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[alsa-devel] [PATCH] ASoC: Keep index within stac9766_reg[]
by Roel Kluin 13 Jul '09

13 Jul '09

Keep index within stac9766_reg[] Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- please ignore the patch lacking the second hunk diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 8ad4b7b..befc648 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -149,7 +149,7 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val); @@ -168,7 +168,7 @@ static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return val; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||

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Re: [alsa-devel] [PATCH] New ASoC Drivers for ADI AD1938 codec
by Mark Brown 13 Jul '09

13 Jul '09

On Mon, Jul 13, 2009 at 04:12:14PM +0800, 宋宝华 wrote: > I want to use ADC/DAC widgets. > static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { > SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AD1938_DAC_CTRL0, 0, 1), > SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AD1938_ADC_CTRL0, 0, 1), > }; > But for this AD1938 codec, DAC's work depends on ADC is powered on in > hardware. I think there is no any mechanism to handle this kind of > strange depending now. So is there a generic way to handle this? Make the ADC power a SND_SOC_DAPM_SUPPLY() supplying both the DAC and the ADC, with the ADC widget marged as having no power management.

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[alsa-devel] [PATCH] ASoC: Keep index within stac9766_reg[]
by Roel Kluin 13 Jul '09

13 Jul '09

Keep index within stac9766_reg[] Signed-off-by: Roel Kluin <roel.kluin(a)gmail.com> --- diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 8ad4b7b..11f467e 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -149,7 +149,7 @@ static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, stac9766_ac97_write(codec, AC97_INT_PAGING, 1); return 0; } - if (reg / 2 > ARRAY_SIZE(stac9766_reg)) + if (reg / 2 >= ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val);

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 12 Jul '09

12 Jul '09

Hello, unfortunately I cannot playback anything (PCM) with the new SB X-Fi driver. Driver: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- driver-snapshot.tar.bz2 Distribution: openSuSE 11.1 Kernel: 2.6.27.23-0.1-xen My card is a "X-Fi Titanium" (PCI device code: 1102:000b). The card is recognized by the driver, and any mixer (alsamixer, kmix) also recognizes the card. But playing back isn't possible. I tried speaker-test, audacity, amarok and aplay. Most applications hang, some also produce an error in the /var/log/messages (see below). I then used "./configure --with-debug=full --with-cards=ctxfi" due to the possible hd-audio conflict, but it didn't make a difference. Is there anything else I could try? Christian --- Console output from speaker-test --- bash # LANG=en speaker-test speaker-test 1.0.18 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 512 to 65536 Period size range from 32 to 32768 Using max buffer size 65536 Periods = 4 was set period_size = 16384 was set buffer_size = 65536 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Operation not permitted ------------------------------------------------------------------------ --- output in /var/log/messages when running speaker-test --- Jun 27 12:52:53 firefly kernel: ALSA /home/chris/Desktop/alsa- driver/acore/../alsa-kernel/core/pcm_lib.c:1693: playback write error (DMA or IRQ trouble?) ------------------------------------------------------------------------ ---output in /var/log/messages when running amarok with pulseudio --- 20 11:11:00 firefly pulseaudio[5649]: module-alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Most likely this is an ALSA driver bug. ------------------------------------------------------------------------ --- Hardware information --- firefly:~ # lspci -nn | grep -i audio 00:07.0 Audio device [0403]: nVidia Corporation MCP65 High Definition Audio [10de:044a] (rev a1) 01:07.0 Multimedia audio controller [0401]: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] [1013:6003] (rev 01) 05:00.0 Audio device [0403]: Creative Labs Device [1102:000b] (rev 03) --- Hardware information from lshal --- udi = '/org/freedesktop/Hal/devices/pci_1102_b_sound_card_0' info.capabilities = {'sound'} (string list) info.category = 'sound' (string) info.parent = '/org/freedesktop/Hal/devices/pci_1102_b' (string) info.product = 'Creative X-Fi 20K2 SB0880 Sound Card' (string) info.subsystem = 'sound' (string) info.udi = '/org/freedesktop/Hal/devices/pci_1102_b_sound_card_0' (string) linux.hotplug_type = 2 (0x2) (int) linux.subsystem = 'sound' (string) linux.sysfs_path = '/sys/devices/pci0000:00/0000:00:0e.0/0000:05:00.0/sound/card0' (string) sound.card = 0 (0x0) (int) sound.card_id = 'Creative X-Fi 20K2 SB0880' (string) sound.originating_device = '/org/freedesktop/Hal/devices/pci_1102_b' (string)

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Paredes 12 Jul '09

12 Jul '09

Hi everyone, So I have an Auzentech X-Fi Forte PCI-E card, and I got it to output sound through the side and center/LFE channels. When I output sound through the center channel, it is correctly routed to the center channel, but when sound is output to the LFE channel, it is output to both of my front speakers. I am running on kernel 2.6.28, fresh install of Ubuntu 9.04. This also happens on kernel 2.6.30 on Arch Linux. Let me know what sort of information you guys need! Thanks, Christian

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[alsa-devel] Let's discuss the state of Audio on Linux - OSS and ALSA people
by apriori＠dkc-clan.de 11 Jul '09

11 Jul '09

Hi all. I would like to encourage all ALSA devs and users to talk about the current state of audio on linux together with the Open Sound System devs at: http://www.4front-tech.com/forum/viewtopic.php?t=3199 I recommend to take this offer since we all and the linux development would definetly benefit if both core audio system developer team would finally have a talk with each other. For the sake of information and knowledge please don't fill up the thread with off-topic comments or bashing comments to either sound system if it doesn't clearly state (in detail) 1) what is wrong and 2) what could be done better and 3) how it could be done better. Criticism should be done in a constructive way. Looking forward to see you in the dicussion apriori

