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[alsa-devel] Custom ioctl
by intellij 30 Mar '07

30 Mar '07

hi, how can I send custom request to my driver? Looking into the sources I suspect that sending ioctl() on opened hardware pcm will not reach the playback_ops.ioctl in my driver. thank you very much.

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[alsa-devel] [PATCH] snd-usb-caiaq: Make playback work
by Karsten Wiese 29 Mar '07

29 Mar '07

Hi Takashi, some typo fixes. best regards, Karsten -----------------------------------------------------------------------> From: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Date: Thu, 29 Mar 2007 16:39:47 +0200 Subject: [PATCH] snd-usb-caiaq: Make playback work Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- sound/usb/caiaq/caiaq-audio.c | 15 ++++++++------- 1 files changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index e80c8db..0414d76 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -388,8 +388,8 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - audio_buf[dev->audio_in_buf_pos[stream]++] - = usb_buf[i]; + audio_buf[dev->audio_in_buf_pos[stream]++] = + usb_buf[i]; dev->period_in_count[stream]++; if (dev->audio_in_buf_pos[stream] == sz) dev->audio_in_buf_pos[stream] = 0; @@ -436,19 +436,21 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, spin_lock(&dev->spinlock); for (i = 0; i < iso->length;) { - for (stream = 0; stream < dev->n_streams; stream++) { + for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_playback[stream]; if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; int sz = frames_to_bytes(rt, rt->buffer_size); - usb_buf[i++] - = audio_buf[dev->audio_out_buf_pos[stream]++]; + usb_buf[i] = + audio_buf[dev->audio_out_buf_pos[stream]]; + dev->period_out_count[stream]++; dev->audio_out_buf_pos[stream]++; if (dev->audio_out_buf_pos[stream] == sz) dev->audio_out_buf_pos[stream] = 0; } else - usb_buf[i++] = 0; + usb_buf[i] = 0; + } /* fill in the check bytes */ if (dev->spec.data_alignment == 2 && @@ -456,7 +458,6 @@ static void fill_out_urb(struct snd_usb_caiaqdev *dev, (dev->n_streams * CHANNELS_PER_STREAM)) for (stream = 0; stream < dev->n_streams; stream++, i++) usb_buf[i] = MAKE_CHECKBYTE(dev, stream, i); - } } spin_unlock(&dev->spinlock);

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[alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 29 Mar '07

29 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hello, I'm looking for the maintainers of the Alsa OSS layer. It seems it is getting more seriously broken each month. The last serious updates were almost 7 months ago by tiwai and perex. Dev must have noticed the 'skips' on the end of an OSS stream. The complete broken big endian emulation and so on. What is currently happening in this area? Yours Sincerely, Stefan de Konink -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGByXUYH1+F2Rqwn0RCmU3AJ9O25z5wcCpXLDUnx61idhB7MGcYACcD0fh Fl95KN+x77vuC006U0mexJ8= =guiG -----END PGP SIGNATURE-----

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[alsa-devel] Function ap192_4wire_start in revo.c
by dustin＠seznam.cz 29 Mar '07

29 Mar '07

Hello, while working on support for AK4114 in Audiotrak MI/ODI/O, I have come accross a bug in revo.c (used for major inspiration :) )The function static unsigned char ap192_4wire_start(struct snd_ice1712 *ice) returns value of current GPIO bits, which is unsigned int. However, return value of the function is unsigned char. This results in clearing all GPIO8-22 in next snd_ice1712_gpio_write call of the 4-wire protocol communication which is a little bit inconvenient :). Thanks for fixing in HG. Best regards, Pavel.

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Re: [alsa-devel] Status of CMI9738 and SiS7012
by Takashi Iwai 29 Mar '07

