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

[alsa-devel] [PATCH 5/16] ALSA: VIA HDA: Fix side channel mute invalid issue
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Fix side channel mute invalid issue. Modify side_mute_channel() and update_side_mute_status() functions to fix invalid side channel mute issue of VT2002P, VT1812 and VT1802 codecs. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 24 +++++++++++++++++++----- 1 file changed, 19 insertions(+), 5 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -698,6 +698,9 @@ case VT1709_10CH: return 0x29; case VT1708B_8CH: /* fall thru */ case VT1708S: return 0x27; + case VT2002P: return 0x19; + case VT1802: return 0x15; + case VT1812: return 0x15; default: return 0; } } @@ -706,13 +709,24 @@ { /* mute side channel */ struct via_spec *spec = codec->spec; - unsigned int parm = spec->hp_independent_mode - ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; + unsigned int parm; hda_nid_t sw3 = side_mute_channel(spec); - if (sw3) - snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, - parm); + if (sw3) { + if (spec->codec_type == VT2002P || + spec->codec_type == VT1802 || + spec->codec_type == VT1812) + parm = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + else + parm = spec->hp_independent_mode ? + AMP_OUT_MUTE : AMP_OUT_UNMUTE; + snd_hda_codec_write(codec, sw3, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + if (spec->codec_type == VT1812) + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm); + } return 0; }

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[alsa-devel] [PATCH 3/16] ALSA: VIA HDA: Update unsolicited event function
by Lydia Wang 21 Mar '11

21 Mar '11

From: Lydia Wang <lydiawang(a)viatech.com.cn> Subject: ALSA: VIA HDA: Update unsolicited event function. Update unsolicited event function via_unsol_event() to process more unsolicited events. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> --- sound/pci/hda/patch_via.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -224,12 +224,12 @@ return codec_type; }; +#define VIA_JACK_EVENT 0x20 #define VIA_HP_EVENT 0x01 #define VIA_GPIO_EVENT 0x02 -#define VIA_JACK_EVENT 0x04 -#define VIA_MONO_EVENT 0x08 -#define VIA_SPEAKER_EVENT 0x10 -#define VIA_BIND_HP_EVENT 0x20 +#define VIA_MONO_EVENT 0x03 +#define VIA_SPEAKER_EVENT 0x04 +#define VIA_BIND_HP_EVENT 0x05 enum { VIA_CTL_WIDGET_VOL, @@ -1705,17 +1705,21 @@ unsigned int res) { res >>= 26; - if (res & VIA_HP_EVENT) - via_hp_automute(codec); - if (res & VIA_GPIO_EVENT) - via_gpio_control(codec); + if (res & VIA_JACK_EVENT) set_widgets_power_state(codec); - if (res & VIA_MONO_EVENT) + + res &= ~VIA_JACK_EVENT; + + if (res == VIA_HP_EVENT) + via_hp_automute(codec); + else if (res == VIA_GPIO_EVENT) + via_gpio_control(codec); + else if (res == VIA_MONO_EVENT) via_mono_automute(codec); - if (res & VIA_SPEAKER_EVENT) + else if (res == VIA_SPEAKER_EVENT) via_speaker_automute(codec); - if (res & VIA_BIND_HP_EVENT) + else if (res == VIA_BIND_HP_EVENT) via_hp_bind_automute(codec); }

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[alsa-devel] [PATCH 0/16] ALSA: HDA VIA: patch series
by Lydia Wang 21 Mar '11

21 Mar '11

Following patch series mainly includes: -support VT1802, VT1705,... -new power management functions of different codecs -fix some issues of old driver

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Re: [alsa-devel] VIA VT1708 bugs
by Raymond Yau 21 Mar '11

