December 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Add HP Acacia detection
by Chris Bagwell 01 Jan '09

01 Jan '09

Add automatic mapping of HP Acacia motherboards to 3stack-hp. Allows for greater then 2 channel audio by enabling Channel Mode option in mixer. Motherboard specs: http://h10025.www1.hp.com/ewfrf/wc/document?docname=c01321559&lc=en&dlc=en&… Signed-off-by: Chris Bagwell <chris at cnpbagwell dot com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..410be84 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -8444,6 +8444,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),

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[alsa-devel] [PATCH 0/2] ASoC: TWL4030: Add DAPM based capture implementation
by Peter Ujfalusi 31 Dec '08

31 Dec '08

Hello, It has been a long time... Finally I have managed to get the DAPM based capture implementation working. I have not counted, but there were several dead end trials before I ended up with the current implementation. I have tested the capture implementation with the following configs: digimic0, Headset mic, AUX connected (not on the same board). So far the capture path behaved in a sane way, powering/enabling only the needed components in the capture path. All of the user controls are showing under the 'Capture' tab in alsamixer. Example for the board file maintainers to get the capture working: Let's say the board have the following input selection: Headset mic, AUXL/R (Line in), digimic0 static const struct snd_soc_dapm_widget BOARD_dapm_widgets[] = { SND_SOC_DAPM_MIC("Digital Mic", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_LINE("Line in", NULL); }; static const struct snd_soc_dapm_route audio_map[] = { {"DIGIMIC0", NULL, "Mic Bias 1"}, {"Mic Bias 1", NULL, "Digital Mic"}, {"HSMIC", NULL, "Headset Mic Bias"}, {"Headset Mic Bias", NULL, "Headset Mic"}, {"AUXL", NULL, "Line in"}, {"AUXR", NULL, "Line in"}, }; To record from the Headset mic: amixer sset 'Analog Left' 'Headset mic' # select the Headset for Analog left amixer sset 'TX1' 'Analog' # Select the Analog path for TX1 amixer sset 'TX2' 'Digimic1' # will disable the TX2 path, since digimic1 is # not connected amixer sset 'Analog' 2,2 # set the analog capture volume Have a Happy New Year! --- Peter Ujfalusi (2): ASoC: TWL4030: Make the enum filter generic for twl4030 ASoC: TWL4030: DAPM based capture implementation sound/soc/codecs/twl4030.c | 377 ++++++++++++++++++++++---------------------- sound/soc/codecs/twl4030.h | 7 + 2 files changed, 196 insertions(+), 188 deletions(-)

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[alsa-devel] Is Asoc suitable for my device?
by Xuesong Yang 31 Dec '08

31 Dec '08

hi all: I'm new to ALSA and still study it. My problem is I don't know if it suitable for my device use Asoc. In my platform, there is only a I2S bypass module, It has a route, select one of the I2S source as input,(from HDMI or cpu data) delay the data for some time and then output it to the backend. In theory, this device is very sample, but I find I can't get a easy way to implement it in ALSA, I'm lost in ALSA. Any suggestion will be greatly appreciated. Thanks.

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[alsa-devel] problems with 2 outputs on snd-hda-intel
by Alex Ander 29 Dec '08

29 Dec '08

My configuration: http://www.alsa-project.org/db/?f=e33841b3aa560b031a3723ecf2e2c5040d2c1a16 Ever since I came back from the holidays I've been having this problem: My laptop has 3 jacks: 2 line out and 1 line in that can be switched to out. But, I can only get sound over the laptop-speakers and the line-in switched to out. The speaker stops playing when a jack is connected to one of the two outs, but I don't get any sound over these outputs. I have checked all mixer-settings in alsamixer, nothing works (except switching the mic to lineout, which doesn't fix the other 2 outs) As you can see in alsa-info I have forced a model-type to the module, I have found this on http://www.mjmwired.net/kernel/Documentation/sound/alsa/ALSA-Configuration.… I have tried all 4 modules listed at line 1015 (which should correspond to the SigmaTel STAC9228) of which 3stack seems to be the one set by default (with empty /etc/modprobe.conf) and is the only one that works with the mic-as-output option. I can give more info if you need it.

