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[alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 24 Apr '07

24 Apr '07

Greetings, I'm setting out to write an ALSA driver for a very limited device thats behind a PLD. The actual codec is a uda1380 and does 16bit MSB stereo encode/decode. The way the device is driven the sampling rate is limited to 8KHz and data transfer to/from the device I have to handle manually. I've read over the 'writing an ALSA driver' tutorial and have started the driver but have run into a few questions: - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB samples does the following snd_pcm_hardware_t look right?: static snd_pcm_hardware_t my_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? .rates = SNDRV_PCM_RATE_8000, // seems redundant? .rate_min = 8000, .rate_max = 8000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 32768, .period_bytes_min = 4096, .period_bytes_max = 32768, .periods_min = 1, .periods_max = 1024, }; - I'm a little confused at how to deal with copying data from/to the device. It looks like I would implement the copy callback and copy the passed buffer to the device in the device specific manner I need to (playback), but I'm not clear where to put data that I receive from the device (capture). I notice that when the hw_params callback is called a dma buffer has been created by ALSA in the substream->runtime struct. I assume this is created based on hte buffer_bytes_max field above and I'm to copy data to this buffer? How do I know where to copy it within the buffer and what data ALSA has already consumed from the buffer? - I might need to create an intermediate buffer between ALSA's rather large buffers and my hardware buffer. Is there any driver that I could look at that does this? The tutorial talks about the vxpocket driver but that code looks very foriegn from the rest of the ALSA drivers. - I'm not sure I understand what the 'period' is about. Its clear to me that I need to call snd_pcm_period_elapsed() periodically from either an IRQ or other timed callback, but I'm not clear what the hardware pointer is or how that converts to 'frames' and 'periods'. - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA provides a mechanism for converting data from other formats? Thanks for any pointers, Tim

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[alsa-devel] What does start_threshold mean, and what value should it contain?
by j t 24 Apr '07

24 Apr '07

Hi. I'm trying to debug a problem with xruns which happen when playing a wav file with vlc, but I can't reproduce the problems when using "aplay -M" to play the same wav file. (logged as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3031). I've checked the contents of /proc/asound/card0/pcm0p/sub0/ in both cases - they appear to be very similar except for the contents of sw_params: when I use vlc, start_threshold says 1, but when I use "aplay -M", start_threshold says 16384. So, what does start_threshold mean, is there a particular value that start_threshold "should" contain, and could this difference be the cause of the xruns? (The aplay man page doesn't seem to have any flags for controlling the value of this parameter, so it doesn't appear to be "user-configurable"). (Apologies if this is the wrong mailing list - I though that -devel would be better than -user, but I could be wrong). Thanks, Jaime

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Re: [alsa-devel] Fwd: No sound with ATI SB450
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 22:45:00 +1000, Ben DiDonc wrote: > > Tobin, Takashi, > Thank you very much for your help. > Takashi's last suggestion did it and I finally got a working ATI SB450 > sound card with Ubtuntu Feisty (kernel 2.6.20 and > alsa-driver-hg20070421). I did a (last) alsa-info.sh in case it helps > others, see http://pastebin.ca/454115 > Here's what it took: > modprobe snd_hda_intel probe_mask=8 model=auto > Without probe_mask, the ALC861 is not properly detected. Without > model=auto, playback doesnt work. > Let me know If I can be of any help or if you want me to try something else. As I requested, get /proc/asound/card0/codec#* files for both working and non-working states. Then we can compare what is not set up properly. probe_mask is a bit hard to fix right now. A least, we need to know which modem codec is harmful. Please check the codec#* file again with probe_mask=1. Takashi > Thanks again. > Ben > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 23 Apr 2007 21:58:17 +1000, > > Ben DiDonc wrote: > > > > > > Tobin, Takashi, > > > upon Takashi's advice, I tried loading hda-intel > > > (alsa-driver-hg20070421) using the probe_mask argument under 2.6.20. > > > With probe_mask=1, the codec isnt detected and only the modem shows up > > > (just like before). However, with probe_mask=8, the codec IS correctly > > > detected (alsa-info.sh at http://pastebin.ca/454059 ). syslog shows: > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 19 > > > ALSA .../alsa-driver-hg20070421/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:696: > > > codec_mask = 0x9 > > > ALSA .../alsa/alsa-driver-hg20070421/pci/hda/hda_codec.c:1819: > > > hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) > > > > > > However, there is still no playback (no muted control, all volumes at > > > 100% as shown by amixer in alsa-info.sh). Here again, capture works, I > > > successfully captured from the mic with arecord. > > > > Now you can try what I suggested, model=auto. And compare with > > codec#* files. If you have /proc/asound/card0/codec#* files on the > > working system, it'd be better, in addition... > > > > > > Takashi > > > > > > > > Can I draw your attention on something ? I m not sure about what i m > > > going to say (specially the part on NIDs) but i ll try. I know Tobin > > > already explained that this was normal, but if you have a look at the > > > post where I loaded edgy (2 posts ago), where the playback > > > works(2.6.17 with 1.0.12), you ll see that when I aplay something, > > > syslog shows only a single call to hda_codec_setup_stream with > > > NID=0x3. In all cases where playback doesnt work (1.0.14rcX under > > > 2.6.20), running aplay makes 4 calls to hda_codec_setup_stream with > > > NID=3, 6 ,5 and 4 in turn. Why does it try 6 5 and 4 ? Is it because 3 > > > fails ? We know that 3 should be used as its the one that works (under > > > 2.6.17). Now arent NIDs node IDs of codecs connected to the HDA > > > controller ? Is it possible that 6, 5 and 4 are mistakenly detected > > > and cannot be used for playback ? > > > Let me know if I m way off track. > > > > > Ben > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Mon, 23 Apr 2007 11:05:19 +0000, > > > > Ben DiDonc wrote: > > > > > > > > > > Takashi, > > > > > I did try passing model=auto to all 14 release candidates, > > > > > > > > Did you already try the HG version (or daily snapshot), right? > > > > > > > > > but without > > > > > success. the codec is never detected (see my previous posts for > > > > > alsa-info.sh outputs with both 2.6.20 and 2.6.17). > > > > > My problem is I cant get my sound card to work with kernel 2.6.20, > > > > > when it used to work with 2.6.17. > > > > > > > > The codec mask probed is 0x09, so you can pass either probe_mask=8 or > > > > probe_mask=1 to skip the modem detection (if it matters). > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > Ben > > > > > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > At Mon, 23 Apr 2007 10:50:18 +0000, > > > > > > Ben DiDonc wrote: > > > > > > > > > > > > > > Tobin, > > > > > > > Unfortunately, ACPI was on at all times. My laptop hardly ever boots > > > > > > > without ACPI (randomly, the SATA driver fails before it can even mount > > > > > > > the root partition). > > > > > > > I ran some more tests today: > > > > > > > I loaded Ubuntu Edgy from the live CD where the sound chip used to > > > > > > > work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works > > > > > > > straight away. From there , I ran the same series of test: download > > > > > > > and build (--with-debug=detect) the next version of alsa-driver, run a > > > > > > > capture from mic and try playing it back, check syslog and run > > > > > > > alsa-info.sh. I did this for successive versions of alsa-driver > > > > > > > (starting with 1.0.12) until it fails. > > > > > > > Here are the results: > > > > > > > > > > > > > > Alsa driver 1.0.12rc1(shipped with edgy): > > > > > > > capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 > > > > > > > > > > > > > > Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) > > > > > > > syslog lines after modprobe: > > > > > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: > > > > > > > codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: > > > > > > > line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 > > > > > > > (0x0/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, > > > > > > > dig_out=0x0, din_in=0x0 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, > > > > > > > fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 > > > > > > > > > > > > > > capture: working > > > > > > > //capture starting > > > > > > > hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, > > > > > > > channel=0, format=0x11 > > > > > > > //capture stopping > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > playback: working > > > > > > > //starting > > > > > > > ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, > > > > > > > fragsize=0x1000, format=0x11 > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, > > > > > > > channel=0, format=0x11 > > > > > > > //stopping > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) > > > > > > > modprobe snd-hda-intel: > > > > > > > hda_intel.c:680: codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, > > > > > > > cd=0x11, aux=0x0 > > > > > > > capture works fine, syslog output similar to 1.0.12 plus an extra > > > > > > > "hda-intel: Invalid position buffer, using LPIB read method instead" > > > > > > > in the middle of the capture. Playback works fine too, with the same > > > > > > > syslog entries as 1.0.12 > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) > > > > > > > modprobe: > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works (output similar to 1.0.12) > > > > > > > Playback does NOT work this time: > > > > > > > syslog shows a call to azx_pcm_prepare: and 4 calls to > > > > > > > hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out > > > > > > > my previous post) > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) > > > > > > > modprobe: the codec is correctly detected > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works. Playback does NOT work, same syslog entries as > > > > > > > 1.0.14rc1 with multiple calls to hda_codec_setup_stream > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) > > > > > > > modprobe: the codec isnt detected. > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > capture and playback dont work. > > > > > > > > > > > > > > Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy > > > > > > > (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but > > > > > > > not playback, 14rc3 doesnt detect the codec. > > > > > > > What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel > > > > > > > between releases 13 and 14 ... > > > > > > > > > > > > Try to pass model=auto to snd-hda-intel module, and check whether it > > > > > > works (make sure that you have unmuted/adjusted mixer elements before > > > > > > that). Then get /proc/asound/card0/codec#* files, and compare with > > > > > > the files without model option. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > > > >

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[alsa-devel] Compilation of latest alsa-hg
by dustin＠seznam.cz 24 Apr '07

24 Apr '07

Hello, please could you point me at a functioning description of latest complete hg tree compilation? The guides e.g. http://www.suseforums.net/index.php?showtopic=30156&pid=170707&st=0&#entry1… end up with /home/pavel/tmp/alsa-hg/alsa/alsa-driver/soc/codecs/../../alsa-kernel/soc/codecs/wm8753.c:1707: error: ‘I2C_DRIVERID_WM8753’ undeclared here (not in a function) Thanks a lot. Pavel.

