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

[alsa-devel] [PATCH] ALSA: usb - prevent kernel panic on disconnect
by Pierre-Louis Bossart 10 Feb '11

10 Feb '11

Fix 100% reproducible kernel panic when a USB audio device is disconnected during playback. When the PCM substream is NULL, any dereference causes a kernel panic. This isn't a recent problem, the crash happened with a 2.6.35 kernel as well Note that this is only a work-around, it does not address the root cause of this inconsistency between urbs and PCM states. The dmesg below shows two calls to snd_urb_complete, the substream is NULL and the state is either running or stopped. This doesn't make any sense. Thanks to Sarah Sharp <sarah.a.sharp(a)linux.intel.com> for her help. usb 6-1: USB disconnect, address 16 ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:492: frame 0 active: -84 ALSA urb.c:197: cannot submit urb (err = -19) ALSA urb.c:186: NULL substream (subs->running 1) <- How is this possible? ALSA urb.c:186: NULL substream (subs->running 0) Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)intel.com> --- usb/urb.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/usb/urb.c b/usb/urb.c index e184349..1a264c2 100644 --- a/usb/urb.c +++ b/usb/urb.c @@ -170,6 +170,12 @@ static void snd_complete_urb(struct urb *urb) struct snd_pcm_substream *substream = ctx->subs->pcm_substream; int err = 0; + if (substream == NULL) { + snd_printd(KERN_ERR "NULL substream (subs->running %d)\n", subs->running); + clear_bit(ctx->index, &subs->active_mask); + return; + } + if ((subs->running && subs->ops.retire(subs, substream->runtime, urb)) || !subs->running || /* can be stopped during retire callback */ (err = subs->ops.prepare(subs, substream->runtime, urb)) < 0 || @@ -193,6 +199,12 @@ static void snd_complete_sync_urb(struct urb *urb) struct snd_pcm_substream *substream = ctx->subs->pcm_substream; int err = 0; + if (substream == NULL) { + snd_printd(KERN_ERR "NULL substream (subs->running %d)\n", subs->running); + clear_bit(ctx->index + 16, &subs->active_mask); + return; + } + if ((subs->running && subs->ops.retire_sync(subs, substream->runtime, urb)) || !subs->running || /* can be stopped during retire callback */ (err = subs->ops.prepare_sync(subs, substream->runtime, urb)) < 0 || -- 1.7.4

2 1

[alsa-devel] [PATCH] ASoC: add support for multiple jack types
by Koul, Vinod 10 Feb '11

10 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> --- include/sound/soc.h | 23 +++++++++++++++++++++++ sound/soc/soc-jack.c | 46 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4b6c0a8..4ccf1e4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -234,6 +234,7 @@ struct snd_soc_codec; struct snd_soc_codec_driver; struct soc_enum; struct snd_soc_jack; +struct snd_soc_jack_zone; struct snd_soc_jack_pin; struct snd_soc_cache_ops; #include <sound/soc-dapm.h> @@ -307,6 +308,9 @@ void snd_soc_jack_notifier_register(struct snd_soc_jack *jack, struct notifier_block *nb); void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack, struct notifier_block *nb); +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones); +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage); #ifdef CONFIG_GPIOLIB int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); @@ -407,6 +411,24 @@ struct snd_soc_jack_pin { }; /** + * struct snd_soc_jack_zone - Describes voltage zones of jack detection + * + * @min_mv: start voltage in mv + * @max_mv: end voltage in mv + * @jack_type: type of jack that is expected for this voltage + * @debounce_time: debounce_time for jack, codec driver should wait for this + * duration before reading the adc for voltages + * @:list: list container + */ +struct snd_soc_jack_zone { + unsigned int min_mv; + unsigned int max_mv; + unsigned int jack_type; + unsigned int debounce_time; + struct list_head list; +}; + +/** * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * * @gpio: gpio number @@ -435,6 +457,7 @@ struct snd_soc_jack { struct list_head pins; int status; struct blocking_notifier_head notifier; + struct list_head jack_zones; }; /* SoC PCM stream information */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index ac5a5bc..99dbaf7 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -37,6 +37,7 @@ int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type, { jack->codec = codec; INIT_LIST_HEAD(&jack->pins); + INIT_LIST_HEAD(&jack->jack_zones); BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier); return snd_jack_new(codec->card->snd_card, id, type, &jack->jack); @@ -112,6 +113,51 @@ out: EXPORT_SYMBOL_GPL(snd_soc_jack_report); /** + * snd_soc_jack_add_zones - Associate voltage zones with jack + * + * @jack: ASoC jack + * @count: Number of zones + * @zone: Array of zones + * + * After this function has been called the zones specified in the + * array will be associated with the jack. + */ +int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count, + struct snd_soc_jack_zone *zones) +{ + int i; + + for (i = 0; i < count; i++) { + INIT_LIST_HEAD(&zones[i].list); + list_add(&(zones[i].list), &jack->jack_zones); + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones); + +/** + * snd_soc_jack_get_type - Based on the mic bias value, this function returns + * the type of jack from the zones delcared in the jack type + * + * @micbias_voltage: mic bias voltage at adc channel when jack is plugged in + * + * Based on the mic bias value passed, this function helps identify + * the type of jack from the already delcared jack zones + */ +int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage) +{ + struct snd_soc_jack_zone *zone; + + list_for_each_entry(zone, &jack->jack_zones, list) { + if (micbias_voltage >= zone->min_mv && + micbias_voltage < zone->max_mv) + return zone->jack_type; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); + +/** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * * @jack: ASoC jack -- 1.7.2.3

