_i2s_dai); -static int __devinit davinci_i2s_init(void) +static int __init davinci_i2s_init(void) { return snd_soc_register_dai(&davinci_i2s_dai); } diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index f1b6e02d24e..bc83e1cbd17 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -384,7 +384,7 @@ struct snd_soc_platform davinci_soc_platform = { }; EXPORT_SYMBOL_GPL(davinci_soc_platform); -static int __devinit davinci_soc_platform_init(void) +static int __init davinci_soc_platform_init(void) { return snd_soc_register_platform(&davinci_soc_platform); } diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 646c807163a..64993eda567 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -853,7 +853,7 @@ int fsl_dma_configure(struct fsl_dma_info *dma_info) } EXPORT_SYMBOL_GPL(fsl_dma_configure); -static int __devinit fsl_soc_platform_init(void) +static int __init fsl_soc_platform_init(void) { return snd_soc_register_platform(&fsl_soc_platform); } diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 41cab203416..39cc57ce4bf 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -499,7 +499,7 @@ struct snd_soc_dai omap_mcbsp_dai[] = { EXPORT_SYMBOL_GPL(omap_mcbsp_dai); -static int __devinit omap_mcbsp_init(void) +static int __init omap_mcbsp_init(void) { return snd_soc_register_dais(omap_mcbsp_dai, ARRAY_SIZE(omap_mcbsp_dai)); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 9940de29631..ce580a97cbd 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -354,7 +354,7 @@ struct snd_soc_platform omap_soc_platform = { }; EXPORT_SYMBOL_GPL(omap_soc_platform); -static int __devinit omap_soc_platform_init(void) +static int __init omap_soc_platform_init(void) { return snd_soc_register_platform(&omap_soc_platform); } diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 3587f2fae5f..73cb6b4c2f2 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -913,7 +913,7 @@ struct snd_soc_dai pxa_ssp_dai[] = { }; EXPORT_SYMBOL_GPL(pxa_ssp_dai); -static int __devinit pxa_ssp_init(void) +static int __init pxa_ssp_init(void) { return snd_soc_register_dais(pxa_ssp_dai, ARRAY_SIZE(pxa_ssp_dai)); } diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f6249d5b492..780db6757ad 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -228,7 +228,7 @@ struct snd_soc_dai pxa_ac97_dai[] = { EXPORT_SYMBOL_GPL(pxa_ac97_dai); EXPORT_SYMBOL_GPL(soc_ac97_ops); -static int __devinit pxa_ac97_init(void) +static int __init pxa_ac97_init(void) { return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai)); } diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 4fa1578f5d4..c670d08e7c9 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -118,7 +118,7 @@ struct snd_soc_platform pxa2xx_soc_platform = { }; EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); -static int __devinit pxa2xx_soc_platform_init(void) +static int __init pxa2xx_soc_platform_init(void) { return snd_soc_register_platform(&pxa2xx_soc_platform); } diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 2cf05079156..f3fc0aba0aa 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -736,7 +736,7 @@ struct snd_soc_dai s3c2412_i2s_dai = { }; EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); -static int __devinit s3c2412_i2s_init(void) +static int __init s3c2412_i2s_init(void) { return snd_soc_register_dai(&s3c2412_i2s_dai); } diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 7360c6d913b..723779a3058 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -396,7 +396,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = { EXPORT_SYMBOL_GPL(s3c2443_ac97_dai); EXPORT_SYMBOL_GPL(soc_ac97_ops); -static int __devinit s3c2443_ac97_init(void) +static int __init s3c2443_ac97_init(void) { return snd_soc_register_dai(&s3c2443_ac97_dai); } diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 897b1ac92ce..6f4d439b57a 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -482,7 +482,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = { }; EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai); -static int __devinit s3c24xx_i2s_init(void) +static int __init s3c24xx_i2s_init(void) { return snd_soc_register_dai(&s3c24xx_i2s_dai); } diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index ea5a9caec13..7c64d31d067 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -465,7 +465,7 @@ struct snd_soc_platform s3c24xx_soc_platform = { }; EXPORT_SYMBOL_GPL(s3c24xx_soc_platform); -static int __devinit s3c24xx_soc_platform_init(void) +static int __init s3c24xx_soc_platform_init(void) { return snd_soc_register_platform(&s3c24xx_soc_platform); } diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index 39ffca0933a..0dad3a0bb92 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -348,7 +348,7 @@ struct snd_soc_platform sh7760_soc_platform = { }; EXPORT_SYMBOL_GPL(sh7760_soc_platform); -static int __devinit sh7760_soc_platform_init(void) +static int __init sh7760_soc_platform_init(void) { return snd_soc_register_platform(&sh7760_soc_platform); } diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 9169bad1acf..eab31838bad 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -314,7 +314,7 @@ struct snd_soc_dai sh4_hac_dai[] = { }; EXPORT_SYMBOL_GPL(sh4_hac_dai); -static int __devinit sh4_hac_init(void) +static int __init sh4_hac_init(void) { return snd_soc_register_dais(sh4_hac_dai, ARRAY_SIZE(sh4_hac_dai)); } diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c index 9093588d4d0..d1e5390fdde 100644 --- a/sound/soc/sh/ssi.c +++ b/sound/soc/sh/ssi.c @@ -392,7 +392,7 @@ struct snd_soc_dai sh4_ssi_dai[] = { }; EXPORT_SYMBOL_GPL(sh4_ssi_dai); -static int __devinit sh4_ssi_init(void) +static int __init sh4_ssi_init(void) { return snd_soc_register_dais(sh4_ssi_dai, ARRAY_SIZE(sh4_ssi_dai)); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 76a89eb65ba..4d2db7cfaf4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2224,7 +2224,7 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform) } EXPORT_SYMBOL_GPL(snd_soc_unregister_platform); -static int __devinit snd_soc_init(void) +static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS debugfs_root = debugfs_create_dir("asoc", NULL); -- cgit v1.2.3-70-g09d2 From 1e297a19252a6792c4479b300020f7f63eeb56ef Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Dec 2008 11:08:33 +0000 Subject: ASoC: Work around warnings from some build environments BUG() should be marked as not returning but for at least some configurations (including some widely deployed compilers) that's either not happening or being forgotten by the compiler. Add some extra return statements to the affected paths. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0b5bea37e3d..b1f5cf77a87 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -392,6 +392,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, break; default: BUG(); + return -EINVAL; /* Spurious warning from some compilers */ } switch (w->shift) { @@ -403,6 +404,7 @@ static int wm8903_output_event(struct snd_soc_dapm_widget *w, break; default: BUG(); + return -EINVAL; /* Spurious warning from some compilers */ } if (event & SND_SOC_DAPM_PRE_PMU) { -- cgit v1.2.3-70-g09d2 From 6a1bee4a9cae13aa73abd8f724bada213a38eb63 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 10 Dec 2008 12:51:46 +0200 Subject: ASoC: TWL4030: Add missing Carkit output SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 71ab5342bea..13d5a12ac29 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -761,6 +761,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("PREDRIVER"), SND_SOC_DAPM_OUTPUT("HSOL"), SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("CARKITL"), + SND_SOC_DAPM_OUTPUT("CARKITR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), -- cgit v1.2.3-70-g09d2 From d4a73131a56e906b8f65e20934516adcad68b524 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 10 Dec 2008 12:51:47 +0200 Subject: ASoC: TWL4030: Small cleanup The mux switch related texts fits to on line, no need to wrap them. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 13d5a12ac29..b321928152d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,8 +192,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* Earpiece */ static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", - "DACR1"}; + {"Off", "DACL1", "DACL2", "Invalid", "DACR1"}; static const struct soc_enum twl4030_earpiece_enum = SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, @@ -205,8 +204,7 @@ SOC_DAPM_ENUM("Route", twl4030_earpiece_enum); /* PreDrive Left */ static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", - "DACR2"}; + {"Off", "DACL1", "DACL2", "Invalid", "DACR2"}; static const struct soc_enum twl4030_predrivel_enum = SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, @@ -218,8 +216,7 @@ SOC_DAPM_ENUM("Route", twl4030_predrivel_enum); /* PreDrive Right */ static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "Invalid", - "DACL2"}; + {"Off", "DACR1", "DACR2", "Invalid", "DACL2"}; static const struct soc_enum twl4030_predriver_enum = SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, -- cgit v1.2.3-70-g09d2 From 1e5fa31f96d558e53fe80e943305104bf4339711 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 10 Dec 2008 12:51:48 +0200 Subject: ASoC: TWL4030: Change the name for the DACs To avoid confusion the names for the DACs changed: DACL1 -> DAC Left1 ... Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index b321928152d..13773028f6f 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -764,13 +764,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), /* DACs */ - SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", + SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", TWL4030_REG_AVDAC_CTL, 0, 0), - SND_SOC_DAPM_DAC("DACL1", "Left Front Playback", + SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", TWL4030_REG_AVDAC_CTL, 1, 0), - SND_SOC_DAPM_DAC("DACR2", "Right Rear Playback", + SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", TWL4030_REG_AVDAC_CTL, 2, 0), - SND_SOC_DAPM_DAC("DACL2", "Left Rear Playback", + SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", TWL4030_REG_AVDAC_CTL, 3, 0), /* Analog PGAs */ @@ -816,10 +816,10 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { - {"ARXL1_APGA", NULL, "DACL1"}, - {"ARXR1_APGA", NULL, "DACR1"}, - {"ARXL2_APGA", NULL, "DACL2"}, - {"ARXR2_APGA", NULL, "DACR2"}, + {"ARXL1_APGA", NULL, "DAC Left1"}, + {"ARXR1_APGA", NULL, "DAC Right1"}, + {"ARXL2_APGA", NULL, "DAC Left2"}, + {"ARXR2_APGA", NULL, "DAC Right2"}, /* Internal playback routings */ /* Earpiece */ -- cgit v1.2.3-70-g09d2 From 78e19a39d3985e2a06354493a70a200c0d432de5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Dec 2008 15:38:36 +0000 Subject: ASoC: Convert WM8900 to do more work at I2C probe time Redo the instantiation of the WM8900 to do most of the initialisation work when the I2C driver probes rather than when the ASoC device is instantiated, registering the codec with the ASoC core when done. Also move all dynamic allocations into a single kmalloc() to simplify error handling and rename the I2C driver to make output more sensible. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8900.c | 159 +++++++++++++++++++++++----------------------- sound/soc/codecs/wm8900.h | 7 -- 2 files changed, 81 insertions(+), 85 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index ebf58fba1be..6767de10ded 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -138,6 +138,10 @@ struct snd_soc_codec_device soc_codec_dev_wm8900; struct wm8900_priv { + struct snd_soc_codec codec; + + u16 reg_cache[WM8900_MAXREG]; + u32 fll_in; /* FLL input frequency */ u32 fll_out; /* FLL output frequency */ }; @@ -1282,16 +1286,28 @@ static int wm8900_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8900 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8900_init(struct snd_soc_device *socdev) +static struct snd_soc_codec *wm8900_codec; + +static int wm8900_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { - struct snd_soc_codec *codec = socdev->codec; - int ret = 0; + struct wm8900_priv *wm8900; + struct snd_soc_codec *codec; unsigned int reg; - struct i2c_client *i2c_client = socdev->codec->control_data; + int ret; + + wm8900 = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); + if (wm8900 == NULL) + return -ENOMEM; + + codec = &wm8900->codec; + codec->private_data = wm8900; + codec->reg_cache = &wm8900->reg_cache[0]; + codec->reg_cache_size = WM8900_MAXREG; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); codec->name = "WM8900"; codec->owner = THIS_MODULE; @@ -1299,33 +1315,28 @@ static int wm8900_init(struct snd_soc_device *socdev) codec->write = wm8900_write; codec->dai = &wm8900_dai; codec->num_dai = 1; - codec->reg_cache_size = WM8900_MAXREG; - codec->reg_cache = kmemdup(wm8900_reg_defaults, - sizeof(wm8900_reg_defaults), GFP_KERNEL); - - if (codec->reg_cache == NULL) - return -ENOMEM; + codec->hw_write = (hw_write_t)i2c_master_send; + codec->control_data = i2c; + codec->set_bias_level = wm8900_set_bias_level; + codec->dev = &i2c->dev; reg = wm8900_read(codec, WM8900_REG_ID); if (reg != 0x8900) { - dev_err(&i2c_client->dev, "Device is not a WM8900 - ID %x\n", - reg); - return -ENODEV; - } - - codec->private_data = kzalloc(sizeof(struct wm8900_priv), GFP_KERNEL); - if (codec->private_data == NULL) { - ret = -ENOMEM; - goto priv_err; + dev_err(&i2c->dev, "Device is not a WM8900 - ID %x\n", reg); + ret = -ENODEV; + goto err; } /* Read back from the chip */ reg = wm8900_chip_read(codec, WM8900_REG_POWER1); reg = (reg >> 12) & 0xf; - dev_info(&i2c_client->dev, "WM8900 revision %d\n", reg); + dev_info(&i2c->dev, "WM8900 revision %d\n", reg); wm8900_reset(codec); + /* Turn the chip on */ + wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Latch the volume update bits */ wm8900_write(codec, WM8900_REG_LINVOL, wm8900_read(codec, WM8900_REG_LINVOL) | 0x100); @@ -1351,52 +1362,43 @@ static int wm8900_init(struct snd_soc_device *socdev) /* Set the DAC and mixer output bias */ wm8900_write(codec, WM8900_REG_OUTBIASCTL, 0x81); - /* Register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - dev_err(&i2c_client->dev, "Failed to register new PCMs\n"); - goto pcm_err; - } + wm8900_dai.dev = &i2c->dev; - /* Turn the chip on */ - codec->bias_level = SND_SOC_BIAS_OFF; - wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8900_codec = codec; - wm8900_add_controls(codec); - wm8900_add_widgets(codec); + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + goto err; + } - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&i2c_client->dev, "Failed to register card\n"); - goto card_err; + ret = snd_soc_register_dai(&wm8900_dai); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; } - return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); -priv_err: - kfree(codec->private_data); return ret; -} - -static struct i2c_client *wm8900_client; -static int wm8900_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - wm8900_client = i2c; - wm8900_dai.dev = &i2c->dev; - return snd_soc_register_dai(&wm8900_dai); +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8900); + wm8900_codec = NULL; + return ret; } static int wm8900_i2c_remove(struct i2c_client *client) { snd_soc_unregister_dai(&wm8900_dai); + snd_soc_unregister_codec(wm8900_codec); + + wm8900_set_bias_level(wm8900_codec, SND_SOC_BIAS_OFF); + wm8900_dai.dev = NULL; - wm8900_client = NULL; + kfree(wm8900_codec->private_data); + wm8900_codec = NULL; + return 0; } @@ -1408,7 +1410,7 @@ MODULE_DEVICE_TABLE(i2c, wm8900_i2c_id); static struct i2c_driver wm8900_i2c_driver = { .driver = { - .name = "WM8900 I2C codec", + .name = "WM8900", .owner = THIS_MODULE, }, .probe = wm8900_i2c_probe, @@ -1422,30 +1424,36 @@ static int wm8900_probe(struct platform_device *pdev) struct snd_soc_codec *codec; int ret = 0; - if (!wm8900_client) { + if (!wm8900_codec) { dev_err(&pdev->dev, "I2C client not yet instantiated\n"); return -ENODEV; } - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); - + codec = wm8900_codec; socdev->codec = codec; - codec->set_bias_level = wm8900_set_bias_level; + /* Register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register new PCMs\n"); + goto pcm_err; + } - codec->hw_write = (hw_write_t)i2c_master_send; - codec->control_data = wm8900_client; + wm8900_add_controls(codec); + wm8900_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register card\n"); + goto card_err; + } - ret = wm8900_init(socdev); - if (ret != 0) - kfree(codec); + return ret; +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: return ret; } @@ -1453,14 +1461,9 @@ static int wm8900_probe(struct platform_device *pdev) static int wm8900_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct snd_soc_codec *codec = socdev->codec; - - if (codec->control_data) - wm8900_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - kfree(codec); return 0; } diff --git a/sound/soc/codecs/wm8900.h b/sound/soc/codecs/wm8900.h index 2249a446ad3..fd15007d10c 100644 --- a/sound/soc/codecs/wm8900.h +++ b/sound/soc/codecs/wm8900.h @@ -52,13 +52,6 @@ #define WM8900_DAC_CLKDIV_5_5 0x14 #define WM8900_DAC_CLKDIV_6 0x18 -#define WM8900_ - -struct wm8900_setup_data { - int i2c_bus; - unsigned short i2c_address; -}; - extern struct snd_soc_dai wm8900_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8900; -- cgit v1.2.3-70-g09d2 From d58d5d5567ea9483346f57c83a94ce05992cd47c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Dec 2008 18:36:42 +0000 Subject: ASoC: Convert WM8903 driver to register at I2C probe time The driver now registers the codec and DAI when probed as an I2C device. Also convert the driver to use a single dynamic allocation to simplify error handling. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 230 +++++++++++++++++++--------------------------- sound/soc/codecs/wm8903.h | 5 - 2 files changed, 97 insertions(+), 138 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b1f5cf77a87..c80968fe326 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -33,19 +33,6 @@ #include "wm8903.h" -struct wm8903_priv { - int sysclk; - - /* Reference counts */ - int charge_pump_users; - int class_w_users; - int playback_active; - int capture_active; - - struct snd_pcm_substream *master_substream; - struct snd_pcm_substream *slave_substream; -}; - /* Register defaults at reset */ static u16 wm8903_reg_defaults[] = { 0x8903, /* R0 - SW Reset and ID */ @@ -223,6 +210,23 @@ static u16 wm8903_reg_defaults[] = { 0x0000, /* R172 - Analogue Output Bias 0 */ }; +struct wm8903_priv { + struct snd_soc_codec codec; + u16 reg_cache[ARRAY_SIZE(wm8903_reg_defaults)]; + + int sysclk; + + /* Reference counts */ + int charge_pump_users; + int class_w_users; + int playback_active; + int capture_active; + + struct snd_pcm_substream *master_substream; + struct snd_pcm_substream *slave_substream; +}; + + static unsigned int wm8903_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) { @@ -360,6 +364,8 @@ static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache) static void wm8903_reset(struct snd_soc_codec *codec) { wm8903_write(codec, WM8903_SW_RESET_AND_ID, 0); + memcpy(codec->reg_cache, wm8903_reg_defaults, + sizeof(wm8903_reg_defaults)); } #define WM8903_OUTPUT_SHORT 0x8 @@ -1563,39 +1569,48 @@ static int wm8903_resume(struct platform_device *pdev) return 0; } -/* - * initialise the WM8903 driver - * register the mixer and dsp interfaces with the kernel - */ -static int wm8903_init(struct snd_soc_device *socdev) +static struct snd_soc_codec *wm8903_codec; + +static int wm8903_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) { - struct snd_soc_codec *codec = socdev->codec; - struct i2c_client *i2c = codec->control_data; - int ret = 0; + struct wm8903_priv *wm8903; + struct snd_soc_codec *codec; + int ret; u16 val; - val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID); - if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) { - dev_err(&i2c->dev, - "Device with ID register %x is not a WM8903\n", val); - return -ENODEV; - } + wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); + if (wm8903 == NULL) + return -ENOMEM; + codec = &wm8903->codec; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->dev = &i2c->dev; codec->name = "WM8903"; codec->owner = THIS_MODULE; codec->read = wm8903_read; codec->write = wm8903_write; + codec->hw_write = (hw_write_t)i2c_master_send; codec->bias_level = SND_SOC_BIAS_OFF; codec->set_bias_level = wm8903_set_bias_level; codec->dai = &wm8903_dai; codec->num_dai = 1; - codec->reg_cache_size = ARRAY_SIZE(wm8903_reg_defaults); - codec->reg_cache = kmemdup(wm8903_reg_defaults, - sizeof(wm8903_reg_defaults), - GFP_KERNEL); - if (codec->reg_cache == NULL) { - dev_err(&i2c->dev, "Failed to allocate register cache\n"); - return -ENOMEM; + codec->reg_cache_size = ARRAY_SIZE(wm8903->reg_cache); + codec->reg_cache = &wm8903->reg_cache[0]; + codec->private_data = wm8903; + + i2c_set_clientdata(i2c, codec); + codec->control_data = i2c; + + val = wm8903_hw_read(codec, WM8903_SW_RESET_AND_ID); + if (val != wm8903_reg_defaults[WM8903_SW_RESET_AND_ID]) { + dev_err(&i2c->dev, + "Device with ID register %x is not a WM8903\n", val); + return -ENODEV; } val = wm8903_read(codec, WM8903_REVISION_NUMBER); @@ -1604,13 +1619,6 @@ static int wm8903_init(struct snd_soc_device *socdev) wm8903_reset(codec); - /* register pcms */ - ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if (ret < 0) { - dev_err(&i2c->dev, "failed to create pcms\n"); - goto pcm_err; - } - /* SYSCLK is required for pretty much anything */ wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA); @@ -1648,47 +1656,45 @@ static int wm8903_init(struct snd_soc_device *socdev) val |= WM8903_DAC_MUTEMODE; wm8903_write(codec, WM8903_DAC_DIGITAL_1, val); - wm8903_add_controls(codec); - wm8903_add_widgets(codec); - ret = snd_soc_init_card(socdev); - if (ret < 0) { - dev_err(&i2c->dev, "wm8903: failed to register card\n"); - goto card_err; + wm8903_dai.dev = &i2c->dev; + wm8903_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&wm8903_dai); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; } return ret; -card_err: - snd_soc_free_pcms(socdev); - snd_soc_dapm_free(socdev); -pcm_err: - kfree(codec->reg_cache); +err_codec: + snd_soc_unregister_codec(codec); +err: + wm8903_codec = NULL; + kfree(wm8903); return ret; } -static struct snd_soc_device *wm8903_socdev; - -static int wm8903_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) +static int wm8903_i2c_remove(struct i2c_client *client) { - struct snd_soc_device *socdev = wm8903_socdev; - struct snd_soc_codec *codec = socdev->codec; - int ret; + struct snd_soc_codec *codec = i2c_get_clientdata(client); - i2c_set_clientdata(i2c, codec); - codec->control_data = i2c; + snd_soc_unregister_dai(&wm8903_dai); + snd_soc_unregister_codec(codec); - ret = wm8903_init(socdev); - if (ret < 0) - dev_err(&i2c->dev, "Device initialisation failed\n"); + wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); - return ret; -} + kfree(codec->private_data); + + wm8903_codec = NULL; + wm8903_dai.dev = NULL; -static int wm8903_i2c_remove(struct i2c_client *client) -{ - struct snd_soc_codec *codec = i2c_get_clientdata(client); - kfree(codec->reg_cache); return 0; } @@ -1712,75 +1718,37 @@ static struct i2c_driver wm8903_i2c_driver = { static int wm8903_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); - struct wm8903_setup_data *setup; - struct snd_soc_codec *codec; - struct wm8903_priv *wm8903; - struct i2c_board_info board_info; - struct i2c_adapter *adapter; - struct i2c_client *i2c_client; int ret = 0; - setup = socdev->codec_data; - - if (!setup->i2c_address) { - dev_err(&pdev->dev, "No codec address provided\n"); - return -ENODEV; + if (!wm8903_codec) { + dev_err(&pdev->dev, "I2C device not yet probed\n"); + goto err; } - codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; + socdev->codec = wm8903_codec; - wm8903 = kzalloc(sizeof(struct wm8903_priv), GFP_KERNEL); - if (wm8903 == NULL) { - ret = -ENOMEM; - goto err_codec; + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + goto err; } - codec->private_data = wm8903; - socdev->codec = codec; - mutex_init(&codec->mutex); - INIT_LIST_HEAD(&codec->dapm_widgets); - INIT_LIST_HEAD(&codec->dapm_paths); + wm8903_add_controls(socdev->codec); + wm8903_add_widgets(socdev->codec); - wm8903_socdev = socdev; - - codec->hw_write = (hw_write_t)i2c_master_send; - ret = i2c_add_driver(&wm8903_i2c_driver); - if (ret != 0) { - dev_err(&pdev->dev, "can't add i2c driver\n"); - goto err_priv; - } else { - memset(&board_info, 0, sizeof(board_info)); - strlcpy(board_info.type, "wm8903", I2C_NAME_SIZE); - board_info.addr = setup->i2c_address; - - adapter = i2c_get_adapter(setup->i2c_bus); - if (!adapter) { - dev_err(&pdev->dev, "Can't get I2C bus %d\n", - setup->i2c_bus); - ret = -ENODEV; - goto err_adapter; - } - - i2c_client = i2c_new_device(adapter, &board_info); - i2c_put_adapter(adapter); - if (i2c_client == NULL) { - dev_err(&pdev->dev, - "I2C driver registration failed\n"); - ret = -ENODEV; - goto err_adapter; - } + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "wm8903: failed to register card\n"); + goto card_err; } return ret; -err_adapter: - i2c_del_driver(&wm8903_i2c_driver); -err_priv: - kfree(codec->private_data); -err_codec: - kfree(codec); +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +err: return ret; } @@ -1795,10 +1763,6 @@ static int wm8903_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); - i2c_unregister_device(socdev->codec->control_data); - i2c_del_driver(&wm8903_i2c_driver); - kfree(codec->private_data); - kfree(codec); return 0; } @@ -1813,13 +1777,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8903); static int __init wm8903_modinit(void) { - return snd_soc_register_dai(&wm8903_dai); + return i2c_add_driver(&wm8903_i2c_driver); } module_init(wm8903_modinit); static void __exit wm8903_exit(void) { - snd_soc_unregister_dai(&wm8903_dai); + i2c_del_driver(&wm8903_i2c_driver); } module_exit(wm8903_exit); diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index cec622f2f66..0ea27e2b996 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -18,11 +18,6 @@ extern struct snd_soc_dai wm8903_dai; extern struct snd_soc_codec_device soc_codec_dev_wm8903; -struct wm8903_setup_data { - int i2c_bus; - int i2c_address; -}; - #define WM8903_MCLK_DIV_2 1 #define WM8903_CLK_SYS 2 #define WM8903_BCLK 3 -- cgit v1.2.3-70-g09d2 From 3b1228abc93f7ab0aa28c46341d6a0f7e2cade70 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 10 Dec 2008 19:27:10 +0000 Subject: ASoC: Stop WM8903 SYSCLK when suspending This will save some additional power. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index c80968fe326..bde74546db4 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -997,6 +997,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->bias_level == SND_SOC_BIAS_OFF) { + wm8903_write(codec, WM8903_CLOCK_RATES_2, + WM8903_CLK_SYS_ENA); + wm8903_run_sequence(codec, 0); wm8903_sync_reg_cache(codec, codec->reg_cache); @@ -1027,6 +1030,9 @@ static int wm8903_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: wm8903_run_sequence(codec, 32); + reg = wm8903_read(codec, WM8903_CLOCK_RATES_2); + reg &= ~WM8903_CLK_SYS_ENA; + wm8903_write(codec, WM8903_CLOCK_RATES_2, reg); break; } @@ -1619,9 +1625,6 @@ static int wm8903_i2c_probe(struct i2c_client *i2c, wm8903_reset(codec); - /* SYSCLK is required for pretty much anything */ - wm8903_write(codec, WM8903_CLOCK_RATES_2, WM8903_CLK_SYS_ENA); - /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -- cgit v1.2.3-70-g09d2 From 49d92c7d5bbd158734bc34ed578a68b214a48583 Mon Sep 17 00:00:00 2001 From: "Stanley.Miao" Date: Thu, 11 Dec 2008 23:28:10 +0800 Subject: ASoC: TWL4030: hands-free start-up sequence. A special start-up sequence is required to reduce the pop-noise of Class D amplifier when enable hands-free on TWL4030. Signed-off-by: Stanley.Miao Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 6 ++++++ 2 files changed, 36 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 13773028f6f..51848880504 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -322,6 +322,30 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, return ret; } +static int handsfree_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; + unsigned char hs_ctl; + + hs_ctl = twl4030_read_reg_cache(w->codec, e->reg); + + if (hs_ctl & TWL4030_HF_CTL_REF_EN) { + hs_ctl |= TWL4030_HF_CTL_RAMP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_LOOP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_HB_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + } else { + hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN + | TWL4030_HF_CTL_HB_EN); + twl4030_write(w->codec, e->reg, hs_ctl); + } + + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -806,10 +830,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitr_control), /* HandsfreeL/R */ - SND_SOC_DAPM_MUX("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, - &twl4030_dapm_handsfreel_control), - SND_SOC_DAPM_MUX("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, - &twl4030_dapm_handsfreer_control), + SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, + &twl4030_dapm_handsfreel_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, + &twl4030_dapm_handsfreer_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index a2065d417c2..54615c76802 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -191,6 +191,12 @@ #define TWL4030_RAMP_DELAY_2581MS 0x1C #define TWL4030_RAMP_EN 0x02 +/* HFL_CTL (0x29, 0x2A) Fields */ +#define TWL4030_HF_CTL_HB_EN 0x04 +#define TWL4030_HF_CTL_LOOP_EN 0x08 +#define TWL4030_HF_CTL_RAMP_EN 0x10 +#define TWL4030_HF_CTL_REF_EN 0x20 + /* APLL_CTL (0x3A) Fields */ #define TWL4030_APLL_EN 0x10 -- cgit v1.2.3-70-g09d2 From 40aa4a30d0fd075fb934de4ee8163056827052ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 16 Dec 2008 10:15:12 +0000 Subject: ASoC: Add WM8350 AudioPlus codec driver The WM8350 is an integrated audio and power management subsystem which provides a single-chip solution for portable audio and multimedia systems. The integrated audio CODEC provides all the necessary functions for high-quality stereo recording and playback. Programmable on-chip amplifiers allow for the direct connection of headphones and microphones with a minimum of external components. A programmable low-noise bias