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[PATCH 1/1] ALSA: hda: Fixup commit 79c944ad (caused codec initialization failure)
by David Henningsson 27 Jul '10

27 Jul '10

Commit 79c944ad caused codecs not to be detected in some cases, found on several Realtek codecs. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_codec.c | 13 +++++++------ 1 files changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8cd5da8..710f61a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -724,23 +724,24 @@ static int get_codec_name(struct hda_codec *codec) */ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) { - int i, total_nodes, function_id; + int i, total_nodes, function_id, param_val; hda_nid_t nid; total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - function_id = snd_hda_param_read(codec, nid, + param_val = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); + function_id = param_val & 0xff; switch (function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; - codec->afg_function_id = function_id & 0xff; - codec->afg_unsol = (function_id >> 8) & 1; + codec->afg_function_id = function_id; + codec->afg_unsol = (param_val >> 8) & 1; break; case AC_GRP_MODEM_FUNCTION: codec->mfg = nid; - codec->mfg_function_id = function_id & 0xff; - codec->mfg_unsol = (function_id >> 8) & 1; + codec->mfg_function_id = function_id; + codec->mfg_unsol = (param_val >> 8) & 1; break; default: break; -- 1.7.0.4 --------------040206080406060608060907--

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[alsa-devel] asoc multi-component: CPU DAI name is wrong
by Timur Tabi 26 Jul '10

26 Jul '10

Liam, I have this in my machine driver: machine_data->dai[0].cpu_dai_name = strrchr(np->full_name, '/') + 1; printk(KERN_INFO "%s:%u cpu_dai_name=%s\n", __func__, __LINE__, machine_data->dai[0].cpu_dai_name); ... platform_set_drvdata(sound_device, &machine_data->card); ret = platform_device_add(sound_device); which displays this message: mpc8610_hpcd_probe:316 cpu_dai_name=ssi@16000 So my machine driver says the CPU DAI name is "ssi@16000", and my CPU driver does the same thing. However, I put some printks in soc_bind_dai_link, and this is what I see: soc_bind_dai_link:1136 cpu_dai->name=e0016000.ssi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1136 cpu_dai->name=cs4270-hifi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1159 codec->name=0-004f dai_link->codec_name=cs4270-codec.0-004f soc_bind_dai_link:1193 platform->name=e0021180.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@0 soc_bind_dai_link:1193 platform->name=e0021100.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@0 soc_bind_dai_link:1136 cpu_dai->name=e0016000.ssi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1136 cpu_dai->name=cs4270-hifi dai_link->cpu_dai_name=ssi@16000 soc_bind_dai_link:1159 codec->name=0-004f dai_link->codec_name=cs4270-codec.0-004f soc_bind_dai_link:1193 platform->name=e0021180.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@1 soc_bind_dai_link:1193 platform->name=e0021100.dma-channel dai_link->platform_name=/soc@e0000000/dma@21300/dma-channel@1 Where in the world is "e0016000.ssi" coming from? Why is ASoC ignoring my .cpu_dai_name value?

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[alsa-devel] hda_intel: too quiet sound (regression)
by Jiri Slaby 26 Jul '10

26 Jul '10

Hi, after I upgraded from 2.6.34-rc5-mm1 to 2.6.35-rc5-mm1 (there were other issues in between them), the sound from my soundcard is very very quiet. alsa-info: http://www.alsa-project.org/db/?f=dcadb29a2d5d047ecf5545792ac3239dc0c8c7f0 Any ideas? thanks, -- js

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[alsa-devel] announcing envy24control, mudita (*) edition.
by Niels Mayer 26 Jul '10

