December 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 10/16] ASoC: twl6040: Enable automatic power for phoenix 1.1
by Olaya, Margarita 11 Dec '10

11 Dec '10

Phoenix 1.1 supports automatic power on sequence, a verification is added to use it with new revision of the chip. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 13 +++++++++---- 1 files changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 27fc1b9..46e8e93 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1136,6 +1136,7 @@ static int twl6040_probe(struct snd_soc_codec *codec) struct twl6040_data *priv; int audpwron, naudint; int ret = 0; + u8 icrev; priv = kzalloc(sizeof(struct twl6040_data), GFP_KERNEL); if (priv == NULL) @@ -1144,13 +1145,17 @@ static int twl6040_probe(struct snd_soc_codec *codec) priv->codec = codec; - if (twl_codec) { + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &icrev, TWL6040_REG_ASICREV); + + if (twl_codec && (icrev > 0)) audpwron = twl_codec->audpwron_gpio; - naudint = twl_codec->naudint_irq; - } else { + else audpwron = -EINVAL; + + if (twl_codec) + naudint = twl_codec->naudint_irq; + else naudint = 0; - } priv->audpwron = audpwron; priv->naudint = naudint; -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 09/16] ASoC: twl6040: Fix analog Mic L & R mux controls
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Francois Mazard <f-mazard(a)ti.com> The mux control has 4 elements not 3 Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 9776715..27fc1b9 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -508,8 +508,8 @@ static const char *twl6040_amicr_texts[] = {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; static const struct soc_enum twl6040_enum[] = { - SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 3, twl6040_amicl_texts), - SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 3, twl6040_amicr_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts), }; static const char *twl6040_hs_texts[] = { -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 08/16] ASoC: twl6040: Support other sample rates.
by Olaya, Margarita 11 Dec '10

11 Dec '10

The twl6040 can support more sample rates other than 88.2 and 96k. Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 9 ++++++++- 1 files changed, 8 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a89e7b8..9776715 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -39,7 +39,7 @@ #include "twl6040.h" -#define TWL6040_RATES (SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) +#define TWL6040_RATES SNDRV_PCM_RATE_8000_96000 #define TWL6040_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) struct twl6040_jack_data { @@ -890,10 +890,17 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { + case 11250: + case 22500: + case 44100: case 88200: lppllctl |= TWL6040_LPLLFIN; priv->sysclk = 17640000; break; + case 8000: + case 16000: + case 32000: + case 48000: case 96000: lppllctl &= ~TWL6040_LPLLFIN; priv->sysclk = 19200000; -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 07/16] ASoC: twl6040: Fix PCM error handling ops
by Olaya, Margarita 11 Dec '10

11 Dec '10

This patch moves all the PCM error handling for clock config out of trigger() and startup() and into prepare(). Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 55 +++++++++++++++++-------------------------- 1 files changed, 22 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d15ea44..a89e7b8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -865,23 +865,6 @@ static int twl6040_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = rtd->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - if (!priv->sysclk) { - dev_err(codec->dev, - "no mclk configured, call set_sysclk() on init\n"); - return -EINVAL; - } - - /* - * capture is not supported at 17.64 MHz, - * it's reserved for headset low-power playback scenario - */ - if ((priv->sysclk == 17640000) && substream->stream) { - dev_err(codec->dev, - "capture mode is not supported at %dHz\n", - priv->sysclk); - return -EINVAL; - } - snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, priv->sysclk_constraints); @@ -925,31 +908,37 @@ static int twl6040_hw_params(struct snd_pcm_substream *substream, return 0; } -static int twl6040_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) +static int twl6040_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - /* - * low-power playback mode is restricted - * for headset path only - */ - if ((priv->sysclk == 17640000) && priv->non_lp) { + if (!priv->sysclk) { + dev_err(codec->dev, + "no mclk configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* + * capture is not supported at 17.64 MHz, + * it's reserved for headset low-power playback scenario + */ + if ((priv->sysclk == 17640000) && + substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + dev_err(codec->dev, + "capture mode is not supported at %dHz\n", + priv->sysclk); + return -EINVAL; + } + + if ((priv->sysclk == 17640000) && priv->non_lp) { dev_err(codec->dev, "some enabled paths aren't supported at %dHz\n", priv->sysclk); return -EPERM; - } - break; - default: - break; } - return 0; } @@ -1063,7 +1052,7 @@ static int twl6040_set_dai_sysclk(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6040_dai_ops = { .startup = twl6040_startup, .hw_params = twl6040_hw_params, - .trigger = twl6040_trigger, + .prepare = twl6040_prepare, .set_sysclk = twl6040_set_dai_sysclk, }; -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 06/16] ASoC: twl6040: Restore bias level at resume
by Olaya, Margarita 11 Dec '10

11 Dec '10

This patch restores the CODEC bias level at resume(). Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b713d13..d15ea44 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1125,6 +1125,7 @@ static int twl6040_suspend(struct snd_soc_codec *codec, pm_message_t state) static int twl6040_resume(struct snd_soc_codec *codec) { twl6040_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + twl6040_set_bias_level(codec, codec->dapm.suspend_bias_level); return 0; } -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 05/16] ASoC: twl6040: Support all DAIs
by Olaya, Margarita 11 Dec '10