voltage is available to feed one or more electret microphones. Additional audio features include programmable high-pass filter in the ADC input path. This driver was originally written by Liam Girdwood with further updates from me. Signed-off-by: Mark Brown --- include/linux/mfd/wm8350/audio.h | 38 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8350.c | 1583 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 20 + 5 files changed, 1643 insertions(+), 4 deletions(-) create mode 100644 sound/soc/codecs/wm8350.c create mode 100644 sound/soc/codecs/wm8350.h (limited to 'sound/soc/codecs') diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h index 217bb22ebb8..af95a1d2f3a 100644 --- a/include/linux/mfd/wm8350/audio.h +++ b/include/linux/mfd/wm8350/audio.h @@ -1,7 +1,7 @@ /* * audio.h -- Audio Driver for Wolfson WM8350 PMIC * - * Copyright 2007 Wolfson Microelectronics PLC + * Copyright 2007, 2008 Wolfson Microelectronics PLC * * 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 @@ -70,9 +70,9 @@ #define WM8350_CODEC_ISEL_0_5 3 /* x0.5 */ #define WM8350_VMID_OFF 0 -#define WM8350_VMID_500K 1 -#define WM8350_VMID_100K 2 -#define WM8350_VMID_10K 3 +#define WM8350_VMID_300K 1 +#define WM8350_VMID_50K 2 +#define WM8350_VMID_5K 3 /* * R40 (0x28) - Clock Control 1 @@ -591,8 +591,38 @@ #define WM8350_IRQ_CODEC_MICSCD 41 #define WM8350_IRQ_CODEC_MICD 42 +/* + * WM8350 Platform data. + * + * This must be initialised per platform for best audio performance. + * Please see WM8350 datasheet for information. + */ +struct wm8350_audio_platform_data { + int vmid_discharge_msecs; /* VMID --> OFF discharge time */ + int drain_msecs; /* OFF drain time */ + int cap_discharge_msecs; /* Cap ON (from OFF) discharge time */ + int vmid_charge_msecs; /* vmid power up time */ + u32 vmid_s_curve:2; /* vmid enable s curve speed */ + u32 dis_out4:2; /* out4 discharge speed */ + u32 dis_out3:2; /* out3 discharge speed */ + u32 dis_out2:2; /* out2 discharge speed */ + u32 dis_out1:2; /* out1 discharge speed */ + u32 vroi_out4:1; /* out4 tie off */ + u32 vroi_out3:1; /* out3 tie off */ + u32 vroi_out2:1; /* out2 tie off */ + u32 vroi_out1:1; /* out1 tie off */ + u32 vroi_enable:1; /* enable tie off */ + u32 codec_current_on:2; /* current level ON */ + u32 codec_current_standby:2; /* current level STANDBY */ + u32 codec_current_charge:2; /* codec current @ vmid charge */ +}; + +struct snd_soc_codec; + struct wm8350_codec { struct platform_device *pdev; + struct snd_soc_codec *codec; + struct wm8350_audio_platform_data *platform_data; }; #endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bf68052d692..c41289b5f58 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -13,6 +13,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C + select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8510 if (I2C || SPI_MASTER) select SND_SOC_WM8580 if I2C select SND_SOC_WM8728 if (I2C || SPI_MASTER) @@ -100,6 +101,9 @@ config SND_SOC_UDA134X config SND_SOC_UDA1380 tristate +config SND_SOC_WM8350 + tristate + config SND_SOC_WM8510 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9a20fddd09c..c4ddc9aa2bb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -12,6 +12,7 @@ snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-twl4030-objs := twl4030.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o +snd-soc-wm8350-objs := wm8350.o snd-soc-wm8510-objs := wm8510.o snd-soc-wm8580-objs := wm8580.o snd-soc-wm8728-objs := wm8728.o @@ -39,6 +40,7 @@ obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o +obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o obj-$(CONFIG_SND_SOC_WM8580) += snd-soc-wm8580.o obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c new file mode 100644 index 00000000000..4bbfb5a5894 --- /dev/null +++ b/sound/soc/codecs/wm8350.c @@ -0,0 +1,1583 @@ +/* + * wm8350.c -- WM8350 ALSA SoC audio driver + * + * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. + * + * Author: Liam Girdwood + * + * 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 "wm8350.h" + +#define WM8350_OUTn_0dB 0x39 + +#define WM8350_RAMP_NONE 0 +#define WM8350_RAMP_UP 1 +#define WM8350_RAMP_DOWN 2 + +/* We only include the analogue supplies here; the digital supplies + * need to be available well before this driver can be probed. + */ +static const char *supply_names[] = { + "AVDD", + "HPVDD", +}; + +struct wm8350_output { + u16 active; + u16 left_vol; + u16 right_vol; + u16 ramp; + u16 mute; +}; + +struct wm8350_data { + struct snd_soc_codec codec; + struct wm8350_output out1; + struct wm8350_output out2; + struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)]; +}; + +static unsigned int wm8350_codec_cache_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct wm8350 *wm8350 = codec->control_data; + return wm8350->reg_cache[reg]; +} + +static unsigned int wm8350_codec_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct wm8350 *wm8350 = codec->control_data; + return wm8350_reg_read(wm8350, reg); +} + +static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + struct wm8350 *wm8350 = codec->control_data; + return wm8350_reg_write(wm8350, reg, value); +} + +/* + * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int wm8350_out1_ramp_step(struct snd_soc_codec *codec) +{ + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out1 = &wm8350_data->out1; + struct wm8350 *wm8350 = codec->control_data; + int left_complete = 0, right_complete = 0; + u16 reg, val; + + /* left channel */ + reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME); + val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + + if (out1->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out1->left_vol) { + val++; + reg &= ~WM8350_OUT1L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else if (out1->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT1L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else + return 1; + + /* right channel */ + reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME); + val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + if (out1->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out1->right_vol) { + val++; + reg &= ~WM8350_OUT1R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } else if (out1->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT1R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } + + /* only hit the update bit if either volume has changed this step */ + if (!left_complete || !right_complete) + wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU); + + return left_complete & right_complete; +} + +/* + * Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown. + */ +static inline int wm8350_out2_ramp_step(struct snd_soc_codec *codec) +{ + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out2 = &wm8350_data->out2; + struct wm8350 *wm8350 = codec->control_data; + int left_complete = 0, right_complete = 0; + u16 reg, val; + + /* left channel */ + reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME); + val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + if (out2->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out2->left_vol) { + val++; + reg &= ~WM8350_OUT2L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else if (out2->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT2L_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, + reg | (val << WM8350_OUT1L_VOL_SHIFT)); + } else + left_complete = 1; + } else + return 1; + + /* right channel */ + reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME); + val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + if (out2->ramp == WM8350_RAMP_UP) { + /* ramp step up */ + if (val < out2->right_vol) { + val++; + reg &= ~WM8350_OUT2R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } else if (out2->ramp == WM8350_RAMP_DOWN) { + /* ramp step down */ + if (val > 0) { + val--; + reg &= ~WM8350_OUT2R_VOL_MASK; + wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, + reg | (val << WM8350_OUT1R_VOL_SHIFT)); + } else + right_complete = 1; + } + + /* only hit the update bit if either volume has changed this step */ + if (!left_complete || !right_complete) + wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU); + + return left_complete & right_complete; +} + +/* + * This work ramps both output PGAs at stream start/stop time to + * minimise pop associated with DAPM power switching. + * It's best to enable Zero Cross when ramping occurs to minimise any + * zipper noises. + */ +static void wm8350_pga_work(struct work_struct *work) +{ + struct snd_soc_codec *codec = + container_of(work, struct snd_soc_codec, delayed_work.work); + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out1 = &wm8350_data->out1, + *out2 = &wm8350_data->out2; + int i, out1_complete, out2_complete; + + /* do we need to ramp at all ? */ + if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE) + return; + + /* PGA volumes have 6 bits of resolution to ramp */ + for (i = 0; i <= 63; i++) { + out1_complete = 1, out2_complete = 1; + if (out1->ramp != WM8350_RAMP_NONE) + out1_complete = wm8350_out1_ramp_step(codec); + if (out2->ramp != WM8350_RAMP_NONE) + out2_complete = wm8350_out2_ramp_step(codec); + + /* ramp finished ? */ + if (out1_complete && out2_complete) + break; + + /* we need to delay longer on the up ramp */ + if (out1->ramp == WM8350_RAMP_UP || + out2->ramp == WM8350_RAMP_UP) { + /* delay is longer over 0dB as increases are larger */ + if (i >= WM8350_OUTn_0dB) + schedule_timeout_interruptible(msecs_to_jiffies + (2)); + else + schedule_timeout_interruptible(msecs_to_jiffies + (1)); + } else + udelay(50); /* doesn't matter if we delay longer */ + } + + out1->ramp = WM8350_RAMP_NONE; + out2->ramp = WM8350_RAMP_NONE; +} + +/* + * WM8350 Controls + */ + +static int pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8350_data *wm8350_data = codec->private_data; + struct wm8350_output *out; + + switch (w->shift) { + case 0: + case 1: + out = &wm8350_data->out1; + break; + case 2: + case 3: + out = &wm8350_data->out2; + break; + + default: + BUG(); + return -1; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + out->ramp = WM8350_RAMP_UP; + out->active = 1; + + if (!delayed_work_pending(&codec->delayed_work)) + schedule_delayed_work(&codec->delayed_work, + msecs_to_jiffies(1)); + break; + + case SND_SOC_DAPM_PRE_PMD: + out->ramp = WM8350_RAMP_DOWN; + out->active = 0; + + if (!delayed_work_pending(&codec->delayed_work)) + schedule_delayed_work(&codec->delayed_work, + msecs_to_jiffies(1)); + break; + } + + return 0; +} + +static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8350_data *wm8350_priv = codec->private_data; + struct wm8350_output *out = NULL; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int ret; + unsigned int reg = mc->reg; + u16 val; + + /* For OUT1 and OUT2 we shadow the values and only actually write + * them out when active in order to ensure the amplifier comes on + * as quietly as possible. */ + switch (reg) { + case WM8350_LOUT1_VOLUME: + out = &wm8350_priv->out1; + break; + case WM8350_LOUT2_VOLUME: + out = &wm8350_priv->out2; + break; + default: + break; + } + + if (out) { + out->left_vol = ucontrol->value.integer.value[0]; + out->right_vol = ucontrol->value.integer.value[1]; + if (!out->active) + return 1; + } + + ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + if (ret < 0) + return ret; + + /* now hit the volume update bits (always bit 8) */ + val = wm8350_codec_read(codec, reg); + wm8350_codec_write(codec, reg, val | WM8350_OUT1_VU); + return 1; +} + +static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm8350_data *wm8350_priv = codec->private_data; + struct wm8350_output *out1 = &wm8350_priv->out1; + struct wm8350_output *out2 = &wm8350_priv->out2; + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int reg = mc->reg; + + /* If these are cached registers use the cache */ + switch (reg) { + case WM8350_LOUT1_VOLUME: + ucontrol->value.integer.value[0] = out1->left_vol; + ucontrol->value.integer.value[1] = out1->right_vol; + return 0; + + case WM8350_LOUT2_VOLUME: + ucontrol->value.integer.value[0] = out2->left_vol; + ucontrol->value.integer.value[1] = out2->right_vol; + return 0; + + default: + break; + } + + return snd_soc_get_volsw_2r(kcontrol, ucontrol); +} + +/* double control with volume update */ +#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \ + xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_VOLATILE, \ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_2r, \ + .get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = reg_left, .rreg = reg_right, .shift = xshift, \ + .rshift = xshift, .max = xmax, .invert = xinvert}, } + +static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" }; +static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" }; +static const char *wm8350_dacmutem[] = { "Normal", "Soft" }; +static const char *wm8350_dacmutes[] = { "Fast", "Slow" }; +static const char *wm8350_dacfilter[] = { "Normal", "Sloping" }; +static const char *wm8350_adcfilter[] = { "None", "High Pass" }; +static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" }; +static const char *wm8350_lr[] = { "Left", "Right" }; + +static const struct soc_enum wm8350_enum[] = { + SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp), + SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol), + SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem), + SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes), + SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 12, 2, wm8350_dacfilter), + SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter), + SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp), + SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol), + SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr), +}; + +static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525); +static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600); +static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1); +static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1); +static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1); + +static const unsigned int capture_sd_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1), + 13, 15, TLV_DB_SCALE_ITEM(0, 0, 0), +}; + +static const struct snd_kcontrol_new wm8350_snd_controls[] = { + SOC_ENUM("Playback Deemphasis", wm8350_enum[0]), + SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]), + SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume", + WM8350_DAC_DIGITAL_VOLUME_L, + WM8350_DAC_DIGITAL_VOLUME_R, + 0, 255, 0, dac_pcm_tlv), + SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]), + SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]), + SOC_ENUM("Playback PCM Filter", wm8350_enum[4]), + SOC_ENUM("Capture PCM Filter", wm8350_enum[5]), + SOC_ENUM("Capture PCM HP Filter", wm8350_enum[6]), + SOC_ENUM("Capture ADC Inversion", wm8350_enum[7]), + SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume", + WM8350_ADC_DIGITAL_VOLUME_L, + WM8350_ADC_DIGITAL_VOLUME_R, + 0, 255, 0, adc_pcm_tlv), + SOC_DOUBLE_TLV("Capture Sidetone Volume", + WM8350_ADC_DIVIDER, + 8, 4, 15, 1, capture_sd_tlv), + SOC_WM8350_DOUBLE_R_TLV("Capture Volume", + WM8350_LEFT_INPUT_VOLUME, + WM8350_RIGHT_INPUT_VOLUME, + 2, 63, 0, pre_amp_tlv), + SOC_DOUBLE_R("Capture ZC Switch", + WM8350_LEFT_INPUT_VOLUME, + WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0), + SOC_SINGLE_TLV("Left Input Left Sidetone Volume", + WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Left Input Right Sidetone Volume", + WM8350_OUTPUT_LEFT_MIXER_VOLUME, + 5, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Left Input Bypass Volume", + WM8350_OUTPUT_LEFT_MIXER_VOLUME, + 9, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Right Input Left Sidetone Volume", + WM8350_OUTPUT_RIGHT_MIXER_VOLUME, + 1, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Right Input Right Sidetone Volume", + WM8350_OUTPUT_RIGHT_MIXER_VOLUME, + 5, 7, 0, out_mix_tlv), + SOC_SINGLE_TLV("Right Input Bypass Volume", + WM8350_OUTPUT_RIGHT_MIXER_VOLUME, + 13, 7, 0, out_mix_tlv), + SOC_SINGLE("Left Input Mixer +20dB Switch", + WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0), + SOC_SINGLE("Right Input Mixer +20dB Switch", + WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0), + SOC_SINGLE_TLV("Out4 Capture Volume", + WM8350_INPUT_MIXER_VOLUME, + 1, 7, 0, out_mix_tlv), + SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume", + WM8350_LOUT1_VOLUME, + WM8350_ROUT1_VOLUME, + 2, 63, 0, out_pga_tlv), + SOC_DOUBLE_R("Out1 Playback ZC Switch", + WM8350_LOUT1_VOLUME, + WM8350_ROUT1_VOLUME, 13, 1, 0), + SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume", + WM8350_LOUT2_VOLUME, + WM8350_ROUT2_VOLUME, + 2, 63, 0, out_pga_tlv), + SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME, + WM8350_ROUT2_VOLUME, 13, 1, 0), + SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0), + SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME, + 5, 7, 0, out_mix_tlv), + + SOC_DOUBLE_R("Out1 Playback Switch", + WM8350_LOUT1_VOLUME, + WM8350_ROUT1_VOLUME, + 14, 1, 1), + SOC_DOUBLE_R("Out2 Playback Switch", + WM8350_LOUT2_VOLUME, + WM8350_ROUT2_VOLUME, + 14, 1, 1), +}; + +/* + * DAPM Controls + */ + +/* Left Playback Mixer */ +static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", + WM8350_LEFT_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", + WM8350_LEFT_MIXER_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", + WM8350_LEFT_MIXER_CONTROL, 12, 1, 0), + SOC_DAPM_SINGLE("Left Sidetone Switch", + WM8350_LEFT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("Right Sidetone Switch", + WM8350_LEFT_MIXER_CONTROL, 1, 1, 0), +}; + +/* Right Playback Mixer */ +static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", + WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", + WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", + WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Left Sidetone Switch", + WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("Right Sidetone Switch", + WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0), +}; + +/* Out4 Mixer */ +static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = { + SOC_DAPM_SINGLE("Right Playback Switch", + WM8350_OUT4_MIXER_CONTROL, 12, 1, 0), + SOC_DAPM_SINGLE("Left Playback Switch", + WM8350_OUT4_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Right Capture Switch", + WM8350_OUT4_MIXER_CONTROL, 9, 1, 0), + SOC_DAPM_SINGLE("Out3 Playback Switch", + WM8350_OUT4_MIXER_CONTROL, 2, 1, 0), + SOC_DAPM_SINGLE("Right Mixer Switch", + WM8350_OUT4_MIXER_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("Left Mixer Switch", + WM8350_OUT4_MIXER_CONTROL, 0, 1, 0), +}; + +/* Out3 Mixer */ +static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", + WM8350_OUT3_MIXER_CONTROL, 11, 1, 0), + SOC_DAPM_SINGLE("Left Capture Switch", + WM8350_OUT3_MIXER_CONTROL, 8, 1, 0), + SOC_DAPM_SINGLE("Out4 Playback Switch", + WM8350_OUT3_MIXER_CONTROL, 3, 1, 0), + SOC_DAPM_SINGLE("Left Mixer Switch", + WM8350_OUT3_MIXER_CONTROL, 0, 1, 0), +}; + +/* Left Input Mixer */ +static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("L2 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE_TLV("L3 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE("PGA Capture Switch", + WM8350_LEFT_INPUT_VOLUME, 14, 1, 0), +}; + +/* Right Input Mixer */ +static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = { + SOC_DAPM_SINGLE_TLV("L2 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE_TLV("L3 Capture Volume", + WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv), + SOC_DAPM_SINGLE("PGA Capture Switch", + WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0), +}; + +/* Left Mic Mixer */ +static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = { + SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0), + SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0), + SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0), +}; + +/* Right Mic Mixer */ +static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = { + SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0), + SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0), + SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0), +}; + +/* Beep