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

[alsa-devel] need input on USB Audio class device.
by Sakthi Subramanian 31 Jan '11

31 Jan '11

Hi All, i am trying to develop a ALSA record application for USB - Audio device. This USB Audio device has stereo recording and 4 inputs (4 mics). I am trying to capture PCM audio from the 4 mics using the simple record application. Need inputs on the below questions, thanks in advance for the inputs. 1. I have used the ALSA API to set the number of channels as 8. snd_pcm_hw_params_set_channels (capture_handle, hw_params, 8); I have started capturing the data with the above change, and opened the same using Audacity, it says 16 bit PCM Mono channel Little Endian. in my opinion, that must show as 8 channel. Am i calling the right API with parameters? or Please let me know how to get the multi channel PCM capture data from ALSA? This is the output of arecord -l: # arecord -l **** List of CAPTURE Hardware Devices **** card 0: Device [USB PnP Audio Device], device 0: USB Audio [USB Audio] Subdevices: 0/1 Subdevice #0: subdevice #0 Once again thanks a lot for ur inputs.. Regards Sakthi SG

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[alsa-devel] [PATCH] MFD: DaVinci: Fix voice codec device name
by Manjunathappa, Prakash 31 Jan '11

31 Jan '11

Fix the device name in DaVinci Voice Codec MFD driver to load davinci-vcif and cq93vc codec client drivers. Signed-off-by: Manjunathappa, Prakash <prakash.pm(a)ti.com> --- This patch applies to Linus's tree at [1]. [1] http://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux-2.6.git --- drivers/mfd/davinci_voicecodec.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/drivers/mfd/davinci_voicecodec.c b/drivers/mfd/davinci_voicecodec.c index 33c923d..fdd8a1b 100644 --- a/drivers/mfd/davinci_voicecodec.c +++ b/drivers/mfd/davinci_voicecodec.c @@ -118,12 +118,12 @@ static int __init davinci_vc_probe(struct platform_device *pdev) /* Voice codec interface client */ cell = &davinci_vc->cells[DAVINCI_VC_VCIF_CELL]; - cell->name = "davinci_vcif"; + cell->name = "davinci-vcif"; cell->driver_data = davinci_vc; /* Voice codec CQ93VC client */ cell = &davinci_vc->cells[DAVINCI_VC_CQ93VC_CELL]; - cell->name = "cq93vc"; + cell->name = "cq93vc-codec"; cell->driver_data = davinci_vc; ret = mfd_add_devices(&pdev->dev, pdev->id, davinci_vc->cells, -- 1.7.1

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[alsa-devel] Bad sound quality
by Stanislav Kučera 31 Jan '11

31 Jan '11

Hi, In linux compared to windows the sound quality is much worse. I have tried different sound cards, different distributions during past 3 years and the sound is still not good. In linux the sound is somewhat towards mids, less bass and is overall dirtier compared to windows (xp or 7). Problem is not driver specific as I have tried different sound cards on different computers. I can hear the difference even on laptop speakers. When using jack server sounds is little better, but still no way same as in windows. For me it is major obstacle not using linux as I would love to, because I listen to music so much with quality hifi system. Sound cards which were tested are: onboard VIA, onboard Intel ACL889, Sound Blaster Live, Echo Gina20. Thank. Standa

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[alsa-devel] [Patch] sound: silent echo'ed messages in Makefile
by Amerigo Wang 31 Jan '11

31 Jan '11

Silent these echo's, please. Signed-off-by: WANG Cong <amwang(a)redhat.com> --- diff --git a/sound/oss/Makefile b/sound/oss/Makefile index 96f14dc..90ffb99 100644 --- a/sound/oss/Makefile +++ b/sound/oss/Makefile @@ -87,7 +87,7 @@ ifeq ($(CONFIG_PSS_HAVE_BOOT),y) $(obj)/bin2hex pss_synth < $< > $@ else $(obj)/pss_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * pss_synth = NULL;'; \ echo 'static int pss_synthLen = 0;'; \ ) > $@ @@ -102,7 +102,7 @@ ifeq ($(CONFIG_TRIX_HAVE_BOOT),y) $(obj)/hex2hex -i trix_boot < $< > $@ else $(obj)/trix_boot.h: - ( \ + $(Q)( \ echo 'static unsigned char * trix_boot = NULL;'; \ echo 'static int trix_boot_len = 0;'; \ ) > $@

