April 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Allow DAPM pin operations to match any context
by Mark Brown 08 Apr '11

08 Apr '11

The DAPM pin operations currently require that the specific DAPM context that the pin being operated in is contained in be specified. With multi component and especially with the addition of a per-card DAPM context this isn't ideal as it means that things like disabling unused pins on CODECs require looking up the CODEC DAPM context. Fix this by falling back to matching a widget in any context if there isn't a match in the current context. The code isn't ideal currently but will do the job. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 14dbba7..5fe750c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1477,6 +1477,19 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, } } + /* Try again in other contexts */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strcmp(w->name, pin)) { + dev_dbg(w->dapm->dev, "dapm: pin %s = %d\n", + pin, status); + w->connected = status; + /* Allow disabling of forced pins */ + if (status == 0) + w->force = 0; + return 0; + } + } + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } @@ -2332,6 +2345,17 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, } } + /* Try again with other contexts */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strcmp(w->name, pin)) { + dev_dbg(w->dapm->dev, + "dapm: force enable pin %s\n", pin); + w->connected = 1; + w->force = 1; + return 0; + } + } + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Force all DAPM contexts into the same bias state
by Mark Brown 08 Apr '11

08 Apr '11

Currently we allow all DAPM contexts to determine their own bias level. While this should in general work in most situations and will deliver the lowest possible power it causes problems for our integration with the card bias level as we're calling the card bias level functions for each DAPM context even though they're card wide but don't say which CODEC we're calling for. Mitigate against this by forcing everything to be in the same state. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b193a95..14dbba7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1101,6 +1101,15 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } + /* Force all contexts in the card to the same bias state */ + power = 0; + list_for_each_entry(d, &card->dapm_list, list) + if (d->dev_power) + power = 1; + list_for_each_entry(d, &card->dapm_list, list) + d->dev_power = power; + + /* Run all the bias changes in parallel */ list_for_each_entry(d, &dapm->card->dapm_list, list) async_schedule_domain(dapm_pre_sequence_async, d, -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Add data based control initialisation for CODECs and cards
by Mark Brown 08 Apr '11

08 Apr '11

Allow CODEC and card drivers to point to an array of controls from their driver structure rather than explicitly calling snd_soc_add_controls(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 7 ++++++- sound/soc/soc-core.c | 11 +++++++++++ 2 files changed, 17 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 2720a9f..435cb83 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -577,7 +577,9 @@ struct snd_soc_codec_driver { pm_message_t state); int (*resume)(struct snd_soc_codec *); - /* Default DAPM setup, added after probe() is run */ + /* Default control and setup, added after probe() is run */ + const struct snd_kcontrol_new *controls; + int num_controls; const struct snd_soc_dapm_widget *dapm_widgets; int num_dapm_widgets; const struct snd_soc_dapm_route *dapm_routes; @@ -747,6 +749,9 @@ struct snd_soc_card { struct snd_soc_pcm_runtime *rtd_aux; int num_aux_rtd; + const struct snd_kcontrol_new *controls; + int num_controls; + /* * Card-specific routes and widgets. */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0cf6997..d95ff2a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1491,6 +1491,9 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + if (driver->controls) + snd_soc_add_controls(codec, driver->controls, + driver->num_controls); if (driver->dapm_widgets) snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); @@ -1888,6 +1891,14 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) } } + /* We should have a non-codec control add function but we don't */ + if (card->controls) + snd_soc_add_controls(list_first_entry(&card->codec_dev_list, + struct snd_soc_codec, + card_list), + card->controls, + card->num_controls); + if (card->dapm_widgets) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Allow DAPM pin operations to match any context
by Mark Brown 08 Apr '11

