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

[alsa-devel] [PATCH v2 2/3] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/sdp4430.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..ca7fb90 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) { @@ -160,7 +176,19 @@ 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); + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: Fix snd_soc_instantiate_card error path
by Axel Lin 07 Dec '10

07 Dec '10

>From 0443551476ee6fabe45a0fbc7460dca8e0f6a965 Mon Sep 17 00:00:00 2001 From: Axel Lin <axel.lin(a)gmail.com> Date: Tue, 7 Dec 2010 16:12:29 +0800 Subject: [PATCH] ASoC: Fix snd_soc_instantiate_card error path Properly free the resources in the case of snd_card_register failure and soc_register_ac97_dai_link failure. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- This patch is against linux-next tree. In the case of soc_register_ac97_dai_link failure, I think we need below fix for Linus's tree: Add while (--i >= 0) soc_unregister_ac97_dai_link(&card->rtd[i]); just before goto probe_dai_err; Should I also send a patch against Linus's tree? Regards, Axel sound/soc/soc-core.c | 14 ++++++++------ 1 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 392d336..b4c8c38 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1839,18 +1839,20 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) ret = snd_card_register(card->snd_card); if (ret < 0) { printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name); - goto probe_dai_err; + goto probe_aux_dev_err; } #ifdef CONFIG_SND_SOC_AC97_BUS /* register any AC97 codecs */ for (i = 0; i < card->num_rtd; i++) { - ret = soc_register_ac97_dai_link(&card->rtd[i]); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); - goto probe_dai_err; - } + ret = soc_register_ac97_dai_link(&card->rtd[i]); + if (ret < 0) { + printk(KERN_ERR "asoc: failed to register AC97 %s\n", card->name); + while (--i >= 0) + soc_unregister_ac97_dai_link(&card->rtd[i]); + goto probe_aux_dev_err; } + } #endif card->instantiated = 1; -- 1.7.2

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[alsa-devel] [RFC v2 0/3] ASoC: Option to reorder widget power sequence
by Peter Ujfalusi 07 Dec '10

07 Dec '10

Hello, Chasnges sicne v1: - Flag moved to per widget - Flag name changed to: route_change_before_powerup - Comments from Mark has been addressed in the soc-dapm.c file - New helper function to set the reorder power up sequence. - tlv320dac33 to be the first codec to use the reversed power up order Note: power down sequence remains the same for the reordered widgets as well. Intro from the v1: Quite some time ago I have faced similar issue with the sequence of the DAPM register writes [1] For that issue, I used a workaround within the twl4030 codec driver. Now I have again issue with the register write sequence. The dac33 codec emmits a pop noise, when user enable the analog bypass path. The route cause in this codec are the DAPM update power and register write sequence: 1. User enable the bypass 2. DAPM power up sequence takes place 2.1 codec DAPM widgets got powered on, codec is enabled 2.2 External speaker enabled 3. The bit enabling the bypass got changed. Step 3 cause a pop, which propagate to the speaker, since it is already enabled. The first patch implements a new flag for the core to reorder the sequence to: 1. User enable the bypass 2. The bit enabling the bypass got changed. 3. DAPM power up sequence takes place 3.1 codec DAPM widgets got powered on, codec is enabled 3.2 External speaker enabled The default sequence is kept, codec driver must ask to have the reordered sequence to avoid any side effect on other codecs. [1]: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-August/030024.html Peter --- Peter Ujfalusi (3): ASoC: core: DAPM widget flag for reversed power up sequence ASoC: core: Helper function to request reversed widget power up ASoC: tlv320dac33: Reverse the power up sequence for bypass switches include/sound/soc-dapm.h | 3 ++ sound/soc/codecs/tlv320dac33.c | 4 +++ sound/soc/soc-dapm.c | 54 ++++++++++++++++++++++++++++++++++++++- 3 files changed, 59 insertions(+), 2 deletions(-) -- 1.7.3.2

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[alsa-devel] snd_hda_intel error
by Jack Byer 07 Dec '10