21 Mar '11

>> *Independent* headphone >> >> http://www.viaarena.com/forums/showthread.p*hp*?t=41015 <http://www.viaarena.com/forums/showthread.php?t=41015> >Ok, that makes some sense. It translates to me as >* With *independent* *HP* turned off - activates auto-mute between >speaker-out/line-out and headphone out. Note the difference between VIA >and Realtek/Sigmatel here as line-out are auto-muted as well. >* With *independent* *HP* turned on - headphone out and speaker-out/line-out >should take output from different PCM subdevices. Practically, it is more complex than the above since the desktop motherboard can have 3 , 5 or 6 audio jacks at rear panel since those hda codecs only have 8 channels instead of 10 Enable Independent HP on a 6stack model should decrease spec->mult_out.max_channels from 8 to 6 for subdevice 0 For 5 jacks model , the driver need to retask blue jack as side only when independent HP switch is off To support 8 channels for a 3stack model need to retask the front HP as side

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[alsa-devel] QasMixer 0.10.0 released
by Sebastian H. 20 Mar '11

20 Mar '11

Hello Alsa and Qt users Version 0.10.0 of QasMixer is now available. This is a polishing release Changes: * Start minimized if a session is restored * Alsa configuration browser in a separate top menu entry * Slider style tweaks - curves instead of triangles * Check button style tweaks Homepage with more informations and screenshots http://xwmw.org/qasmixer Files are hosted at sourceforge http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

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[alsa-devel] [PATCH - Set EAPD for Realtek ALC665 (Vendor Id: 0x10eSet EAPD for Realtek ALC665 (Vendor Id: 0x10ec0665).
by Andres Mejia 20 Mar '11

20 Mar '11

Signed-off-by: Andres Mejia <mcitadel(a)gmail.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 50855cf..be910fa6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1264,6 +1264,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 1); @@ -4236,6 +4237,7 @@ static void alc_power_eapd(struct hda_codec *codec) case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: + case 0x10ec0665: case 0x10ec0862: case 0x10ec0889: set_eapd(codec, 0x14, 0); -- 1.7.4.1

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[alsa-devel] snd_pcm_writei returns -EAGAIN after poll
by Ivan Sorokin 20 Mar '11

20 Mar '11

Hello! I use slightly modified test/pcm.c in write_and_poll mode. My program looks like this: snd_pcm_open with SND_PCM_NONBLOCK snd_pcm_set_params infinite loop poll for snd_pcm_poll_descriptors snd_pcm_poll_descriptors_revents if POLLOUT flag is set snd_pcm_writei Everything works, but snd_pcm_writei sometimes returns -EAGAIN. The probability of when snd_pcm_writei returns -EAGAIN depends on buffer size that I pass to it. ExperimentallyI found out that when I call snd_pcm_writei with 940 bytes buffers (maybe something related to period size?) it never returns -EAGAIN. Why poll returns when there is no place for new data? Is it ok to ignore EAGAIN and just start another poll? Is there any recommended buffer size for snd_pcm_writei? My test program: https://github.com/sorokin/asoundpp/blob/master/sine_async.cpp (94 LOC)

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[alsa-devel] ALSA/pulseaudio problem workaround
by Craig Bourne 19 Mar '11