26 Jul '10

Summary of updates from envy24control 0.6.0 (GIT HEAD) to "1.0.0": (0) After a decade, incremented version to 1.0.0 (**) (1) Implemented missing "Peak Hold" functionality in meters and reimplemented meters for increased efficiency and lower X resource usage. (see http://www.linuxaudio.org/mailarchive/lad/2010/7/12/171535 & https://bugzilla.redhat.com/show_bug.cgi?id=602903 ) (2) All volumes are represented as decibels, including the 0 to -48dB range of the hardware peak-meters, the 0 -to- -144dB attenuation for all inputs to the digital mixer, the 0 -to- -63dB attenuation of the analog DAC, and the +18 -to- -63dB attenuation/amplification of the analog ADC. (3) All gtk "scale" widgets have dB legends; the "PageUp" "PageDown" keys allow rapid movement between the marked levels. (4) Got rid of myriad compile warnings and other minor fixes across codebase. Some screenshots: http://nielsmayer.com/envy24control/Screenshot-Envy24Control-AnalogVolume.p… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorInputs.… http://nielsmayer.com/envy24control/Screenshot-Envy24Control-MonitorPCM.png ------------ To the ALSA project: please consider this patch to alsa-tools' envy24control (**): http://nielsmayer.com/envy24control/envy24control-0.6-to-1.0.patch (patch to 'envy24control' from GIT trunk/head of alsa-tools) http://nielsmayer.com/envy24control/envy24control-1.0.README (summary of changes from 0.6.0 to 1.0.0) ------------ Those wanting to compile directly, or run a 64 bit linux binary I've built: http://nielsmayer.com/envy24control/envy24control-1.0.tar.gz (full directory, just follow README directions to build/install) http://nielsmayer.com/envy24control/envy24control-1.0-fc12-x86_64.tar.gz (x86_64 binary that should work on fedora12 and equivalent OpenSuse release) I'd appreciate any testing results or comments on this "1.0.0" release. In particular, I'd like some assurance that the dB markings on sliders in "Analog Volume" panel are correct (compared to values reported by 'alsamixer'). I'm looking for testing with following devices (as I think my testing covers code for M-Audio Delta 44 & Delta 66, Terratec Dmx6fire & EWX2496) specifically: M-Audio Delta 1010, M-Audio Audiophile 2496, M-Audio Delta 1010LT TerraTec EWS 88MT, TerraTec EWS 88D, TerraTec Phase 88, Hoontech SoundTrack DSP 24 (all variants). Thanks, Niels http://nielsmayer.com PS: (*) http://en.wikipedia.org/wiki/Envy#In_philosophy :: "mudita, taking joy in the good fortune of another. This virtue is considered the antidote to envy and the opposite of schadenfreude." (**) http://git.alsa-project.org/?p=alsa-tools.git;a=tree;f=envy24control;h=d5a5…

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Re: [alsa-devel] [Fwd: Re: [pulseaudio-discuss] pulseaudio debug mode]
by Raymond Yau 26 Jul '10

26 Jul '10

2010/7/26 Chris <cpollock(a)embarqmail.com> > Raymond, attached is a post I made today to the pulseaudio list. > > From: Chris <cpollock(a)embarqmail.com> > To: pulseaudio-discuss(a)mail.0pointer.de > Date: Sun, 25 Jul 2010 12:15:02 -0500 > Subject: Re: [pulseaudio-discuss] pulseaudio debug mode > On Fri, 2010-07-23 at 08:58 +0100, Colin Guthrie wrote: > > 'Twas brillig, and Chris at 23/07/10 02:04 did gyre and gimble: > > > What is the best way to start PA for debugging and still have all the > > > usual clients running? > > > > If you mean having all the clients connect (e.g. applications with > > libcanberra support or similar for sound events), then there are > > basically two ways. > > > > The first is as Luke suggests. These clients will automatically > > reconnect to PA if they need to (provided you have a vaguely recent > > libcanberra), after it is restarted and run in debug mode. > > > > Alternatively you can simply set debug-level to "debug" in daemon.conf > > (in /etc/pulse or ~/.pulse), and then "grep pulseaudio /var/log/messages" > > > > Col > > > > Colin, link below is for debug output also some other output. Anything > look out of place that would cause the overruns? > > http://pastebin.com/ZWSWmXZt > -- > Most likely you will need to set debug-level to "debug" in daemon.conf logout and login , and check the /var/log/messages I suspect it occur after PA server abort and auto spawn , so please don't just post the PA log starting from "asyncq overrun" , you need to post from the last two PA startup sequence before the error occur since we need to know why PA auto spawn

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[alsa-devel] [PATCH] eukrea-tlv320: add support for cpuimx35sd
by Eric Bénard 26 Jul '10

26 Jul '10

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 2 +- sound/soc/imx/eukrea-tlv320.c | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 2f0d6d3..52dac5e 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -32,6 +32,6 @@ config SND_SOC_EUKREA_TLV320 select SND_SOC_TLV320AIC23 help Enable I2S based access to the TLV320AIC23B codec attached - to the SSI4 interface + to the SSI interface endif # SND_IMX_SOC diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index 45f5e4b..f15dfbd 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -101,7 +101,8 @@ static int __init eukrea_tlv320_init(void) { int ret; - if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()) + if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd() + && !machine_is_eukrea_cpuimx35sd()) /* return happy. We might run on a totally different machine */ return 0; -- 1.6.3.3

