February 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] MTP-AV Bugs
by Holger Dehnhardt 12 Feb '09

12 Feb '09

Hi everybody, I recently started again to use my older music equipment and try to get it work under Linux. One of these things is an Motu Midi Express XT (8 Port Midi Interface at parallel port), which is listed as supported along with the Motu MidiTimePiece AV. After upgrading to a recent Kernel, sending Midi from an Sequencer Programm to the Interface was working, but connecting an input port resulted in an immediate kernel freeze. After trying some different parallel port settings and some other things, I upgraded to the latest alsa release (alsa-driver-1.0.19) and after the problem remains, started debugging the driver. (Because I never did any driver programming I only used print statements and returns to locate the point of failure..., if anyone of you has a good tutorial about debugging kernel modules - please sent a private Mail...) After some time I found the following bug: static void snd_mtpav_inmidi_h(struct mtpav *mcrd, u8 inbyte) { if (inbyte >= 0xf8) { /* real-time midi code */ --> snd_mtpav_inmidi_process(mcrd, inbyte); //here the inmidiport is not translated to the prtnumber, so // snd_mtpav_inmidi_process the port with an index of some 80000 // in the ports array of the mtpav struct was called return; } if (mcrd->inmidistate == 0) { // awaiting command if (inbyte == 0xf5) // MTP port # mcrd->inmidistate = 1; else snd_mtpav_inmidi_process(mcrd, inbyte); } else if (mcrd->inmidistate) { mcrd->inmidiport = translate_hwport_to_subdevice(mcrd, inbyte); mcrd->inmidistate = 0; } The following change made the driver work: static void snd_mtpav_inmidi_h(struct mtpav *mcrd, u8 inbyte) { if (inbyte >= 0xf8) { /* real-time midi code */ mcrd->inmidiport = translate_hwport_to_subdevice(mcrd, inbyte); snd_mtpav_inmidi_process(mcrd, inbyte); return; } .... After that I realised, that with the recent driver - even without my change - playing midi files din not work anymore. Only the first midi event was sent to the interface, everything else was dropped somewhere. So a diff between the last workin version and the recent version showed only one change: 699,701c703,705 < card = snd_card_new(index, id, THIS_MODULE, sizeof(*mtp_card)); < if (! card) < return -ENOMEM; --- > err = snd_card_create(index, id, THIS_MODULE, sizeof(*mtp_card), &card); > if (err < 0) > return err; which seems due to an api change. After rolling that back to the old (deprecated) way of creating the sound card, everything works fine. Could someone with a little bit more experience in driver programming please look into the first change. For the second change I would be happy about every tip why the old way works and the new way not. I'm willing to correct this as well. Thank you for listening! Holger

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[alsa-devel] [PATCH] ALSA: hda - Cleanup setting of pin_configs in patch_stac927x
by Herton Ronaldo Krzesinski 12 Feb '09

12 Feb '09

After commit "ALSA: hda - Fix restore of pin configs at resume for STAC/IDT codecs", the introduced stac_save_pin_cfgs function checks already for pins == NULL case, saving then default pin configs from machine with stac92xx_save_bios_config_regs. So we can remove the extra checks when stac927x_brd_tbl[spec->board_config] == NULL. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_sigmatel.c | 7 +++---- 1 files changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 12b3088..6fa064d 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5292,10 +5292,9 @@ static int patch_stac927x(struct hda_codec *codec) stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) { - if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - "STAC927x, using BIOS defaults\n"); + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); } else err = stac_save_pin_cfgs(codec, -- 1.6.1.3

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[alsa-devel] snd_pcm_runtime.buffer_size question
by Tim Blechmann 11 Feb '09

11 Feb '09

hi all, i am currently fighting with the snd_pcm_runtime.buffer_size. how is the value defined? i would expect: channels * periods * period_size * sample_bits/8 (in my case 64*5*128*3) however in the code, i am working on, it seems to be: periods * period_size (in my case 5*128) am i missing something obvious? thanks, tim -- tim(a)klingt.org http://tim.klingt.org Just what the hell is the experimental tradition? Morton Feldman

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[alsa-devel] Please Help
by constantine 11 Feb '09