19 Mar '11

Here is a workaround for the problem I described in my email to you yesterday: Background My past experience with ALSA, JACK and many of the Linux audio tools that depend on them is that often some few applications in the tool-chain of a complex task will not work properly (or at all) unless invoked with *root* privileges. One is now frustrated in getting such things to work Fedora 14 as they have for the past decade or so in that the most basic tools (e.g., alsactl, QjackCtl) no longer work for root. Indeed, audio also does not work for unprivileged users on Fedora 14 either (though this evidently fails at a different level and will be taken up in a subsequent communication). Symptoms alsactl fails carping about pulseaudio, audacity can be started on its own but no longer wants to cooperate with JACK and ALSA resulting in horribly distorted sound from something horked together in the background as a "favor" to the user, QjackCtl runs, produces messages, and blinks its cute little meter display but evidently does this off on the sidelines reporting 0 XRUNS even as every little jot and tiddle of computation produces a surprising audio result. Strategy Based on my observation that PULSEAUDIO is evidently being allowed to interpose itself between the user (even the superuser) and ALSA, and since this user has been kept largely ignorant of the workings of all this and does not know the secret handshake, I will attempt to at least minimize the damage to audio and regain some measure of control while not attempting to disable the PULSEAUDIO/ALSA cabal. Workaround Install the following if they are not already on your system: - avahi - padevchooser - pulseaudio-libs-zeroconf - pulseaudio-module-zeroconf Enable and start the Avahi Daemon Create these two shell scripts (that will be used by QjackCtl), make them executable, and place them in a directory where they will stay to be invoked each time QjackCtl is started: PAjack_startup #!/bin/bash # Name: PAjack_startup pulseaudio -D --system #perhaps more fastidious than the above line is #pulseaudio -D --system --disallow-exit --disallow-module-loading padevchooser& PAjack_shutdown #!/bin/bash #Name: PAjack_shutdown pulseaudio --kill killall padevchooser Invoke QjackCtl click Setup select the Options tab tick the box "Execute script after Startup" and enter ~/bin/PAjack_startup [Note: using, of course the path to your script] tick the box "Execute script on Shutdown" and enter ~/bin/PAjack_shutdown [Note: as before, using your path to the file] Quit QjackCtl -- your scripts will now be invoked the next time it is used. A partial workaround for the unprivileged user, at least to the point of getting QjackCtl and audacious working places a slightly modified PAjack_startup script in a place where QjackCtl can find it when it starts with the changes used above for root (e.g., insure that both acounts have a subdirectory named "bin" in their home directory and have QjackCtl look for ~/bin/PAjack_startup when it is invoked). Here is the script for unprivileged users: #!/bin/bash #Name: PAjack_startup pulseaudio -D padevchooser& Limitations This is not a general solution to the ALSA/PULSEAUDIO problems I am experiencing but it is a workaround that at least allows me to have some minimal functionality back from basic audio tools that I have come to rely on in my work as a composer. -- Comradely, Craig Bourne

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[alsa-devel] syslog requested reporting this to ALSA developers (ubuntu 10.04 inside virtualbox)
by Jason E. Aten 19 Mar '11