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[alsa-devel] snd-via82xx setting
by Benjamin Hovey 31 Jan '11

31 Jan '11

FYI, the below system required the following non-default settings for ALSA to play any SPDIF output correctly. Prior to finding this setting, output from different sources, WAV, MP3, CD, etc, were played at a slightly faster speed than they should have. snd_via82xx: dxs_support=3 When installed, the default setting was dxs_support=5. Hope this helps. I am new to Linux and it took me quite a while to find out how to set this setting (I put it in /etc/modprobe.d/alsa-base.conf) Cheers upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Mon Jan 31 02:37:03 UTC 2011 !!Linux Distribution !!------------------ Ubuntu 10.04.1 LTS \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 10.04.1 LTS" !!DMI Information !!--------------- Manufacturer: VIA Technologies, Inc. Product Name: KT600-8237 Product Version: !!Kernel Information !!------------------ Kernel release: 2.6.32-28-generic Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.21 Library version: 1.0.23 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_via82xx snd_emu10k1 !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [V8237 ]: VIA8237 - VIA 8237 VIA 8237 with VT1616i at 0xe400, irq 22 1 [Live ]: EMU10K1 - SB Live! Platinum [CT4760P] SB Live! Platinum [CT4760P] (rev.8, serial:0x80401102) at 0xb000, irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:09.0 Multimedia audio controller: Creative Labs SB Live! EMU10k1 (rev 08) 00:0a.0 Multimedia controller: Sigma Designs, Inc. REALmagic Hollywood Plus DVD Decoder (rev 02) 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 60) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 00:09.0 0401: 1102:0002 (rev 08) Subsystem: 1102:8040 -- 00:11.5 0401: 1106:3059 (rev 60) Subsystem: 147b:1408 !!Modprobe options (Sound related) !!-------------------------------- snd_via82xx: index=0 snd_emu10k1: index=1 snd_via82xx: dxs_support=4 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd_via82xx: index=0 snd_emu10k1: index=1 snd_via82xx: dxs_support=3 snd_via82xx: ac97_clock=48000

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[alsa-devel] alsamixer don´t start
by kvoigtlaender 31 Jan '11

31 Jan '11

I described my problem here (in german): http://www.vdr-portal.de/board/thread.php?postid=974850#post974850 Mobo Gigabyte M68M-S2P with nVidia chip and ALC883 Tried with headphone on rear and front socket. Error "unknown symbol" in dmesg. Card is not recognized.

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Re: [alsa-devel] Any OSS changes from kernel 2.6.21 to 2.6.23? Something broke.
by Raymond Yau 31 Jan '11