08 Apr '11

The DAPM pin operations currently require that the specific DAPM context that the pin being operated in is contained in be specified. With multi component and especially with the addition of a per-card DAPM context this isn't ideal as it means that things like disabling unused pins on CODECs require looking up the CODEC DAPM context. Fix this by falling back to matching a widget in any context if there isn't a match in the current context. The code isn't ideal currently but will do the job. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 24 ++++++++++++++++++++++++ 1 files changed, 24 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 790ffa9..f651a71 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1478,6 +1478,19 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, } } + /* Try again in other contexts */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strcmp(w->name, pin)) { + dev_dbg(w->dapm->dev, "dapm: pin %s = %d\n", + pin, status); + w->connected = status; + /* Allow disabling of forced pins */ + if (status == 0) + w->force = 0; + return 0; + } + } + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } @@ -2333,6 +2346,17 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, } } + /* Try again with other contexts */ + list_for_each_entry(w, &dapm->card->widgets, list) { + if (!strcmp(w->name, pin)) { + dev_dbg(w->dapm->dev, + "dapm: force enable pin %s\n", pin); + w->connected = 1; + w->force = 1; + return 0; + } + } + dev_err(dapm->dev, "dapm: unknown pin %s\n", pin); return -EINVAL; } -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: Set left channel volume update bits for WM8994
by Mark Brown 08 Apr '11

08 Apr '11

Ensures that we apply volume updates that don't affect the right channel. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 16 ++++++++++++++++ 1 files changed, 16 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 03ef624..b6d47e7 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2977,20 +2977,36 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Latch volume updates (right only; we always do left then right). */ + snd_soc_update_bits(codec, WM8994_AIF1_DAC1_LEFT_VOLUME, + WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU, WM8994_AIF1DAC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_DAC2_LEFT_VOLUME, + WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU, WM8994_AIF1DAC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_DAC_LEFT_VOLUME, + WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); snd_soc_update_bits(codec, WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU, WM8994_AIF2DAC_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC1_LEFT_VOLUME, + WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU, WM8994_AIF1ADC1_VU); + snd_soc_update_bits(codec, WM8994_AIF1_ADC2_LEFT_VOLUME, + WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_AIF2_ADC_LEFT_VOLUME, + WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); snd_soc_update_bits(codec, WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF2ADC_VU, WM8994_AIF1ADC2_VU); + snd_soc_update_bits(codec, WM8994_DAC1_LEFT_VOLUME, + WM8994_DAC1_VU, WM8994_DAC1_VU); snd_soc_update_bits(codec, WM8994_DAC1_RIGHT_VOLUME, WM8994_DAC1_VU, WM8994_DAC1_VU); + snd_soc_update_bits(codec, WM8994_DAC2_LEFT_VOLUME, + WM8994_DAC2_VU, WM8994_DAC2_VU); snd_soc_update_bits(codec, WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU, WM8994_DAC2_VU); -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: SSM2602: add SPI support
by Mike Frysinger 08 Apr '11

08 Apr '11

The ssm2602 codec has a SPI interface as well as I2C, so add the simple bit of glue to make it usable. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/ssm2602.c | 56 ++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 53 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e24d075..910bb66 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 select SND_SOC_SPDIF - select SND_SOC_SSM2602 if I2C + select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_SSM2604 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 6b12cff..86fb64c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -32,6 +32,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/spi/spi.h> #include <linux/platform_device.h> #include <linux/slab.h> #include <sound/core.h> @@ -504,6 +505,43 @@ static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { .reg_cache_default = ssm2602_reg, }; +#if defined(CONFIG_SPI_MASTER) +static int __devinit ssm2602_spi_probe(struct spi_device *spi) +{ + struct ssm2602_priv *ssm2602; + int ret; + + ssm2602 = kzalloc(sizeof(struct ssm2602_priv), GFP_KERNEL); + if (ssm2602 == NULL) + return -ENOMEM; + + spi_set_drvdata(spi, ssm2602); + ssm2602->control_type = SND_SOC_SPI; + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_ssm2602, &ssm2602_dai, 1); + if (ret < 0) + kfree(ssm2602); + return ret; +} + +static int __devexit ssm2602_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + kfree(spi_get_drvdata(spi)); + return 0; +} + +static struct spi_driver ssm2602_spi_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_spi_probe, + .remove = __devexit_p(ssm2602_spi_remove), +}; +#endif + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * ssm2602 2 wire address is determined by GPIO5 @@ -560,19 +598,29 @@ static struct i2c_driver ssm2602_i2c_driver = { static int __init ssm2602_modinit(void) { int ret = 0; + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&ssm2602_spi_driver); + if (ret) + return ret; +#endif + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ssm2602_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register SSM2602 I2C driver: %d\n", - ret); - } + if (ret) + return ret; #endif + return ret; } module_init(ssm2602_modinit); static void __exit ssm2602_exit(void) { +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&ssm2602_spi_driver); +#endif + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&ssm2602_i2c_driver); #endif -- 1.7.4.1