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Re: [alsa-devel] "Playback open error: Inappropriate ioctl for device" in alsa-lib-1.0.22/test/*.c
by John Simpson 26 Jul '10

26 Jul '10

On 7/21/10, John Simpson <john(a)swajime.com> wrote: > On 7/21/10, Clemens Ladisch <clemens(a)ladisch.de> wrote: >> John Simpson wrote: >>> Playback open error: Inappropriate ioctl for device >> >> Make sure that both your kernel and your alsa-lib use the correct >> compiler and compiler flags (where "correct" usually means "whatever >> the vendor's distribution uses"). > > You've gone way over my head ... > How do I find out what flags were used to compile the kernel? > >> >>> This time, checking out https://vcs.maemo.org/svn/dsp-alsa >> >> It would be nice to know what the changes from the original alsa-lib >> are; in theory, they should go upstream. > > I agree ... but it looks like that is an abandoned project; at > https://garage.maemo.org/plugins/scmsvn/viewcvs.php/?root=dsp-alsa it > says the last modification was 3 years ago. The text file "version" > says 1.0.10 > > I found that "apt-get source libasound2" gives me a newer version: > [1|user at Nokia-N810-43-7|~/src/alsa-lib-1.0.14sa/test]cat > ../../alsa-lib_1.0.14sa-2maemo2.dsc > Format: 1.0 > Source: alsa-lib > Version: 1.0.14sa-2maemo2 > Binary: lib32asound2-dev, lib64asound2, lib32asound2, libasound2-dev, > libasound2, libasound2-doc, libasound2-dbg, lib64asound2-dev > Maintainer: Debian ALSA Maintainers <pkg-alsa-devel at > lists.alioth.debian.org> > Architecture: any > Standards-Version: 3.7.2 > Build-Depends: debhelper (>= 4.2.32), quilt (>= 0.40), autotools-dev, > doxygen, libc6-dev-powerpc [ppc64], libc6-dev-i386 [amd64], > libc6-dev-ppc64 [powerpc], libc6-dev-s390x [s390], libc6-dev-sparc64 > [sparc], lib32gcc1 [amd64 ppc64], libc6-dev (>= 2.5) [armel] > Vcs-Browser: http://svn.debian.org/wsvn/pkg-alsa/trunk/alsa-lib/ > Uploaders: Jordi Mallach <jordi at debian.org>, Mikael Magnusson > <mikma at users.sourceforge.net>, Elimar Riesebieter <riesebie at > lxtec.de>, > Marc-Andre Lureau <marc-andre.lureau at nokia.com> > Vcs-Svn: svn://svn.debian.org/pkg-alsa/trunk/alsa-lib > Files: > 5542a18a1479e4de4b9890e30e94588e 1099165 alsa-lib_1.0.14sa.orig.tar.gz > fc02aa3dbf2f66fb090e02ada703cc1d 47455 alsa-lib_1.0.14sa-2maemo2.diff.gz > > This version seems to install correctly, but still gives me no sound. > The full log of my latest effort is here -> > http://www.swajime.com/n810/alsa-lib_1.0.14sa-2maemo2.log >> >>> $ cd dsp-alsa/trunk/alsa-lib >>> #### cvscompile does not succeed >> >> Why? > > I honestly don't know. :-( It looks like a bug in m4? Here is the > log for cvscompile: > [1|root@Nokia-N810-43-7|/home/user/src/dsp-alsa/trunk/alsa-lib]cat > cvscompile > #!/bin/bash > > touch ltconfig > libtoolize --force --copy --automake > aclocal $ACLOCAL_FLAGS > autoheader > automake --foreign --copy --add-missing > touch depcomp # seems to be missing for old automake > autoconf > export CFLAGS='-O2 -Wall -W -pipe -g' > echo "CFLAGS=$CFLAGS" > echo "./configure $@" > ./configure $@ || exit 1 > unset CFLAGS > if [ -z "$CVSCOMPILE_NO_MAKE" ]; then > make > fi > [1|user@Nokia-N810-43-7|~/src/dsp-alsa/trunk/alsa-lib]bash -v ./cvscompile > #!/bin/bash > > touch ltconfig > libtoolize --force --copy --automake > aclocal $ACLOCAL_FLAGS > aclocal: configure.in: 41: macro `AM_PROG_LIBTOOL' not found in library > autoheader > configure.in:41: /usr/local/bin/m4: ERROR: Copying inserted file: No > space left on device > autom4te: /usr/local/bin/m4 failed with exit status: 1 > autoheader: /usr/local/bin/autom4te failed with exit status: 1 > automake --foreign --copy --add-missing > configure.in:41: /usr/local/bin/m4: ERROR: Copying inserted file: No > space left on device > autom4te: /usr/local/bin/m4 failed with exit status: 1 > configure.in: no proper invocation of AM_INIT_AUTOMAKE was found. > configure.in: You should verify that configure.in invokes AM_INIT_AUTOMAKE, > configure.in: that aclocal.m4 is present in the top-level directory, > configure.in: and that aclocal.m4 was recently regenerated (using aclocal). > automake: no `Makefile.am' found or specified > touch depcomp # seems to be missing for old automake > autoconf > configure.in:41: /usr/local/bin/m4: ERROR: Copying inserted file: No > space left on device > autom4te: /usr/local/bin/m4 failed with exit status: 1 > export CFLAGS='-O2 -Wall -W -pipe -g' > echo "CFLAGS=$CFLAGS" > CFLAGS=-O2 -Wall -W -pipe -g > echo "./configure $@" > ./configure > ./configure $@ || exit 1 > ./cvscompile: line 13: ./configure: No such file or directory > [1|user@Nokia-N810-43-7|~/src/dsp-alsa/trunk/alsa-lib] > >> >>> $ .libs/pcm -D 'default' -vf 440 >>> Playback device is default >>> Stream parameters are 44100Hz, S16_LE, 1 channels >>> Sine wave rate is 440.0000Hz >>> Using transfer method: write >>> >>> Still no sound :-( >> >> Mixer controls not set? (amixer and alsamixer are in alsa-utils) > > I haven't been able to find alsa-utils in any maemo/diablo repositories. I found that alsa-tools is in the fremantle repo: http://repository.maemo.org/pool/fremantle/free/a/alsa-utils/ It bricks the n810 :-( Fremantle is not compatible with the n810 as I remember.