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Re: [alsa-devel] Functioning support for Prodigy 192
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 14:00:22 +0200 (CEST), dustin(a)seznam.cz wrote: > > Hello, > > Back here again :) > > While comparing media reading precision of various CD players I hit the following issues: > > 1. De-emphasis filter in AK4114 is turned on, distorting incoming data. A patch to disable: > > diff -r aed6a15c9398 pci/ice1712/prodigy192.c > --- a/pci/ice1712/prodigy192.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/prodigy192.c Mon Apr 23 12:47:01 2007 +0200 > @@ -714,7 +714,7 @@ static int prodigy192_ak4114_init(struct > { > static const unsigned char ak4114_init_vals[] = { > AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1, > - AK4114_DIF_I24I2S, /* ice1724 expects I2S and provides clock */ > + AK4114_DIF_I24I2S | AK4114_DEM0 , /* ice1724 expects I2S and provides clock, DEM0 disables deemphasis filter */ > AK4114_TX1E, > AK4114_EFH_1024 | AK4114_DIT, /* default input RX0 */ > 0, Looks OK. > 2. All data recorded when master clock was provided by AK4114 (Multi Track Internal Clock= IEC958 Input) had double the sampling rate. For each input sample there were two identical output samples in the file. > > Here is where the FPGA on MIODIO gets introduced. Both MCKO1 and MCKO2 of ak4114 are fed to FPGA, which outputs master clock to SPMCLKIN of ice1724. Unfortunately my oscilloscope has just died, so experimentally I found out that only a combination of OCKS0=1, OCKS1 = 1 (128fs, 64fs output) and ice1724- VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct sampling rate. In order for this to work the FPGA must double the MCK01 rate (I cannot check the frequency now). > > It may actually make sense as it is probably the only way to support input sampling frequencies all the way up to 192kHz (128fs is max ak4114 can output at 192kHz) without having to switch the VT1724_MT_I2S_MCLK_128X bit in ice1724. In consumer mode the chips have no information about the sampling frequency (ak4114 does no detection) and the driver would not know when to set the bit. > > The problem is that due to switching sequence of Multi Track Internal Clock enum the VT1724_MT_I2S_MCLK_128X stays set (from the previous 192kHz option). I think it should not be set for ice1724 slave mode at all, here is the patch which fixes the sampling rate. Looks fine, too. Could you post a patch with a short patch summary and sign-off to apply to the upstream tree? thanks, Takashi > > Thanks, > > Pavel. > > diff -r aed6a15c9398 pci/ice1712/ice1724.c > --- a/pci/ice1712/ice1724.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/ice1724.c Mon Apr 23 13:57:19 2007 +0200 > @@ -1666,7 +1666,11 @@ static int snd_vt1724_pro_internal_clock > spin_lock_irq(&ice->reg_lock); > oval = inb(ICEMT1724(ice, RATE)); > if (ucontrol->value.enumerated.item[0] == spdif) { > + unsigned char i2s_oval; > outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); > + /* setting 256fs */ > + i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); > + outb( i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); > } else { > rate = rates[ucontrol->value.integer.value[0] % 15]; > if (rate <= get_max_rate(ice)) { > >

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[alsa-devel] Buffer Overrun Messages
by John Dey 24 Apr '07

24 Apr '07

Hi, I have been driving fluidsynth with with aplaymidi and get the following: fluidsynth: ALSAsequencer buffer overrun, lost event. I have played around with buffer size and buffer number and also run with and without jackd. The sound out doesn't seem to be affect. Can anyone give me some guidance on how to eliminate the error. Could the error be caused by fluidsynth? I have also run the program on PS2/Yellowdog linux and CCRMA fc5 (x86) both with similar results. Thanks. John

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[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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Re: [alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, Unfortunately, ACPI was on at all times. My laptop hardly ever boots without ACPI (randomly, the SATA driver fails before it can even mount the root partition). I ran some more tests today: I loaded Ubuntu Edgy from the live CD where the sound chip used to work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works straight away. From there , I ran the same series of test: download and build (--with-debug=detect) the next version of alsa-driver, run a capture from mic and try playing it back, check syslog and run alsa-info.sh. I did this for successive versions of alsa-driver (starting with 1.0.12) until it fails. Here are the results: Alsa driver 1.0.12rc1(shipped with edgy): capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) syslog lines after modprobe: ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, dig_out=0x0, din_in=0x0 ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture: working //capture starting hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 //capture stopping hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 playback: working //starting ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 //stopping ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) modprobe snd-hda-intel: hda_intel.c:680: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture works fine, syslog output similar to 1.0.12 plus an extra "hda-intel: Invalid position buffer, using LPIB read method instead" in the middle of the capture. Playback works fine too, with the same syslog entries as 1.0.12 alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) modprobe: ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works (output similar to 1.0.12) Playback does NOT work this time: syslog shows a call to azx_pcm_prepare: and 4 calls to hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out my previous post) alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) modprobe: the codec is correctly detected ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works. Playback does NOT work, same syslog entries as 1.0.14rc1 with multiple calls to hda_codec_setup_stream alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) modprobe: the codec isnt detected. ALSA hda_intel.c:696: codec_mask = 0x9 capture and playback dont work. Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but not playback, 14rc3 doesnt detect the codec. What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel between releases 13 and 14 ...

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[alsa-devel] ad1986a snd_hda_intel patch
by Dmitry Ursegov 23 Apr '07

23 Apr '07

Hello! I have an asus a8jn laptop and a integrated audio with Analog Devices AD1986A chip. $ lspci -vnn 0:1b.0 Audio device [0403]: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: ASUSTeK Computer Inc. Unknown device [1043:1447] Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at febfc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> The audio seems to work properly on the right channel, but not on the left. All I get on the left channel is a loud screeching noise followed by a pop when the audio turns off. The microphon jack don't work also. I make a patch from the solution described in this http://mailman.alsa-project.org/pipermail/alsa-devel/2007-March/000093.html message: diff -Nur alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c --- alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c 2007-03-06 15:26:32.000000000 +0300 +++ alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c 2007-04-20 23:32:19.000000000 +0400 @@ -873,7 +873,7 @@ board_config = snd_hda_check_board_config(codec, AD1986A_MODELS, ad1986a_models, ad1986a_cfg_tbl); - switch (board_config) { +/* switch (board_config) { case AD1986A_3STACK: spec->num_mixers = 2; spec->mixers[1] = ad1986a_3st_mixers; @@ -912,6 +912,18 @@ spec->multiout.dig_out_nid = 0; break; } +*/ + + spec->num_mixers = 2; + spec->mixers[1] = ad1986a_3st_mixers; + spec->num_init_verbs = 3; + spec->init_verbs[1] = ad1986a_3st_init_verbs; + spec->init_verbs[2] = ad1986a_ch2_init; + spec->channel_mode = ad1986a_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); + spec->need_dac_fix = 1; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; return 0; } I don't know, why snd_hda_check_board_config() return wrong value, but that workaround works. Thanks! Dmitry Ursegov

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[alsa-devel] Strange code in
by Pete Zaitcev 23 Apr '07

23 Apr '07

Dear Takashi: Hans de Goede (on cc:) hits a problem when snd-usb-audio breaks with: "USB device is in the shutdown state, cannot create a card instance". This is how the place looks: chip = NULL; for (i = 0; i < SNDRV_CARDS; i++) { if (usb_chip[i] && usb_chip[i]->dev == dev) { if (usb_chip[i]->shutdown) { snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n"); goto __error; } chip = usb_chip[i]; break; } } The code looks broken to me, because dev at this time is a new device structure, freshly allocated. You cannot compare it with an old instance, and I don't know why you would want to. Unfortunately (for fortunately), I have zero clue about the way ALSA works, specifically rules for snd_device_ops.dev_free, so I am asking you to look at this problem. Yours, -- Pete

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[alsa-devel] [PATCH] sound: fix incorrect use of platform_device_register()
by Akinobu Mita 23 Apr '07

23 Apr '07

The platform_device allocated by platform_device_alloc() should be added to the device hierarchy by platform_device_add() instead of platform_device_register(). Otherwise it will hit WARN_ON() in platform_device_register(). by illegal refcount. This patch fixes such incorrect usages in portman2x4 and mts64 drivers. Also it removes unnecessary trailing whitespaces. Cc: Jaroslav Kysela <perex(a)suse.cz> Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Index: 2.6-mm/sound/drivers/portman2x4.c =================================================================== --- 2.6-mm.orig/sound/drivers/portman2x4.c +++ 2.6-mm/sound/drivers/portman2x4.c @@ -676,13 +676,13 @@ static void __devinit snd_portman_attach struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; } Index: 2.6-mm/sound/drivers/mts64.c =================================================================== --- 2.6-mm.orig/sound/drivers/mts64.c +++ 2.6-mm/sound/drivers/mts64.c @@ -892,13 +892,13 @@ static void __devinit snd_mts64_attach(s struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; }