Switch */ +static const struct snd_kcontrol_new wm8350_beep_switch_controls = +SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1); + +/* Out4 Capture Mux */ +static const struct snd_kcontrol_new wm8350_out4_capture_controls = +SOC_DAPM_ENUM("Route", wm8350_enum[8]); + +static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = { + + SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL, + 0, pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL, + 0, pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0, + pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2, + 7, 0, &wm8350_right_capt_mixer_controls[0], + ARRAY_SIZE(wm8350_right_capt_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2, + 6, 0, &wm8350_left_capt_mixer_controls[0], + ARRAY_SIZE(wm8350_left_capt_mixer_controls)), + + SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0, + &wm8350_out4_mixer_controls[0], + ARRAY_SIZE(wm8350_out4_mixer_controls)), + + SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0, + &wm8350_out3_mixer_controls[0], + ARRAY_SIZE(wm8350_out3_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0, + &wm8350_right_play_mixer_controls[0], + ARRAY_SIZE(wm8350_right_play_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0, + &wm8350_left_play_mixer_controls[0], + ARRAY_SIZE(wm8350_left_play_mixer_controls)), + + SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0, + &wm8350_left_mic_mixer_controls[0], + ARRAY_SIZE(wm8350_left_mic_mixer_controls)), + + SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0, + &wm8350_right_mic_mixer_controls[0], + ARRAY_SIZE(wm8350_right_mic_mixer_controls)), + + /* virtual mixer for Beep and Out2R */ + SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0, + &wm8350_beep_switch_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", + WM8350_POWER_MGMT_4, 3, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", + WM8350_POWER_MGMT_4, 2, 0), + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", + WM8350_POWER_MGMT_4, 5, 0), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", + WM8350_POWER_MGMT_4, 4, 0), + + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0), + + SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0, + &wm8350_out4_capture_controls), + + SND_SOC_DAPM_OUTPUT("OUT1R"), + SND_SOC_DAPM_OUTPUT("OUT1L"), + SND_SOC_DAPM_OUTPUT("OUT2R"), + SND_SOC_DAPM_OUTPUT("OUT2L"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("OUT4"), + + SND_SOC_DAPM_INPUT("IN1RN"), + SND_SOC_DAPM_INPUT("IN1RP"), + SND_SOC_DAPM_INPUT("IN2R"), + SND_SOC_DAPM_INPUT("IN1LP"), + SND_SOC_DAPM_INPUT("IN1LN"), + SND_SOC_DAPM_INPUT("IN2L"), + SND_SOC_DAPM_INPUT("IN3R"), + SND_SOC_DAPM_INPUT("IN3L"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + /* left playback mixer */ + {"Left Playback Mixer", "Playback Switch", "Left DAC"}, + {"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"}, + {"Left Playback Mixer", "Right Playback Switch", "Right DAC"}, + {"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"}, + {"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"}, + + /* right playback mixer */ + {"Right Playback Mixer", "Playback Switch", "Right DAC"}, + {"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"}, + {"Right Playback Mixer", "Left Playback Switch", "Left DAC"}, + {"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"}, + {"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"}, + + /* out4 playback mixer */ + {"Out4 Mixer", "Right Playback Switch", "Right DAC"}, + {"Out4 Mixer", "Left Playback Switch", "Left DAC"}, + {"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"}, + {"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"}, + {"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"}, + {"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"}, + {"OUT4", NULL, "Out4 Mixer"}, + + /* out3 playback mixer */ + {"Out3 Mixer", "Left Playback Switch", "Left DAC"}, + {"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"}, + {"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"}, + {"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"}, + {"OUT3", NULL, "Out3 Mixer"}, + + /* out2 */ + {"Right Out2 PGA", NULL, "Right Playback Mixer"}, + {"Left Out2 PGA", NULL, "Left Playback Mixer"}, + {"OUT2L", NULL, "Left Out2 PGA"}, + {"OUT2R", NULL, "Right Out2 PGA"}, + + /* out1 */ + {"Right Out1 PGA", NULL, "Right Playback Mixer"}, + {"Left Out1 PGA", NULL, "Left Playback Mixer"}, + {"OUT1L", NULL, "Left Out1 PGA"}, + {"OUT1R", NULL, "Right Out1 PGA"}, + + /* ADCs */ + {"Left ADC", NULL, "Left Capture Mixer"}, + {"Right ADC", NULL, "Right Capture Mixer"}, + + /* Left capture mixer */ + {"Left Capture Mixer", "L2 Capture Volume", "IN2L"}, + {"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"}, + {"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"}, + {"Left Capture Mixer", NULL, "Out4 Capture Channel"}, + + /* Right capture mixer */ + {"Right Capture Mixer", "L2 Capture Volume", "IN2R"}, + {"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"}, + {"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"}, + {"Right Capture Mixer", NULL, "Out4 Capture Channel"}, + + /* L3 Inputs */ + {"IN3L PGA", NULL, "IN3L"}, + {"IN3R PGA", NULL, "IN3R"}, + + /* Left Mic mixer */ + {"Left Mic Mixer", "INN Capture Switch", "IN1LN"}, + {"Left Mic Mixer", "INP Capture Switch", "IN1LP"}, + {"Left Mic Mixer", "IN2 Capture Switch", "IN2L"}, + + /* Right Mic mixer */ + {"Right Mic Mixer", "INN Capture Switch", "IN1RN"}, + {"Right Mic Mixer", "INP Capture Switch", "IN1RP"}, + {"Right Mic Mixer", "IN2 Capture Switch", "IN2R"}, + + /* out 4 capture */ + {"Out4 Capture Channel", NULL, "Out4 Mixer"}, + + /* Beep */ + {"Beep", NULL, "IN3R PGA"}, +}; + +static int wm8350_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8350_snd_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + + return 0; +} + +static int wm8350_add_widgets(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, + wm8350_dapm_widgets, + ARRAY_SIZE(wm8350_dapm_widgets)); + if (ret != 0) { + dev_err(codec->dev, "dapm control register failed\n"); + return ret; + } + + /* set up audio paths */ + ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + if (ret != 0) { + dev_err(codec->dev, "DAPM route register failed\n"); + return ret; + } + + return snd_soc_dapm_new_widgets(codec); +} + +static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8350 *wm8350 = codec->control_data; + u16 fll_4; + + switch (clk_id) { + case WM8350_MCLK_SEL_MCLK: + wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1, + WM8350_MCLK_SEL); + break; + case WM8350_MCLK_SEL_PLL_MCLK: + case WM8350_MCLK_SEL_PLL_DAC: + case WM8350_MCLK_SEL_PLL_ADC: + case WM8350_MCLK_SEL_PLL_32K: + wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1, + WM8350_MCLK_SEL); + fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + ~WM8350_FLL_CLK_SRC_MASK; + wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, fll_4 | clk_id); + break; + } + + /* MCLK direction */ + if (dir == WM8350_MCLK_DIR_OUT) + wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2, + WM8350_MCLK_DIR); + else + wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2, + WM8350_MCLK_DIR); + + return 0; +} + +static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 val; + + switch (div_id) { + case WM8350_ADC_CLKDIV: + val = wm8350_codec_read(codec, WM8350_ADC_DIVIDER) & + ~WM8350_ADC_CLKDIV_MASK; + wm8350_codec_write(codec, WM8350_ADC_DIVIDER, val | div); + break; + case WM8350_DAC_CLKDIV: + val = wm8350_codec_read(codec, WM8350_DAC_CLOCK_CONTROL) & + ~WM8350_DAC_CLKDIV_MASK; + wm8350_codec_write(codec, WM8350_DAC_CLOCK_CONTROL, val | div); + break; + case WM8350_BCLK_CLKDIV: + val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + ~WM8350_BCLK_DIV_MASK; + wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + break; + case WM8350_OPCLK_CLKDIV: + val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + ~WM8350_OPCLK_DIV_MASK; + wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + break; + case WM8350_SYS_CLKDIV: + val = wm8350_codec_read(codec, WM8350_CLOCK_CONTROL_1) & + ~WM8350_MCLK_DIV_MASK; + wm8350_codec_write(codec, WM8350_CLOCK_CONTROL_1, val | div); + break; + case WM8350_DACLR_CLKDIV: + val = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + ~WM8350_DACLRC_RATE_MASK; + wm8350_codec_write(codec, WM8350_DAC_LR_RATE, val | div); + break; + case WM8350_ADCLR_CLKDIV: + val = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + ~WM8350_ADCLRC_RATE_MASK; + wm8350_codec_write(codec, WM8350_ADC_LR_RATE, val | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + ~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK); + u16 master = wm8350_codec_read(codec, WM8350_AI_DAC_CONTROL) & + ~WM8350_BCLK_MSTR; + u16 dac_lrc = wm8350_codec_read(codec, WM8350_DAC_LR_RATE) & + ~WM8350_DACLRC_ENA; + u16 adc_lrc = wm8350_codec_read(codec, WM8350_ADC_LR_RATE) & + ~WM8350_ADCLRC_ENA; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + master |= WM8350_BCLK_MSTR; + dac_lrc |= WM8350_DACLRC_ENA; + adc_lrc |= WM8350_ADCLRC_ENA; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x2 << 8; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x1 << 8; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x3 << 8; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x3 << 8; /* lg not sure which mode */ + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= WM8350_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= WM8350_AIF_LRCLK_INV; + break; + default: + return -EINVAL; + } + + wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + wm8350_codec_write(codec, WM8350_AI_DAC_CONTROL, master); + wm8350_codec_write(codec, WM8350_DAC_LR_RATE, dac_lrc); + wm8350_codec_write(codec, WM8350_ADC_LR_RATE, adc_lrc); + return 0; +} + +static int wm8350_pcm_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int master = wm8350_codec_cache_read(codec, WM8350_AI_DAC_CONTROL) & + WM8350_BCLK_MSTR; + int enabled = 0; + + /* Check that the DACs or ADCs are enabled since they are + * required for LRC in master mode. The DACs or ADCs need a + * valid audio