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[alsa-devel] ca0106 SPDIF defaults
by Ben Stanley 19 Dec '08

19 Dec '08

Hi, I previously reported problems that are still plaguing me [1] to do with SPDIF settings that don't reset properly. I am now looking at doing a patch to the ca0106 driver to fix this problem. It appears I have to add a new control to be able to set the default settings. I presume this introduces complications with existing settings in asound.state? Is there some collected wisdom on how to add new controls while providing a migration/upgrade/interoperability path? Ben Stanley. [1] http://article.gmane.org/gmane.linux.alsa.devel/55406/match=independent+pcm…

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[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 19 Dec '08

19 Dec '08

The following changes since commit 49d92c7d5bbd158734bc34ed578a68b214a48583: Stanley.Miao (1): ASoC: TWL4030: hands-free start-up sequence. are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Alexander Beregalov (1): ASoC: switch davinci DPRINTK to pr_debug() Mark Brown (3): ASoC: Ease merge difficulties from new architectures ASoC: Complain if we fail to create DAPM controls ASoC: Add WM8350 AudioPlus codec driver include/linux/mfd/wm8350/audio.h | 38 +- sound/soc/Kconfig | 10 +- sound/soc/Makefile | 12 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8350.c | 1583 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8350.h | 20 + sound/soc/davinci/davinci-pcm.c | 18 +- sound/soc/soc-dapm.c | 6 +- 9 files changed, 1669 insertions(+), 24 deletions(-) create mode 100644 sound/soc/codecs/wm8350.c create mode 100644 sound/soc/codecs/wm8350.h

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[alsa-devel] [ASoC] TWL4030 underrun error
by naveenkrishna.ch 18 Dec '08

18 Dec '08

Hello, I am working on OMAP3 + TWL4030 combination for ASoC codec The omap generic omap-pcm.c file has static const struct snd_pcm_hardware omap_pcm_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE, .rates = SNDRV_PCM_RATE_8000_96000, .period_bytes_min = 32, .period_bytes_max = 64 * 1024, .periods_min = 2, .periods_max = 255, .buffer_bytes_max = 128 * 1024, }; following structure, When i play with aplay XXX.wav Rare but i used to get some underrun messages i found thses message are coming from aplay. underrun!!! (at least 10271.545 ms long) underrun!!! (at least 904.663 ms long) underrun!!! (at least 771.790 ms long) underrun!!! (at least 3393.433 ms long) underrun!!! (at least 777.314 ms long) I dint found much info about this in net too. How can i solve this, can any one help me out -- Thanks, (: Naveen Krishna Ch :)

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[alsa-devel] [PATCH] ALSA: Add support for TV-out to the jack reporting API
by Jani Nikula 18 Dec '08

18 Dec '08

Add support for reporting new jack types SND_JACK_TVOUT and SND_JACK_AVOUT (a combination of LINEOUT and TVOUT) to the jack reporting API. Also add the corresponding SW_TVOUT_INSERT switch to the input system header. Signed-off-by: Jani Nikula <ext-jani.1.nikula(a)nokia.com> --- include/linux/input.h | 1 + include/sound/jack.h | 2 ++ sound/core/jack.c | 6 ++++++ 3 files changed, 9 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index abd223b..ea5b224 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -646,6 +646,7 @@ struct input_absinfo { #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ #define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ +#define SW_TVOUT_INSERT 0x08 /* set = inserted */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 2e0315c..69fc1f0 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -37,6 +37,8 @@ enum snd_jack_types { SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ + SND_JACK_TVOUT = 0x0010, + SND_JACK_AVOUT = SND_JACK_LINEOUT | SND_JACK_TVOUT, }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index 7ca905c..23c3f3e 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -105,6 +105,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_LINEOUT) input_set_capability(jack->input_dev, EV_SW, SW_LINEOUT_INSERT); + if (type & SND_JACK_TVOUT) + input_set_capability(jack->input_dev, EV_SW, + SW_TVOUT_INSERT); if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); @@ -162,6 +165,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_LINEOUT) input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, status & SND_JACK_LINEOUT); + if (jack->type & SND_JACK_TVOUT) + input_report_switch(jack->input_dev, SW_TVOUT_INSERT, + status & SND_JACK_TVOUT); if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); -- 1.6.0.4

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[alsa-devel] aplay..no sound..no error..
by vasudha rao 17 Dec '08

17 Dec '08

Hi all.. I have installed alsa for my embedded board..(at91sam9263) when I use aplay utility I got, $ aplay a.wav Playing WAVE 'a.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Mono I am not getting any error message.. But no sound from speaker also.. What else I can try to listen sound from speaker..?? Thanks,