07 Dec '10

I found this message in my system logs: Dec 04 07:25:52 [pulseaudio] alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 8. Dec 04 07:25:52 [pulseaudio] alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Dec 04 07:25:52 [pulseaudio] alsa-util.c: snd_pcm_dump(): Dec 04 07:25:52 [pulseaudio] alsa-util.c: Soft volume PCM Dec 04 07:25:52 [pulseaudio] alsa-util.c: Control: PCM Playback Volume Dec 04 07:25:52 [pulseaudio] alsa-util.c: min_dB: -51 Dec 04 07:25:52 [pulseaudio] alsa-util.c: max_dB: 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: resolution: 256 Dec 04 07:25:52 [pulseaudio] alsa-util.c: Its setup is: Dec 04 07:25:52 [pulseaudio] alsa-util.c: stream : CAPTURE Dec 04 07:25:52 [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED Dec 04 07:25:52 [pulseaudio] alsa-util.c: format : S16_LE Dec 04 07:25:52 [pulseaudio] alsa-util.c: subformat : STD Dec 04 07:25:52 [pulseaudio] alsa-util.c: channels : 2 Dec 04 07:25:52 [pulseaudio] alsa-util.c: rate : 44100 Dec 04 07:25:52 [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) Dec 04 07:25:52 [pulseaudio] alsa-util.c: msbits : 16 Dec 04 07:25:52 [pulseaudio] alsa-util.c: buffer_size : 16384 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_size : 8192 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_time : 185759 Dec 04 07:25:52 [pulseaudio] alsa-util.c: tstamp_mode : ENABLE Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_step : 1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: avail_min : 15502 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_event : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: start_threshold : -1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: stop_threshold : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_threshold: 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_size : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: boundary : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: Its setup is: Dec 04 07:25:52 [pulseaudio] alsa-util.c: stream : CAPTURE Dec 04 07:25:52 [pulseaudio] alsa-util.c: access : MMAP_INTERLEAVED Dec 04 07:25:52 [pulseaudio] alsa-util.c: format : S16_LE Dec 04 07:25:52 [pulseaudio] alsa-util.c: subformat : STD Dec 04 07:25:52 [pulseaudio] alsa-util.c: channels : 2 Dec 04 07:25:52 [pulseaudio] alsa-util.c: rate : 44100 Dec 04 07:25:52 [pulseaudio] alsa-util.c: exact rate : 44100 (44100/1) Dec 04 07:25:52 [pulseaudio] alsa-util.c: msbits : 16 Dec 04 07:25:52 [pulseaudio] alsa-util.c: buffer_size : 16384 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_size : 8192 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_time : 185759 Dec 04 07:25:52 [pulseaudio] alsa-util.c: tstamp_mode : ENABLE Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_step : 1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: avail_min : 15502 Dec 04 07:25:52 [pulseaudio] alsa-util.c: period_event : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: start_threshold : -1 Dec 04 07:25:52 [pulseaudio] alsa-util.c: stop_threshold : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_threshold: 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: silence_size : 0 Dec 04 07:25:52 [pulseaudio] alsa-util.c: boundary : 4611686018427387904 Dec 04 07:25:52 [pulseaudio] alsa-util.c: appl_ptr : 135040 Dec 04 07:25:52 [pulseaudio] alsa-util.c: hw_ptr : 135040 I am using the driver from the 2.6.31.1 kernel

