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[alsa-devel] Problem restarting alsa: can't set buffer size
by Juan Manuel Lopez Baio 06 Jun '07

06 Jun '07

Hello. I'm developing an application that records input from ALSA and process it. To do so I user the API starting with snd_pcm_open and follow with the parameters settings and so on, and once the app is done and exits, it closes with snd_pcm_hw_params_free and snd_pcm_close. The problem is that afterwards, if I restart the application, I can't set up ALSA, becausea continuous calls to snd_pcm_hw_params_set_buffer_size always fail (I start with a buffer size of 8192 and keep dividing it by 2). I finally have to restart the system in order to get it to work again. I'm on a Mandriva box with gcc 4.1.2 ¿Any thoughts on this? Thanks in advance! juan

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[alsa-devel] using 'type softvol' in a card conf file in /usr/share/alsa/cards, am i doing this right?
by John L. Utz III 05 Jun '07

05 Jun '07

Hello; I am working with an CMEDIA 8738-MC6 and i am trying to gain control of the second DAC by way of softvolume. but i seem to not be succeeding because my application cant find my 'Rear' control, am i doing this correctly? the original /usr/share/alsa/cards/CMI8738-MC6.conf CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 1 } i have modified it to look thusly: notice that the type is now softvol and the original hardware description is encapsulated in slave.pcm # 2nd DAC CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type softvol slave.pcm { type hw card $CARD device 1 } control { name "Rear Playback Volume" card $CARD } } Shouldnt i now be able to see 'Rear Playback Volume' in the output of amixer controls ?? as it stands right now, i can not. what am i doing incorrectly? tnx! johnu

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Re: [alsa-devel] [2/5] 2.6.22-rc4: known regressions
by Michal Piotrowski 05 Jun '07

05 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc4. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions ACPI Subject : ACPI_CPUFREQ fails to load - HP ML110 G4 Xeon 3040 References : http://bugzilla.kernel.org/show_bug.cgi?id=8570 Submitter : Marc Tuerschmann <marc(a)tuerschmann.de> Status : Unknown (possible hardware issues) Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown File systems Subject : OOPS (NULL pointer dereference) in v2.6.22-rc3 References : http://lkml.org/lkml/2007/6/1/389 Submitter : Alex Riesen <raa.lkml(a)gmail.com> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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[alsa-devel] Alsa-OSS Duplex bug (revisited)
by Alan Horstmann 05 Jun '07

05 Jun '07

About a year ago I posted an explaination of the cause of non-functioning with Alsa-OSS in duplex, together with a workaround patch. After some discussion the thread ended. However since then there has been a steady trickle of downloads of the patches from my host (it can also be found elsewhere) and some reports of great success,eg http://audacityteam.org/forum/thread/1388 (2nd page especially) In attempting to tidy up some loose ends, I hope you can bear with me revisiting this and making the case for it's acceptance. I am aware the Alsa-OSS is not of much interest to developers, and it might be said 'use native Alsa'; however that is not possible or desirable in all applications, and anyway, why provide an OSS emulation that cannot be made to work properly? To summarise the problem first: OSS provides a single SNDCTL_DSP_CHANNELS ioctl; where a device is duplex the number of capture and playback channels cannot be set separately. With original OSS this was not a problem AFAIK because separate devices are created for capture and playback, ie they are not used duplex. However Alsa-OSS in most cases creates a single combined device, which only operates correctly when an equal number of capture and playback channels are used. The Workaround: The workaround I proposed and now use constantly is like a secret trapdoor. Where the existing interface is adequate (ie captue channels == playback channels, or non-duplex) nothing changes, and SNDCTL_DSP_CHANNELS is set with a number between 0 and 128. It is fully compatible with apps that don't know of the secret door -the front door is used! However, if an app needs to set capture channels != playback channels, in duplex, at present the stream will malfunction every time. But with this patch applied, if the app knows the workaround, then it can simply set SNDCTL_DSP_CHANNELS as playback + 256*capture channels, and this will be interpreted correctly in the patched Alsa-OSS and the stream functions correctly. As I say, there is therefore no loss of compatibility in either direction. Providing the app only uses the special formula when necessary rather than always, it is only providing a way to fix something that is broken. It would seem to me of value to add the patch to Alsa so that the workaround can be effected by changes to the app only, without also having to patch and compile a replacement Alsa. Whilst it is by no means ideal, it does overcome the underlying problem successfully. Details of the workaround could be added to the OSS-Emulation.html document. Patch for 1.0.14rc4 is attached for consideration, together with example portaudio patch. Regards Alan

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[alsa-devel] ALSA 1.0.14 release
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hi all, finally, ALSA 1.0.14 is out. For list of changes, please, check: http://www.alsa-project.org/changes/v1-0-13--v1-0-14.txt http://www.alsa-project.org/changes/v1-0-14rc4--v1-0-14.txt Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Alsa guru needed for short-term contracting gig
by Darren Gibbs 04 Jun '07

04 Jun '07

Are there any SF Bay Area local Alsa experts out there that could provide assistance bringing up Alsa with driver for an I2S codec on an ARM embedded system?