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[alsa-devel] [PATCH V2 0/5] Adding ASoC drivers for SPEAr13XX platform
by Rajeev Kumar 08 Apr '11

08 Apr '11

This patchset contains five patches which add support for ASoC sound drivers on ST's SPEAr13XX platform. Details of the SPEAr13XX platforms can be seen here: http://www.st.com/internet/mcu/product/250658.jsp The ARCH and PLATFORM specific code for SPEAr13XX are already under review in Russell King's ARM mailing list. This patchset has been tested for linux-kernel-version 2.6.38 and on ARM platform. Please review the same and consider for mainline inclusion. changes in platform code since V1: 1. Spilt machine driver from platform driver. 2. Updated rate supported. 3. Moved clock specific part from "spear13xx_i2s_set_dai_sysclk". function to platform code. This function is removed now. 4. Function "spear13xx_i2s_startup" removed. 5. Function "spear13xx_i2s_shutdown" removed. Changes in codec code since V1: 1. Removed sta529 version number. 2. Removed "sta529_read_reg_cache". 3. Removed function "sta529_write_reg_cache". 4. Removed function "sta529_write". 5. Moved sta529.h file contents to sta529.c file. 6. Move clock specific part from "spear_sta_set_dai_sysclk". function to platform code. This function is removed now. 7. Renamed function names from spear_sta_* to sta_*. 8. Modified probe() function to avoid rewriting default chip values to hardware. Rajeev Kumar (5): sound: asoc: Adding support for STA529 Audio Codec sound: asoc: Adding support for SPEAr13XX ASoC platform driver sound: asoc: Adding support for SPEAr13XX ASoC machine driver sound: asoc: Adding Kconfig and Makefile to support SPEAr13XX ASoC driver sound: asoc: Adding support for SPEAr13XX in soc sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta529.c | 359 +++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 19 ++ sound/soc/spear/Makefile | 6 + sound/soc/spear/evb_sta529.c | 104 ++++++++ sound/soc/spear/spear13xx-i2s.c | 523 +++++++++++++++++++++++++++++++++++++++ sound/soc/spear/spear13xx-i2s.h | 19 ++ sound/soc/spear/spear13xx-pcm.c | 500 +++++++++++++++++++++++++++++++++++++ sound/soc/spear/spear13xx-pcm.h | 50 ++++ 12 files changed, 1589 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/evb_sta529.c create mode 100644 sound/soc/spear/spear13xx-i2s.c create mode 100644 sound/soc/spear/spear13xx-i2s.h create mode 100644 sound/soc/spear/spear13xx-pcm.c create mode 100644 sound/soc/spear/spear13xx-pcm.h

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[alsa-devel] [RESEND for-2.6.40 PATCH] sst: fix compile error
by Lu Guanqun 08 Apr '11

08 Apr '11

Add several include files to fix the below compile error. drivers/staging/intel_sst/intelmid.c: In function ‘snd_intelmad_sst_register’: drivers/staging/intel_sst/intelmid.c:805:2: error: ‘sst_drv_ctx’ undeclared (first use in this function) drivers/staging/intel_sst/intelmid.c:805:2: note: each undeclared identifier is reported only once for each function it appears in Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- drivers/staging/intel_sst/intelmid.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/drivers/staging/intel_sst/intelmid.c b/drivers/staging/intel_sst/intelmid.c index 1fb39d4..4e4e4a9 100644 --- a/drivers/staging/intel_sst/intelmid.c +++ b/drivers/staging/intel_sst/intelmid.c @@ -32,6 +32,7 @@ #include <linux/platform_device.h> #include <linux/interrupt.h> #include <linux/sched.h> +#include <linux/firmware.h> #include <sound/control.h> #include <asm/mrst.h> #include <sound/pcm.h> @@ -40,6 +41,8 @@ #include <sound/initval.h> #include "intel_sst.h" #include "intel_sst_ioctl.h" +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" #include "intelmid_snd_control.h" #include "intelmid.h"

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[alsa-devel] [PATCH] sst: fix compile error
by Lu Guanqun 08 Apr '11

