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

[alsa-devel] 1.0.16 final release
by Jaroslav Kysela 06 Feb '08

06 Feb '08

Hello all, I am happy to announce ALSA 1.0.16 final release. All packages except alsa-oss (no changes) are available for download. To see changes, please, follow this URL: http://www.alsa-project.org/main/index.php/Changes_v1.0.15_v1.0.16 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

2 1

[alsa-devel] [PATCH] aplay/arecord - Add support for IEEE float 32-bit WAV files
by Andrew Paprocki 06 Feb '08

06 Feb '08

This patch modifies aplay/arecord to support playing/capturing IEEE float 32-bit WAV files. Tested on HDA hardware in both stereo and multi-channel modes. Added the WAV file constant for Dolby AC-3 S/PDIF passthrough to formats.h for future use when AC-3 passthrough is better supported. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 675baf17b7c9 -r 711f7227eb13 aplay/aplay.c --- a/aplay/aplay.c Mon Jan 21 10:42:48 2008 +0100 +++ b/aplay/aplay.c Sat Feb 02 17:28:32 2008 -0500 @@ -745,8 +745,9 @@ static ssize_t test_wavefile(int fd, u_c check_wavefile_space(buffer, len, blimit); test_wavefile_read(fd, buffer, &size, len, __LINE__); f = (WaveFmtBody*) buffer; - if (LE_SHORT(f->format) != WAV_PCM_CODE) { - error(_("can't play not PCM-coded WAVE-files")); + if (LE_SHORT(f->format) != WAV_FMT_PCM && + LE_SHORT(f->format) != WAV_FMT_IEEE_FLOAT) { + error(_("can't play WAVE-file format 0x%04x which is not PCM or FLOAT encoded"), LE_SHORT(f->format)); exit(EXIT_FAILURE); } if (LE_SHORT(f->modus) < 1) { @@ -788,7 +789,10 @@ static ssize_t test_wavefile(int fd, u_c } break; case 32: - hwparams.format = SND_PCM_FORMAT_S32_LE; + if (LE_SHORT(f->format) == WAV_FMT_PCM) + hwparams.format = SND_PCM_FORMAT_S32_LE; + else if (LE_SHORT(f->format) == WAV_FMT_IEEE_FLOAT) + hwparams.format = SND_PCM_FORMAT_FLOAT_LE; break; default: error(_(" can't play WAVE-files with sample %d bits wide"), @@ -1778,6 +1782,7 @@ static void begin_wave(int fd, size_t cn bits = 16; break; case SND_PCM_FORMAT_S32_LE: + case SND_PCM_FORMAT_FLOAT_LE: bits = 32; break; case SND_PCM_FORMAT_S24_LE: @@ -1796,7 +1801,10 @@ static void begin_wave(int fd, size_t cn cf.type = WAV_FMT; cf.length = LE_INT(16); - f.format = LE_SHORT(WAV_PCM_CODE); + if (hwparams.format == SND_PCM_FORMAT_FLOAT_LE) + f.format = LE_SHORT(WAV_FMT_IEEE_FLOAT); + else + f.format = LE_SHORT(WAV_FMT_PCM); f.modus = LE_SHORT(hwparams.channels); f.sample_fq = LE_INT(hwparams.rate); #if 0 diff -r 675baf17b7c9 -r 711f7227eb13 aplay/formats.h --- a/aplay/formats.h Mon Jan 21 10:42:48 2008 +0100 +++ b/aplay/formats.h Sat Feb 02 17:28:32 2008 -0500 @@ -64,7 +64,11 @@ typedef struct voc_ext_block { #define WAV_WAVE COMPOSE_ID('W','A','V','E') #define WAV_FMT COMPOSE_ID('f','m','t',' ') #define WAV_DATA COMPOSE_ID('d','a','t','a') -#define WAV_PCM_CODE 1 + +/* WAVE fmt block constants from Microsoft mmreg.h header */ +#define WAV_FMT_PCM 0x0001 +#define WAV_FMT_IEEE_FLOAT 0x0003 +#define WAV_FMT_DOLBY_AC3_SPDIF 0x0092 /* it's in chunks like .voc and AMIGA iff, but my source say there are in only in this combination, so I combined them in one header;

4 3

[alsa-devel] DSP2000 Midi In - Not a correct fix??
by Alan Horstmann 06 Feb '08

06 Feb '08

I think it is time to post my investigations so far... re: Hoontech DSP2000 Midi In non-functional see : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=55 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1161 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1846 I missed a fix patch in March 07 for this as it was put on a later duplicate of the bug report I had contributed to. However I noticed JK closing -> resolved these in Jan 08, so tried the patch: http://hg.alsa-project.org/alsa-kernel/rev/319688a93b6e From my testing, it seems the only part necessary is:- - for (box = 0; box < 4; box++) { + for (box = 0; box < 1; box++) { as it is successful without the rest. However I feel this is not really a correct fix, preventing use of multiple boxes in code that originally allowed it. The code is a little strange and has taken me a while to approximately grasp. What seems to be going wrong is that when a second box is configured (ie box = 1) the function snd_ice1712_stdsp24_box_midi(...) switches the midi input off box[0]. If box[1] has ICE1712_STDSP24_BOX_MIDI1 set in it's config, then midi in and out can work on box[1] instead of box[0], but otherwise it is just stopped on box[0]. After many hours/days of tests I still can't propose a definite way to fix this function to provide correct configuration to all 4 allowed boxes, as I really don't understand why the code does all the gpio writes that it does. Unfortunately I do not have any circuit information either. In some way it seems to be latching some of the configuration data, but not all. Clemens, can you remember anything about this change: http://hg.alsa-project.org/alsa-kernel/rev/a42043505e46 -was it someone else's patch or did you work on this? I am hoping that someone can contribute enough of an idea about this to help formulate a proper fix. Unfortunately, as the bugs are closed, I can not continue the discussions there. Alan