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[alsa-devel] Realtek alc269 questions
by David Henningsson 26 Jul '10

26 Jul '10

I'm trying to fix the sound on some Dell Optiplex machines, which are running the ALC269 chip. I can "fix" them with some hda-verb commands, but that won't get me auto-mute, so I'll have to make a model (or possibly more than one) for them. Anyway, I have a pair of questions: 1) In dmesg, there is a "SKU not ready" message at startup. What is SKU, and is it a problem that it is not ready? 2) There are two DACs, and all models so far seem to use NID 0x03 for headphones and NID 0x02 for everything else. But is there a single stream going through the controller and then splitted into two on the codec side, or is just one of the DACs activated at a time, depending on the jack sensing state? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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[alsa-devel] problems about dummy alsa driver
by longer 26 Jul '10

26 Jul '10

Hi all , I've written an ALSA driver(only support 8K sample rate), which register two dummy sound cards.The main function of this driver is to exchange audio data of that two dummy sound cards. In other words , the driver route audio data of dummy sound card (PLAYBACK) to the other (CAPTURE),and vice versa. And I've written some test programs based on my ALSA driver . When the sample rate in application is set to be 8K , the driver work well. But when requiring rate conversion , the driver can't get any data. And , when tracing the code of alsa-lib (1.0.23), I found that : In case of rate conversion, It mmap an address from my driver, but writing data to another address instead of that mmaped address. I think this is why my driver don't get any audio data. But I don't know why it don't write audio data to that mmaped address. I am looking forward to your reply. And any advices will help me a lot. Thank you, longer zhou

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[alsa-devel] Compiling alsa-utils
by Pavel Hofman 26 Jul '10

26 Jul '10

Hi, after clean git clone of git://git.alsa-project.org/alsa-utils.git I got the following error: pavel@sara:~/install/alsa/alsa-utils$ autoheader configure.in:268: error: `po/Makefile.in' is already registered with AC_CONFIG_FILES. ../../lib/autoconf/status.m4:311: AC_CONFIG_FILES is expanded from... configure.in:268: the top level autom4te: /usr/bin/m4 failed with exit status: 1 autoheader: '/usr/bin/autom4te' failed with exit status: 1 After deleting po/Makefile.in from the list of AC_OUTPUT in configure.in:268 the compilation proceeded fine. Ubuntu 9.10. Thanks, Pavel Hofman.

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