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[alsa-devel] SND_SOC_DAIFMT_NB_NF meaning
by Troy Kisky 17 Dec '08

17 Dec '08

A quick question here. Does SND_SOC_DAIFMT_NB_NF have a standard meaning or does it depend on what mode your in (I2S, DSP_A, right just). From looking at the sources, SND_SOC_DAIFMT_NB_NF usually means that data is valid on the rising BCLK and start of frame is rising edge if DSP mode. The reason I ask, is that I'd like to pass the same flags to the codec and the cpu_dai and have them agree with the interpretation. Thanks Troy

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Re: [alsa-devel] 1.0.18: /proc/asound/card0/stream0 no longer existing?
by Takashi Iwai 17 Dec '08

17 Dec '08

At Wed, 17 Dec 2008 10:49:08 +0100, Klaus Schulz wrote: > > Takashi. > > As you can see below SND_VERBOSE_PROCFS is y. But DEBUG and VERBOSE_PRINTK is > NO! These are irrelevant. > BTW: I am running a 2.6.28-rc6-zen1 kernel. I don't know what it means. But the stream* proc file should be created for snd-usb-audio when CONFIG_PROC_FS and CONFIG_SND_VERBOSE_PROCFS are set. At least, there is no change in the ALSA upstream side. Takashi > > Cheers > Klaus > > CONFIG_SOUND=m > CONFIG_SOUND_OSS_CORE=y > CONFIG_SND=m > CONFIG_SND_TIMER=m > CONFIG_SND_PCM=m > CONFIG_SND_HWDEP=m > CONFIG_SND_RAWMIDI=m > CONFIG_SND_SEQUENCER=m > CONFIG_SND_SEQ_DUMMY=m > CONFIG_SND_OSSEMUL=y > CONFIG_SND_MIXER_OSS=m > CONFIG_SND_PCM_OSS=m > CONFIG_SND_PCM_OSS_PLUGINS=y > CONFIG_SND_SEQUENCER_OSS=y > CONFIG_SND_DYNAMIC_MINORS=y > CONFIG_SND_SUPPORT_OLD_API=y > CONFIG_SND_VERBOSE_PROCFS=y > # CONFIG_SND_VERBOSE_PRINTK is not set > # CONFIG_SND_DEBUG is not set > CONFIG_SND_VMASTER=y > CONFIG_SND_MPU401_UART=m > CONFIG_SND_OPL3_LIB=m > CONFIG_SND_OPL4_LIB=m > CONFIG_SND_VX_LIB=m > CONFIG_SND_AC97_CODEC=m > CONFIG_SND_DRIVERS=y > CONFIG_SND_PCSP=m > CONFIG_SND_DUMMY=m > CONFIG_SND_VIRMIDI=m > CONFIG_SND_MTPAV=m > CONFIG_SND_MTS64=m > CONFIG_SND_SERIAL_U16550=m > CONFIG_SND_MPU401=m > CONFIG_SND_PORTMAN2X4=m > CONFIG_SND_AC97_POWER_SAVE=y > CONFIG_SND_AC97_POWER_SAVE_DEFAULT=0 > CONFIG_SND_WSS_LIB=m > CONFIG_SND_SB_COMMON=m > CONFIG_SND_SB8_DSP=m > CONFIG_SND_SB16_DSP=m > CONFIG_SND_ISA=y > CONFIG_SND_ADLIB=m > CONFIG_SND_AD1816A=m > CONFIG_SND_AD1848=m > CONFIG_SND_ALS100=m > CONFIG_SND_AZT2320=m > CONFIG_SND_CMI8330=m > CONFIG_SND_CS4231=m > CONFIG_SND_CS4232=m > CONFIG_SND_CS4236=m > CONFIG_SND_DT019X=m > CONFIG_SND_ES968=m > CONFIG_SND_ES1688=m > CONFIG_SND_ES18XX=m > CONFIG_SND_SC6000=m > CONFIG_SND_GUSCLASSIC=m > CONFIG_SND_GUSEXTREME=m > CONFIG_SND_GUSMAX=m > CONFIG_SND_INTERWAVE=m > CONFIG_SND_INTERWAVE_STB=m > CONFIG_SND_OPL3SA2=m > CONFIG_SND_OPTI92X_AD1848=m > CONFIG_SND_OPTI92X_CS4231=m > CONFIG_SND_OPTI93X=m > CONFIG_SND_MIRO=m > CONFIG_SND_SB8=m > CONFIG_SND_SB16=m > CONFIG_SND_SBAWE=m > CONFIG_SND_SB16_CSP=y > CONFIG_SND_SGALAXY=m > CONFIG_SND_SSCAPE=m > CONFIG_SND_WAVEFRONT=m > CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL=y > CONFIG_SND_PCI=y > CONFIG_SND_AD1889=m > CONFIG_SND_ALS300=m > CONFIG_SND_ALS4000=m > CONFIG_SND_ALI5451=m > CONFIG_SND_ATIIXP=m > CONFIG_SND_ATIIXP_MODEM=m > CONFIG_SND_AU8810=m > CONFIG_SND_AU8820=m > CONFIG_SND_AU8830=m > CONFIG_SND_AW2=m > CONFIG_SND_AZT3328=m > CONFIG_SND_BT87X=m > # CONFIG_SND_BT87X_OVERCLOCK is not set > CONFIG_SND_CA0106=m > CONFIG_SND_CMIPCI=m > CONFIG_SND_OXYGEN_LIB=m > CONFIG_SND_OXYGEN=m > CONFIG_SND_CS4281=m > CONFIG_SND_CS46XX=m > CONFIG_SND_CS46XX_NEW_DSP=y > CONFIG_SND_CS5530=m > CONFIG_SND_CS5535AUDIO=m > CONFIG_SND_DARLA20=m > CONFIG_SND_GINA20=m > CONFIG_SND_LAYLA20=m > CONFIG_SND_DARLA24=m > CONFIG_SND_GINA24=m > CONFIG_SND_LAYLA24=m > CONFIG_SND_MONA=m > CONFIG_SND_MIA=m > CONFIG_SND_ECHO3G=m > CONFIG_SND_INDIGO=m > CONFIG_SND_INDIGOIO=m > CONFIG_SND_INDIGODJ=m > CONFIG_SND_EMU10K1=m > CONFIG_SND_EMU10K1X=m > CONFIG_SND_ENS1370=m > CONFIG_SND_ENS1371=m > CONFIG_SND_ES1938=m > CONFIG_SND_ES1968=m > CONFIG_SND_FM801=m > CONFIG_SND_FM801_TEA575X_BOOL=y > CONFIG_SND_FM801_TEA575X=m > CONFIG_SND_HDA_INTEL=m > # CONFIG_SND_HDA_HWDEP is not set > # CONFIG_SND_HDA_INPUT_BEEP is not set > CONFIG_SND_HDA_CODEC_REALTEK=y > CONFIG_SND_HDA_CODEC_ANALOG=y > CONFIG_SND_HDA_CODEC_SIGMATEL=y > CONFIG_SND_HDA_CODEC_VIA=y > CONFIG_SND_HDA_CODEC_ATIHDMI=y > CONFIG_SND_HDA_CODEC_NVHDMI=y > CONFIG_SND_HDA_CODEC_CONEXANT=y > CONFIG_SND_HDA_CODEC_CMEDIA=y > CONFIG_SND_HDA_CODEC_SI3054=y > CONFIG_SND_HDA_GENERIC=y > CONFIG_SND_HDA_POWER_SAVE=y > CONFIG_SND_HDA_POWER_SAVE_DEFAULT=0 > CONFIG_SND_HDSP=m > CONFIG_SND_HDSPM=m > CONFIG_SND_HIFIER=m > CONFIG_SND_ICE1712=m > CONFIG_SND_ICE1724=m > CONFIG_SND_INTEL8X0=m > CONFIG_SND_INTEL8X0M=m > CONFIG_SND_KORG1212=m > CONFIG_SND_MAESTRO3=m > CONFIG_SND_MIXART=m > CONFIG_SND_NM256=m > CONFIG_SND_PCXHR=m > CONFIG_SND_RIPTIDE=m > CONFIG_SND_RME32=m > CONFIG_SND_RME96=m > CONFIG_SND_RME9652=m > CONFIG_SND_SIS7019=m > CONFIG_SND_SONICVIBES=m > CONFIG_SND_TRIDENT=m > CONFIG_SND_VIA82XX=m > CONFIG_SND_VIA82XX_MODEM=m > CONFIG_SND_VIRTUOSO=m > CONFIG_SND_VX222=m > CONFIG_SND_YMFPCI=m > CONFIG_SND_SPI=y > CONFIG_SND_USB=y > CONFIG_SND_USB_AUDIO=m > CONFIG_SND_USB_USX2Y=m > CONFIG_SND_USB_CAIAQ=m > CONFIG_SND_USB_CAIAQ_INPUT=y > # CONFIG_SND_USB_US122L is not set > CONFIG_SND_PCMCIA=y > CONFIG_SND_VXPOCKET=m > CONFIG_SND_PDAUDIOCF=m > CONFIG_SND_SOC=m > # CONFIG_SND_SOC_ALL_CODECS is not set > CONFIG_SOUND_PRIME=m > CONFIG_SOUND_MSNDCLAS=m > CONFIG_MSNDCLAS_INIT_FILE="/etc/sound/msndinit.bin" > CONFIG_MSNDCLAS_PERM_FILE="/etc/sound/msndperm.bin" > CONFIG_SOUND_MSNDPIN=m > CONFIG_MSNDPIN_INIT_FILE="/etc/sound/pndspini.bin" > CONFIG_MSNDPIN_PERM_FILE="/etc/sound/pndsperm.bin" > CONFIG_SOUND_OSS=m > # CONFIG_SOUND_TRACEINIT is not set > CONFIG_SOUND_DMAP=y > CONFIG_SOUND_SSCAPE=m > CONFIG_SOUND_VMIDI=m > CONFIG_SOUND_TRIX=m > CONFIG_SOUND_MSS=m > CONFIG_SOUND_MPU401=m > CONFIG_SOUND_PAS=m > CONFIG_SOUND_PSS=m > CONFIG_PSS_MIXER=y > # CONFIG_PSS_HAVE_BOOT is not set > CONFIG_SOUND_SB=m > CONFIG_SOUND_YM3812=m > CONFIG_SOUND_UART6850=m > CONFIG_SOUND_AEDSP16=m > CONFIG_SC6600=y > CONFIG_SC6600_JOY=y > CONFIG_SC6600_CDROM=4 > CONFIG_SC6600_CDROMBASE=0 > CONFIG_SOUND_KAHLUA=m > CONFIG_AC97_BUS=m > CONFIG_HID_SUPPORT=y > CONFIG_HID=m > # CONFIG_HID_DEBUG is not set > CONFIG_HIDRAW=y > > On Wed, Dec 17, 2008 at 9:40 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 17 Dec 2008 08:37:58 +0100, > Klaus Schulz wrote: > > > > Hi there. > > > > Can anybody tell me if /proc/asound/card0/stream0 is not existing any > longer > > and what has replaced it? > > It's built only when kernel config CONFIG_SND_VERBOSE_PROCFS=y. > In the case of alsa-driver tarball, this is enabled unless you add > --disable-verbose-procfs configure option. > > Takashi > >