3 2

[alsa-devel] [PATCH 0/3] add support for jack detection in mid-x86
by Koul, Vinod 09 Feb '11

09 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch set adds support for jack detection in mid-x6 machine and TI sn95031 codec drivers. This also adds support to read mic bias values thru ADC This version fixes namespace issue in patch1, and removes gpio bits in machine driver pending gpiolib support, which will be added separately Vinod Koul (3): ASoC: sn95031: Add jack support in the codec ASoC: mfld_machine: Add support for jack detection ASoC: sn95031: Add support for reading mic bias sound/soc/codecs/sn95031.c | 183 +++++++++++++++++++++++++++++++++- sound/soc/codecs/sn95031.h | 33 ++++++ sound/soc/mid-x86/mfld_machine.c | 203 +++++++++++++++++++++++++++++++++----- 3 files changed, 388 insertions(+), 31 deletions(-) -- 1.7.2.3

3 5

[alsa-devel] [PATCH 1/3] ASoC: Davinci: Use resource_size() helper function
by Vaibhav Bedia 09 Feb '11

09 Feb '11

This patch modifies the Davinci i2s and mcasp drivers to make use of the resource_size() helper function for readability. Signed-off-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- sound/soc/davinci/davinci-i2s.c | 6 +++--- sound/soc/davinci/davinci-mcasp.c | 6 +++--- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 9e0e565..f8c8487 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -658,7 +658,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) return -ENODEV; } - ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1, + ioarea = request_mem_region(mem->start, resource_size(mem), pdev->name); if (!ioarea) { dev_err(&pdev->dev, "McBSP region already claimed\n"); @@ -731,7 +731,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) err_free_mem: kfree(dev); err_release_region: - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); return ret; } @@ -747,7 +747,7 @@ static int davinci_i2s_remove(struct platform_device *pdev) dev->clk = NULL; kfree(dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); return 0; } diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index fb55d2c..9b5d5ca 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -868,7 +868,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } ioarea = request_mem_region(mem->start, - (mem->end - mem->start) + 1, pdev->name); + resource_size(mem), pdev->name); if (!ioarea) { dev_err(&pdev->dev, "Audio region already claimed\n"); ret = -EBUSY; @@ -933,7 +933,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return 0; err_release_region: - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); err_release_data: kfree(dev); @@ -951,7 +951,7 @@ static int davinci_mcasp_remove(struct platform_device *pdev) dev->clk = NULL; mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - release_mem_region(mem->start, (mem->end - mem->start) + 1); + release_mem_region(mem->start, resource_size(mem)); kfree(dev); -- 1.6.2.4

3 4

[alsa-devel] [PATCH 1/3] ASoC: Remove redundant -codec from WM8903 driver name
by Mark Brown 09 Feb '11

09 Feb '11

It causes noisy -codecs to appear in things like .codec_name. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 2 +- sound/soc/tegra/harmony.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 27cd8f5..04c3256 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2134,7 +2134,7 @@ MODULE_DEVICE_TABLE(i2c, wm8903_i2c_id); static struct i2c_driver wm8903_i2c_driver = { .driver = { - .name = "wm8903-codec", + .name = "wm8903", .owner = THIS_MODULE, }, .probe = wm8903_i2c_probe, diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c index 11e2cb8..d702abe 100644 --- a/sound/soc/tegra/harmony.c +++ b/sound/soc/tegra/harmony.c @@ -189,7 +189,7 @@ static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link harmony_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", - .codec_name = "wm8903-codec.0-001a", + .codec_name = "wm8903.0-001a", .platform_name = "tegra-pcm-audio", .cpu_dai_name = "tegra-i2s.0", .codec_dai_name = "wm8903-hifi", -- 1.7.2.3