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[alsa-devel] [PATCH for-2.6.32] sound dapm: fix checking for external widgets bug
by Rongrong Cao 11 Jul '09

11 Jul '09

From: Rongrong Cao rrcao(a)ambarella.com This patch is based on for-2.6.32 In SOC DAPM layer of SOUND subsystem, when add signal route (in the function snd_soc_dapm_add_route() ), the original code has wrong logic when dapm layer check each widget whether an external one. --------------------------------------------------------------------------------------------------------- --- for-2.6.32/sound/soc/soc-dapm.c.orig 2009-07-08 13:33:41.000000000 +0800 +++ for-2.6.32/sound/soc/soc-dapm.c 2009-07-08 13:34:50.000000000 +0800 @@ -1290,8 +1290,8 @@ static int snd_soc_dapm_add_route(struct if (wsink->id == snd_soc_dapm_input) { if (wsource->id == snd_soc_dapm_micbias || wsource->id == snd_soc_dapm_mic || - wsink->id == snd_soc_dapm_line || - wsink->id == snd_soc_dapm_output) + wsource->id == snd_soc_dapm_line || + wsource->id == snd_soc_dapm_output) wsink->ext = 1; } if (wsource->id == snd_soc_dapm_output) { --------------------------------------------------------------------------------------------------------- Signed-off-by: Rongrong Cao <rrcao(a)ambarella.com> ******************** Best regards! Allen (Rongrong) Cao Ambarella Shanghai Ltd. Tel: (86-21) 50942311 Fax: (86-21) 50942321

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[alsa-devel] codec drivers for Cirrus CS8406 or Texas Instruments PCM1748
by just4emmett 10 Jul '09

10 Jul '09

Hello all. Is anyone working on codec drivers for these devices, or know where to find them if they exist already? Checked http://opensource.wolfsonmicro.com/cgi-bin/gitweb.cgi?p=linux-2.6-asoc.git;… and didn't see anything. Thanks!

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Re: [alsa-devel] [Jack-Devel] Says: server is not connected, even though it has been!
by Guilherme Longo 10 Jul '09

10 Jul '09

Yes Patrick.... same user for both Really strange, isn't it! Patrick Shirkey wrote: > > On 07/10/2009 05:20 PM, Guilherme Longo wrote: >> Hi all! >> >> I couldn't find where this problem comes from. > > Hi, > > Sorry, to ask the obvious but, are you running as the same user for > the server and the simple_client? > > > Patrick Shirkey > Boost Hardware Ltd > > > >> I am connecting jack with the command: >> >> jackd -d alsa -d default >> >> and appears right below: >> >> bash-3.1# jackd -d alsa -d default >> jackdmp 1.9.2 >> Copyright 2001-2005 Paul Davis and others. >> Copyright 2004-2008 Grame. >> jackdmp comes with ABSOLUTELY NO WARRANTY >> This is free software, and you are welcome to redistribute it >> under certain conditions; see the file COPYING for details >> JACK server starting in non-realtime mode >> creating alsa driver ... >> default|default|1024|2|48000|0|0|nomon|swmeter|-|32bit >> Using ALSA driver EMU10K1 running on SB Live [Unknown] (rev.10, >> serial:0x80671102) at 0xd000, irq 18 >> configuring for 48000Hz, period = 1024 frames (21.3 ms), buffer = 2 >> periods >> ALSA: final selected sample format for capture: 32bit float >> little-endian >> You appear to be using the ALSA software "plug" layer, probably >> a result of using the "default" ALSA device. This is less >> efficient than it could be. Consider using a hardware device >> instead rather than using the plug layer. Usually the name of the >> hardware device that corresponds to the first soun >> ALSA: use 2 periods for capture >> ALSA: final selected sample format for playback: 32bit float >> little-endian >> You appear to be using the ALSA software "plug" layer, probably >> a result of using the "default" ALSA device. This is less >> efficient than it could be. Consider using a hardware device >> instead rather than using the plug layer. Usually the name of the >> hardware device that corresponds to the first soun >> ALSA: use 2 periods for playback >> >> >> In another terminal, I am running a simple_cliente, and when I run it >> with the code: >> >> quart$: simple_client audiofile >> jack server not running? >> >> It is resulting of: >> >> if ((client = jack_client_open(argv[1], JackNoStartServer, NULL)) == >> 0) { >> fprintf (stderr, "jack server not running?\n"); >> return 1; >> } >> >> >> Plz, can I get som help here. >> Tks. >> _______________________________________________ >> Jack-Devel mailing list >> Jack-Devel(a)lists.jackaudio.org >> http://lists.jackaudio.org/listinfo.cgi/jack-devel-jackaudio.org >

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[alsa-devel] Bug report: Random MIDI output data drop-outs/corruption
by Sean M. Pappalardo 10 Jul '09

10 Jul '09

Hello there. I just wanted to call your attention to a bug report I recently filed, because I'm not sure I put it in the right place: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4606 In a nutshell, ALSA seems to be dropping or corrupting output MIDI messages (both "short" and SysEx) at random as evidenced by the devices not lighting some LEDs or changing modes (sysex) on occasion. The problem is positively affected, but not completely mitigated, by the use of a real-time kernel. aseqdump -d shows everything correctly even when the device response is incorrect. All details are in that bug report including links to alsa-info output in the notes. Thanks for your time. Let me know if I can provide any further information. Sincerely, Sean M. Pappalardo "D.J. Pegasus" <<--------------------------------------------------------------------------------->> This E-Mail message has been scanned for viruses and cleared by >>SmartMail<< from Smarter Technology, Inc. <<--------------------------------------------------------------------------------->>

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