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Re: [alsa-devel] Problem with Conexant
by Tobin Davis 23 Apr '07

23 Apr '07

Did you try adjusting the PCM volume levels? They may be set too low. That particular sound chip required a lot of manipulating to get functional, and is still very problematic. Unfortunately, I have no way of testing it, so users (like yourself) have to be my ears. The last time I worked with that particular configuration, the user that was testing said that it took a bit of tweaking with the PCM and other volume levels to get good sound. Tobin On Mon, 2007-04-23 at 08:11 +0307, Black X wrote: > after I install the last alsa-driver from the link you give me it > ( alsa-driver-hg20070421.tar.bz2 ) I still have the same problem. now > what I most to do to inform you about this issue? > > > Thank you. > > On Sun, 2007-04-22 at 19:27 -0700, Tobin Davis wrote: > > Could you try the latest alsa-driver snapshot from > > ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver? > > > > I can't tell what revision of Conexant code you are using based on > > your email. Anything after 20070413 should work better, and it will > > also be the most current. I haven't made any changes to the code > > since ~April 6. If you are still having audio issues, let me know. > > > > Tobin > > > > > > On Sun, 2007-04-22 at 23:31 +0307, Black X wrote: > > > Hello > > > > > > First at all I wanna thanks Tobin Davis for his support to get Conexant > > > audio devices working under Linux, and I was testing his last patchs, > > > but I had a little problems with my sound card, the first one is about > > > headphone, it is works but with noise, and when I'm maximize the volume > > > of headphone the sound quality starting so weak. > > > > > > The Second problem I had it is when I run the 3D Desktop Effects (XGL > > > with Compiz), that when I play any sound clip and moving any window or > > > rotate desktop cube that's making a noise with the sound, and the > > > quality of the sound clip it is so weak with slow playing in the sound. > > > > > > I make a little screen cast about this problem and you can see it at > > > this link: > > > http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html > > > > > > any way this is my alsa information: http://pastebin.ca/452453 > > > finaly my device information : > > > http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… > > > > > > I 'm sorry, but the English it's not my mother language. > > > > > > Thank you. > > > > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > > Tobin Davis <tdavis(a)dsl-only.net> > -- Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Ok, I try that before and try it now, still no change, and still have the same problem. When I highest the sound volume of headphone this makes sound starting noisy (and that's the same with PCM). About speaker it's work fine, but IF I moving any window or rotate the desktop cube this makes the sound very slow with noise.

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[alsa-devel] partial fix for bug #2708: [ALSA - driver] PCI - hda-intel
by Jeffrey Arbuckle 23 Apr '07

23 Apr '07

Using alsa-driver-1.0.14rc3 with an iMac Core2 Duo 24" results in audio only available through the headphones. Additionally the sound is very low, even at the highest mixer setting. This problem is very similar to the bug recently fixed in 1.0.14rc2 with the audio on the MacPro. I was able to make a patch that applies the same fix for the MacPro to my iMac. The patch fixes the audio so that the built in speakers work normally, but now the headphones do not work at all. Obviously some more work is needed on this. The patch I've attached just adds my device's' PCI SSID to an existing switch statement. I am willing to provide some more testing if necessary. Here is my systems info: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Realtek Semiconductor Co., Ltd. Unknown device 0885 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 92400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 # cat /proc/asound/card0/codec#0 Codec: Realtek ALC885 Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x106b1000 Revision Id: 0x100103 # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3. Compiled on Apr 22 2007 for kernel 2.6.20-15-generic (SMP).

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[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Hello First at all I wanna thanks Tobin Davis for his support to get Conexant audio devices working under Linux, and I was testing his last patchs, but I had a little problems with my sound card, the first one is about headphone, it is works but with noise, and when I'm maximize the volume of headphone the sound quality starting so weak. The Second problem I had it is when I run the 3D Desktop Effects (XGL with Compiz), that when I play any sound clip and moving any window or rotate desktop cube that's making a noise with the sound, and the quality of the sound clip it is so weak with slow playing in the sound. I make a little screen cast about this problem and you can see it at this link: http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html any way this is my alsa information: http://pastebin.ca/452453 finaly my device information : http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… I 'm sorry, but the English it's not my mother language. Thank you.

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[alsa-devel] Qustions on dmix - aplay - hw_ptr - appl_ptr
by Matthew Foster 23 Apr '07

23 Apr '07

I am a driver developer that is new to ALSA. I have read the tutorial as an example on how to write an ALSA driver. I appreciate the documentation you have provided online. I was wondering if you might have time to possibly answer a question. I am having trouble understanding things in the memory/buffer management area. I am not writing a driver for a PCI device, I am writing a driver that acts as an interface from the ALSA to a render device. So I essentially have a circular buffer I am transferring the ALSA data to. I set up the ALSA buffer like this: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), ALSA_BUF_SIZE, ALSA_BUF_SIZE)) < 0); And later on in my driver after open is called I have a buffer polling loop that does the period notification and constantly monitors my render (ouput) buffer. I have my asound.conf setup like this: pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } This is to enable the dmix plugin so I can run aplay more than once and have the two streams mixed in. The problem I am having is that it seems that when I run aplay like this: "aplay close.wav" I only see this pointer moving: substream->runtime->status->hw_ptrbir And if I run aplay like this: "aplay -D plughw:0,0 close.wav" I only see this pointer moving: substream->runtime->control->appl_ptr This seems odd to me and currently in my code I just check which pointer moves and use that to calculate how much data I need to write to my render device. This is so I can support both use cases. Another major problem is that when I play a short wav file 25KB or so and use the dmix "aplay close.wav" it randomly doesn't play at sometimes. Also if I run "aplay close.wav" and then very quickly run "aplay close.wav" again at the same prompt I get corruption of the data which produces screeching. In the case where I run aplay one right after the other I notice that the hw_ptr moves differently (less) than if I run it, wait, run again. I am also noticing also that when I do the aplay over and over quickly that these values change (get lower) in my snd_pcm_prepare callback: chip_data->buf_size = snd_pcm_lib_buffer_bytes(substream); chip_data->period_buf_size = snd_pcm_lib_period_bytes(substream); It seems like those value should remain constant, and I use those values to calculate later in my polling thread to notify of a period lapse. It's almost like the runtime is telling me the period has changed and the buffer size has changed. But this only happens in this case with a short wav file and running aplay over and over quickly. Is there a way to make sure the hw_ptr doesn't get affected by multiple instances of aplay trying to render out small clips. It seems like maybe I am setting up something incorrectly or maybe not blocking/ waiting correctly. I am at a loss in trying to figure this out, wondering if you would be so kind as to comment on my situation. I really appreciate your time in reading this email. Matt

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[alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, I downloaded and installed alsa-driver, alsa-lib and alsa-utils, but still no sound coming out of those speakers... I ran alsa-info.sh and the results are at http://pastebin.ca/451128 I am clueless as to what to do next, but I m ready to try anything. Any help would be greatly appreciated. Ben On 4/21/07, Tobin Davis < tdavis(a)dsl-only.net> wrote: > > Try downloading a current daily snapshot from ftp://ftp.suse.com/pub/projects/alsa/snapshot/ > . > > There was an issue with the ATI HD Audio detection that was fixed after > 14rc3 was released. > > Tobin > > On Sat, 2007-04-21 at 16:00 +1000, Ben DiDonc wrote: > > The output of alsa-info.sh is available at http://pastebin.ca/451007 > > > On 4/21/07, *Tobin Davis* < tdavis(a)dsl-only.net> wrote: > > Could you download an run http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > , then post the resulting link? This will help determine what setup you > have. > > Tobin Davis > > > > On Sat, 2007-04-21 at 14:56 +1000, Ben DiDonc wrote: > > Hi I have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450. > Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy). I > just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa 1.0.14rc1) > and the only mixer settings available are those of the modem, named off-hook > and caller-id (see output of amixer, aplay -l , and the rest in attached > file output_feisty) I decided to upgrade alsa-driver and alsa-lib to > 1.0.14rc2. after downloading the package, I unloaded all sound related > modules (everything with "snd_" in it) in the right order and did: > ./configure --with-oss=yes --with-cards=hda-intel;make; sudo make install; > sudo modprobe snd_hda_intel A minor improvement: the mixer now shows > controls for both the modem and the sound card (see > output_after_upgrade_to_rc2), but unfortunately, there is still no sound > coming out of those speakers (even though alsamixer shows control for PCM > and the rest...). Finally, I downloaded alsa-driver-1.0.14rc3 and gave it > a shot. This time again, only modem related settings appeared in alsamixer, > and no sound card stuff (see aplay and amixer in > output_after_upgrade_to_rc3), and no sound either. Does anyone know how to > get some sound out of these speakers (given that it used to work with > alsa1.0.12rc1...) ? Thanks for your help Ben > > _______________________________________________Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > Tobin Davis <tdavis(a)dsl-only.net> > > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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Re: [alsa-devel] iec958 switch uneffective while playing ac3 stream
by Takashi Iwai 22 Apr '07