11 Feb '09

I'm having trouble with youtube in Ubuntu 8.10 I've tried this guide: https://help.ubuntu.com/community/SoundTroubleshooting to no success and submitted a bug report as suggested. I then entered #alsa and here's the result of that chat: (I am Constantine) <constantine_> hi, I'm having problems with youtube sound and already troubleshot it <constantine_> I am trying to manually start the driver at this point but can't get sudo modprobe snd <constantine_> to work with the file name I put after it <constantine_> all other audio worked fine before the switch to alsa and still does, except youtube <gnubien> constantine_ run this command: cat /proc/asound/cards #and then paste output to this channel; its OK to paste up to 10 lines here <constantine_> 0 [Intel ]: HDA-Intel - HDA Intel <constantine_> HDA Intel at 0xfea38000 irq 16 <gnubien> constantine_ close all sound apps and firefox and then ONLY report you have: sound, no sound or errors after you run this command: speaker-test -c2 -D hw:0,0 -t wav -l1 <constantine_> sound <gnubien> constantine_: ok, only problem is no sound when playing flash in firefox? <constantine_> only 'certain' flash, ie youtube <gnubien> constantine_: ok, alsa is setup and working ok so flash is the problem so time to do some searching... Thanks, Constantine

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[alsa-devel] using ALSA one call
by kallipygos＠inbox.lv 11 Feb '09

11 Feb '09

Hi Developers. My name is Alf and i new in this list . I physics student , and i write some sound programs under linux. Big sorry for my english , in school i learn only deutsch. So far i used old /dev/dsp open funntion also under ALSA. /*--OSD--*/#define RATE 44100 /* the sampling rate */ /*--OSD--*/#define SIZE 16 /* sample size: 8 or 16 bits */ /*--OSD--*/#define CHANNELS 2 /* 1 = mono 2 = stereo */ /*--OSD--*/void open_sound_device_wr(void) /*--OSD--*/{// fdo = open("/dev/dsp", O_RDWR ); /*--OSD--*/ fdo = open("/dev/dsp", O_WRONLY ); /*--OSD--*/ if (fdo < 0) { perror("open of /dev/dsp failed"); exit(1); } /*--OSD--*/ /*--OSD--*/ arg = SIZE; //// sample size /*--OSD--*/ status = ioctl(fdo, SOUND_PCM_WRITE_BITS, &arg); /*--OSD--*/ if (status == -1) perror("SOUND_PCM_WRITE_BITS ioctl failed"); /*--OSD--*/ if (arg != SIZE) perror("unable to set sample size"); /*--OSD--*/ /*--OSD--*/ arg = CHANNELS; //// mono or stereo /*--OSD--*/ status = ioctl(fdo, SOUND_PCM_WRITE_CHANNELS, &arg); /*--OSD--*/ if (status == -1) perror("SOUND_PCM_WRITE_CHANNELS ioctl failed"); /*--OSD--*/ if (arg != CHANNELS) perror("unable to set number of channels"); /*--OSD--*/ /*--OSD--*/ arg = RATE; //// sampling rate /*--OSD--*/ status = ioctl(fdo, SOUND_PCM_WRITE_RATE, &arg); /*--OSD--*/ if (status == -1) perror("SOUND_PCM_WRITE_WRITE ioctl failed"); /*--OSD--*/} ---- Now i wana try da same with ALSA , for example all parameters put in one struckture and then simple call one universal alsa_open funktion. Is it posssible with ALSA ? Is here copy-paste example/function for fast, comfortable and easy ALSA use ? for example so : ==== #include "alsa/asoundlib.h" ... alsa_parm_struckt.device = "default"; // "hw:0,0"; "hw:1,0"; alsa_parm_struckt.play_capt = SND_PCM_STREAM_PLAYBACK; alsa_parm_struckt.format = SND_PCM_FORMAT_S16_LE; alsa_parm_struckt.access = SND_PCM_ACCESS_RW_INTERLEAVED; //alsa_parm_struckt.samplerate = 48000; alsa_parm_struckt.samplerate = 44100; alsa_parm_struckt.channelz =2; alsa_parm_struckt.bytes_per_sample = 2; alsa_parm_struckt.latency = 0; // ?? alsa_parm_struckt.nonblock = 1; // ?? ... other parameters set ... ... alsa_open_function( & handle , & alsa_parm_struckt [ , ...] ); and after this write data blocks frames_2_write = ... for(;;) { ... frames = snd_pcm_writei(handle, buffer, frames_2_write ); ... } ==== OK , this example works http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm__min_8c-example.h… but only if samplerate = 48000; if samplerate = 44100 - then not :( 44k file play in 48k mode :D it sounds funnee :D I remixed this program so that it read data from wave file. gcc pcm_min01.c -lasound -O2 -o pcm_min01 ./pcm_min01 '/mnt/hda2/Booty Luv - Shine/Booty Luv - Shine (Ian Carey remix).wav' i gotta this : ALSA lib pcm.c:7160:(snd_pcm_set_params) Rate doesn't match (requested 44100Hz, get 0Hz) Playback open error: Invalid argument With aplay no problems, it is played in da correct speed. -- aplay '/mnt/hda2/Booty Luv - Shine/Booty Luv - Shine (Ian Carey remix).wav' Playing WAVE '/mnt/hda2/Booty Luv - Shine/Booty Luv - Shine (Ian Carey remix).wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo -- Any ideas/examples/... welcome Tnx in advance Alf ==== ----