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[alsa-devel] MIDI In on Audigy 2 ZS Live Drive
by Carl Bauer 28 Dec '08

28 Dec '08

On the Live Drive CD Bay thingy: MIDI in does not work. MIDI out (and everything else on the Live Drive) does work. For details, I have posted this problem before: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg23618.html http://forums.fedoraforum.org/showthread.php?t=198240 and seen it elsewhere: http://ubuntuforums.org/showthread.php?t=476633 I don't know if it's an ALSA bug or a kernel thing. My aadebug output is in the first of the above links. Or, my alsa-info.sh output is here: http://www.alsa-project.org/db/?f=5990a1a1b7476778a6cc82fff91997567db43a14 Thanks. Forgive me if this is the wrong list to post to. -Carl

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[alsa-devel] [PATCH] alsa-info.sh - added extra checks
by Travis Place 28 Dec '08

28 Dec '08

Changelog: Added checks for Pulseaudio/Esound/aRts 'sound servers', Check if they are 1) installed 2) running A few small changes for the locale, and timezone - keep things more uniform. Bumped version to 0.4.53 Signed-off-by: Travis Place <wishie(a)wishie.net>

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[alsa-devel] S/PDIF-Out on Asus M3A-H/HDMI with ALC1200
by Thomas Schneider 25 Dec '08

25 Dec '08

Hi, i am looking for a way to make Digital-Audio work with the external and internal S/PDIF-Output of my Motherboard Asus M3A-H/HDMI with ALC1200 Soundard. I use the latest unstable snapshot and i can´t make it work at the same time. On the external S/PDIF-Output is my Digital Receiver connected, on the internal one my GeForce 8500GT to get sound over HDMI on my LCT-TV. When i load the driver without any model-options my Soundcard is detected as Asus P5Q-EM HDMI ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:2716: hda_codec: model 'asus-p5q' is selected for config 1043:82fe (Asus P5Q-EM HDMI) [ 6.721700] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Headphone Playback Volume, skipped [ 6.721714] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Speaker Playback Volume, skipped [ 6.721716] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Mono Playback Volume, skipped [ 6.721719] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Line-Out Playback Volume, skipped [ 6.721721] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave PCM Playback Volume, skipped [ 6.721729] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Speaker Playback Switch, skipped [ 6.721731] ALSA /usr/local/src/alsa-driver-unstable/pci/hda/hda_codec.c:1339: Cannot find slave Mono Playback Switch, skipped Now i have sound on my LCD-TV over HDMI (internal S/PDIF-Out) When i load the driver with model=6stack-dig i have sound on my Digital-Receiver (external S/PDIF-Out). In the mailinglist-archive or in the doku i have seen, that the only difference between this models is the DIGOUT_NID asus-p5q = 0x10 6stack-dig = 0x06 I there a way to get both output to work at the same time so i have sound on my LCD-TV and my Digital-Receiver ? Regards Thomas

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[alsa-devel] [PATCH 0/8]ALSA: ASoc: DaVinci: cleanup
by Troy Kisky 24 Dec '08

24 Dec '08

Sorry, for the long series to do something so simple, but there were unresolved objections when I first posted this to the davinci list. Hopefully, having this split into tiny pieces will make resolving those differences easier.

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[alsa-devel] [PATCH] ALSA: hda - Add missing terminators in patch_sigmatel.c
by Herton Ronaldo Krzesinski 24 Dec '08

24 Dec '08

Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 71d3a77..35b83dc 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -841,6 +841,7 @@ static struct hda_verb stac92hd83xxx_core_init[] = { /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, + {} }; static struct hda_verb stac92hd71bxx_core_init[] = { @@ -850,6 +851,7 @@ static struct hda_verb stac92hd71bxx_core_init[] = { { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} }; #define HD_DISABLE_PORTF 2 @@ -1681,6 +1683,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + {} /* terminator */ }; static unsigned int ref92hd71bxx_pin_configs[11] = {

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[alsa-devel] Urgent Help Needed:ALsa Querries
by SUBHRANIL CHOUDHURY 23 Dec '08

23 Dec '08

I have some doubts on the ALSA programming, 1)My ALSA driver supports minimum Period Bytes size as 32 bytes, but when i set the period size as 512 bytes it is always setting to 1024 bytes.I am using the function "snd_pcm_hw_params_set_period_size_near()" . What might be wrong here?Is the function what i am using correct? 2)How the Library functions and the Driver functions interact, what is the interface between the library functions and driver functions?Is there any document which explains this. I got a document which explains how to write ALSA driver.But it does not mention the interface with the library. 3)How Can we achive MIC and LINEIN switching using the mixer?What library function do i need to use to achive this? 4)If i want to get the Timestamp for each buffer captured, what library function should be used? 5)I am trying to capture and play the same audio (ALSA duplex). When i capture and store it in a file and play from the file i am able to hear the sound properly. If i capture in a buffer and play the same buffer directly it says Broken PIPE ERROR in the playout. I tried the capture and playout in two different threads, i am able to hear the sound and i am not getting any Broken PIPE error. But along with the audio i am hearing some noise. Do we need to use some kind of synchronisation between the threads for capture and playout? Any help on these things is appreciated. Thank you in advance. Regards, Subhranil

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