22 Apr '07

[Nice analysis cut out..] At Tue, 03 Apr 2007 10:17:16 +0200, Dominique Dumont wrote: > > If the stream format is not valid when the digi converter is set up, > there's a chance that the digi converter goes belly up. > > I reckon that setting the stream *then* the digi converter would be safer. OK, now question is whether we need to turn off DIGI_CONV* once before setting it up again. Anyway, a test patch is below. It forces to set DIGI_CONV* verbs in prepare. Give it a try. thanks, Takashi diff -r 05ecca0fba92 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_codec.c Tue Apr 03 15:28:47 2007 +0200 @@ -1083,6 +1083,19 @@ static int snd_hda_spdif_default_put(str return change; } +static void setup_digi_convert(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned short val = codec->spdif_ctls; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | + AC_AMP_SET_OUTPUT | + ((val & 1) ? 0 : 0x80)); +} + static int snd_hda_spdif_out_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1114,10 +1127,7 @@ static int snd_hda_spdif_out_switch_put( change = codec->spdif_ctls != val; if (change || codec->in_resume) { codec->spdif_ctls = val; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | - AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80)); + setup_digi_convert(codec, nid); } mutex_unlock(&codec->spdif_mutex); return change; @@ -1901,6 +1911,21 @@ int snd_hda_multi_out_dig_open(struct hd return 0; } +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + mutex_lock(&codec->spdif_mutex); + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + stream_tag, 0, format); + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} + /* * release the digital out */ @@ -1949,6 +1974,8 @@ int snd_hda_multi_out_analog_prepare(str mout->dig_out_used = 0; snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); } + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); } mutex_unlock(&codec->spdif_mutex); diff -r 05ecca0fba92 pci/hda/hda_local.h --- a/pci/hda/hda_local.h Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_local.h Tue Apr 03 15:31:29 2007 +0200 @@ -148,6 +148,11 @@ struct hda_multi_out { int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout); +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, diff -r 05ecca0fba92 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/patch_realtek.c Tue Apr 03 15:32:23 2007 +0200 @@ -1916,6 +1916,17 @@ static int alc880_dig_playback_pcm_open( return snd_hda_multi_out_dig_open(codec, &spec->multiout); } +static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1984,7 +1995,8 @@ static struct hda_pcm_stream alc880_pcm_ /* NID is set in alc_build_pcms */ .ops = { .open = alc880_dig_playback_pcm_open, - .close = alc880_dig_playback_pcm_close + .close = alc880_dig_playback_pcm_close, + .prepare = alc880_dig_playback_pcm_prepare }, };

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[alsa-devel] Please ignore, test.
by Rene Herman 22 Apr '07

22 Apr '07

Body Attachment

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Re: [alsa-devel] No sound with ATI SB450
by Ben DiDonc 21 Apr '07

21 Apr '07

The output of alsa-info.sh is available at http://pastebin.ca/451007 On 4/21/07, Tobin Davis <tdavis(a)dsl-only.net> wrote: > > Could you download an run > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh , then post the > resulting link? This will help determine what setup you have. > > Tobin Davis > > On Sat, 2007-04-21 at 14:56 +1000, Ben DiDonc wrote: > > HiI have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450.Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy).I just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa1.0.14rc1) and the only mixer settings available are those of themodem, named off-hook and caller-id (see output of amixer, aplay -l ,and the rest in attached file output_feisty) > I decided to upgrade alsa-driver and alsa-lib to 1.0.14rc2. afterdownloading the package, I unloaded all sound related modules(everything with "snd_" in it) in the right order and did:./configure --with-oss=yes --with-cards=hda-intel;make; sudo makeinstall; sudo modprobe snd_hda_intelA minor improvement: the mixer now shows controls for both the modemand the sound card (see output_after_upgrade_to_rc2), butunfortunately, there is still no sound coming out of those speakers(even though alsamixer shows control for PCM and the rest...). > Finally, I downloaded alsa-driver-1.0.14rc3 and gave it a shot. Thistime again, only modem related settings appeared in alsamixer, and nosound card stuff (see aplay and amixer inoutput_after_upgrade_to_rc3), and no sound either. > Does anyone know how to get some sound out of these speakers (giventhat it used to work with alsa1.0.12rc1...) ? > Thanks for your helpBen > _______________________________________________Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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[alsa-devel] No sound with ATI SB450
by Ben DiDonc 21 Apr '07

21 Apr '07

Hi I have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450. Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy). I just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa 1.0.14rc1) and the only mixer settings available are those of the modem, named off-hook and caller-id (see output of amixer, aplay -l , and the rest in attached file output_feisty) I decided to upgrade alsa-driver and alsa-lib to 1.0.14rc2. after downloading the package, I unloaded all sound related modules (everything with "snd_" in it) in the right order and did: ./configure --with-oss=yes --with-cards=hda-intel;make; sudo make install; sudo modprobe snd_hda_intel A minor improvement: the mixer now shows controls for both the modem and the sound card (see output_after_upgrade_to_rc2), but unfortunately, there is still no sound coming out of those speakers (even though alsamixer shows control for PCM and the rest...). Finally, I downloaded alsa-driver-1.0.14rc3 and gave it a shot. This time again, only modem related settings appeared in alsamixer, and no sound card stuff (see aplay and amixer in output_after_upgrade_to_rc3), and no sound either. Does anyone know how to get some sound out of these speakers (given that it used to work with alsa1.0.12rc1...) ? Thanks for your help Ben

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[alsa-devel] [PATCHv2] ALSA: Support Media Vision Jazz16 chips
by Rask Ingemann Lambertsen 20 Apr '07