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[alsa-devel] Juli@ ICE1724 and 24 bit audio
by Demian Martin 11 Feb '09

11 Feb '09

I have the driver working fine for sample rates from 44.1K to 196K however it seems to be truncating the data to 16 bits. I wasn't sure until I started testing with the HRx 176.4K 24 bit files that have an embedded HDCD flag in the LSB. The files work OK on Windoze systems (with a lot of low level settings tweaked) and they play fine but the flag isn't detected by a system that can detect them. Further looking at the data stream with a scope it seems the last 8 bits aren't changing. Is there anything I can do to control the driver to confirm this problem or change the playback settings to make it work? Demian Martin Product Design Services 784 Cary Drive San Leandro, CA 94577 209 613 6990

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Re: [alsa-devel] Juli@ ICE1724 and 24 bit audio
by Pavel Hofman 11 Feb '09

11 Feb '09

Pavel Hofman napsal(a): > Demian Martin napsal(a): >> I confirmed that aplay and plughw:0,1 work fine. Aplay and hw:0,1 >> don't work >> at all. >> >> And that Xine is the culprit. Now to find out how to fix/work around >> Xine. >> >> Demian Martin >> Product Design Services >> >> > > The older aplay you most likely use does not support 24bit properly. > Install the latest alsa-utils from git and you will be fine. > Sorry for confusion. Your aplay handles already 24bit correctly. ICE1724 supports only 32bit natively (as most modern cards do), and hw:0 correctly does not accept any other format. That is why plughw converts your 24-bit format to 32bit - see the verbose output. For testing purposes make a 32bit file (sox -V in.wav -4 32.wav), and aplay -v -Dhw:0 32.wav will play fine. Pavel -- ----------------- inSITE, s.r.o. Rubesova 29, 326 00 Plzen Tel. +420 377 449 358 Fax +420 222 745 553 GSM: +420 603 163 973 Email: pavel.hofman(a)insite.cz www.educity.cz, www.jobcity.cz www.meetings.cz, www.hrzive.cz www.comben.cz, www.insite.cz ------------------------------- Navstivte www.educity.cz, server s nejvetsi nabidkou profesniho vzdelavani na ceskem internetu.

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[alsa-devel] [PATCH 0/4] Intel HDMI audio fixes
by Wu Fengguang 11 Feb '09

11 Feb '09

Hi Takashi, Here are four assorted fixes for Intel HDMI audio. They have been tested in G35 SDVO HDMI and G45 integrated HDMI systems, and shall be OK for 2.6.29. [PATCH 1/4] hda - allow multi-channel HDMI audio playback when ELD is not present [PATCH 2/4] hda - enable HDMI audio pin out at module loading time [PATCH 3/4] hda - compute checksum in HDMI audio infoframe [PATCH 4/4] hda - add id for Intel IbexPeak integrated HDMI codec Shane: the first two patches should address the two bugs you reported, i.e. - multi-channel HDMI audio playback problem - first half-second audio samples missing These two bugs was reproduced and fixed in the combination of G35 SDVO HDMI + Yamaha RX-V1800 AV receiver. The first problem cannot be verified in G45 system because we still cannot make G45 work with Yamaha. The second problem is interesting in that it can be fixed by the second patch in G35, but still remains in G45. Thanks, Fengguang --

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[alsa-devel] ASoC update for 2.6.29
by Mark Brown 11 Feb '09

11 Feb '09

The following changes since commit 397d5aeeb5a2c9ca6108899a04b35a51cd904503: Jarkko Nikula (1): ASoC: WM8990: Fix kcontrol's private value use in put callback are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.29 Lopez Cruz, Misael (1): ASoC: Update SDP3430 machine driver for snd_soc_card sound/soc/omap/sdp3430.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

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[alsa-devel] Issue with Alsa and PulseAudio
by Ash 11 Feb '09

11 Feb '09

Hello, I have a media application that uses alsa as its sound renderer. It has worked on numerous systems without any problems until I came across a mandriva distro that bundles with PulseAudio. The alsa pulseaudio plugin is installed correctly, however I am having weird problems. When I am setting up my hw params, and I call snd_pcm_hw_params_any(), I get an "Operation not permitted" error. I am not sure why this is happening, anyone else see this before? I am using pulseaudio version 0.9.9 and alsa 1.0.16. Thanks in advance! _________________________________________________________________ Be the filmmaker you always wanted to be—learn how to burn a DVD with Windows®. http://clk.atdmt.com/MRT/go/108588797/direct/01/

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