path i.e. pin -> ADC or DAC -> pin before + * the LRC will be enabled in master mode. */ + if (!master && cmd != SNDRV_PCM_TRIGGER_START) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & + (WM8350_ADCR_ENA | WM8350_ADCL_ENA); + } else { + enabled = wm8350_codec_cache_read(codec, WM8350_POWER_MGMT_4) & + (WM8350_DACR_ENA | WM8350_DACL_ENA); + } + + if (!enabled) { + dev_err(codec->dev, + "%s: invalid audio path - no clocks available\n", + __func__); + return -EINVAL; + } + return 0; +} + +static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = wm8350_codec_read(codec, WM8350_AI_FORMATING) & + ~WM8350_AIF_WL_MASK; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x1 << 10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x2 << 10; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x3 << 10; + break; + } + + wm8350_codec_write(codec, WM8350_AI_FORMATING, iface); + return 0; +} + +static int wm8350_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8350 *wm8350 = codec->control_data; + + if (mute) + wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + else + wm8350_clear_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + return 0; +} + +/* FLL divisors */ +struct _fll_div { + int div; /* FLL_OUTDIV */ + int n; + int k; + int ratio; /* FLL_FRATIO */ +}; + +/* The size in bits of the fll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_FLL_SIZE ((1 << 16) * 10) + +static inline int fll_factors(struct _fll_div *fll_div, unsigned int input, + unsigned int output) +{ + u64 Kpart; + unsigned int t1, t2, K, Nmod; + + if (output >= 2815250 && output <= 3125000) + fll_div->div = 0x4; + else if (output >= 5625000 && output <= 6250000) + fll_div->div = 0x3; + else if (output >= 11250000 && output <= 12500000) + fll_div->div = 0x2; + else if (output >= 22500000 && output <= 25000000) + fll_div->div = 0x1; + else { + printk(KERN_ERR "wm8350: fll freq %d out of range\n", output); + return -EINVAL; + } + + if (input > 48000) + fll_div->ratio = 1; + else + fll_div->ratio = 8; + + t1 = output * (1 << (fll_div->div + 1)); + t2 = input * fll_div->ratio; + + fll_div->n = t1 / t2; + Nmod = t1 % t2; + + if (Nmod) { + Kpart = FIXED_FLL_SIZE * (long long)Nmod; + do_div(Kpart, t2); + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + fll_div->k = K; + } else + fll_div->k = 0; + + return 0; +} + +static int wm8350_set_fll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, + unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8350 *wm8350 = codec->control_data; + struct _fll_div fll_div; + int ret = 0; + u16 fll_1, fll_4; + + /* power down FLL - we need to do this for reconfiguration */ + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, + WM8350_FLL_ENA | WM8350_FLL_OSC_ENA); + + if (freq_out == 0 || freq_in == 0) + return ret; + + ret = fll_factors(&fll_div, freq_in, freq_out); + if (ret < 0) + return ret; + dev_dbg(wm8350->dev, + "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d", + freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div, + fll_div.ratio); + + /* set up N.K & dividers */ + fll_1 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_1) & + ~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_1, + fll_1 | (fll_div.div << 8) | 0x50); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_2, + (fll_div.ratio << 11) | (fll_div. + n & WM8350_FLL_N_MASK)); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_3, fll_div.k); + fll_4 = wm8350_codec_read(codec, WM8350_FLL_CONTROL_4) & + ~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF); + wm8350_codec_write(codec, WM8350_FLL_CONTROL_4, + fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) | + (fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0)); + + /* power FLL on */ + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA); + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA); + + return 0; +} + +static int wm8350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8350 *wm8350 = codec->control_data; + struct wm8350_data *priv = codec->private_data; + struct wm8350_audio_platform_data *platform = + wm8350->codec.platform_data; + u16 pm1; + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_VMID_50K | + platform->codec_current_on << 14); + break; + + case SND_SOC_BIAS_PREPARE: + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1); + pm1 &= ~WM8350_VMID_MASK; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_VMID_50K); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret != 0) + return ret; + + /* Enable the system clock */ + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, + WM8350_SYSCLK_ENA); + + /* mute DAC & outputs */ + wm8350_set_bits(wm8350, WM8350_DAC_MUTE, + WM8350_DAC_MUTE_ENA); + + /* discharge cap memory */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + platform->dis_out1 | + (platform->dis_out2 << 2) | + (platform->dis_out3 << 4) | + (platform->dis_out4 << 6)); + + /* wait for discharge */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform-> + cap_discharge_msecs)); + + /* enable antipop */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + (platform->vmid_s_curve << 8)); + + /* ramp up vmid */ + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + (platform-> + codec_current_charge << 14) | + WM8350_VMID_5K | WM8350_VMIDEN | + WM8350_VBUFEN); + + /* wait for vmid */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform-> + vmid_charge_msecs)); + + /* turn on vmid 300k */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK); + pm1 |= WM8350_VMID_300K | + (platform->codec_current_standby << 14); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1); + + + /* enable analogue bias */ + pm1 |= WM8350_BIASEN; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1); + + /* disable antipop */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0); + + } else { + /* turn on vmid 300k and reduce current */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_VMID_300K | + (platform-> + codec_current_standby << 14)); + + } + break; + + case SND_SOC_BIAS_OFF: + + /* mute DAC & enable outputs */ + wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA); + + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3, + WM8350_OUT1L_ENA | WM8350_OUT1R_ENA | + WM8350_OUT2L_ENA | WM8350_OUT2R_ENA); + + /* enable anti pop S curve */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + (platform->vmid_s_curve << 8)); + + /* turn off vmid */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~WM8350_VMIDEN; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1); + + /* wait */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform-> + vmid_discharge_msecs)); + + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, + (platform->vmid_s_curve << 8) | + platform->dis_out1 | + (platform->dis_out2 << 2) | + (platform->dis_out3 << 4) | + (platform->dis_out4 << 6)); + + /* turn off VBuf and drain */ + pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) & + ~(WM8350_VBUFEN | WM8350_VMID_MASK); + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, + pm1 | WM8350_OUTPUT_DRAIN_EN); + + /* wait */ + schedule_timeout_interruptible(msecs_to_jiffies + (platform->drain_msecs)); + + pm1 &= ~WM8350_BIASEN; + wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1); + + /* disable anti-pop */ + wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0); + + wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME, + WM8350_OUT1L_ENA); + wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME, + WM8350_OUT1R_ENA); + wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME, + WM8350_OUT2L_ENA); + wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME, + WM8350_OUT2R_ENA); + + /* disable clock gen */ + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, + WM8350_SYSCLK_ENA); + + regulator_bulk_disable(ARRAY_SIZE(priv->supplies), + priv->supplies); + break; + } + codec->bias_level = level; + return 0; +} + +static int wm8350_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8350_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + if (codec->suspend_bias_level == SND_SOC_BIAS_ON) + wm8350_set_bias_level(codec, SND_SOC_BIAS_ON); + + return 0; +} + +static struct snd_soc_codec *wm8350_codec; + +static int wm8350_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + struct wm8350 *wm8350; + struct wm8350_data *priv; + int ret; + struct wm8350_output *out1; + struct wm8350_output *out2; + + BUG_ON(!wm8350_codec); + + socdev->codec = wm8350_codec; + codec = socdev->codec; + wm8350 = codec->control_data; + priv = codec->private_data; + + /* Enable the codec */ + wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); + + /* Enable robust clocking mode in ADC */ + wm8350_codec_write(codec, WM8350_SECURITY, 0xa7); + wm8350_codec_write(codec, 0xde, 0x13); + wm8350_codec_write(codec, WM8350_SECURITY, 0); + + /* read OUT1 & OUT2 volumes */ + out1 = &priv->out1; + out2 = &priv->out2; + out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) & + WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) & + WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) & + WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT; + out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) & + WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT; + wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0); + wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0); + wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0); + wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0); + + /* Latch VU bits & mute */ + wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, + WM8350_OUT1_VU | WM8350_OUT1L_MUTE); + wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, + WM8350_OUT2_VU | WM8350_OUT2L_MUTE); + wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME, + WM8350_OUT1_VU | WM8350_OUT1R_MUTE); + wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, + WM8350_OUT2_VU | WM8350_OUT2R_MUTE); + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(&pdev->dev, "failed to create pcms\n"); + return ret; + } + + wm8350_add_controls(codec); + wm8350_add_widgets(codec); + + wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(&pdev->dev, "failed