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[alsa-devel] [PATCH] ALSA: Fix prototype for snd_soc_dai_link.init() function pointer
by Timur Tabi 04 Jun '07

04 Jun '07

Change the snd_soc_dai_link.init() function to take a pointer to the corresponding snd_soc_dai_link structure, instead of a pointer to a snd_soc_codec structure. This allows the initialization function to initialize its own structure. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This is a fix for ALSA bug 3135 (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3135). Since I don't have an ARM cross-compiler, and currently all ASoC drivers are for ARM platforms only, I'm not sure if this fix is correct. include/sound/soc.h | 2 +- sound/soc/at91/eti_b1_wm8731.c | 3 ++- sound/soc/pxa/corgi.c | 3 ++- sound/soc/pxa/poodle.c | 3 ++- sound/soc/pxa/spitz.c | 3 ++- sound/soc/pxa/tosa.c | 3 ++- sound/soc/soc-core.c | 2 +- 7 files changed, 12 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index db6edba..86e1b1d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -410,7 +410,7 @@ struct snd_soc_dai_link { struct snd_soc_ops *ops; /* codec/machine specific init - e.g. add machine controls */ - int (*init)(struct snd_soc_codec *codec); + int (*init)(struct snd_soc_dai_link *codec); }; /* SoC machine */ diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index 820a676..a4f5a99 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -216,9 +216,10 @@ static const char *intercon[][3] = { /* * Logic for a wm8731 as connected on a Endrelia ETI-B1 board. */ -static int eti_b1_wm8731_init(struct snd_soc_codec *codec) +static int eti_b1_wm8731_init(struct snd_soc_dai_link *dai) { int i; + struct snd_soc_codec *codec = dai->codec_dai->codec; DBG("eti_b1_wm8731_init() called\n"); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 5ee51a9..43237f9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -286,9 +286,10 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = { /* * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device */ -static int corgi_wm8731_init(struct snd_soc_codec *codec) +static int corgi_wm8731_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 0915cf7..5e8cb76 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -246,9 +246,10 @@ static const snd_kcontrol_new_t wm8731_poodle_controls[] = { /* * Logic for a wm8731 as connected on a Sharp SL-C7x0 Device */ -static int poodle_wm8731_init(struct snd_soc_codec *codec) +static int poodle_wm8731_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "LLINEIN", 0); snd_soc_dapm_set_endpoint(codec, "RLINEIN", 0); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 80e8210..95dcaa4 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -292,9 +292,10 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = { /* * Logic for a wm8750 as connected on a Sharp SL-Cxx00 Device */ -static int spitz_wm8750_init(struct snd_soc_codec *codec) +static int spitz_wm8750_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; /* NC codec pins */ snd_soc_dapm_set_endpoint(codec, "RINPUT1", 0); diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 5504e30..b438349 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -192,9 +192,10 @@ static const struct snd_kcontrol_new tosa_controls[] = { tosa_set_spk), }; -static int tosa_ac97_init(struct snd_soc_codec *codec) +static int tosa_ac97_init(struct snd_soc_dai_link *dai) { int i, err; + struct snd_soc_codec *codec = dai->codec_dai->codec; snd_soc_dapm_set_endpoint(codec, "OUT3", 0); snd_soc_dapm_set_endpoint(codec, "MONOOUT", 0); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 92d5d91..654a517 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1093,7 +1093,7 @@ int snd_soc_register_card(struct snd_soc_device *socdev) mutex_lock(&codec->mutex); for(i = 0; i < machine->num_links; i++) { if (socdev->machine->dai_link[i].init) { - err = socdev->machine->dai_link[i].init(codec); + err = socdev->machine->dai_link[i].init(&machine->dai_link[i]); if (err < 0) { printk(KERN_ERR "asoc: failed to init %s\n", socdev->machine->dai_link[i].stream_name); -- 1.5.0.2.260.g2eb065

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[alsa-devel] SALSA-Lib version 0.0.1 release
by Takashi Iwai 04 Jun '07

04 Jun '07

Hi all, I created a small replacement of ALSA-library, mainly for embedded systems with limited resources. The library is named as "SALSA-Lib" (too easy, eh?) It has no plugin layer but only hw layer, thus no sexy features like dmix and format conversions are available. It supports no shiny and complex configuration files. And it supports no sequencer. The only goal of this library is the reduction of the size of library and running binaries by inlining functions with direct syscalls. The library is designed to be source-level API compatible with the normal alsa-lib 1.0.14. Note that it is _not_ designed to be binary-compatible at all, and it will be never so. The first released version 0.0.1 is available at: ftp://ftp.suse.com/pub/people/tiwai/salsa-lib/salsa-lib-0.0.1.tar.bz2 The package is released under LGPL as well as the original alsa-lib. I've tested it only on i386, and tests on other architectures would be greatly appreciated. (And, no, I didn't make it on HG tree yet.) Takashi

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[alsa-devel] Datasheet Needed
by chris＠idlelion.net 04 Jun '07

04 Jun '07

(working on bug #305) I'm looking for a proper datasheet on the Cirrus Logic / Crystal Semiconductor CS4624 (or CS4622) sound chip. It's used in a number of popular laptops made around 1999-2001. I don't need the 7- or 8-page feature list, or the 30 (or so) page reference design document. I need register lists and theory of operation and a discussion of power modes. The chip was end-of-lifed in 2004. Many thanks, Chris

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[alsa-devel] v1.0.14 - test packages #2
by Jaroslav Kysela 04 Jun '07

04 Jun '07

Hi, I recreated driver and tools packages in /pub/testing to fix reported problems. Please, test... Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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