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

[alsa-devel] Conexant CX20561 - Microphone problems
by Shawn Starr 19 Apr '11

19 Apr '11

Hello ALSA developers, I am having problems with the microphone on the following hardware this is on a Lenovo W500 laptop. Card: HDA Intel Chip: Conexant CX20561 (Hermosa) Here is the problems im noticing: 1) If I use ALSA 'default' with the microphone I get noise interference on output [high/low/high/low]. If I switch this to use the "ALSA Capture on HDA Intel [CONEXANT Analog] I get interference except it is the standard 60Hz electrical noise [steady noise] 2) Using the internal/external microphones gives an inconsistent volume. It also does not seem to work with the microphone cutting out so much. I need to adjust Digital + another microphone but it still drops capturing. Alsamixer shows: Digital/Docking Microphone/External Microphone/Internal Microphone Here is the module options I give to the sound driver: options snd-intel-hda enable_msi=1 power_save=10 power_save_controller=true model=laptop index=0 Assuming power saving/controller causes the microphone to cut out or not. 3) I'm now aware that this card has no pinouts for PCM capture https://bugzilla.redhat.com/show_bug.cgi?id=502293 Thanks for any help, Shawn.

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[alsa-devel] alsa-lib: snd_mixer_selem_is_active changes - which event?
by Sebastian H. 17 Apr '11

17 Apr '11

Hello I wonder which event mask notifies an application that snd_mixer_selem_is_active() may have changed. SND_CTL_EVENT_MASK_VALUE or SND_CTL_EVENT_MASK_INFO or something else? I'd be glad for any insights? Sebastian H.

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[alsa-devel] [PATCH 1/7] ASoC: neo1973_wm8753: Remove scenario management code.
by Lars-Peter Clausen 14 Apr '11

14 Apr '11

It has been proven to be inflexible to do scenario management in kernel space. Since actual neo1973 board support has not been merged in mainline and this patch has been in the neo1973 tree for some time now it should be safe to remove this functionality without breaking existing userspace. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/samsung/neo1973_wm8753.c | 148 +++--------------------------------- 1 files changed, 11 insertions(+), 137 deletions(-) diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index d3cd688..fe92946 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -37,17 +37,6 @@ #include "dma.h" #include "s3c24xx-i2s.h" -/* define the scenarios */ -#define NEO_AUDIO_OFF 0 -#define NEO_GSM_CALL_AUDIO_HANDSET 1 -#define NEO_GSM_CALL_AUDIO_HEADSET 2 -#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3 -#define NEO_STEREO_TO_SPEAKERS 4 -#define NEO_STEREO_TO_HEADPHONES 5 -#define NEO_CAPTURE_HANDSET 6 -#define NEO_CAPTURE_HEADSET 7 -#define NEO_CAPTURE_BLUETOOTH 8 - static struct snd_soc_card neo1973; static struct i2c_client *i2c; @@ -224,113 +213,6 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -static int neo1973_scenario; - -static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.integer.value[0] = neo1973_scenario; - return 0; -} - -static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - pr_debug("Entered %s\n", __func__); - - switch (neo1973_scenario) { - case NEO_AUDIO_OFF: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_enable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HANDSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_HEADSET: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - break; - default: - snd_soc_dapm_disable_pin(dapm, "Audio Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - } - - snd_soc_dapm_sync(dapm); - - return 0; -} - -static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - - pr_debug("Entered %s\n", __func__); - - if (neo1973_scenario == ucontrol->value.integer.value[0]) - return 0; - - neo1973_scenario = ucontrol->value.integer.value[0]; - set_scenario_endpoints(codec, neo1973_scenario); - return 1; -} - static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) @@ -454,26 +336,16 @@ static const struct soc_enum lm4857_mode_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode), }; -static const char *neo_scenarios[] = { - "Off", - "GSM Handset", - "GSM Headset", - "GSM Bluetooth", - "Speakers", - "Headphones", - "Capture Handset", - "Capture Headset", - "Capture Bluetooth" -}; - -static const struct soc_enum neo_scenario_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), -}; - static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { + SOC_DAPM_PIN_SWITCH("Audio Out"), + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Call Mic"), + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, lm4857_get_reg, lm4857_set_reg, stereo_tlv), SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, @@ -482,8 +354,6 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), - SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], - neo1973_get_scenario, neo1973_set_scenario), SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0, lm4857_get_reg, lm4857_set_reg), SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0, @@ -520,7 +390,11 @@ static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(wm8753_dapm_widgets)); /* set endpoints to default mode */ - set_scenario_endpoints(codec, NEO_AUDIO_OFF); + snd_soc_dapm_disable_pin(dapm, "Audio Out"); + snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); + snd_soc_dapm_disable_pin(dapm, "GSM Line In"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(dapm, "Call Mic"); /* add neo1973 specific controls */ err = snd_soc_add_controls(codec, wm8753_neo1973_controls, -- 1.7.2.3