3 5

[alsa-devel] [PATCH 1/3] ASoC: Convert WM8903 to use PGA_S for output stage enables
by Mark Brown 09 Feb '11

09 Feb '11

This simplfies the code and slightly reduces the startup time. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 180 +++++++++++++++------------------------------ 1 files changed, 60 insertions(+), 120 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b20611b..a5b2d07 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -251,6 +251,8 @@ 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: return 1; default: @@ -309,11 +311,6 @@ static void wm8903_reset(struct snd_soc_codec *codec) sizeof(wm8903_reg_defaults)); } -#define WM8903_OUTPUT_SHORT 0x8 -#define WM8903_OUTPUT_OUT 0x4 -#define WM8903_OUTPUT_INT 0x2 -#define WM8903_OUTPUT_IN 0x1 - static int wm8903_cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -324,99 +321,6 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, } /* - * Event for headphone and line out amplifier power changes. Special - * power up/down sequences are required in order to maximise pop/click - * performance. - */ -static int wm8903_output_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - u16 val; - u16 reg; - u16 dcs_reg; - u16 dcs_bit; - int shift; - - switch (w->reg) { - case WM8903_POWER_MANAGEMENT_2: - reg = WM8903_ANALOGUE_HP_0; - dcs_bit = 0 + w->shift; - break; - case WM8903_POWER_MANAGEMENT_3: - reg = WM8903_ANALOGUE_LINEOUT_0; - dcs_bit = 2 + w->shift; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - switch (w->shift) { - case 0: - shift = 0; - break; - case 1: - shift = 4; - break; - default: - BUG(); - return -EINVAL; /* Spurious warning from some compilers */ - } - - if (event & SND_SOC_DAPM_PRE_PMU) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_POST_PMU) { - val = snd_soc_read(codec, reg); - - val |= (WM8903_OUTPUT_IN << shift); - snd_soc_write(codec, reg, val); - - val |= (WM8903_OUTPUT_INT << shift); - snd_soc_write(codec, reg, val); - - /* Turn on the output ENA_OUTP */ - val |= (WM8903_OUTPUT_OUT << shift); - snd_soc_write(codec, reg, val); - - /* Enable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg |= dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Remove the short */ - val |= (WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - } - - if (event & SND_SOC_DAPM_PRE_PMD) { - val = snd_soc_read(codec, reg); - - /* Short the output */ - val &= ~(WM8903_OUTPUT_SHORT << shift); - snd_soc_write(codec, reg, val); - - /* Disable the DC servo */ - dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0); - dcs_reg &= ~dcs_bit; - snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg); - - /* Then disable the intermediate and output stages */ - val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT | - WM8903_OUTPUT_IN) << shift); - snd_soc_write(codec, reg, val); - } - - return 0; -} - -/* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption * when used with digital sources. Enable and disable this mode @@ -918,23 +822,40 @@ 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_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 1, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2, - 0, 0, NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), - -SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0, - NULL, 0, wm8903_output_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD), +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, + 0, 0, NULL, 0), + +SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0, + NULL, 0), +SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 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("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("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, + 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, + NULL, 0), + +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -1050,11 +971,30 @@ static const struct snd_soc_dapm_route intercon[] = { { "Left Speaker PGA", NULL, "Left Speaker Mixer" }, { "Right Speaker PGA", NULL, "Right Speaker Mixer" }, - { "HPOUTL", NULL, "Left Headphone Output PGA" }, - { "HPOUTR", NULL, "Right Headphone Output PGA" }, - - { "LINEOUTL", NULL, "Left Line Output PGA" }, - { "LINEOUTR", NULL, "Right Line Output PGA" }, + { "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_DCS", NULL, "HPL_ENA_DLY" }, + { "HPR_DCS", NULL, "HPR_ENA_DLY" }, + { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, + { "LINEOUTR_DCS", NULL, "LINEOUTR_ENA_DLY" }, + + { "HPL_ENA_OUTP", NULL, "HPL_DCS" }, + { "HPR_ENA_OUTP", NULL, "HPR_DCS" }, + { "LINEOUTL_ENA_OUTP", NULL, "LINEOUTL_DCS" }, + { "LINEOUTR_ENA_OUTP", NULL, "LINEOUTR_DCS" }, + + { "HPL_RMV_SHORT", NULL, "HPL_ENA_OUTP" }, + { "HPR_RMV_SHORT", NULL, "HPR_ENA_OUTP" }, + { "LINEOUTL_RMV_SHORT", NULL, "LINEOUTL_ENA_OUTP" }, + { "LINEOUTR_RMV_SHORT", NULL, "LINEOUTR_ENA_OUTP" }, + + { "HPOUTL", NULL, "HPL_RMV_SHORT" }, + { "HPOUTR", NULL, "HPR_RMV_SHORT" }, + { "LINEOUTL", NULL, "LINEOUTL_RMV_SHORT" }, + { "LINEOUTR", NULL, "LINEOUTR_RMV_SHORT" }, { "LOP", NULL, "Left Speaker PGA" }, { "LON", NULL, "Left Speaker PGA" }, -- 1.7.2.3

