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

[alsa-devel] [PATCH] ALSA: HDA: Fix microphone(s) on Lenovo Edge 13
by David Henningsson 04 Feb '11

04 Feb '11

This Edge 13 model has an internal mic at 0x1a and should therefore use the asus quirk. Since my backport of the asus model is now tested and working, would it be possible to apply that patch as well as this quirk to 2.6.38? Thanks Takashi for keeping up with all the patches coming in! :-) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] [PATCH 1/3] ASoC: soc_jack - add function to add jack zones and determine jack type
by Harsha Priya 04 Feb '11

04 Feb '11

Based on the voltage value in adc channel,jack plugged in needs to be determined This patch adds a helper functions that adds the adc voltage zones and determines the jack type given the mic bias value. When a jack is created, the voltage zones can to be added to it. Later when a jack is inserted, the micbias voltage can be measured and used to find the jack type For any codec that would give the mic bias value on a jack insertion, this function would return the type of jack based on mic bias range Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- include/sound/soc.h | 20 +++++++++++++++ sound/soc/soc-jack.c | 67 ++++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 87 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4b6c0a8..97f2a3a 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); @@ -406,6 +410,21 @@ struct snd_soc_jack_pin { bool invert; }; +/* struct snd_soc_jack_zone - Describes voltage zones of jack detection + * + * @adc_value_start: start range of voltage value in adc channel + * when jack is pluggedin + * @adc_value_end: end range of voltage value in adc channel + * when jack is pluggedin + * @jack_type: type of jack that is expected for this voltage + */ +struct snd_soc_jack_zone { + unsigned int adc_value_start; + unsigned int adc_value_end; + unsigned int jack_type; + struct list_head list; +}; + /** * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection * @@ -435,6 +454,7 @@ 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..66bd68f 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); @@ -111,6 +112,72 @@ out: } EXPORT_SYMBOL_GPL(snd_soc_jack_report); +static int snd_soc_validate_jack_type(int type) +{ + switch (type) { + case SND_JACK_HEADPHONE: + case SND_JACK_MICROPHONE: + case SND_JACK_HEADSET: + case SND_JACK_LINEOUT: + case SND_JACK_MECHANICAL: + case SND_JACK_VIDEOOUT: + case SND_JACK_AVOUT: + break; + default: + return -EINVAL; + + }; + return 0; +} + +/** + * 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++) { + if (snd_soc_validate_jack_type(zones[i].jack_type)) { + printk(KERN_ERR "Unknown jack type given %d\n", i); + return -EINVAL; + } + 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->adc_value_start && + micbias_voltage < zone->adc_value_end) + return zone->jack_type; + } + return -ENODEV; +} +EXPORT_SYMBOL_GPL(snd_soc_jack_get_type); /** * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack * -- 1.7.3.4

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[alsa-devel] NVIDIA HDMI surround sound is broken starting with kernel 2.6.36
by VDR User 04 Feb '11

04 Feb '11

The behavior described below started with 2.6.36 and follows through up to the most recent stable kernel, 2.6.37. The last properly working driver is in kernel 2.6.35.9. I should also mention that while the tests below use the static_hdmi_pcm flag recently introduced to alsa git, I also tested with the in-kernel drivers for each stable kernel. The particular device used for testing is an Nvidia GT220 video card connected directly to my receiver via HDMI. Can someone please look into this? Tested: cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=1, receiver ON cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=1, receiver OFF cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=0, receiver ON cold boot, kernel 2.6.37, alsa snapshot jan.25,2010, static_hdmi_pcm=0, receiver OFF In all cases only the front-left and front-right speakers worked. LFE, rear-left, rear-right, and center were missing. ELD correctly reported channels corresponding to receiver ON/OFF. (receiver on=7.1, receiver off=tv stereo). Also, even though the eld file is removed, the eld info is _not_ updated when stopping alsa & unloading drivers, then reloading. A reboot is required to get the eld info to update. This is why I cold boot the pc for each test. With kernel 2.6.35.9 drivers, front-left & front-right worked with receiver OFF as expected. I then turn the receiver on and all channels worked. I didn't bother to stop alsa, reboot, or even look at eld info since it works. Thanks for any & all help, Derek

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[alsa-devel] [PATCH - hdspmixer 0/4] Beautification patches
by Adrian Knoth 03 Feb '11

03 Feb '11

Hi! Here's a line of code beautification patches for hdspmixer. I broke them up into smaller pieces to ease the review, though one may find four patches for no added functionality a little bit excessive. If so, please feel free simply commit them as a single commit. Cheers Adrian Knoth (4): hdspmixer: retab and reindent code. hdspmixer: Improve readability. hdspmixer: Beautification; Lower nesting depth hdspmixer: Break overlong line and don't plenken. -- 1.7.2.3

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[alsa-devel] QasMixer 0.7.0 released (former Alsamixer-Qt4)
by Sebastian H. 03 Feb '11

03 Feb '11

Hello ALSA users A new version 0.7.0 of QasMixer (formerly Alsamixer-Qt4) is now available at http://xwmw.org/qasmixer or http://sourceforge.net/projects/qasmixer/ QasMixer is a fusion of Alsamixer-Qt4 and AlsaInfo-Qt4. Both will be discontinued because it's easier for me to support just one larger application instead of two smaller ones. Most notably for end users will be the improved desktop integration with single instance mode, a system tray icon, a qasmixer.desktop file for application menus and automatic desktop size detection which should play well with netbooks. Regards, Sebastian Holtermann

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[alsa-devel] [PATCH v2] asoc: davinci: da830/omap-l137: correct cpu_dai_name
by Vaibhav Bedia 03 Feb '11