20 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add support for Jazz16 features to the common SoundBlaster code. - Add a front end for Jazz16 based sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20.6. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch is a prerequisite for the Jazz PnP patch, which needs the patched include/sound/sb.h. Documentation/sound/alsa/ALSA-Configuration.txt | 10 include/sound/sb.h | 5 sound/isa/Kconfig | 15 + sound/isa/sb/Makefile | 2 sound/isa/sb/jazz16.c | 183 ++++++++++++++ sound/isa/sb/sb8_main.c | 98 +++++++ sound/isa/sb/sb_common.c | 12 sound/isa/sb/sb_mixer.c | 3 8 files changed, 323 insertions(+), 5 deletions(-) Changes since the last patch: The same DMA channel is now used for both 8-bit and 16-bit sound. DSP high-speed mode isn't used at all now for the Jazz16. A 16-bit DMA buffer can now be 128 kB instead of 64 kB. diff -rup linux-2.6.20.6-clean/Documentation/sound/alsa/ALSA-Configuration.txt linux-2.6.20.6-ril/Documentation/sound/alsa/ALSA-Configuration.txt --- linux-2.6.20.6-clean/Documentation/sound/alsa/ALSA-Configuration.txt 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/Documentation/sound/alsa/ALSA-Configuration.txt 2007-04-07 17:31:38.000000000 +0200 @@ -1120,6 +1120,16 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based on Media Vision Jazz16 chip (PnP only) + + This module supports multiple cards and PnP. + + Note: You will most likely want to enable Jazz16 PnP support + (CONFIG_JAZZ16PNP) in the kernel. + Module snd-korg1212 ------------------- diff -rup linux-2.6.20.6-clean/include/sound/sb.h linux-2.6.20.6-ril/include/sound/sb.h --- linux-2.6.20.6-clean/include/sound/sb.h 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/include/sound/sb.h 2007-04-07 17:31:38.000000000 +0200 @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ @@ -142,6 +143,7 @@ struct snd_sb { #define SB_DSP_HI_INPUT_AUTO 0x98 #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +267,9 @@ struct snd_sb { #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ diff -rup linux-2.6.20.6-clean/sound/isa/Kconfig linux-2.6.20.6-ril/sound/isa/Kconfig --- linux-2.6.20.6-clean/sound/isa/Kconfig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/Kconfig 2007-04-07 17:31:38.000000000 +0200 @@ -248,6 +248,21 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say Y here to include support for Media Vision Jazz16 based + soundcards. + + You probably want to enable Jazz16 Plug and Play support + (CONFIG_JAZZ16PNP) also. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND diff -rup linux-2.6.20.6-clean/sound/isa/sb/Makefile linux-2.6.20.6-ril/sound/isa/sb/Makefile --- linux-2.6.20.6-clean/sound/isa/sb/Makefile 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/Makefile 2007-04-07 17:31:38.000000000 +0200 @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb8_main.c linux-2.6.20.6-ril/sound/isa/sb/sb8_main.c --- linux-2.6.20.6-clean/sound/isa/sb/sb8_main.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb8_main.c 2007-04-12 00:03:35.000000000 +0200 @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Wed Apr 11 23:56:18 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements (mostly ported from sound/oss/sb_common.c). */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,13 +113,52 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + if (c->max > 1) { + unsigned int num = 0, den = 0; + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char stereo, format; + stereo = runtime->channels > 1; rate = runtime->rate; switch (chip->hardware) { case SB_HW_PRO: @@ -123,6 +174,7 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -134,8 +186,14 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + else + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -167,6 +225,8 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->dma8 > 3) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); @@ -233,7 +293,9 @@ static int snd_sb8_capture_prepare(struc struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char stereo, format; + stereo = runtime->channels > 1; rate = runtime->rate; switch (chip->hardware) { case SB_HW_PRO: @@ -250,6 +312,7 @@ static int snd_sb8_capture_prepare(struc break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -261,10 +324,14 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + else + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (stereo || chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -279,6 +346,8 @@ static int snd_sb8_capture_prepare(struc snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_OUTPUT) { + if (chip->dma8 > 3) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); @@ -443,6 +512,19 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -465,6 +547,14 @@ static int snd_sb8_open(struct snd_pcm_s } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -512,6 +602,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc; if (rpcm) *rpcm = NULL; @@ -524,9 +615,10 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + max_prealloc = (chip->dma8 > 3) ? 128 * 1024 : 64 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb_common.c linux-2.6.20.6-ril/sound/isa/sb/sb_common.c --- linux-2.6.20.6-clean/sound/isa/sb/sb_common.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb_common.c 2007-04-11 20:23:55.000000000 +0200 @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb_mixer.c linux-2.6.20.6-ril/sound/isa/sb/sb_mixer.c --- linux-2.6.20.6-clean/sound/isa/sb/sb_mixer.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb_mixer.c 2007-04-07 17:31:38.000000000 +0200 @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: --- /dev/null 2007-04-11 19:30:17.320748640 +0200 +++ linux-2.6.20.6-ril/sound/isa/sb/jazz16.c 2007-04-05 23:30:27.000000000 +0200 @@ -0,0 +1,183 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * The Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * There is also an MPU-401 interface. + * + * AFAIK, the following systems have an on-board Jazz16 chip: + * IBM PPS Model 6015 + * Texas Intstruments/Acer Travelmate TM5000 series. + * Please tell me if this driver works with these boards or not. + */ + +#include <sound/driver.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Jazz16 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Jazz16 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Jazz16 soundcard."); + +#define PFX "jazz16: " +#define PFX_WARN KERN_WARNING PFX +#define PFX_ERR KERN_ERR PFX + +static struct pnp_driver snd_jazz16_pnp_driver; + +static irqreturn_t jazz16_interrupt(int irq, void *card) +{ + return snd_sb8dsp_interrupt(card); +} + +static struct pnp_device_id snd_jazz16_pnpids[] = { + { .id = "PNPb00f" }, + { .id = "" } +}; +MODULE_DEVICE_TABLE(pnp, snd_jazz16_pnpids); + +static int __devinit snd_jazz16_pnp_probe(struct pnp_dev *pnp_dev, + const struct pnp_device_id *pnp_id) +{ struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int err; + uint sbport, sbirq, sbdma, mpuport, mpuirq; + static uint dev_num = 0; + + if (enable[dev_num]) + card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0); + else + card = NULL; + dev_num++; + if (card == NULL) + return enable[dev_num] ? -ENOMEM : -ENODEV; + snd_card_set_dev(card, &pnp_dev->dev); + pnp_set_drvdata(pnp_dev, card); + /* TODO use pnp_port_valid(), pnp_port_flags(), pnp_port_length()... */ + sbport = pnp_port_start(pnp_dev, 0); + sbirq = pnp_irq(pnp_dev, 0); + sbdma = pnp_dma(pnp_dev, 0); + err = snd_sbdsp_create(card, sbport, sbirq, jazz16_interrupt, + sbdma, -1, SB_HW_AUTO, &chip); + if (err < 0) + goto out_err; + + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at %#x, irq %u, dma %u", + chip->name, sbport, sbirq, chip->dma8); + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(PFX_ERR "Not a Jazz16 chip at %#x.\n", sbport); + snd_card_free(card); + return -ENODEV; + } + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto out_err; + + err = snd_sbmixer_new(chip); + if (err < 0) + goto out_err; + + err = snd_opl3_create(card, sbport, sbport + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(PFX_WARN "No OPL device found, skipping.\n"); + else { + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto out_err; + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto out_err; + } + if (pnp_port_valid(pnp_dev, 1) + && !(pnp_port_flags(pnp_dev, 1) & IORESOURCE_DISABLED)) { + int mpuirqflags; + if (!pnp_irq_valid(pnp_dev, 1) + || pnp_irq_flags(pnp_dev, 1) & IORESOURCE_DISABLED) { + mpuirq = -1; + mpuirqflags = 0; + } else { + mpuirq = pnp_irq(pnp_dev, 1); + mpuirqflags = IRQF_DISABLED; + } + mpuport = pnp_port_start(pnp_dev, 1); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport, 0, mpuirq, + mpuirqflags, NULL); + if (err < 0) + snd_printk(PFX_WARN "MPU-401 device not configured, skipping.\n"); + } + err = snd_card_register(card); + if (err >= 0) + return 0; +out_err: + snd_card_free(card); + return err; +} + +static void __devexit snd_jazz16_pnp_remove(struct pnp_dev *dev) +{ + snd_card_free(pnp_get_drvdata(dev)); + pnp_set_drvdata(dev, NULL); +} + +static struct pnp_driver snd_jazz16_pnp_driver = { + .name = "Jazz16", + .id_table = snd_jazz16_pnpids, + .probe = snd_jazz16_pnp_probe, + .remove = __devexit_p(snd_jazz16_pnp_remove), +}; + +static int __devinit alsa_card_jazz16_init(void) +{ + return pnp_register_driver(&snd_jazz16_pnp_driver); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + pnp_unregister_driver(&snd_jazz16_pnp_driver); +} + +module_init(alsa_card_jazz16_init); +module_exit(alsa_card_jazz16_exit); -- Rask Ingemann Lambertsen

3 3

[alsa-devel] [Alsa-devel] Read the input of a soundcard with ALSA
by Nyx 20 Apr '07

20 Apr '07

Hi, I have some problems to read the input of my soundcard using Alsa and I don't really understand how to have access to the input of my soundcard. I follow : http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht… # So, in my program, I have created two handles for capture and playback : char *device = "hw:0,0"; snd_output_t *output; snd_input_t *input; snd_pcm_t *phandle, *chandle; # then I connect the phandle to the out err = snd_output_stdio_attach(&output, stdout, 0); if (..) # Do I have to connect the chandle with the input ? # err = snd_input_stdio_attach(&input, stdin, 0); # I open the access to PCM err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, 0) if (..) err = snd_pcm_open(&chandle, pdevice, SND_PCM_STREAM_CAPTURE, 0) if(..) # Hardware and software Parameters ## Hardware Parameters snd_pcm_hw_params; int err; err = snd_pcm_hw_params_any(phandle, params); if(..); err = snd_pcm_hw_params_set_access(phandle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if(..); err = snd_pcm_hw_params_set_format(phandle, params,SND_PCM_FORMAT_S16); if(..); err = snd_pcm_hw_params_set_channels(phandle, params, 1); if(..); err = snd_pcm_hw_params_set_rate_near(phandle, params, 44100, 0); if(..); err = snd_pcm_hw_params_set_buffer_time_near(phandle, params, 500000, &dir); if(..); err = snd_pcm_hw_params_set_time_near(phandle, params, 100000, &dir), if(..); err = snd_pcm_hw_params(phandle, params); ##Software parameters err = snd_pcm_sw_params(phandle, params); if(..); err = snd_pcm_sw_params_set_start_threshold(phandle, swparams, 0x7fffffff); if(..); err = snd_pcm_sw_params_set_avail_min(phandle, swparams, 4); # same configuration for phandle # Link output to input and start err = snd_pcm_link(chandle, phandle); if(..); err = snd_pcm_start(chandle); if(..); #Then I want to read my input frames_in = 0; in_max = 0; latency = 28; buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2); while (1) { r = readbuf(chandle, buffer, latency, &frames_in, &in_max)); if(..); } # description of the function long readbuf(snd_pcm_t *handle, char *buf, long len, size_t *frames, size_t *max) { long r; do { r = snd_pcm_readi(handle, buf, len); } while (r == -EAGAIN); if (r > 0) { *frames += r; if ((long)*max < r) *max = r; } printf ("r = %s, len %li and buf[%d] %f \r", snd_strerror(r), len,i, buf[i]); return r; } So I compile it and I run it, and I have $ read = Broken Pipe, len = 28 and buf[..] = 0.000 Anyone can help me to read the input of my sound card ? Thank you -- K

1 0

Re: [alsa-devel] [PATCH 16/16] alsa sound support for mips
by Ralf Baechle 20 Apr '07