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[alsa-devel] [PATCH] ASoC: mcbsp: Add McBSP support for OMAP4
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> This patch adds McBSP support for the OMAP4 CPU Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 38 +++++++++++++++++++++++++++++++++++--- sound/soc/omap/omap-mcbsp.h | 4 ++++ 2 files changed, 39 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 7e84f24..132062c 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -102,6 +102,17 @@ static const int omap24xx_dma_reqs[][2] = { static const int omap24xx_dma_reqs[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP4) +static const int omap44xx_dma_reqs[][2] = { + { OMAP44XX_DMA_MCBSP1_TX, OMAP44XX_DMA_MCBSP1_RX }, + { OMAP44XX_DMA_MCBSP2_TX, OMAP44XX_DMA_MCBSP2_RX }, + { OMAP44XX_DMA_MCBSP3_TX, OMAP44XX_DMA_MCBSP3_RX }, + { OMAP44XX_DMA_MCBSP4_TX, OMAP44XX_DMA_MCBSP4_RX }, +}; +#else +static const int omap44xx_dma_reqs[][2] = {}; +#endif + #if defined(CONFIG_ARCH_OMAP2420) static const unsigned long omap2420_mcbsp_port[][2] = { { OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1, @@ -147,6 +158,21 @@ static const unsigned long omap34xx_mcbsp_port[][2] = { static const unsigned long omap34xx_mcbsp_port[][2] = {}; #endif +#if defined(CONFIG_ARCH_OMAP4) +static const unsigned long omap44xx_mcbsp_port[][2] = { + { OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR }, + { OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR, + OMAP44XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR }, +}; +#else +static const unsigned long omap44xx_mcbsp_port[][2] = {}; +#endif + static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; @@ -224,7 +250,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream, * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words) * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words) */ - if (cpu_is_omap343x()) { + if (cpu_is_omap343x() || cpu_is_omap44xx()) { /* * Rule for the buffer size. We should not allow * smaller buffer than the FIFO size to avoid underruns @@ -332,6 +358,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } else if (cpu_is_omap343x()) { dma = omap24xx_dma_reqs[bus_id][substream->stream]; port = omap34xx_mcbsp_port[bus_id][substream->stream]; + } else if (cpu_is_omap44xx()) { + dma = omap44xx_dma_reqs[bus_id][substream->stream]; + port = omap44xx_mcbsp_port[bus_id][substream->stream]; } else { return -ENODEV; } @@ -498,11 +527,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->spcr2 |= XINTM(3) | FREE; regs->spcr1 |= RINTM(3); /* RFIG and XFIG are not defined in 34xx */ - if (!cpu_is_omap34xx()) { + if (!cpu_is_omap34xx() && !cpu_is_omap44xx()) { regs->rcr2 |= RFIG; regs->xcr2 |= XFIG; } - if (cpu_is_omap2430() || cpu_is_omap34xx()) { + if (cpu_is_omap2430() || cpu_is_omap34xx() || cpu_is_omap44xx()) { regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE; regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE; } @@ -623,6 +652,9 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, if (cpu_class_is_omap1()) { err = -EINVAL; break; + } else if (cpu_is_omap44xx()) { + regs->srgr2 |= CLKSM; + break; } err = omap2_mcbsp_set_clks_src(mcbsp_data->bus_id, MCBSP_CLKS_PRCM_SRC); diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h index ffdcc5a..110c106 100644 --- a/sound/soc/omap/omap-mcbsp.h +++ b/sound/soc/omap/omap-mcbsp.h @@ -50,6 +50,10 @@ enum omap_mcbsp_div { #undef NUM_LINKS #define NUM_LINKS 3 #endif +#if defined(CONFIG_ARCH_OMAP4) +#undef NUM_LINKS +#define NUM_LINKS 4 +#endif #if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP3) #undef NUM_LINKS #define NUM_LINKS 5 -- 1.7.0.4

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[alsa-devel] Bug in alsa drivers reported when launching pulseaudio
by kieran magee 07 Dec '10