03 Feb '11

McASP1 is used on the DA830/OMAP-L137 platform for the codec. This is different from the DA850/OMAP-L138 platform which uses McASP0. This is fixed by adding a new snd_soc_dai_link struct. Signed-off-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> --- v2: Fixed the subject as pointed out by Sergei. sound/soc/davinci/davinci-evm.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index b36f0b3..fe79842 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -218,7 +218,19 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_spdif_ops, }, }; -static struct snd_soc_dai_link da8xx_evm_dai = { + +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai_name = "davinci-mcasp.1", + .codec_dai_name = "tlv320aic3x-hifi", + .codec_name = "tlv320aic3x-codec.1-0018", + .platform_name = "davinci-pcm-audio", + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +static struct snd_soc_dai_link da850_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp.0", @@ -259,13 +271,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { static struct snd_soc_card da830_snd_soc_card = { .name = "DA830/OMAP-L137 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da830_evm_dai, .num_links = 1, }; static struct snd_soc_card da850_snd_soc_card = { .name = "DA850/OMAP-L138 EVM", - .dai_link = &da8xx_evm_dai, + .dai_link = &da850_evm_dai, .num_links = 1, }; -- 1.6.2.4

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[alsa-devel] [PATCH 1/2] ASoC: Create an AIF1ADCDAT signal widget to match AIF2
by Dimitris Papastamos 03 Feb '11

03 Feb '11

From: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Due to the different routing for AIF1 and AIF2 we weren't using a single widget to represent the ADCDAT signal. For consistency add one. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++---- 1 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 0ca81d3..f53b5a9 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1287,9 +1287,9 @@ SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 9, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, @@ -1298,9 +1298,9 @@ SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, 0, WM8994_POWER_MANAGEMENT_4, 11, 0), -SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture", +SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, 0, WM8994_POWER_MANAGEMENT_4, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, @@ -1345,6 +1345,7 @@ SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux), @@ -1546,6 +1547,11 @@ static const struct snd_soc_dapm_route intercon[] = { { "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" }, { "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC1R" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2L" }, + { "AIF1ADCDAT", NULL, "AIF1ADC2R" }, + { "AIF2ADCDAT", NULL, "AIF2ADC Mux" }, /* AIF3 output */ -- 1.7.3.5

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[alsa-devel] [PATCH] ASoC: Sort ALC5623 in Kconfig and Makefile
by Mark Brown 03 Feb '11

03 Feb '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Makefile | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 06894da..46ffb26 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -92,6 +92,7 @@ obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o @@ -103,7 +104,6 @@ obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o -obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o -- 1.7.2.3

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[alsa-devel] UCM list/set/get
by pl bossart 03 Feb '11

03 Feb '11

I've been trying to figure out what exactly needs to be filled in UCM configuration files, and so far it's been a pretty frustrating experience with alsa-utils/alsaucm :-( The interactive mode in alsa-utils/alsaucm is broken, I fixed a couple of segfaults and bad output (patches attached), but so far the only thing I can do is list the verbs...It looks like alsa-lib/include/use-case.h and alsa-utils/alsaucm are not aligned, whenever I query with predefined identifiers alsaucm seg faults or provides an error message that doesn't help much. This happens with the latest code. [ume@plb alsaucm]$ ./alsaucm listcards 0: AudioPC this is my AudioPCI card 1: SDP4430 this is my SDP4430 card 2: TestHDA A test HDA card [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _verbs 0: HiFi Play and record HiFi quality Music. 1: Voice Full duplex low-power voice call So far so good. Now the fun begins [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _devices ./alsaucm: error failed to get list _devices: No such file or directory [ume@plb alsaucm]$ ./alsaucm -c AudioPCI set _verb HiFi ALSA lib main.c:1250:(set_verb_user) error: failed to initialize new use case: HiFi ./alsaucm: error failed to set _verb=HiFi: No such device [ume@plb alsaucm]$ ./alsaucm -c AudioPCI get _verb ./alsaucm: error failed to get _verb: No such file or directory Likewise if the .conf file contains a cset in the default values, then the parser fails (I took the example from src/ucm/parser.c). Commenting out the cset is the only solution. What exactly is this index supposed to mean? SectionDefaults [ cdev "hw:AudioPCI" cset "name='Master Playback Switch',index=2 1,1" ] [ume@plb alsaucm]$ ./alsaucm -c AudioPCI list _verbs ALSA lib main.c:302:(import_master_config) Unable to execute default sequence ALSA lib main.c:637:(snd_use_case_mgr_open) error: failed to import AudioPCI use case configuration -19 ./alsaucm: error failed to open sound card AudioPCI: No such device Ideally I'd like to write an interface between PulseAudio and UCM, there are existing modules I can reuse to set voice or hifi modes, depending on what the client roles are. But if I can't make the examples work I don't think this is going to go anywhere. -Pierre

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[alsa-devel] [PATCH 0/3] ASoC: add support for jack detection in mid-x86 and sn95031 codec
by Harsha, Priya 03 Feb '11

03 Feb '11

Hi Mark, This patch set adds support for jack detection in mid-x6 machine and TI sn95031 codec drivers. Also adds a helper function in soc-jack.c for converting mic bias values to jack types. The comments from previous submission are addressed. [PATCH 1/3] ASoC: soc_jack - add function to add voltage zones and determine jack type This patch adds a helper functions that adds the adc voltage zones and determines the jack type given the mic bias value. When a jack is created, the voltage zones can to be added to it. Later when a jack is inserted, the micbias voltage can be measured and used to find the jack type [PATCH 2/3] ASoC: mid-x86 machine driver - add jack detect support This patch adds the jack detection in mid-86 machine driver [PATCH 3/3] ASoC: sn95031 codec - adding jack detection/reporting This patch adds the jack detection logic in sn95031 codec. This also adds code for reading the ADC to figure out the mic bias and therby report jack type detected -Harsha

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