29 Mar '07

Hi, Please don't drop Cc to alsa-devel... At Wed, 28 Mar 2007 10:23:22 -0600, Rob Franklin wrote: > > On 3/28/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Tue, 27 Mar 2007 10:44:23 -0600, > > Rob Franklin wrote: > > > > > > On 3/27/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Tue, 27 Mar 2007 08:49:05 -0600, > > > > Rob Franklin wrote: > > > > > > > > > > Hi All, > > > > > > > > > > I posted a question a couple of weeks ago to the alsa-users list > > > > > (Can't get spdif working with ALSA and FC6 on ECS L4S5MG/651+ > > > > > motherboard), but never got a response. > > > > > > > > > > I have been looking at the ALSA source code for the AC'97 codec on my > > > > > board (CMI9738) and the source code for the intel8x0 driver (for my > > > > > SiS 7012). However, as far as I can tell from looking at the source, > > > > > S/PDIF is not enabled/supported for this combination of devices. > > > > > Additionally, Googling around I haven't found any mention of people > > > > > getting spdif working with ALSA and a CM9738 / SiS7012 combo. > > > > > > > > > > I am wondering whether anyone on the developer list can confirm that > > > > > this is the case. Also, if this is the case can someone give me an > > > > > idea of what would be required to get this working. The system is > > > > > setup with dual boot and so I could grab some of the values from > > > > > windows using a PCI sniffer, and I might be able to do some hacking on > > > > > the source code and submit a patch if someone could help guide me in > > > > > the right direction. > > > > > > > > Are you sure it's CM9738? AFAIK, CM9738 doesn't support SPDIF at > > > > all, but CM9739 does... > > > > > > > That was my first thought too, but I pulled open the case last night > > > and verified that it was indeed a 9738. The download page from the > > > motherboard manufacturer has a driver for CMI9738 and a CMI9738/9739 > > > driver (http://www.ecsusa.com/downloads/drivers_sound.html). I am > > > wondering if there are two different versions of this chip. > > > > > > Here is the output from lspci -vv for the sound device: > > > > > > 00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS] > > > AC'97 Sound Controller (rev a0) > > > Subsystem: Elitegroup Computer Systems Unknown device 0a44 > > > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > > > ParErr- Stepping- SERR- FastB2B- > > > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium > > > >TAbort- <TAbort- <MAbort- >SERR- <PERR- > > > Latency: 32 (13000ns min, 2750ns max) > > > Interrupt: pin C routed to IRQ 22 > > > Region 0: I/O ports at d000 [size=256] > > > Region 1: I/O ports at d400 [size=128] > > > Capabilities: <access denied> > > > > > > One thought that I had was to force the AC'97 driver to detect the > > > chip as a 9739 instead of a 9738 to see whether my 9738 can work as a > > > 9739. But I'm not sure how to do that short of modifying the source > > > code and recompiling the kernel module. > > > > First check whether its codec id is CMI9738, see > > /proc/asound/card0/codec97#0/* files. If it's really CM9738, you can > > hack pci/ac97/ac97_codec.c to replace patch_cm9738 with patch_cm9739 > > in snd_ac97_codec_ids[]. > > > Here are the first couple of lines from /proc/asound/card0/codec97#0/ac97#0-0 : > > 0-0/0: C-Media Electronics CMI9738 > > PCI Subsys Vendor: 0x1019 > PCI Subsys Device: 0x0a44 > > Last night I hacked the ac97_codec.c and recompiled snd-ac97-codec.ko, > and now the file reads > > 0-0/0: C-Media Electronics -- rob CMI9739 > > Everything seems to be working the same as it was before, and I can > get stereo sound from the line-out jack just fine. However, when I > try to play back sound using the spdif I get the following: > > $ aplay -D plug:iec958 ac3_48khz_diatonis_soal.ac3 > ALSA lib setup.c:555:(add_elem) Cannot obtain info for CTL elem > (MIXER,'IEC958 Playback Default',0,0,0): No such file or directory I guess the control element wasn't created indeed because of lack of SPDIF feature on the codec. patch_cm9739() checks the availability of SPDIF bit (SPCV) and turn off the whole SPDIF feature if the bit isn't set. You can avoid that check just for testing. If this doesn't work, the possible situation is that the SPDIF is transmitted directly from the controller chip without AC97 codec. Unfortunately, we have no hardware information about SIS7012, so don't know whether it's even possible, and how to control it. > Is there a script I need to run or file I need to update my list of > CTLs (I have seen references to alsaconf and snddev on the web but > neither seems to be included with my FC5 install of alsa-utils)? % alsactl -f somefile store > Also, is there any easy way to reload the snd-ac97-codec.ko module. I > tried removing the modules with "rmmod snd-ac97-codec" and "modprobe > snd-ac97-codec". The module came out ok, but I got a bunch of errors > like this when I tried to reinsert the modules: > > Mar 27 21:16:46 XXXX kernel: snd_ac97_codec: Unknown symbol snd_info_register > Mar 27 21:16:46 XXXX kernel: snd_ac97_codec: disagrees about version > of symbol snd_ctl_add > > So right now I am rebooting the machine every time I want to replace > snd-ac97-codec. Seems that you still have the old modules on memory while installing the new modules. First unload all snd related modules, then reload again. Otherwise, something wrong with the build with your kernel tree. I never used FC or RedHat, so can't answer about it. Takashi