2 5

[alsa-devel] [PATCH] ASoC: Sync initial widget state with hardware
by Mark Brown 09 Feb '11

09 Feb '11

ASoC generally uses the register defaults for everything, but in some cases the hardware will default to enabling some of the DAPM widgets (clocks for example). Ensure that DAPM knows about the actual widget state at initialisation by reading the enable bits after instantiating the widgets so they don't get left enabled needlessly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d0342aa..b30eda6 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1664,6 +1664,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; + unsigned int val; list_for_each_entry(w, &dapm->card->widgets, list) { @@ -1712,6 +1713,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_post: break; } + + /* Read the initial power state from the device */ + if (w->reg >= 0) { + val = snd_soc_read(w->codec, w->reg); + val &= 1 << w->shift; + if (w->invert) + val = !val; + + if (val) + w->power = 1; + } + w->new = 1; } -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Actively manage WM8903 DC servo configuration
by Mark Brown 09 Feb '11

09 Feb '11

Explicitly cache the DC servo offsets for digital paths in the driver, allowing them to be preserved over suspend and resume, and ensure that we recalibrate analogue outputs paths when they are in use so that we cover any changes in the input offset. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8903.c | 123 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8903.h | 8 +++ 2 files changed, 127 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 0ade99d..ff1371f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -224,6 +224,9 @@ struct wm8903_priv { int fs; int deemph; + int dcs_pending; + int dcs_cache[4]; + /* Reference count */ int class_w_users; @@ -253,6 +256,10 @@ static int wm8903_volatile_register(struct snd_soc_codec *codec, unsigned int re 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: + case WM8903_DC_SERVO_READBACK_4: return 1; default: @@ -320,6 +327,103 @@ static int wm8903_cp_event(struct snd_soc_dapm_widget *w, return 0; } +static int wm8903_dcs_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + wm8903->dcs_pending |= 1 << w->shift; + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + 1 << w->shift, 0); + break; + } + + return 0; +} + +#define WM8903_DCS_MODE_WRITE_STOP 0 +#define WM8903_DCS_MODE_START_STOP 2 + +static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm, + enum snd_soc_dapm_type event, int subseq) +{ + struct snd_soc_codec *codec = container_of(dapm, + struct snd_soc_codec, dapm); + struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); + int dcs_mode = WM8903_DCS_MODE_WRITE_STOP; + int i, val; + + /* Complete any pending DC servo starts */ + if (wm8903->dcs_pending) { + dev_dbg(codec->dev, "Starting DC servo for %x\n", + wm8903->dcs_pending); + + /* If we've no cached values then we need to do startup */ + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + if (wm8903->dcs_cache[i]) { + dev_dbg(codec->dev, + "Restore DC servo %d value %x\n", + 3 - i, wm8903->dcs_cache[i]); + + snd_soc_write(codec, WM8903_DC_SERVO_4 + i, + wm8903->dcs_cache[i] & 0xff); + } else { + dev_dbg(codec->dev, + "Calibrate DC servo %d\n", 3 - i); + dcs_mode = WM8903_DCS_MODE_START_STOP; + } + } + + /* Don't trust the cache for analogue */ + if (wm8903->class_w_users) + dcs_mode = WM8903_DCS_MODE_START_STOP; + + snd_soc_update_bits(codec, WM8903_DC_SERVO_2, + WM8903_DCS_MODE_MASK, dcs_mode); + + snd_soc_update_bits(codec, WM8903_DC_SERVO_0, + WM8903_DCS_ENA_MASK, wm8903->dcs_pending); + + switch (dcs_mode) { + case WM8903_DCS_MODE_WRITE_STOP: + break; + + case WM8903_DCS_MODE_START_STOP: + msleep(270); + + /* Cache the measured offsets for digital */ + if (wm8903->class_w_users) + break; + + for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) { + if (!(wm8903->dcs_pending & (1 << i))) + continue; + + val = snd_soc_read(codec, + WM8903_DC_SERVO_READBACK_1 + i); + dev_dbg(codec->dev, "DC servo %d: %x\n", + 3 - i, val); + wm8903->dcs_cache[i] = val; + } + break; + + default: + pr_warn("DCS mode %d delay not set\n", dcs_mode); + break; + } + + wm8903->dcs_pending = 0; + } +} + /* * When used with DAC outputs only the WM8903 charge pump supports * operation in class W mode, providing very low power consumption @@ -852,10 +956,15 @@ SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0, SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPL_DCS", 3, WM8903_DC_SERVO_0, 3, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("HPR_DCS", 3, WM8903_DC_SERVO_0, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, WM8903_DC_SERVO_0, 1, 0, NULL, 0), -SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, WM8903_DC_SERVO_0, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA_S("HPL_DCS", 3, SND_SOC_NOPM, 3, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("HPR_DCS", 3, SND_SOC_NOPM, 2, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, SND_SOC_NOPM, 1, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, SND_SOC_NOPM, 0, 0, wm8903_dcs_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0, NULL, 0), @@ -984,6 +1093,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" }, { "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" }, + { "HPL_DCS", NULL, "DCS Master" }, + { "HPR_DCS", NULL, "DCS Master" }, + { "LINEOUTL_DCS", NULL, "DCS Master" }, + { "LINEOUTR_DCS", NULL, "DCS Master" }, + { "HPL_DCS", NULL, "HPL_ENA_DLY" }, { "HPR_DCS", NULL, "HPR_ENA_DLY" }, { "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" }, @@ -2103,6 +2217,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8903 = { .reg_word_size = sizeof(u16), .reg_cache_default = wm8903_reg_defaults, .volatile_register = wm8903_volatile_register, + .seq_notifier = wm8903_seq_notifier, }; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h index a6d63e4..7d96646 100644 --- a/sound/soc/codecs/wm8903.h +++ b/sound/soc/codecs/wm8903.h @@ -81,6 +81,14 @@ extern int wm8903_mic_detect(struct snd_soc_codec *codec, #define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0 0x41 #define WM8903_DC_SERVO_0 0x43 #define WM8903_DC_SERVO_2 0x45 +#define WM8903_DC_SERVO_4 0x47 +#define WM8903_DC_SERVO_5 0x48 +#define WM8903_DC_SERVO_6 0x49 +#define WM8903_DC_SERVO_7 0x4A +#define WM8903_DC_SERVO_READBACK_1 0x51 +#define WM8903_DC_SERVO_READBACK_2 0x52 +#define WM8903_DC_SERVO_READBACK_3 0x53 +#define WM8903_DC_SERVO_READBACK_4 0x54 #define WM8903_ANALOGUE_HP_0 0x5A #define WM8903_ANALOGUE_LINEOUT_0 0x5E #define WM8903_CHARGE_PUMP_0 0x62 -- 1.7.2.3