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[alsa-devel] CA0110 taints kernel
by Vedran Miletić 13 Apr '11

13 Apr '11

Hi, I experience frequent kernel crashes with 2.6.31 and CA0110. I know this card doesn't work, and there is no problem with that; however, it crashing non-stop is not very nice. Also, card doesn't get any controls (and I believe it does get some controls when it's the only card using HDA driver, that is when I disable onboard one). dmesg is here: http://pastebin.com/m7c53562f lspci: 02:00.0 PCI bridge: Creative Labs Device 7006 (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=02, secondary=03, subordinate=03, sec-latency=64 Memory behind bridge: fe900000-fe9fffff Capabilities: [50] Power Management version 3 Capabilities: [60] MSI: Enable- Count=1/16 Maskable- 64bit+ Capabilities: [80] Subsystem: Creative Labs Device 0010 Capabilities: [90] Express PCI/PCI-X Bridge, MSI 00 Capabilities: [100] Advanced Error Reporting Kernel modules: shpchp 03:00.0 Audio device: Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG Subsystem: Creative Labs Device 0018 Flags: bus master, medium devsel, latency 64, IRQ 16 Memory at fe9fc000 (32-bit, non-prefetchable) [size=16K] Capabilities: [dc] Power Management version 3 Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel Any ideas? -- Vedran Miletić P.S. Please CC me, as I'm not subscribed to mailing list at the moment.

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[alsa-devel] GPL/LGPL licensing of the Alsa library
by Ossi Niiranen 11 Apr '11

11 Apr '11

Hi, We recently performed a licensing audit of a larger software project incorporating the Alsa library 1.0.19. One issue we identified was that although the Alsa library purported to state that the library was licensed under the LGPL, there were still several files with third party copyright holders that indicated in their header that the file in question was licensed under the GPL. Does the Alsa project have some sort of documentation somewhere showing that the third parties consented to the re-licensing under the LGPL, or does it have copies of contributor agreements showing that the copyright holders granted the project organisation rights to re-license under the LGPL? I realise nobody here is under any obligation to answer any of my questions, but if someone has the time, it would be greatly appreciated! Regards. --- Ossi Niiranen Lakimies / Attorney-at-law (NY) Neolex AB Tel. +358 400 414 289 http://www.neolex.fi LinkedIn: http://fi.linkedin.com/in/oniiranen Twitter: http://www.twitter.com/oniiranen The information contained in this e-mail is privileged and confidential, intended only for the use of the recipient named above. If the reader of this message is not the above mentioned recipient please note that any saving, distribution or copying of this e-mail is strictly prohibited. Please notify us promptly by e-mail at info(a)neolex.fi should you have received such misdirected messages and kindly destroy this e-mail. Thank You.