to register card\n"); + goto card_err; + } + + return 0; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + return ret; +} + +static int wm8350_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm8350 *wm8350 = codec->control_data; + int ret; + + /* cancel any work waiting to be queued. */ + ret = cancel_delayed_work(&codec->delayed_work); + + /* if there was any work waiting then we run it now and + * wait for its completion */ + if (ret) { + schedule_delayed_work(&codec->delayed_work, 0); + flush_scheduled_work(); + } + + wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF); + + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); + + return 0; +} + +#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000) + +#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +struct snd_soc_dai wm8350_dai = { + .name = "WM8350", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8350_RATES, + .formats = WM8350_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8350_RATES, + .formats = WM8350_FORMATS, + }, + .ops = { + .hw_params = wm8350_pcm_hw_params, + .digital_mute = wm8350_mute, + .trigger = wm8350_pcm_trigger, + .set_fmt = wm8350_set_dai_fmt, + .set_sysclk = wm8350_set_dai_sysclk, + .set_pll = wm8350_set_fll, + .set_clkdiv = wm8350_set_clkdiv, + }, +}; +EXPORT_SYMBOL_GPL(wm8350_dai); + +struct snd_soc_codec_device soc_codec_dev_wm8350 = { + .probe = wm8350_probe, + .remove = wm8350_remove, + .suspend = wm8350_suspend, + .resume = wm8350_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350); + +static int wm8350_codec_probe(struct platform_device *pdev) +{ + struct wm8350 *wm8350 = platform_get_drvdata(pdev); + struct wm8350_data *priv; + struct snd_soc_codec *codec; + int ret, i; + + if (wm8350->codec.platform_data == NULL) { + dev_err(&pdev->dev, "No audio platform data supplied\n"); + return -EINVAL; + } + + priv = kzalloc(sizeof(struct wm8350_data), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(supply_names); i++) + priv->supplies[i].supply = supply_names[i]; + + ret = regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies), + priv->supplies); + if (ret != 0) + goto err_priv; + + codec = &priv->codec; + wm8350->codec.codec = codec; + + wm8350_dai.dev = &pdev->dev; + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->dev = &pdev->dev; + codec->name = "WM8350"; + codec->owner = THIS_MODULE; + codec->read = wm8350_codec_read; + codec->write = wm8350_codec_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8350_set_bias_level; + codec->dai = &wm8350_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8350_MAX_REGISTER; + codec->private_data = priv; + codec->control_data = wm8350; + + /* Put the codec into reset if it wasn't already */ + wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); + + INIT_DELAYED_WORK(&codec->delayed_work, wm8350_pga_work); + ret = snd_soc_register_codec(codec); + if (ret != 0) + goto err_supply; + + wm8350_codec = codec; + + ret = snd_soc_register_dai(&wm8350_dai); + if (ret != 0) + goto err_codec; + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err_supply: + regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies); +err_priv: + kfree(priv); + wm8350_codec = NULL; + return ret; +} + +static int wm8350_codec_remove(struct platform_device *pdev) +{ + struct wm8350 *wm8350 = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = wm8350->codec.codec; + struct wm8350_data *priv = codec->private_data; + + snd_soc_unregister_dai(&wm8350_dai); + snd_soc_unregister_codec(codec); + regulator_bulk_free(ARRAY_SIZE(priv->supplies), priv->supplies); + kfree(priv); + wm8350_codec = NULL; + return 0; +} + +static struct platform_driver wm8350_codec_driver = { + .driver = { + .name = "wm8350-codec", + .owner = THIS_MODULE, + }, + .probe = wm8350_codec_probe, + .remove = __devexit_p(wm8350_codec_remove), +}; + +static __init int wm8350_init(void) +{ + return platform_driver_register(&wm8350_codec_driver); +} +module_init(wm8350_init); + +static __exit void wm8350_exit(void) +{ + platform_driver_unregister(&wm8350_codec_driver); +} +module_exit(wm8350_exit); + +MODULE_DESCRIPTION("ASoC WM8350 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm8350-codec"); diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h new file mode 100644 index 00000000000..cc2887aa6c3 --- /dev/null +++ b/sound/soc/codecs/wm8350.h @@ -0,0 +1,20 @@ +/* + * wm8350.h - WM8903 audio codec interface + * + * Copyright 2008 Wolfson Microelectronics PLC. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef _WM8350_H +#define _WM8350_H + +#include + +extern struct snd_soc_dai wm8350_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8350; + +#endif -- cgit v1.2.3-70-g09d2 From a24f4f682661b8069d374a9197bc491525a7c799 Mon Sep 17 00:00:00 2001 From: Troy Kisky Date: Fri, 19 Dec 2008 13:05:22 -0700 Subject: ALSA: ASoC: tlv320aic3x add dsp_a Add SND_SOC_DAIFMT_DSP_A mode option. Signed-off-by: Troy Kisky Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic3x.c | 4 ++++ sound/soc/codecs/tlv320aic3x.h | 2 ++ 2 files changed, 6 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 8da9e5d2e2f..b47a749c5ea 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -891,6 +891,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = codec->private_data; u8 iface_areg, iface_breg; + int delay = 0; iface_areg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f; iface_breg = aic3x_read_reg_cache(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f; @@ -916,6 +917,8 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, SND_SOC_DAIFMT_INV_MASK)) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; + case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF): + delay = 1; case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF): iface_breg |= (0x01 << 6); break; @@ -932,6 +935,7 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, /* set iface */ aic3x_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg); aic3x_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg); + aic3x_write(codec, AIC3X_ASD_INTF_CTRLC, delay); return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 73e35b6ec92..ac827e578c4 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -35,6 +35,8 @@ #define AIC3X_ASD_INTF_CTRLA 8 /* Audio serial data interface control register B */ #define AIC3X_ASD_INTF_CTRLB 9 +/* Audio serial data interface control register C */ +#define AIC3X_ASD_INTF_CTRLC 10 /* Audio overflow status and PLL R value programming register */ #define AIC3X_OVRF_STATUS_AND_PLLR_REG 11 /* Audio codec digital filter control register */ -- cgit v1.2.3-70-g09d2 From a31501d1041c9d0a6c3f520736ae2b2fa081493a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 20 Dec 2008 16:50:53 +0100 Subject: ALSA: ASoC - Add missing __devexit annotation to wm8350.c Added the missing __devexit annotation to wm8350_codec_remove(): sound/soc/codecs/wm8350.c:1546: warning: 'wm8350_codec_remove' defined but not used Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8350.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 4bbfb5a5894..e3989d406f5 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1542,7 +1542,7 @@ err_priv: return ret; } -static int wm8350_codec_remove(struct platform_device *pdev) +static int __devexit wm8350_codec_remove(struct platform_device *pdev) { struct wm8350 *wm8350 = platform_get_drvdata(pdev); struct snd_soc_codec *codec = wm8350->codec.codec; -- cgit v1.2.3-70-g09d2 From bd25867a6cbe7a00ef7dbe8d9ddebc91b00b9b3f Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 22 Dec 2008 10:21:36 +0200 Subject: ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers - OMAP McBSP DAI driver claims to support DSP_A format which has 1-bit data delay but configures link for 0-bit data delay which is in fact DSP_B - Fix this by changing format from DSP_A to DSP_B - Fix also TLV320AIC23 codec and OSK5912 machine drivers since the same error is populated also there Signed-off-by: Jarkko Nikula Acked-by: Arun KS Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/omap/omap-mcbsp.c | 4 ++-- sound/soc/omap/osk5912.c | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 39f5b981d25..cfdea007c4c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -541,7 +541,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: iface_reg |= TLV320AIC23_FOR_I2S; break; - case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; case SND_SOC_DAIFMT_RIGHT_J: diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7b86373007c..ec5e18a7875 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -270,7 +270,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->srgr2 |= FPER(wlen * 2 - 1); regs->srgr1 |= FWID(wlen - 1); break; - case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: regs->srgr2 |= FPER(wlen * channels - 1); regs->srgr1 |= FWID(wlen * channels - 2); break; @@ -309,7 +309,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; - case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c index 845bf41335b..cd41a948df7 100644 --- a/sound/soc/omap/osk5912.c +++ b/sound/soc/omap/osk5912.c @@ -61,7 +61,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set codec DAI configuration */ err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { @@ -71,7 +71,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream, /* Set cpu DAI configuration */ err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBM_CFM); if (err < 0) { -- cgit v1.2.3-70-g09d2 From c69134858722977a82f58cae88e7ffdb28e1e858 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 22 Dec 2008 10:57:33 +0200 Subject: ASoC: Fix DSP formats in SSM2602 audio codec Thanks to Troy Kisky for noticing. - DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2 - DSP_B has 0-bit data delay which corresponds to submode 1 - Currently driver sets them opposite so swap the submode setting Signed-off-by: Jarkko Nikula Cc: Cliff Cai Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 2325aefea41..cac37361676 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -454,10 +454,10 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; -- cgit v1.2.3-70-g09d2