2 1

[alsa-devel] [PATCH] ASoC: Sync initial widget state with hardware
by Mark Brown 09 Feb '11

09 Feb '11

ASoC generally uses the register defaults for everything, but in some cases the hardware will default to enabling some of the DAPM widgets (clocks for example). Ensure that DAPM knows about the actual widget state at initialisation by reading the enable bits after instantiating the widgets so they don't get left enabled needlessly. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 13 +++++++++++++ 1 files changed, 13 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d0342aa..b4e8755 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1664,6 +1664,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) { struct snd_soc_dapm_widget *w; + unsigned int val; list_for_each_entry(w, &dapm->card->widgets, list) { @@ -1712,6 +1713,18 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_post: break; } + + /* Read the initial power state from the device */ + if (w->reg >= 0) { + val = snd_soc_read(w->codec, w->reg); + val &= 1 << w->shift; + if (w->invert) + val = !val; + + if (val) + w->active = 1; + } + w->new = 1; } -- 1.7.2.3

1 1

[alsa-devel] [PATCH 0/3] add support for jack detection in mid-x86
by Koul, Vinod 09 Feb '11

09 Feb '11

From: Vinod Koul <vinod.koul(a)intel.com> This patch set adds support for jack detection in mid-x6 machine and TI sn95031 codec drivers. This also adds support to read mic bias values thru ADC The comments from previous submission are addressed and patches split Vinod Koul (3): ASoC: sn95031: Add jack support in the codec ASoC: mfld_machine: Add support for jack detection ASoC: sn95031: Add support for reading mic bias sound/soc/codecs/sn95031.c | 184 ++++++++++++++++++++++++++++++++- sound/soc/codecs/sn95031.h | 35 ++++++ sound/soc/mid-x86/mfld_machine.c | 211 +++++++++++++++++++++++++++++++++----- 3 files changed, 399 insertions(+), 31 deletions(-) -- 1.7.2.3

2 12