08 Apr '11

Add several include files to fix the below compile error. drivers/staging/intel_sst/intelmid.c: In function ‘snd_intelmad_sst_register’: drivers/staging/intel_sst/intelmid.c:612:2: error: ‘sst_drv_ctx’ undeclared (first use in this function) drivers/staging/intel_sst/intelmid.c:612:2: note: each undeclared identifier is reported only once for each function it appears in Signed-off-by: Lu Guanqun <guanqun.lu(a)intel.com> --- drivers/staging/intel_sst/intelmid.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/drivers/staging/intel_sst/intelmid.c b/drivers/staging/intel_sst/intelmid.c index 32ded14..bbda30f 100644 --- a/drivers/staging/intel_sst/intelmid.c +++ b/drivers/staging/intel_sst/intelmid.c @@ -37,8 +37,11 @@ #include <sound/pcm_params.h> #include <sound/initval.h> #include <linux/gpio.h> +#include <linux/firmware.h> #include "intel_sst.h" #include "intel_sst_ioctl.h" +#include "intel_sst_fw_ipc.h" +#include "intel_sst_common.h" #include "intelmid_snd_control.h" #include "intelmid_adc_control.h" #include "intelmid.h"

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[alsa-devel] [PATCH] ASoC: WM8903: HP and Line out PGA/mixer DAPM fixes
by Stephen Warren 08 Apr '11

08 Apr '11

From: Dilan Lee <dilee(a)nvidia.com> Update the headphone and line out mixers and PGAs use the same logical set of register bits and sequencing as the speaker mixer/PGA. This allows ALSA controls for mute and volume on headphone and line out to operate correctly. Per conversation on alsa-devel, earlier datasheets indicated that the POWER_MANAGEMENT_* register bits 0 and 1 were aliases to ANALOG_* register bits 0 and 4, and hence only one copy of those bits was programmed. However, later datasheets corrected this. From: Dilan Lee <dilee(a)nvidia.com> [swarren: Applied same change to headphone widgets] Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- Runtime tested in chromeos-2.6.37 which uses ASoC for-2.6.39 backport. Compile tesed in ASoC for-2.6.40; needed to disable USB due to apparent existing build break there. sound/soc/codecs/wm8903.c | 38 +++++++++++++++++++++++--------------- 1 files changed, 23 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index a739e09..d53f206 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -247,8 +247,6 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re case WM8903_REVISION_NUMBER: case WM8903_INTERRUPT_STATUS_1: case WM8903_WRITE_SEQUENCER_4: - case WM8903_POWER_MANAGEMENT_3: - case WM8903_POWER_MANAGEMENT_2: case WM8903_DC_SERVO_READBACK_1: case WM8903_DC_SERVO_READBACK_2: case WM8903_DC_SERVO_READBACK_3: @@ -875,34 +873,40 @@ SND_SOC_DAPM_MIXER("Left Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 1, 0, SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), -SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, - 4, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0, +SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, + 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_POWER_MANAGEMENT_2, 0, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0, +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_POWER_MANAGEMENT_3, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_ENA", 1, WM8903_ANALOGUE_HP_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 2, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_ENA", 1, WM8903_ANALOGUE_HP_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0, +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 5, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 4, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0, NULL, 0), SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 2, WM8903_ANALOGUE_LINEOUT_0, 1, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_ENA", 1, WM8903_ANALOGUE_LINEOUT_0, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), @@ -1037,10 +1041,14 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" }, - { "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" }, - { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, - { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_ENA", NULL, "Left Headphone Output PGA" }, + { "HPR_ENA", NULL, "Right Headphone Output PGA" }, + { "HPL_ENA_DLY", NULL, "HPL_ENA" }, + { "HPR_ENA_DLY", NULL, "HPR_ENA" }, + { "LINEOUTL_ENA", NULL, "Left Line Output PGA" }, + { "LINEOUTR_ENA", NULL, "Right Line Output PGA" }, + { "LINEOUTL_ENA_DLY", NULL, "LINEOUTL_ENA" }, + { "LINEOUTR_ENA_DLY", NULL, "LINEOUTR_ENA" }, { "HPL_DCS", NULL, "DCS Master" }, { "HPR_DCS", NULL, "DCS Master" }, -- 1.7.1

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