31 Jan '11

> Clemens Ladisch wrote: >> Timur Tabi wrote: >>> It turns out that ALSA (when using mplayer to play a divx video file via OSS >>> emulation) is rapidly sending back-to-back SNDRV_PCM_TRIGGER_STOP and >>> SNDRV_PCM_TRIGGER_START commands. Why would it do that? >> >> To recover from underruns. >I typically get the STOP command just milliseconds before the period ends. ALSA >is pretty impatient! How does ALSA (in OSS emulation mode) know that an >underrun has occurred? >> There shouldn't be any difference, but maybe the buffer size of the >> driver has changed. Could you show the output of mplayer with the "-v" >> option for both kernels? >Unfortunately, I can't seem to get the original kernel to work now, so it's >broken on all kernels. Anywhere, here's the mplayer output on out 2.6.23-based >kernel: >Linux MPC8610HPCD 2.6.23-6 #20 Thu Nov 8 10:02:21 CST 2007 ppc GNU/Linux: ># mplayer -v -ao oss Cars480_1\[1\].5M.divx >MPlayer 1.0rc2-3.3.5 (C) 2000-2007 MPlayer Team >CommandLine: '-v' '-ao' 'oss' 'Cars480_1[1].5M.divx' >========================== >Found 'wf', 30 bytes of 18 >======= WAVE Format ======= >Format Tag: 85 (0x55) >Channels: 2 >Samplerate: 44100 >avg byte/sec: 16000 >Block align: 1 >bits/sample: 0 >cbSize: 12 >mp3.wID=256 >mp3.fdwFlags=0x0 >mp3.nBlockSize=41729 >mp3.nFramesPerBlock=256 >mp3.nCodecDelay=28933 >Opening audio decoder: [mp3lib] MPEG layer-2, layer-3 >dec_audio: Allocating 4608 + 65536 = 70144 bytes for output buffer. >mp3lib: using AltiVec optimized decore! >MP3lib: init layer2&3 finished, tables done >MPEG 1.0, Layer III, 44100 Hz 128 kbit Joint-Stereo, BPF: 418 >Channels: 2, copyright: No, original: No, CRC: No, emphasis: 0 >AUDIO: 44100 Hz, 2 ch, s16be, 128.0 kbit/9.07% (ratio: 16000->176400) >Selected audio codec: [mp3] afm: mp3lib (mp3lib MPEG layer-2, layer-3) >========================================================================== >Building audio filter chain for 44100Hz/2ch/s16be -> 0Hz*/0ch/*??... >[libaf] Adding filter dummy >[dummy] Was reinitialized: 44100Hz/2ch/s16be >[dummy] Was reinitialized: 44100Hz/2ch/s16be >ao2: 44100 Hz 2 chans s16be >audio_setup: using '/dev/dsp' dsp device >audio_setup: using '/dev/mixer' mixer device >audio_setup: using 'pcm' mixer device >audio_setup: sample format: s16be (requested: s16be) >audio_setup: using 2 channels (requested: 2) >audio_setup: using 44100 Hz samplerate (requested: 44100) >audio_setup: frags: 2/2 (4104 bytes/frag) free: 8208 >AO: [oss] 44100Hz 2ch s16be (2 bytes per sample) >AO: Description: OSS/ioctl audio output >AO: Author: A'rpi >Building audio filter chain for 44100Hz/2ch/s16be -> 44100Hz/2ch/s16be... >[dummy] Was reinitialized: 44100Hz/2ch/s16be >[dummy] Was reinitialized: 44100Hz/2ch/s16be >Starting playback... I also get shuttering with mplayer SVN-r32829-4.3.2 after mplayer display "Increasing filtered audio buffer size from 65536 to 67584" using oss or alsa with sound card which has max buffer bytes 65536 bytes , but it does not happen with "pulse" or hda driver with max buffer bytes > 65536 bytes dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. audio_setup: frags: 16/16 (4096 bytes/frag) free: 65536 2048 bytes is only half of the frag size 4096 bytes why mplayer increasing filtered audio buffer size by half frag or half period size ? it look like related to R24920 Change decode_audio() interface Rewrite decode_audio to better deal with filters that handle input in large blocks. It now always places output in sh_audio->a_out_buffer (which was always given as a parameter before) and reallocates the buffer if needed. After the changes filters can return arbitrarily large blocks of data without some of it being lost. The new version also allows simplifying some code. mplayer -ao oss stereo.wav ======= WAVE Format ======= Format Tag: 1 (0x1) Channels: 2 Samplerate: 44100 avg byte/sec: 176400 Block align: 4 bits/sample: 16 cbSize: 0 ========================================================================== demux_audio: audio data 0x2C - 0x308002C Audio only file format detected. Load subtitles in ./ get_path('sub/') -> '/home/raymond/.mplayer/sub/' ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'oss', options '[none]' ao2: 44100 Hz 2 chans s16le audio_setup: using '/dev/dsp' dsp device audio_setup: using '/dev/mixer' mixer device audio_setup: using 'pcm' mixer device audio_setup: sample format: s16le (requested: s16le) audio_setup: using 2 channels (requested: 2) audio_setup: using 44100 Hz samplerate (requested: 44100) audio_setup: frags: 16/16 (4096 bytes/frag) free: 65536 AO: [oss] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: OSS/ioctl audio output AO: Author: A'rpi Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 65536 A: 127.7 (02:07.6) of 288.0 (04:48.0) 0.1% Increasing filtered audio buffer size from 65536 to 67584 A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) ds_fill_buffer: EOF reached (stream: audio) A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) A: 288.0 (04:47.9) of 288.0 (04:48.0) 0.1% EOF code: 1 Uninit audio filters... [libaf] Removing filter dummy Uninit audio: pcm mplayer -ao alsa:device=hw=0 stereo.wav ======= WAVE Format ======= Format Tag: 1 (0x1) Channels: 2 Samplerate: 44100 avg byte/sec: 176400 Block align: 4 bits/sample: 16 cbSize: 0 ========================================================================== demux_audio: audio data 0x2C - 0x308002C Audio only file format detected. Load subtitles in ./ get_path('sub/') -> '/home/raymond/.mplayer/sub/' ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options 'device=hw=0' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.23 alsa-init: setup for 1/2 channel(s) alsa-init: using device hw:0 alsa-init: pcm opened in blocking mode alsa-init: got buffersize=65536 alsa-init: got period size 1024 alsa: 44100 Hz/2 channels/4 bpf/65536 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under development Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 65536 A: 3.0 (02.9) of 288.0 (04:48.0) 0.1% Increasing filtered audio buffer size from 65536 to 67584 A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) ds_fill_buffer: EOF reached (stream: audio) A: 287.9 (04:47.9) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) A: 288.0 (04:47.9) of 288.0 (04:48.0) 0.1% EOF code: 1 Uninit audio filters... [libaf] Removing filter dummy Uninit audio: pcm alsa-uninit: pcm closed mplayer -ao alsa:device=pulse stereo.wav ======= WAVE Format ======= Format Tag: 1 (0x1) Channels: 2 Samplerate: 44100 avg byte/sec: 176400 Block align: 4 bits/sample: 16 cbSize: 0 ========================================================================== demux_audio: audio data 0x2C - 0x308002C Audio only file format detected. Load subtitles in ./ get_path('sub/') -> '/home/raymond/.mplayer/sub/' ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options 'device=pulse' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.23 alsa-init: setup for 1/2 channel(s) alsa-init: using device pulse alsa-init: pcm opened in blocking mode alsa-init: got buffersize=88200 alsa-init: got period size 1378 alsa: 44100 Hz/2 channels/4 bpf/88200 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under development Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 65536 A: 287.7 (04:47.7) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) ds_fill_buffer: EOF reached (stream: audio) A: 287.7 (04:47.7) of 288.0 (04:48.0) 0.1% ds_fill_buffer: EOF reached (stream: audio) A: 287.8 (04:47.7) of 288.0 (04:48.0) 0.1% EOF code: 1 Uninit audio filters... [libaf] Removing filter dummy Uninit audio: pcm alsa-uninit: pcm closed