20 Apr '07

On Sun, Apr 15, 2007 at 11:26:05PM +0800, tiansm(a)lemote.com wrote: (Adding a few more people to the cc'list) > sound/core/pcm_native.c | 10 ++++++++++ > sound/core/sgbuf.c | 9 +++++++++ > 2 files changed, 19 insertions(+), 0 deletions(-) > > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index 3e276fc..9005bac 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -3145,7 +3145,11 @@ static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, > return NOPAGE_OOM; /* XXX: is this really due to OOM? */ > } else { > vaddr = runtime->dma_area + offset; > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) Please use CONFIG_MIPS instead of __mips__ in #if / #ifdefs. The question if #ifdefing is the right approach to solve this problem is something else but I think no, .... > + page = virt_to_page(CAC_ADDR(vaddr)); > +#else > page = virt_to_page(vaddr); > +#endif So this is needed because the MIPS virt_to_page is returning a unsuitable value if vaddress is not a KSEG0 (64-bit: cached XKPHYS) address which is what GFP allocations and the slab will return. So now we have to deciede if a) the MIPS __pa() should be changed to handle uncached addresses. b) the sound code here is simply broken. Some drivers seem to allocate runtime->dma_area from vmalloc, so this whole area in the sound code is looking like built on quicksand ... > } > get_page(page); > if (type) > @@ -3261,6 +3265,12 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) > substream = pcm_file->substream; > snd_assert(substream != NULL, return -ENXIO); > > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) > + /* all mmap using uncached mode */ > + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); > + area->vm_flags |= ( VM_RESERVED | VM_IO); VM_RESERVED will prevent the buffer from being freed. I assume that is another workaround for some kernel subsystem blowing up when being fed a pointer to an uncached RAM address? This smells like a memory leak. > +#endif > + > offset = area->vm_pgoff << PAGE_SHIFT; > switch (offset) { > case SNDRV_PCM_MMAP_OFFSET_STATUS: > diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c > index cefd228..535f0bc 100644 > --- a/sound/core/sgbuf.c > +++ b/sound/core/sgbuf.c > @@ -91,12 +91,21 @@ void *snd_malloc_sgbuf_pages(struct device *device, > } > sgbuf->table[i].buf = tmpb.area; > sgbuf->table[i].addr = tmpb.addr; > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) > + sgbuf->page_table[i] = virt_to_page(CAC_ADDR(tmpb.area)); VM_RESERVED will prevent the buffer from being freed. I assume that is another workaround for some kernel subsystem blowing up when being fed a pointer to an uncached RAM address? This smells like a memory leak. > +#else > sgbuf->page_table[i] = virt_to_page(tmpb.area); > +#endif > sgbuf->pages++; > } > > sgbuf->size = size; > +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) > + /* maybe we should use uncached accelerated mode */ > + dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP | VM_IO, pgprot_noncached(PAGE_KERNEL)); > +#else > dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); > +#endif I would suggest to get rid of this ifdef with a new arch-specific function like vmap_io_buffer which will do whatever a platform seems fit for this case? > if (! dmab->area) > goto _failed; > return dmab->area; Ralf

3 2

[alsa-devel] Trying to support hda-intel ALC883_CLEVO
by Pablo N. Hess 19 Apr '07

19 Apr '07

I have a Clevo M600 notebook, which comes with Realtek's HDA Intel audio chip. 'lspci -vn' says: 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1558:0660 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 6c340000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Linux 2.6.20 (Gentoo's gentoo-sources-2.6.20r4) recognizes it as an ALC883 chip, but no parameter combination would make the sound work properly (and that includes 'model=clevo'). As this module ships, no output would come from the built-in speakers, and the headphone volume was miserably low. patch_realtek.c supports vendor=1558 device=0660, but only as a ALC880 chip. However, mine *is* an ALC883 vendor=1558 device=0660. After copying the ALC880_CLEVO support code lines into a new ALC883_CLEVO (and changing every occurence of ALC880 to ALC883 in that new code block), recompiling the kernel and loading the snd-hda-intel with model=clevo, almost everything started working fine. I mean, sound comes as perfectly as it can through the built-in speakers, and the built-in mic also outputs its sound through these speakers. However, I can't capture anything. Alsamixer shows no capture device at all (although all playback devices appear to be present). This is very strange to me, as the built-in mic really *does* output its sound through the speakers. As you can probably see, I don't know anything about sound chips. I just did what seemed understandable (to me) to add proper support for my ALC883_CLEVO device. Thank you very much if you can help me solve this issue, as I really need to speak with my family and girlfriend far, far away. Cheers, Pablo Hess

2 1

[alsa-devel] [KJ][PATCH] sound: SPIN_LOCK_UNLOCKED cleanup
by Milind Arun Choudhary 19 Apr '07

19 Apr '07

SPIN_LOCK_UNLOCKED cleanup,use __SPIN_LOCK_UNLOCKED instead Signed-off-by: Milind Arun Choudhary <milindchoudhary(a)gmail.com> --- soc/at91/at91-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/at91/at91-i2s.c b/sound/soc/at91/at91-i2s.c index 9fc0c03..d65a47d 100644 --- a/sound/soc/at91/at91-i2s.c +++ b/sound/soc/at91/at91-i2s.c @@ -151,20 +151,20 @@ static struct at91_ssc_info { } ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = 0, .initialized = 0, }, #if NUM_SSC_DEVICES == 3 { .name = "ssc1", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = 0, .initialized = 0, }, { .name = "ssc2", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = 0, .initialized = 0, }, -- Milind Arun Choudhary

1 0

[alsa-devel] CS4270 supported?
by Timur Tabi 19 Apr '07

19 Apr '07

I've never worked with ALSA before, but I'm about to start developing on an ALSA driver for our hardware, which has a CS4270 on it attached via an I2S interface. After scanning the source code and mailing lists, I see no mention of the CS4270. So does that mean I would need to write an ASoC codec driver for the CS4270? If so, does the data sheet (http://www.cirrus.com/en/pubs/proDatasheet/CS4270_PP1.pdf) contain enough information? -- Timur Tabi Linux Kernel Developer @ Freescale

4 7

[alsa-devel] Trying to support hda-intel ALC883_CLEVO
by natunobilis＠gentoobr.org 19 Apr '07

19 Apr '07

I have a Clevo M600 notebook, which comes with Realtek's HDA Intel audio chip. 'lspci -vn' says: 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1558:0660 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 6c340000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Linux 2.6.20 (Gentoo's gentoo-sources-2.6.20r4) recognizes it as an ALC883 chip, but no parameter combination would make the sound work properly (and that includes 'model=clevo'). As this module ships, no output would come from the built-in speakers, and the headphone volume was miserably low. patch_realtek.c supports vendor=1558 device=0660, but only as a ALC880 chip. However, mine *is* an ALC883 vendor=1558 device=0660. After copying the ALC880_CLEVO support code lines into a new ALC883_CLEVO (and changing every occurence of ALC880 to ALC883 in that new code block), recompiling the kernel and loading the snd-hda-intel with model=clevo, almost everything started working fine. I mean, sound comes as perfectly as it can through the built-in speakers, and the built-in mic also outputs its sound through these speakers. However, I can't capture anything. Alsamixer shows no capture device at all (although all playback devices appear to be present). This is very strange to me, as the built-in mic really *does* output its sound through the speakers. As you can probably see, I don't know anything about sound chips. I just did what seemed understandable (to me) to add proper support for my ALC883_CLEVO device. Thank you very much if you can help me solve this issue, as I really need to speak with my family and girlfriend far, far away. Cheers, Pablo Hess

1 0

[alsa-devel] Surround problems on ALC883
by Pavel Patz 18 Apr '07

18 Apr '07

Hi all, I have problems with chip Realtek ALC883 (lspci: 00:06.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2)) and surround sound. Stereo sound is working (games, music) fine - all 6 speaker are playing. But when I tried to run "aplay chan-id.wav", i've heard only "front left" (front & rear left + center were playing) and "front right" (right front & rear were playing), then was silence until the end of the file. Running "aplay -Dsurround51 chan-id.wav" causes that aplay hangs-up without any output. Running it with -v parameter produces this output: Playing WAVE 'Desktop/Surround-SDL-testfiles/chan-id.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Channels 6 Soft volume PCM Control: PCM Playback Volume min_dB: -51 resolution: 256 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 Slave: Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 and then hang up until i press ctrl+c. Same results i get with speaker-test. I'm using alsa-lib-1.0.14-0.4.rc3.fc7 I've found similar problem in archive (about half a year old) but it still seems it isn't working. Is there any possibility, that this will be fixed in future? Thanks for all your help (and please leave me in CC, i'm not on mailing-list). Following lsmod & /proc/asound/card0/codec#0 [cdome@andromeda ~]$ /sbin/lsmod |grep snd snd_hda_intel 27936 2 snd_hda_codec 262400 1 snd_hda_intel snd_seq_dummy 11908 0 snd_seq_oss 40448 0 snd_seq_midi_event 16128 1 snd_seq_oss snd_seq 60704 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 16404 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss 50944 0 snd_mixer_oss 24320 2 snd_pcm_oss snd_pcm 90888 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 30088 2 snd_seq,snd_pcm snd 67624 12 snd_hda_intel,snd_hda_codec,snd_seq_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 17505 2 snd snd_page_alloc 16912 2 snd_hda_intel,snd_pcm [cdome@andromeda ~]$ cat /proc/asound/card0/codec#0 Codec: Realtek ALC883 Address: 0 Vendor Id: 0x10ec0883 Subsystem Id: 0x1458e601 Revision Id: 0x100002 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x10 0x10] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x17 0x17] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x04] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01a19c40: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01813c41: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x9933014f: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x01cb7160: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

3 8

[alsa-devel] snd_pcm_hardware_t.info question
by Ciaccia 18 Apr '07

18 Apr '07

Hi all, I am trying to fix (improve?) a driver for an embedded ARM device I bought months ago, and for some reasons some ALSA applications work fine while other ones don't... Looking at the driver I got, I noticed that the info field in the snd_pcm_hardware_t struct just defines: SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE while the info in the tutorial "writing an alsa driver" defines: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID Furthermore, the driver defines a copy callback (actually 2, one for playback and one for capture), where it copies the data from the use space to the dma buffer. Since the dma buffer was preallocated with snd_pcm_lib_preallocate_pages_for_all and is accessible from the "outside", I wonder why the original author did this... I have some questions for you gurus: - what is the difference between SNDRV_PCM_INFO_MMAP and SNDRV_PCM_INFO_MMAP_VALID? What is SNDRV_PCM_INFO_BLOCK_TRANSFER? - The code calls snd_pcm_lib_preallocate_pages_for_all function as following: /* allocate the pcm(DMA) memory */ ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,0, 4*128*1024, 4*128*1024); is the DMA_TYPE right for an ARM device? Is NULL a correct value for data in this case? - In case I get rid of the copy callback, how do I specify the format of the stream? How does Where can I find an example for that? Thanks a lot Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 5