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[alsa-devel] Sigmatel patch for missing
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi folks. Below patch seems to be missing in 1.0.18a! Note that also MCP51 seemed to work afterwards. It needs to be ported to 1.0.18.a though. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3036 Can anybody let me know how the patch would look like in latest driver rev. THX Klaus --- patch_sigmatel.c.orig 2007-09-15 17:57:07.000000000 -0700 +++ patch_sigmatel.c 2007-09-15 17:50:26.000000000 -0700 @@ -2304,6 +2304,15 @@ stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, AC_PINCTL_IN_EN); + switch (codec->subsystem_id) { + case 0x107b0317: // Gateway MX3412, MT3423 + case 0x107b0318: // Gateway MT3707, ML3109 (ATI SB450) + case 0x10de026c: // Gateway MX3410 + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_POWER_STATE, 0x00); + snd_hda_codec_write(codec, 0x08, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); + break; + } + if (spec->gpio_mute) { stac922x_gpio_mute(codec, 0, 0); stac922x_gpio_mute(codec, 1, 0);

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[alsa-devel] 1.0.18: /proc/asound/card0/stream0 no longer existing?
by Klaus Schulz 17 Dec '08

17 Dec '08

Hi there. Can anybody tell me if /proc/asound/card0/stream0 is not existing any longer and what has replaced it? THX a lot Klaus

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Re: [alsa-devel] Trying to figure out what went wrong with my HDMI.
by Clemens Ladisch 17 Dec '08

17 Dec '08

Mark Jenks wrote: > On Tue, Dec 16, 2008 at 7:24 AM, Clemens Ladisch <clemens(a)ladisch.de> wrote: > > Use "plughw:0,3" instead of "hw:0,3". > > What is the difference? Plughw automatically converts to a sample format that the device supports. > And why did hw:0,3 work before? Probably your old hardware did support mono playback. > If I want to use it in Myth, would it be ALSA:plughw:0,3 then? Yes. (It is possible that Myth can do resampling and channel rerouting by itself; in this case, hw would also work.) HTH Clemens

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