11 Dec '10

This patch adds support for all the TWL6040 DAIs. - twl6040-ul - twl6040-dl1 - twl6040-dl2 - twl6040-vib Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 43 ++++++++++++++++++++++++++++++++++++------- 1 files changed, 36 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 27d6d53..b713d13 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1067,23 +1067,51 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { .set_sysclk = twl6040_set_dai_sysclk, }; -static struct snd_soc_dai_driver twl6040_dai = { - .name = "twl6040-hifi", +static struct snd_soc_dai_driver twl6040_dai[] = { +{ + .name = "twl6040-ul", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-dl1", .playback = { - .stream_name = "Playback", + .stream_name = "Headset Playback", .channels_min = 1, - .channels_max = 4, + .channels_max = 2, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, - .capture = { - .stream_name = "Capture", + .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-dl2", + .playback = { + .stream_name = "Handsfree Playback", .channels_min = 1, .channels_max = 2, .rates = TWL6040_RATES, .formats = TWL6040_FORMATS, }, .ops = &twl6040_dai_ops, +}, +{ + .name = "twl6040-vib", + .playback = { + .stream_name = "Vibra Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, }; #ifdef CONFIG_PM @@ -1235,7 +1263,8 @@ static struct snd_soc_codec_driver soc_codec_dev_twl6040 = { static int __devinit twl6040_codec_probe(struct platform_device *pdev) { return snd_soc_register_codec(&pdev->dev, - &soc_codec_dev_twl6040, &twl6040_dai, 1); + &soc_codec_dev_twl6040, twl6040_dai, + ARRAY_SIZE(twl6040_dai)); } static int __devexit twl6040_codec_remove(struct platform_device *pdev) -- 1.7.0.4

1 0

[alsa-devel] [PATCH v2 01/16] ASoC: twl6040: Update twl IO macro
by Olaya, Margarita 11 Dec '10

11 Dec '10

Update the codec to use the new twl core register macros Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/twl6040.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 5d7e2f7..d33d2b4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -210,7 +210,7 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &value, reg); + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &value, reg); twl6040_write_reg_cache(codec, reg, value); return value; @@ -226,7 +226,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); + return twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, value, reg); } static void twl6040_init_vio_regs(struct snd_soc_codec *codec) @@ -439,7 +439,7 @@ static irqreturn_t twl6040_naudint_handler(int irq, void *data) struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); u8 intid; - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); switch (intid) { case TWL6040_THINT: @@ -715,7 +715,7 @@ static int twl6040_power_up_completion(struct snd_soc_codec *codec, msecs_to_jiffies(48)); if (!time_left) { - twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &intid, + twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &intid, TWL6040_REG_INTID); if (!(intid & TWL6040_READYINT)) { dev_err(codec->dev, "timeout waiting for READYINT\n"); -- 1.7.0.4

1 0

[alsa-devel] [PATCH v4 3/3] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output in the sdp4430. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: David Anders <x0132446(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/sdp4430.c | 32 ++++++++++++++++++++++++++++++++ 1 files changed, 32 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..189e039 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -24,6 +24,7 @@ #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/jack.h> #include <asm/mach-types.h> #include <plat/hardware.h> @@ -65,6 +66,21 @@ static struct snd_soc_ops sdp4430_ops = { .hw_params = sdp4430_hw_params, }; +/* Headset jack */ +static struct snd_soc_jack hs_jack; + +/*Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headset Stereophone", + .mask = SND_JACK_HEADPHONE, + }, +}; + static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -160,6 +176,22 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); ret = snd_soc_dapm_sync(dapm); + if (ret) + return ret; + + /* Headset jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", + SND_JACK_HEADSET, &hs_jack); + if (ret) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + + if (machine_is_omap_4430sdp()) + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); + else + snd_soc_jack_report(&hs_jack, SND_JACK_HEADSET, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

1 0

[alsa-devel] [PATCHv4 1/3] ASoC: sdp4430: Enable FM stereo pins
by Olaya, Margarita 11 Dec '10

11 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Add FM stereo pins to the machine driver and add them as a dapm widget. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/omap/sdp4430.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index ebbd62f..0c37c51 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), + SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -123,6 +124,10 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Earphone speaker */ {"Earphone Spk", NULL, "EP"}, + + /* Aux/FM Stereo In: AFML, AFMR */ + {"AFML", NULL, "Aux/FM Stereo In"}, + {"AFMR", NULL, "Aux/FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) @@ -149,13 +154,11 @@ static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) /* SDP4430 connected pins */ snd_soc_dapm_enable_pin(dapm, "Ext Mic"); snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "AFML"); + snd_soc_dapm_enable_pin(dapm, "AFMR"); snd_soc_dapm_enable_pin(dapm, "Headset Mic"); snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); - /* TWL6040 not connected pins */ - snd_soc_dapm_nc_pin(dapm, "AFML"); - snd_soc_dapm_nc_pin(dapm, "AFMR"); - ret = snd_soc_dapm_sync(dapm); return ret; -- 1.7.0.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: Implement WM8903 oversampling rate controls
by Mark Brown 11 Dec '10

11 Dec '10

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 797992c..542c7c4 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -481,6 +481,16 @@ static const char *hpf_mode_text[] = { static const struct soc_enum hpf_mode = SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static const char *osr_text[] = { + "Low power", "High performance" +}; + +static const struct soc_enum adc_osr = + SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text); + +static const struct soc_enum dac_osr = + SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text); + static const char *drc_slope_text[] = { "1", "1/2", "1/4", "1/8", "1/16", "0" }; @@ -619,6 +629,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, 6, 1, 0), /* ADCs */ +SOC_ENUM("ADC OSR", adc_osr), SOC_SINGLE("HPF Switch", WM8903_ADC_DIGITAL_0, 4, 1, 0), SOC_ENUM("HPF Mode", hpf_mode), SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), @@ -650,6 +661,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8, 12, 0, digital_sidetone_tlv), /* DAC */ +SOC_ENUM("DAC OSR", dac_osr), SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT, WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv), SOC_ENUM("DAC Soft Mute Rate", soft_mute), -- 1.7.1

2 5