[alsa-devel] [PATCH] sound: convert "sound" subdirectory to UTF-8
by John Anthony Kazos Jr. 17 Apr '07

17 Apr '07

From: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> Convert the "sound" subdirectory to UTF-8. The following files are changed. sound/oss/{trident.c,es1371.c,pas2_pcm.c} sound/oss/emu10k1/{passthrough.h,passthrough.c} sound/pci/mixart/mixart.c Signed-off-by: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> --- I can't get my mail client to send in ISO-8859-1 explicitly instead of whatever it decides I want, so the actual patch is attached in octet-stream format. The below patch is just for reference and will very likely NOT work, unless this time it picks ISO-8859-1. Also, since the patch includes both encodings, whichever encoding you select while viewing it will display one correct and one garbled line. However, the bytecodes -are- correct. diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.c -- Emu10k1 digital passthrough - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.h -- Emu10k1 digital passthrough header file - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/es1371.c linux-2.6.21-rc7.mod/sound/oss/es1371.c --- linux-2.6.21-rc7.orig/sound/oss/es1371.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/es1371.c 2007-04-17 13:55:45.000000000 -0400 @@ -100,7 +100,7 @@ * Tjeerd Mulder <tjeerd.mulder(a)fujitsu-siemens.com> * 05.01.2001 0.29 Hopefully updates will not be required anymore when Creative bumps * the CT5880 revision. - * suggested by Stephan Müller <smueller(a)chronox.de> + * suggested by Stephan MÃ¼ller <smueller(a)chronox.de> * 31.01.2001 0.30 Register/Unregister gameport * Fix SETTRIGGER non OSS API conformity * 14.07.2001 0.31 Add list of laptops needing amplifier control diff -uprN linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c --- linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c 2007-04-17 13:55:45.000000000 -0400 @@ -85,7 +85,7 @@ static int pcm_set_speed(int arg) * come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/. * * I cleared bit 5 of these values, since that bit controls the master - * mute flag. (Olav Wölfelschneider) + * mute flag. (Olav WÃ¶lfelschneider) * */ #if !defined NO_AUTO_FILTER_SET diff -uprN linux-2.6.21-rc7.orig/sound/oss/trident.c linux-2.6.21-rc7.mod/sound/oss/trident.c --- linux-2.6.21-rc7.orig/sound/oss/trident.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/trident.c 2007-04-17 13:55:45.000000000 -0400 @@ -18,7 +18,7 @@ * Ollie Lho <ollie(a)sis.com.tw> SiS 7018 Audio Core Support * Ching-Ling Lee <cling-li(a)ali.com.tw> ALi 5451 Audio Core Support * Matt Wu <mattwu(a)acersoftech.com.cn> ALi 5451 Audio Core Support - * Peter Wächtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support + * Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support * Muli Ben-Yehuda <mulix(a)mulix.org> * * @@ -89,7 +89,7 @@ * use set_current_state, properly release resources on failure in * trident_probe, get rid of check_region * v0.14.9c - * August 10 2001 Peter Wächtler <pwaechtler(a)loewe-komp.de> + * August 10 2001 Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> * added support for Tvia (formerly Integraphics/IGST) CyberPro5050 * this chip is often found in settop boxes (combined video+audio) * v0.14.9b diff -uprN linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c --- linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c 2007-04-17 13:55:45.000000000 -0400 @@ -472,7 +472,7 @@ static int snd_mixart_prepare(struct snd struct snd_mixart *chip = snd_pcm_substream_chip(subs); struct mixart_stream *stream = subs->runtime->private_data; - /* TODO de façon non bloquante, réappliquer les hw_params (rate, bits, codec) */ + /* TODO de faÃ§on non bloquante, rÃ©appliquer les hw_params (rate, bits, codec) */ snd_printdd("snd_mixart_prepare\n");

1 0

[alsa-devel] Request for CS4280 Manual
by Amar singh 17 Apr '07

17 Apr '07

Hello Sir, I'm a newbie in Linux Deivce Drivers and Kernel Development. We're writing our own driver for sound card - Cirrus Logic, Sound Fusion CS4280. in Fedora Core 6 Linux flavour. For that I've been looking for the programming manual of the above said sound card, for a long time now, but didn't succeed in getting one. Since nothing being fruitful for me in this case, we're stuck up with our project, at the start. Therefore I'd kindly like to request you to plz, send me, if you can, the programming manual in the pdf/doc format or a link to the place from where we can get that document. We'd really appreciate your help for the matter. Looking ahead for positive reply from you. Thanks and Regards, Amar singh Emertxe Information Technologies (P) Ltd., Bangalore - 56. India.

3 3

[alsa-devel] [PATCH] ASoC Kconfig description.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch makes the ASoC Kconfig descriptions a little more meaningful. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

2 1

[alsa-devel] Why do I not see patches in the emails?
by Timur Tabi 17 Apr '07

17 Apr '07

Can someone tell me why I can't see that actual patches in the posts that include them? I'm guessing it's some odd configuration option, but I went to http://mailman.alsa-project.org/mailman/options/alsa-devel and I didn't see anything applicable. Here's an example. As you can see, there is a text/plain attachment where the patch would be, but it's empty. Sender: alsa-devel-bounces(a)alsa-project.org Errors-To: alsa-devel-bounces(a)alsa-project.org Return-Path: alsa-devel-bounces(a)alsa-project.org X-OriginalArrivalTime: 16 Apr 2007 13:52:51.0309 (UTC) FILETIME=[864861D0:01C7802E] --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain Content-Transfer-Encoding: 7bit This patch by Frank Madarino updates the AT91xxxx Makefile and Kconfig to build the renamed SSC files. Changes:- o Rename various i2s labels to ssc Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel --=-Uhr6lvaXHMfZKXRJXsBj-- -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] FW: Individual PCM stream control
by Nagalla, Hari 17 Apr '07

17 Apr '07

Thanks for the reply.. We followed your suggestions and we see the PCM stream controls from the user space after we have created the controls at the driver using IFACE_PCM tags.. But when I try to set the 'index' in 'snd_ctl_elem_id' to reflect the PCM substream number, it seems to always get reset to '0'. Can you please suggest, how to set the index. --- snapshot of the code -------- snd_ctl_elem_info_alloca(&info); snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_numid(id, ctl_num); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_PCM); fprintf(stdout,"setting the mute/unmute for stream index 0x%x \n", (unsigned int) stream_id); snd_ctl_elem_id_set_index(id,(unsigned int) stream_id); if (handle == NULL && (err = snd_ctl_open(&handle, card, 0)) < 0) { fprintf(stderr,"Control %s open error: %s\n", card, snd_strerror(err)); return err; } snd_ctl_elem_info_set_id(info, id); if ((err = snd_ctl_elem_info(handle, info)) < 0) { fprintf(stderr,"Cannot find the given element from control %s\n", card); return err; } snd_ctl_elem_info_get_id(info, id); /* FIXME: Remove it when hctl find works ok !!! */ type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); snd_ctl_elem_value_set_id(control, id); show_control_id(id); printf("\n"); fprintf(stdout, "Retrieving the stream index 0x%x \n",snd_ctl_elem_id_get_index(id)); ---------------------------------------------------------------- Thanks a bunch !! Hari > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, April 04, 2007 11:21 AM > To: Nagalla, Hari > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Individual PCM stream control > > At Fri, 30 Mar 2007 05:47:11 -0500, > Nagalla, Hari wrote: > > > > Thanks for the advice.. Couple of inline questions below.. > > > > > > > -----Original Message----- > > > > I am new to ALSA, so some help from experienced folks would help.. > > > > > > > > We are trying to develop ALSA support for our DSP based mixer. In > > our > > > > architecture DSP performs the mixing, sample rate conversion etc. My > > > > question is how, can does the user space applications exercise > > > > controlling individual PCM stream to the mixer? Ex: individual > > stream > > > > gain control, individual stream mute, etc.. Any user space example > > code > > > > would be useful too.. > > > > > > Most drivers create PCM volume controls with IFACE_PCM tag, and set up > > > the volume appropriately at each PCM open, or let alsa-lib configs to > > > set up the initial volumes. Usually, the control element index > > > corresponds to the PCM substream number. > > > > Which alsa library API lets us to set up the PCM stream volume? And can > > it be used dynamically when the stream is active? > > It is controlled via the normal control API. Open the snd_ctl_t > handle for the same card (e.g. "hw:0" or "default"), and change the > value of the element. However, there is no standard name for the > individual PCM volume controls, and currrently depending on the driver > implementation... > > > I could n't figure out 'index' of the control element. > > It's the index in snd_ctl_elem_id. You can access via > snd_ctl_elem_id_{get|set}_index() in alsa-lib. > > > Takashi

