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[alsa-devel] S/PDIF on NVidia CK804 not working
by Mitko Haralanov 21 Sep '09

21 Sep '09

Hi all, I am new to the list and was hoping that someone here could give me some help and advise as I am stuck and none of my research has turned anything up. I am having trouble getting any sound out of the S/PDIF port on my built-in sound card. The card is recognized and 'aplay -L' shows the iec958 device (as evident by the attached alsa-info.txt file) but when I try to play any sound through it (using aplay) I get one of two errors: 1. If I use the iec958 device directly: $ aplay -D iec958 /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:984: Channels count non available 2. If I try to use the plug:iec958: $ aplay -D plug:iec958 /usr/share/sounds/alsa/Front_Center.wav Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono (the sound file seems to play but there is no audio coming out of the receiver) Looking through the ALSA website, I see that my sound device is supported. Configuring /etc/asound.conf and /etc/modprobe.conf as per the ALSA website yields the same results as above. Can anyone please, help me get my S/PDIF working? Thank you very much! -- Mitko Haralanov

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[alsa-devel] Application and hw mixing for VoIP (modified)
by Fabien COMTE 21 Sep '09

21 Sep '09

Hi, I need some help please. I have an application that does VoIP and alsa interface. I want to : - send through VoIP application a mixed sound from mic and an another application (aplay should be ok for testing). mic ---------- soundcard 0 -------------> mix ---- alsa interface -----> voip application application 1 --- alsa interface ------> (aplay) - use sound received from VoIP with HP output and an another application (arecord should be ok for testing). ----- alsa interface ------> application 2 (arecord) voip application ----- alsa interface ----> share ------ soundcard 0 --------> HP output And a little question : How to define an application interface in asoundrc ? I tried a lot of asoundrc but they doesn't work. Could someone help me ? Thank you. Fabien

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[alsa-devel] [PATCH] ALSA: lx6464es - remove unused struct member
by Tim Blechmann 21 Sep '09

21 Sep '09

we cannot set the sampling rate of the device, but can only read it from the board, so we don't need the member for it. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.h | 1 - 1 files changed, 0 insertions(+), 1 deletions(-)

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[alsa-devel] [PATCH] ALSA: lx6464es - cleanup of rmh message bus function
by Tim Blechmann 21 Sep '09

21 Sep '09

the rmh bus is not used asynchronously, so it is safe to remove the specific code pieces. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.h | 1 - sound/pci/lx6464es/lx_core.c | 98 +---------------------------------------- 2 files changed, 1 insertions(+), 98 deletions(-)

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[alsa-devel] [PATCH] sb16: request_region failure ignored in snd_sb16_isa_probe1()
by Roel Kluin 21 Sep '09

21 Sep '09

request_region() may fail. clean up if it does, also a release region later on was missing. Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- Found with sed: http://kernelnewbies.org/roelkluin build, sparse and checkpatch tested. diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 519c363..f50ed6f 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -502,6 +502,11 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev) fm_port[dev] = port[dev]; /* block the 0x388 port to avoid PnP conflicts */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); + if (acard->fm_res == NULL) { + snd_card_free(card); + return -ENOMEM; + } + #ifdef SNDRV_SBAWE_EMU8000 /* non-PnP AWE port address is hardwired with base port address */ awe_port[dev] = port[dev] + 0x400; @@ -509,6 +514,7 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev) snd_card_set_dev(card, pdev); if ((err = snd_sb16_probe(card, dev)) < 0) { + release_region(0x388, 4); snd_card_free(card); return err; }

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[alsa-devel] [PATCH] some minor changes for AD1836 and AD1938 codec drivers
by Barry Song 21 Sep '09

21 Sep '09

1. delete redundant assignment to bus field in spi_driver structure 2. fix lost assignment to set_bias_level entry in ad1938 codec dai 3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836" Signed-off-by: Barry Song <barry.song(a)analog.com> --- sound/soc/codecs/ad1836.c | 3 +-- sound/soc/codecs/ad1938.c | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 01343dc..c48485f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -251,8 +251,7 @@ static int __devexit ad1836_spi_remove(struct spi_device *spi) static struct spi_driver ad1836_spi_driver = { .driver = { - .name = "ad1836-spi", - .bus = &spi_bus_type, + .name = "ad1836", .owner = THIS_MODULE, }, .probe = ad1836_spi_probe, diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 9a049a1..34b30ef 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -456,7 +456,6 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { .name = "ad1938", - .bus = &spi_bus_type, .owner = THIS_MODULE, }, .probe = ad1938_spi_probe, @@ -515,6 +514,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) codec->num_dai = 1; codec->write = ad1938_write_reg; codec->read = ad1938_read_reg_cache; + codec->set_bias_level = ad1938_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); -- 1.5.6.3

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[alsa-devel] Multiple, independent streams with HDA
by Stefan Bruens 20 Sep '09

20 Sep '09

Hi as HDA supports up to 16 independent streams (sample rate, resolution), is there a possibility to expose this to user space? Use cases: - Simultaneous playback of music to speaker and connecting HPs for VoIP - Multiseat - pair of I/O for each seat ... I think there is already limited support for this, as e.g. SPDIF and analog expose independent PCMS (a least on STAC9271), although it is just a single codec. Maybe even make this configurable, either at run time, or at module load time. E.g. "option hda-intel pcm_out_nids=0x2:0x3,0x4:0x1e pcm_in_nids=0x7:0x20" creates pcmC0D{0,1,2}p and pcmC0D{0,1}c, where 0p is analog stereo out, 1p is analog 4chan out, 2p digital out, 0c is analog stereo in, and finally 1c is digital stereo in. (This example is for STAC9271: cat /proc/asound/Intel/codec#2 | egrep "Node|Converter" | grep Conv -B1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x06 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Converter: stream=0, channel=0 Node 0x07 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 Node 0x09 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 -- Node 0x1e [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 -- Node 0x20 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 ) Stefan -- Stefan Brüns / Bergstraße 21 / 52062 Aachen phone: +49 241 53809034 mobile: +49 151 50412019