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[alsa-devel] VIA VT1828S: crash in alsamixer, no mic input
by Wolfram Nyaa~ 29 Jan '11

29 Jan '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. I've found crash in alsamixer and described it in comment 0023300 in the following issue thread at bugtracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5249 Here is copypaste from it: > Also I've found crash in alsamixer: when I'm trying to unmute "Smart 5.1" control it crashes with message "alsamixer: hcontrol.c:764: snd_hctl_handle_event: Assertion `elem' failed.", but control really unmutes. Muting control doesn't cause crash. > > (gdb) bt > #0 0x00007ffff6d45335 in raise () from /lib/libc.so.6 > #1 0x00007ffff6d465cf in abort () from /lib/libc.so.6 > #2 0x00007ffff6d3e4c1 in __assert_fail () from /lib/libc.so.6 > #3 0x00007ffff72d00d3 in snd_hctl_handle_events () from /usr/lib/libasound.so.2 > #4 0x00007ffff72d98c9 in snd_mixer_handle_events () from /usr/lib/libasound.so.2 > #5 0x0000000000405af1 in ?? () > #6 0x0000000000405096 in ?? () > #7 0x00007ffff6d31d2d in __libc_start_main () from /lib/libc.so.6 > #8 0x0000000000404679 in ?? () > #9 0x00007fffffffd688 in ?? () > #10 0x000000000000001c in ?? () > #11 0x0000000000000001 in ?? () > #12 0x00007fffffffda39 in ?? () > #13 0x0000000000000000 in ?? () Also could anybody give me some hints how to get microphone working with my soundcard (problem is described in thread)? -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.16 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iQEbBAEBAgAGBQJNQ5rfAAoJEN9tENx5d7sA1gUH9itsA4Hnly22lkei9FRRM22d 0ztVKKFiJq2kz4FAHgvUKr97OlW30b7O5ojbAXU3s6+//w8g//wt4+UsLHdWVKfC RvFIb5sEUh2kdiwqcfAbNqrU27favRmKX9+ZjBXnw4Z6RNGiY8v/StwiuK/uB82h xmC2Bx79Y5jHUnhSSS/kmSMgtiTxFqZrqxYyd/zA/yvth1CLIBgqKrl9oni29Y1n VRFn45NB5DMbNREi4qS17lr37iUZ1j/VB6c9Odd4IWSnTCDn30QPZb8WZJNj2hHn vqHemKdFKnPsMavs3OCjxcEplpoea3kQQjpfXjElPg+O3ZErcGvYqojsSW80fg== =KPqa -----END PGP SIGNATURE-----

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[alsa-devel] Fixed sampling freq UAC2 doubt
by Jassi Brar 29 Jan '11

29 Jan '11

Hi, While testing my under-development generic USB Audio Class 2.0 _gadget_ driver with ALSA's USB Audio Class driver, I came across an apparent discrepency. Or so do I think. Page-98 Section 5.2.5.1.1 of 'Audio 20 final.pdf' specifies :- [In many cases, the Clock Source Entity represents a crystal oscillator based generator with a single fixed frequency. In that case, the Set request is not supported.] Here 'not supported' means setup requests for CUR/RANGE CS_SAM_FREQ_CONTROL should stall. Right ? The snd_usb_hw_params in sound/usb/pcm.c apparently attempts to do that with disregard to Clock-Frequency-Control bitmask in bmControls field of the Clock Source Descriptor. The confusion is, subs->cur_rate is initialized only if snd_usb_init_sample_rate succeeds. Which does SET control tranfer, albeit with the supported frequency. Couldn't a fixed sampling freq UAC2 compliant device refuse to work in that case, thereby failing any attempt to use the USB card? Could someone please point where I got it wrong and/or if such a card works with ALSA USB? Thanks.

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[alsa-devel] [PATCH] sound, hda_eld: suspicious sizeof() in call to snd_print_pcm_bits()
by Jesper Juhl 29 Jan '11

29 Jan '11

In hdmi_show_short_audio_desc() we call snd_print_pcm_bits() like this: snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); that last argument is supposed to be the size of the buffer that snd_print_pcm_bits() writes to, so I believe it should be this: snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) Compile tested only. I don't have relevant hardware. diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 4a66347..74b0560 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -381,7 +381,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) snd_print_pcm_rates(a->rates, buf, sizeof(buf)); if (a->format == AUDIO_CODING_TYPE_LPCM) - snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2 - 8)); + snd_print_pcm_bits(a->sample_bits, buf2 + 8, sizeof(buf2) - 8); else if (a->max_bitrate) snprintf(buf2, sizeof(buf2), ", max bitrate = %d", a->max_bitrate); -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/~esr/jargon/html/T/top-post.html Plain text mails only, please.

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