September 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa-lib 1.0.21a release
by Jaroslav Kysela 22 Sep '09

22 Sep '09

Hello, I made a quick release 1.0.21a of the alsa-lib package to fix regression for the configuration parsing (makes some device names unavailable). These patches were included: 19abc37 PCM - Change the hw_params determination order d1b70af Change dmix.conf to accept user configuration from defaults.dmix.<driver_id>.xxx eb9ae58 Revert "Fix driver conf parsing in snd_config_hook_load_for_all_cards()" f0a32a6 mixer: Add Speaker and Beep names to the weight list Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] (no subject)
by Delusion's Master 22 Sep '09

22 Sep '09

Hi, I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Windows Live™: Keep your life in sync. Check it out! http://windowslive.com/explore?ocid=TXT_TAGLM_WL_t1_allup_explore_012009

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[alsa-devel] [PATCH] wm8753: fix mapping when MONOMIX is set to Stereo
by Phil Vandry 22 Sep '09

22 Sep '09

When MONOMIX is set to Stereo, Left PGA was not powered on but should be. Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue. Signed-off-by: Phil Vandry <vandry(a)TZoNE.ORG> --- sound/soc/codecs/wm8753.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d80d414..5ad677c 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -595,6 +595,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Mono Capture mixer-mux */ {"Capture Right Mixer", "Stereo", "Capture Right Mux"}, + {"Capture Left Mixer", "Stereo", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"}, {"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"}, {"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},

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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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