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[alsa-devel] [[ALSA] PATCH - /pci/hda/patch_conexant.c 1/1] patch_conexant.c: cd pin added to mixer for benq r55 laptop,
by nowymarluk＠wp.pl 20 Sep '09

20 Sep '09

From: edoiks <nowymarluk(a)wp.pl> Signed-off-by: edoiks <nowymarluk(a)wp.pl> diff --git a/pci/hda/patch_conexant.c b/pci/hda/patch_conexant.c index 9d899ed..700a430 100644 --- a/pci/hda/patch_conexant.c +++ b/pci/hda/patch_conexant.c @@ -687,6 +687,8 @@ static struct hda_input_mux cxt5045_capture_source_benq = { { "IntMic", 0x1 }, { "ExtMic", 0x2 }, { "LineIn", 0x3 }, + { "CD Pin", 0x4 }, + { "Mixer ", 0x0 }, } }; @@ -816,6 +818,14 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), + {} }; -- 1.6.0.4

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 20 Sep '09

20 Sep '09

The following changes since commit cdc65fbe18aef15e92d2ebb410a189fbf956fb06: Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 The only really big thing here is the addition of the AK4671 driver, other than that it's minor tweaks. Barry Song (4): ASoC: Provide API for reordering channels ASoC: use set_channel_map api to reorder channels for AD1938 and AD1836 ASoC: fix typos in Blackfin headers ASoC: Blackfin I2S: add lost platform_device parameter to resume function Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration Cliff Cai (2): ASoC: Blackfin AC97: add a few missing multichannel define handling ASoC: Blackfin I2S: fix resuming when device hasn't been used Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> Jassi (3): ASoC: S3C lrsync function made to work with IRQs disabled. ASoC: S3C I2S LRCLK polarity option. ASoC: Return correct codec clock in s3c64xx-i2s Joonyoung Shim (2): ASoC: Factor out I2C 8 bit address 8 bit data I/O ASoC: AK4671: add ak4671 codec driver Julia Lawall (1): ASoC: Clean up error handling in MPC5200 DMA setup Mark Brown (5): ASoC: Add source argument to PLL configuration ASoC: Allow per-route connectedness checks for supplies ASoC: Fix display of stream name in DAPM debugfs ASoC: Add S3C64xx IIS CDCLK source selection Merge branch 'for-2.6.32' into for-2.6.33 Miguel Aguilar (1): ASoC: Davinci: Add audio codec support for DM365 EVM Mike Frysinger (1): ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 2 + include/sound/soc-dai.h | 13 +- include/sound/soc-dapm.h | 7 + sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 8 + sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-ad1836.c | 7 + sound/soc/blackfin/bf5xx-ad1938.c | 9 +- sound/soc/blackfin/bf5xx-i2s.c | 22 +- sound/soc/blackfin/bf5xx-i2s.h | 2 +- sound/soc/blackfin/bf5xx-sport.c | 2 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 9 +- sound/soc/blackfin/bf5xx-tdm.c | 45 +- sound/soc/blackfin/bf5xx-tdm.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad1938.c | 1 - sound/soc/codecs/ak4671.c | 825 +++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 ++++ sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 4 +- sound/soc/codecs/wm8580.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8900.c | 4 +- sound/soc/codecs/wm8940.c | 4 +- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8974.c | 5 +- sound/soc/codecs/wm8990.c | 4 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm9713.c | 4 +- sound/soc/davinci/Kconfig | 4 +- sound/soc/davinci/davinci-evm.c | 7 +- sound/soc/davinci/davinci-mcasp.c | 24 +- sound/soc/fsl/mpc5200_dma.c | 33 +- sound/soc/imx/mx27vis_wm8974.c | 2 +- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/pxa/zylonite.c | 5 +- sound/soc/s3c24xx/neo1973_gta02_wm8753.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c-i2s-v2.c | 46 +- sound/soc/s3c24xx/s3c64xx-i2s.c | 19 +- sound/soc/s3c24xx/s3c64xx-i2s.h | 1 + sound/soc/soc-cache.c | 30 + sound/soc/soc-core.c | 32 +- sound/soc/soc-dapm.c | 26 +- 48 files changed, 1312 insertions(+), 101 deletions(-) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 20 Sep '09

20 Sep '09

The following changes since commit d4e54e871f4d2ca29df081abf8e0d5209d252979: Huang Weiyi (1): ASoC: remove unused #include <linux/version.h> are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (2): ASoC: fix typos in Blackfin headers ASoC: Blackfin I2S: add lost platform_device parameter to resume function Chaithrika U S (1): ASoC: DaVinci: Fixes to McASP configuration Cliff Cai (2): ASoC: Blackfin AC97: add a few missing multichannel define handling ASoC: Blackfin I2S: fix resuming when device hasn't been used Mike Frysinger (1): ASoC: bf5xx-sport: the irq save/restore funcs take an unsigned long sound/soc/blackfin/bf5xx-ac97.c | 8 ++++++++ sound/soc/blackfin/bf5xx-ac97.h | 2 +- sound/soc/blackfin/bf5xx-i2s.c | 22 +++++++++------------- sound/soc/blackfin/bf5xx-i2s.h | 2 +- sound/soc/blackfin/bf5xx-sport.c | 2 +- sound/soc/davinci/davinci-mcasp.c | 24 ++++++++++++++++++++++-- 6 files changed, 42 insertions(+), 18 deletions(-)

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