4 5

[alsa-devel] ALC268 (no sound)
by Marco Schuster 06 Feb '08

06 Feb '08

Hi all, maybe this topic is a bit outdated, but still, when using the current hg code of the kernel module together with 2.6.24 (stable), I can't get any sound :( Do I have also to upgrade the ALSA libs and programs? Marco

3 7

[alsa-devel] [PATCH][sound/soc/fsl/mpc8610_hpcd.c] duplicate strcasecmp test for "rj-master" in mpc8610_hpcd_probe()
by Roel Kluin 06 Feb '08

06 Feb '08

In linus' git tree I found this problem. Is it also in the alsa tree? please confirm it's the right fix. The patch was not yet tested. -- duplicate test for "rj-master" Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index f26c4b2..a00aac7 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -314,9 +314,9 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev, } else if (strcasecmp(sprop, "lj-master") == 0) { machine_data->dai_format = SND_SOC_DAIFMT_LEFT_J; machine_data->codec_clk_direction = SND_SOC_CLOCK_IN; machine_data->cpu_clk_direction = SND_SOC_CLOCK_OUT; - } else if (strcasecmp(sprop, "rj-master") == 0) { + } else if (strcasecmp(sprop, "rj-slave") == 0) { machine_data->dai_format = SND_SOC_DAIFMT_RIGHT_J; machine_data->codec_clk_direction = SND_SOC_CLOCK_OUT; machine_data->cpu_clk_direction = SND_SOC_CLOCK_IN; } else if (strcasecmp(sprop, "rj-master") == 0) {

4 4

[alsa-devel] How to change the sound volume?
by Cosmin Humeniuc 06 Feb '08

06 Feb '08

Hi, How would a simple alsa-based sound player change the playback volume? I've been looking into the mixer interface, but there's no connection to snd_pcm_t*, so I'm not sure this is what I need to use. I only want to change the sound volume of my player, without altering the sound streams of the other applications. Thanks, Cosmin

2 1

Re: [alsa-devel] [alsa-cvslog] alsa-tools: hdspmixer - Fix Digiface channel map for ADAT speed mode 1
by Thierry Vignaud 05 Feb '08

05 Feb '08

Takashi Iwai <tiwai(a)suse.de> writes: > changeset: 365:9d1e48f8dd5d > tag: tip > user: tiwai > date: Tue Feb 05 13:30:45 2008 +0100 > files: hdspmixer/src/HDSPMixerCard.cxx > description: > hdspmixer - Fix Digiface channel map for ADAT speed mode 1 > > Fixed the channel map for ADAT speed mode 1 (rate > 48000). > ALSA bug#3732 > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3732 Will you move the 1.0.16 tag so that it got included in final release? > diff -r 7f30f325ab54 -r 9d1e48f8dd5d hdspmixer/src/HDSPMixerCard.cxx > --- a/hdspmixer/src/HDSPMixerCard.cxx Tue Feb 05 10:08:12 2008 +0100 > +++ b/hdspmixer/src/HDSPMixerCard.cxx Tue Feb 05 13:30:45 2008 +0100 > @@ -299,7 +299,7 @@ void HDSPMixerCard::adjustSettings() { > break; > case 1: > channels = 14; > - channel_map = channel_map_ds; > + channel_map = meter_map_ds; > dest_map = dest_map_ds; > meter_map = meter_map_ds; > lineouts = 2;

2 1

[alsa-devel] Salsa-lib memleak
by Matthieu CASTET 05 Feb '08

05 Feb '08

Hi, it seems Salsa-lib leak memory in case of errors. For snd_pcm_open, snd_pcm_t is allocated, but not free in case of snd_pcm_hw_mmap_status error. The same things happen in _snd_pcm_mmap, there error return in middle of the code, without freeing the allocated memory. Thanks Matthieu

2 1

Re: [alsa-devel] M-Delta Audiophile 192 (ice1724) Recording
by Computer AG im FSK 04 Feb '08

04 Feb '08

Hi, unfortunately it didn't solve the issue. Same problem: no audio signal can be recorded. I provided all the information at http://pastebin.ca/886740 If we can't solve the issue this way, maybe we could lend you one of our cards ? Greeting, Toast

3 5

[alsa-devel] HDA - Add support for the OQO Model 2
by Tobin Davis 04 Feb '08

04 Feb '08

This patch adds support for the OQO Model 2 Ultra Mobile PC. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

3 11