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[alsa-devel] Enabling SPDIF output on Turtle Beach Audio Advantage Micro (CMEDIA CM102S)
by Eric Sandeen 28 Mar '07

28 Mar '07

I have this Turtle Beach Audio Advantage Micro usb sound card, er, stick. Inside it has pretty much nothing but a C-Media CM102S chip. TB claims, as does C-Media, that it can do S/PDIF out, and I believe them based on the glowing red TOSLINK light. :) I found previous emails on this list (http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg09979.html) suggesting that there is probably some magic to be applied to tell usb hardware to go into S/PDIF mode. I've captured some traces under windows, linked from http://sandeen.net/tbaam/index.html Specifically, http://sandeen.net/tbaam/2spkr-spdif.txt seems to be a fairly short burst of USB activity when I used the control panel to switch from 2 speaker mode to spdif mode. I see the exact same activity when switching from headphone mode to spdif mode. Strangely, switching from, say, spdif back to 2 speaker seems to generate a lot more traffic. The full annotated trace is here: http://sandeen.net/tbaam/usbsnoop1.txt In any case, are these traces helpful to anyone who understands how to talk to a C-Media CM102S usb sound card? Does it look possible to poke this hardware into S/PDIF mode with this information? Thanks, -Eric

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[alsa-devel] how to subscribe to new maillist?
by Songmao Tian 28 Mar '07

28 Mar '07

Hello, I have tried several times on the web page http://mailman.alsa-project.org/mailman/listinfo/alsa-devel, but i can't receive the confirm mail, so how to subscribe? Tian

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[alsa-devel] New code for conexant chipsets in -mm tree
by Rodrigo Luiz 28 Mar '07

28 Mar '07

Hi. I have a HDA NVidia, with a Conexant chipset CX20549 Venice (CXT5045) I tried to test the last alsa-kernel in -mm tree to support my mic. (2.6.21-rc4-mm1 and 2.6.21-rc5-mm2) When I try to load the hda-intel, I receive this message: Mar 27 11:29:08 dig kernel: hda_intel: azx_get_response timeout, switching to polling mode Mar 27 11:29:09 dig kernel: hda_intel: azx_get_response timeout, switching to single_cmd mode After loaded, the speakers doesn't work. Headphone works ok. Mic too. Look in code, I apply this comment and now the speakers works for me: --- patch_conexant.c.orig 2007-03-27 11:37:43.000000000 -0300 +++ patch_conexant.c 2007-03-27 11:38:02.000000000 -0300 @@ -822,7 +822,7 @@ case CXT5045_FUJITSU: spec->input_mux = &cxt5045_capture_source; spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; +// spec->init_verbs[1] = cxt5045_mic_sense_init_verbs; spec->mixers[0] = cxt5045_mixers; codec->patch_ops.init = cxt5045_init; break; But, I don't understand why, because this code is not executed for my chipset. Thanks, Rodrigo Luiz

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[alsa-devel] snd-es1968 capture broken
by Rene Herman 28 Mar '07

28 Mar '07

Good day. Capture is broken on snd-es1968, at least if it is driving a ES1970MS-3D (ESS Canyon 3D). The captured stream is "stuttering" like mad. Kernel 2.6.20 (ALSA 1.0.14-rc1) with alsa-lib 1.0.11 (my distribution's version). Does not happen with a CS4236 in the same machine and same software version(s). Don't know if this driver has a specific maintainer. If needed I guess I can try to pin it down to some change. I hardly use capture on this card but it wasn't always broken... Rene.

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Re: [alsa-devel] Basic DMA question
by Sven Johnsson 28 Mar '07

28 Mar '07

Rene, I have already read the ALSA driver document. What I really like to understand is when DMA transfers are invoked, and what the source and destination for the transfer is. Regards, Sven 2007/3/26, Rene Herman <rene.herman(a)gmail.com>: > On 03/26/2007 03:13 AM, Sven Johnsson wrote: > > > I'm new to writing ALSA drivers, and I could use some help on > > clarifying how DMA is used in ALSA. > > "Writing an ALSA Driver" is a useful document. Google will find it... > > By the way, the alsa-devel mailinglist recently moved from > sourceforge.net to alsa-project.org directly and the new list is > moderated for non-subscribers. I just approved your message from the > moderation queue manually, but please consider subscribing, at: > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Thanks, > Rene > > > > > > > > Regards, > > Sven > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > >

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