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[alsa-devel] Question about snd_mixer_selem_get_playback_dB_range()
by Tanu Kaskinen 11 Apr '11

11 Apr '11

Hi, I have added code to Pulseaudio that checks that the kernel driver returns the same values if I call snd_mixer_selem_get_playback_dB_range() or snd_mixer_selem_ask_playback_vol_dB() with the lowest and highest volume steps. The reason I do this is that I've seen a driver that didn't return correct values from _dB_range() (it's fixed nowadays), so it's possible that the same kind of bugs will be seen in the future too. Doing this check at Pulseaudio startup helps to detect such bugs early. My question is about the case where the lowest volume step is mute. Is it a bug in the driver if _dB_range() returns something else than SND_CTL_TLV_DB_GAIN_MUTE as the min gain in such case? -- Tanu

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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] [PATCH 1/3] ASoC: Blackfin: drop "-codec" from codec names
by Mike Frysinger 08 Apr '11

08 Apr '11

The recent multi-component patch incorrectly added "-codec" suffixes to parts which are not MFD. Drop the suffix from the machine drivers too. Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- sound/soc/blackfin/bf5xx-ad1836.c | 2 +- sound/soc/blackfin/bf5xx-ad193x.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 +- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index 83012da..826bae6 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -85,7 +85,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai = { .cpu_dai_name = "bf5xx-tdm", .codec_dai_name = "ad1836-hifi", .platform_name = "bf5xx-tdm-pcm-audio", - .codec_name = "ad1836-codec.0", + .codec_name = "ad1836.0", .ops = &bf5xx_ad1836_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c index 3367882..5794d09 100644 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ b/sound/soc/blackfin/bf5xx-ad193x.c @@ -113,7 +113,7 @@ static struct snd_soc_dai_link bf5xx_ad193x_dai = { .cpu_dai_name = "bf5xx-tdm", .codec_dai_name ="ad193x-hifi", .platform_name = "bf5xx-tdm-pcm-audio", - .codec_name = "ad193x-codec.5", + .codec_name = "ad193x.5", .ops = &bf5xx_ad193x_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c index d57c9c9..c147130 100644 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ b/sound/soc/blackfin/bf5xx-ad1980.c @@ -73,7 +73,7 @@ static struct snd_soc_dai_link bf5xx_board_dai = { .cpu_dai_name = "bfin-ac97", .codec_dai_name = "ad1980-hifi", .platform_name = "bfin-pcm-audio", - .codec_name = "ad1980-codec", + .codec_name = "ad1980", .ops = &bf5xx_board_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732fb8b..1655f95 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -186,7 +186,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ad73311-hifi", .platform_name = "bfin-pcm-audio", - .codec_name = "ad73311-codec", + .codec_name = "ad73311", .ops = &bf5xx_ad73311_ops, }; diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index ad28663..732a02f 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -119,7 +119,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai = { .cpu_dai_name = "bf5xx-i2s", .codec_dai_name = "ssm2602-hifi", .platform_name = "bf5xx-pcm-audio", - .codec_name = "ssm2602-codec.0-001b", + .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, }; -- 1.7.4.1

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[alsa-devel] spi codec names in machine drivers
by Mike Frysinger 07 Apr '11

07 Apr '11

after the recent multi-component patch, spi codec names were changed to "<name>.<spics>" in the machine drivers. so if we have an ad193x codec on spi bus 0 cs 5, the name is now "ad193x.5". this also seems to be how fmt_single_name in soc-core.c is doing things. am i reading this right ? what if i have two codecs on two different spi busses but happen to have the same cs ? shouldnt the spi code do the same as i2c and include the bus in the id naming ? -mike

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[alsa-devel] [RFC Patch] ALSA - EMU10K1 - Remove Center LFE Wave Center and Wave LFE control for SB Live! Platinum
by Raymond Yau 06 Apr '11

06 Apr '11

SB Live! Platinum CT4760P is just a 4 channels sound card with STAC9721 and Philips UDA1334 DAC Is it feasible to remove "Center" , "LFE", "Wave Cetner" and "Wave LFE" Playback Volume control ? or add a bitmap to disable the creation of those controls Anyone know whether this patch has any side effect for the user with Live Drive AFAIK, speaker-test will fail speaker-test -d -c 6 -t wav -Dsurround51:1 speaker-test 1.0.24.2 Playback device is surround51:1 Stream parameters are 48000Hz, S16_LE, 6 channels WAV file(s) ALSA lib setup.c:565:(add_elem) Cannot obtain info for CTL elem (MIXER,'Headphone Center Playback Switch',1,0,0): No such file or directory Playback open error: -2,No such file or directory

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