07 Dec '10

Some strange anomalies were occuring in my audio systems, namely, volume fluctuations during playback. Attempted to access sound in preferences, had disabled pulseaudio a while back with no ill effects, (weeks back), I couldn't access the menu Item. I was instructed that pulseaudio not running may be the issue, so attempted to launch that and came upon this worrying error message, E: alsa-util.c: snd_pcm_avail_delay() returned strange values: delay 0 is less than avail 16. E: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. E: alsa-util.c: snd_pcm_dump(): E: alsa-util.c: Soft volume PCM E: alsa-util.c: Control: PCM Playback Volume E: alsa-util.c: min_dB: -51 E: alsa-util.c: max_dB: 0 E: alsa-util.c: resolution: 256 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat : STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 88192 E: alsa-util.c: period_size : 44096 E: alsa-util.c: period_time : 999909 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min : 87310 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 6205960286516543488 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 6205960286516543488 E: alsa-util.c: Slave: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 E: alsa-util.c: Its setup is: E: alsa-util.c: stream : CAPTURE E: alsa-util.c: access : MMAP_INTERLEAVED E: alsa-util.c: format : S16_LE E: alsa-util.c: subformat : STD E: alsa-util.c: channels : 2 E: alsa-util.c: rate : 44100 E: alsa-util.c: exact rate : 44100 (44100/1) E: alsa-util.c: msbits : 16 E: alsa-util.c: buffer_size : 88192 E: alsa-util.c: period_size : 44096 E: alsa-util.c: period_time : 999909 E: alsa-util.c: tstamp_mode : ENABLE E: alsa-util.c: period_step : 1 E: alsa-util.c: avail_min : 87310 E: alsa-util.c: period_event : 0 E: alsa-util.c: start_threshold : -1 E: alsa-util.c: stop_threshold : 6205960286516543488 E: alsa-util.c: silence_threshold: 0 E: alsa-util.c: silence_size : 0 E: alsa-util.c: boundary : 6205960286516543488 E: alsa-util.c: appl_ptr : 87328 E: alsa-util.c: hw_ptr : 87328 If you need any system information, please feel free to ask. I'm not subscribed to the mailing list, as a note, so responses need to be cc'd directly to me if I am to get them. Thanks for any help you can give me, Best, Kieran Brantner-Magee (the system is ubuntu 10.10 kernel 2.6.35-23 )

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[alsa-devel] [PATCH 00/15] twl6040 updates
by Olaya, Margarita 07 Dec '10

07 Dec '10

This patch series contains several fixes and adds new features for twl6040 (phoenix) codec. Francois Mazard (1): ASoC: twl6040: Fix analog Mic L & R mux controls Jorge Eduardo Candelaria (3): ASoC: twl6040: Add headset and handset mux controls ASoC: twl6040: Clear interrupt status at boot time ASoC: twl6040: Increase timeout for power up Margarita Olaya Cabrera (9): ASoC: twl6040: Update twl IO macro ASoC: twl6040: Modify the IRQ handler ASoC: twl6040: Support all DAIs ASoC: twl6040: Restore bias level at resume ASoC: twl6040: Fix PCM error handling ops ASoC: twl6040: Support other sample rates. ASoC: twl6040: Enable automatic power for phoenix 1.1 ASoC: twl6040: Add ramp up/down volume for HS and HF ASoC: twl6040: Fix TLV dB step values for gains. Misael Lopez Cruz (2): ASoC: twl6040: Enable plug detection interrupts ASoC: twl6040: Use correct offset for LineInAmp Right sound/soc/codecs/twl6040.c | 815 +++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/twl6040.h | 1 + 2 files changed, 695 insertions(+), 121 deletions(-) Regards, Margarita

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[alsa-devel] [PATCH v2 1/3] ASoC: sdp4430: Enable FM stereo pins
by Olaya, Margarita 07 Dec '10

07 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

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[alsa-devel] [PATCH v2 0/3] Add Jack support for
by Olaya, Margarita 07 Dec '10

07 Dec '10

This patch series adds the support for headset jack detection v2: Include codec patch in the series Jorge Eduardo Candelaria (3): ASoC: sdp4430: Enable FM stereo pins ASoC: sdp4430: Add Jack support ASoC: twl6040: Add jack support for headset and handset sound/soc/codecs/twl6040.c | 26 ++++++++++++++++++++++++++ sound/soc/codecs/twl6040.h | 12 ++++++++++++ sound/soc/omap/sdp4430.c | 39 +++++++++++++++++++++++++++++++++++---- 3 files changed, 73 insertions(+), 4 deletions(-) Regards, Margarita

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[alsa-devel] [PATCH 2/2] ASoC: sdp4430: Add Jack support
by Olaya, Margarita 07 Dec '10

07 Dec '10

From: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Use jack framework to enable detection for the headset microphone and stereo output. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria(a)ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya(a)ti.com> --- sound/soc/omap/sdp4430.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 0c37c51..ca7fb90 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) { @@ -160,7 +176,19 @@ 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); + twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET); return ret; } -- 1.7.0.4

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