19 Mar '11

from my syslog, it requests a report to ALSA developers. Let me know if I can provide further assistance. -Jason $ tail /var/log/syslog Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: ALSA woke us up to read new data from the device, but there was actually nothing to read! Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:25 virtub pulseaudio[1038]: alsa-source.c: We were woken up with POLLIN set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:26 virtub pulseaudio[1038]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:58:29 virtub gdm-session-worker[1030]: GLib-GObject-CRITICAL: g_value_get_boolean: assertion `G_VALUE_HOLDS_BOOLEAN (value)' failed Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1256 of process 1256 (n/a) owned by '1000' high priority at nice level -11. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 4 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1257 of process 1256 (n/a) owned by '1000' RT at priority 5. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 5 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-util.c: Disabling timer-based scheduling because running inside a VM. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Sucessfully made thread 1258 of process 1256 (n/a) owned by '1000' RT at priority 5. Mar 17 13:58:35 virtub rtkit-daemon[1043]: Supervising 6 threads of 2 processes of 2 users. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_intel8x0'. Please report this issue to the ALSA developers. Mar 17 13:58:35 virtub pulseaudio[1256]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Mar 17 13:59:40 virtub AptDaemon: INFO: Initializing daemon root@virtub:/var/log# uname -a Linux virtub 2.6.32-29-generic #58-Ubuntu SMP Fri Feb 11 20:52:10 UTC 2011 x86_64 GNU/Linux root@virtub:/var/log# dpkg -s pulseaudio Package: pulseaudio Status: install ok installed Priority: optional Section: sound Installed-Size: 4540 Maintainer: Ubuntu Core Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Architecture: amd64 Version: 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14 Replaces: pulseaudio-module-hal, pulseaudio-module-rygel-media-server (<< 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14), pulseaudio-module-udev (<< 1:0.9.22~0.9.21+stable-qu\ eue-32-g8478-0ubuntu14) Provides: pulseaudio-module-hal, pulseaudio-module-rygel-media-server, pulseaudio-module-udev Depends: libasound2 (>> 1.0.22), libc6 (>= 2.9), libdbus-1-3 (>= 1.0.2), libice6 (>= 1:1.0.0), libltdl7 (>= 2.2.6b), libpulse0 (= 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu1\ 4), libsamplerate0, libsm6, libsndfile1 (>= 1.0.20), libspeexdsp1 (>= 1.2~beta3.2-1), libtdb1 (>= 1.1.2~git20080520), libudev0 (>= 147), libx11-6 (>= 0), libxtst6, adduser, lsb\ -base (>= 3), consolekit, libasound2-plugins, pulseaudio-utils Recommends: pulseaudio-module-x11, gstreamer0.10-pulseaudio, pulseaudio-esound-compat, rtkit Suggests: pavumeter, paman, paprefs Conflicts: libltdl3 (<< 1.5.24-1), pulseaudio-module-rygel-media-server (<< 1:0.9.22~0.9.21+stable-queue-32-g8478-0ubuntu14), pulseaudio-module-udev (<< 1:0.9.22~0.9.21+stable-\ queue-32-g8478-0ubuntu14) Conffiles: /etc/default/pulseaudio 5ac1e3004508adbb816e9b1af352617e /etc/init.d/pulseaudio bd6ccc065d494d8baac1d823521833b4 /etc/pulse/default.pa d2d623c97aef9f81f5b55f2d7e736c87 /etc/pulse/system.pa 182fbc3ba3449027c4d1e39a150c8396 /etc/pulse/daemon.conf ed3746b9cd3ba486184128fd611b73c9 /etc/dbus-1/system.d/pulseaudio-system.conf 69d56dc5342bf589a2e199541709a452 /etc/xdg/autostart/pulseaudio.desktop 1310d377ad0d534f699e864037f66bd2 Description: PulseAudio sound server PulseAudio, previously known as Polypaudio, is a sound server for POSIX and WIN32 systems. It is a drop in replacement for the ESD sound server with much better latency, mixing/re-sampling quality and overall architecture. . These are some of PulseAudio's features: . * High quality software mixing of multiple audio streams with support for more than one sink/source. May be used to combine multiple sound cards into one (with sample rate adjustment). . * Wide range of supported client libraries. ESD, ALSA, oss, libao and GStreamer client applications are supported as-is. Native PulseAudio plug-ins are also available for xmms and mplayer. . * Good low latency behaviour and very accurate latency measurement for playback and recording. Ability to fully synchronize multiple playback streams. . * Network transparency, allowing an application to play back or record audio on a different machine than the one it is running on. . * Extensible plug-in architecture with plug-ins for jackd, multicast-rtp lirc and avahi, just to name a few. . This package contains the daemon and basic module set. Homepage: http://www.pulseaudio.org Original-Maintainer: Pulseaudio maintenance team < pkg-pulseaudio-devel(a)lists.alioth.debian.org> root@virtub:/var/log#

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Re: [alsa-devel] Microphone not working on PCI ID 8086:3b56
by Jeff Chua 18 Mar '11

18 Mar '11

On Tue, Sep 28, 2010 at 9:42 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > Enable all CONFIG_SND_HDA_*. Done that. Still no mic. On Tue, Sep 28, 2010 at 9:42 PM, Wu Fengguang <fengguang.wu(a)intel.com> wrote: > It's easier to enable all of them. They are small modules. Tried just about everything, still no mic. Sounds works in all cases. I found that with SND_HDA_CODEC_INTELHDMI selected, SND_DYNAMIC_MINORS will be enabled by default which won't work in my case as I'm not using udev. Here's a patch to take away the default so non-udev will still works. Do I really need udev to get mic to work? Thanks, Jeff --- lx/sound/pci/hda/Kconfig.org 2010-09-14 16:34:52.000000000 +0800 +++ lx/sound/pci/hda/Kconfig 2010-09-14 16:35:09.000000000 +0800 @@ -145,7 +145,6 @@ config SND_HDA_CODEC_INTELHDMI bool "Build INTEL HDMI HD-audio codec support" - select SND_DYNAMIC_MINORS default y help Say Y here to include INTEL HDMI HD-audio codec support in

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