1 0

Re: [alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 17 Apr '07

17 Apr '07

I applied the first hint from Jaroslav (at the end of this post the new asound.conf), but the problem seems not gone: one time each 256 seconds I get a very strong cpu load spike, that is maybe related (my guess) to the usage of plugin (maybe periodic cleaning each 256 secs?). When there is no spike, the cpu load is 0.0% with occasional 0.1% any 60-90 secs. I ran the latency test code with plughw:0 (I can't use latency.c with my virtual cards because they can't link the in and out togheter) using the blocking mode, real time, no poll. Any 256 secs, spike! (a cpu load for latency.c of 45.00%, for roughly a sec, with the cumulated cpu time of the latency.c app reported from top -d 1.0 passing from: 0:00.01 to: 0:00.64 to: 0:01.26 etc, any 256 seconds. latency.c output: ./latency -P plughw:0 -C plughw:0 -f S16_LE -c 1 -r 8000 -m 320 -M 960 -s 600 -b Playback device is plughw:0 Capture device is plughw:0 Parameters are 8000Hz, S16_LE, 1 channels, blocking mode Wanted tick time: 0us, poll mode: no Loop limit is 4800000 frames, minimum latency = 320, maximum latency = 960 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 2 <- 0 3 <- 0 4 <- 0 5 <- 0 6 <- 0 7 <- 0 8 <- 0 9 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 10 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0*0.0833333 + 1*0.0833333 + 2*0.0833333 + 3*0.0833333 + 4*0.0833333 + 5*0.0833333 + 6*0.0833333 + 7*0.0833333 + 8*0.0833333 + 9*0.0833333 + 10*0.0833333 + 11*0.0833333 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 12 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Trying latency 320 frames, 40000.000us, 40.000000ms (25.0000Hz) Success Playback: *** frames = 4800320 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482713139 delay : 317 avail : 3 avail_max : 192 Capture: *** frames = 4800000 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482757926 delay : 1 avail : 1 avail_max : 189 Maximum read: 160 frames Maximum read latency: 20000.000us, 20.000000ms (50.0000Hz) Hardware sync Playback time = 1176471431.639255, Record time = 1176471431.639255, diff = 0 new asound.conf: pcm.c3 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 2 } } pcm.capt3 { type plug slave.pcm c3 } pcm.c4 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 3 } } pcm.capt4 { type plug slave.pcm c4 } pcm.c5 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 4 } } pcm.capt5 { type plug slave.pcm c5 } pcm.c6 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 5 } } pcm.capt6 { type plug slave.pcm c6 } pcm.c7 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 6 } } pcm.capt7 { type plug slave.pcm c7 } pcm.c8 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 7 } } pcm.capt8 { type plug slave.pcm c8 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 } On 4/13/07, Giovanni Maruzzelli <gmaruzz.lists(a)gmail.com> wrote: > Thanks a lot Jaroslav, I'll apply your suggestions and I'll report here > > On 4/13/07, Jaroslav Kysela <perex(a)suse.cz> wrote: > > On Fri, 13 Apr 2007, Giovanni Maruzzelli wrote: > > > > > Dear Alsa developers, > > > > > > I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. > > > > > > I want to use the /etc/asound.conf (.asoundrc) to split the card (that > > > has 8 in and 8 out - but I use just 6 in/out), with minimal latency. > > > > > > I mean, I want to record and play (duplex) from/to each virtual card > > > concurrently and from separate application instances. > > > > > > My app works with S16_LE, 8000, mono. > > > > > > Is the following asound.conf (.asoundrc) correct and efficient for this? > > > > Use dsnoop bindings rather than routing (ttable) for capture too, > > otherwise alsa-lib works with all channels internally. > > > > > I got spikes of cpu load (at 5-6%) on each virtual card any roughly > > > 256 seconds when using a(play|record) to record or play from/to any > > > > aplay|arecord is not very good test. Use latency test code - available in > > alsa-lib/test directory. > > > > Jaroslav > > > > ----- > > Jaroslav Kysela <perex(a)suse.cz> > > Linux Kernel Sound Maintainer > > ALSA Project, SUSE Labs > > >

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[alsa-devel] [PATCH 1/2] ASoC Samsung S3c24xx updates - audio DMA cleanup.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch cleans up the audio DMA for the Samsung S3C24xx platform. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH] [ALSA] Add first generation macbook subsystem ID
by bainonline＠gmail.com 17 Apr '07

17 Apr '07

From: Abhijit Bhopatkar <bainonline(a)gmail.com> First generation MacBooks were getting ignored by sigmatel drivers and wrongly being identified as MACMINI. This patch makes them identify as MACBOOK. Signed-off-by: Abhijit Bhopatkar <bainonline(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c94291b..cb99df5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1905,6 +1905,9 @@ static int patch_stac922x(struct hda_codec *codec) */ printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id); switch (codec->subsystem_id) { + case 0x106b0a00: /* MacBook First generatoin */ + spec->board_config = STAC_MACBOOK; + break; case 0x106b0200: /* MacBook Pro first generation */ spec->board_config = STAC_MACBOOK_PRO_V1; break; -- 1.4.4.2

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[alsa-devel] [1/2] 2.6.21-rc7: known regressions
by Adrian Bunk 17 Apr '07

17 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Status : unknown Subject : ali_pata: boot from CD fails References : http://lkml.org/lkml/2007/3/31/160 Submitter : Stephen Clark <Stephen.Clark(a)seclark.us> Status : unknown Subject : kernels fail to boot with drives on ATIIXP controller (ACPI/IRQ related) References : https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=229621 http://lkml.org/lkml/2007/3/4/257 Submitter : Michal Jaegermann <michal(a)ellpspace.math.ualberta.ca> Status : unknown Subject : boot failure: rtl8139: exception in interrupt routine References : http://lkml.org/lkml/2007/3/31/160 Submitter : Stephen Clark <Stephen.Clark(a)seclark.us> Status : unknown Subject : laptops with e1000: lockups References : https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=229603 Submitter : Dave Jones <davej(a)redhat.com> Handled-By : Jesse Brandeburg <jesse.brandeburg(a)intel.com> Status : problem is being debugged Subject : forcedeth: interface hangs under load References : http://lkml.org/lkml/2007/4/3/39 http://www.spinics.net/lists/netdev/msg28981.html Submitter : Ingo Molnar <mingo(a)elte.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Ayaz Abdulla <aabdulla(a)nvidia.com> David Miller <davem(a)davemloft.net> Status : problem is being debugged Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available

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[alsa-devel] [PATCH] ASoC export AC97 DAI
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch exports the ASoC AC97 Digital Audio Interface as a GPL'ed symbol. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 1/2] ASoC WM8753 codec support.
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch series adds support for the WM8753 codec as found on the OpenMoko Neo 1973 (other Neo 1973 and Samsung S3C24xx patches to follow today) as well other new devices. Features:- o HiFi and Voice DAI supported (inc runtime switching of DAI mode) o DAPM o All mixers o PLL calculator o 16,20 and 24bit samples. o WM8753 I2C ID added to include/linux/i2c-id.h Signed-off-by: Harald Welte <laforge(a)openmoko.org> Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Seth Forshee <seth.forshee(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH 2/2] ASoC Samsung S3C24xx updates - i2s
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch adds DMA size fields to the S3C24xx audio DMA params and exports the S3C24xx I2S digital audio interface. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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Re: [alsa-devel] [PATCH] Re: iec958 switch uneffective while playing ac3 stream
by Takashi Iwai 16 Apr '07

16 Apr '07

At Thu, 05 Apr 2007 10:47:52 +0200, Dominique Dumont wrote: > > Takashi Iwai <tiwai(a)suse.de> writes: > > > The patch is a bit dangerous because it unconditionally resets the > > DIGI_CONVERT verb for each widget. > > Agreed. We could also test the NID before switching on and off DigEn > in the DIGI_CONVERT. > > OTOH the switch would occur for *all* HDA chips which may not be what > you want. I'm not 100% sure that this particular procedure is necessary for all codecs. So, I prefer having it in the hda-codec part rather in hda-intel part. > > The below is my modified version. > > It doesn't turn off before the controller side reset as you did. So, > > if my version doesn't work, it implies that the SPDIF reset has to be > > done before the controller reset. Please give it a try. > > It works also ! I've tested AC3 and 44KHz. I cannot test 96KHz or > 196Khz (my amp is too old). Good to hear. > > + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ > > I had more the impression that the SPDIF stream is messed up if you > change the stream format while the digital converter is working... > > > > + if (codec->spdif_ctls & 1) > > Shouldn't you use AC_DIG1_ENABLE for a proper patch ? > > if (codec->spdif_ctls & AC_DIG1_ENABLE) > > > + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, > > + codec->spdif_ctls & 0xfe); > > likewise: codec->spdif_ctls & 0xff & ~AC_DIG1_ENABLE ? > > > + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); > > + /* turn on again (if needed) */ > > + if (codec->spdif_ctls & 1) > > if (codec->spdif_ctls & AC_DIG1_ENABLE) ? Yeah, that'd be better. I'll fix and commit to the HG tree later. thanks, Takashi

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[alsa-devel] [PATCH 1/3] ASoC AT91xxxx - SSC port DSP support
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch series by Frank Madarino updates the AT91xxxx core to add DSP/PCM audio hardware formats. Changes:- o Rename at19-i2s.c -> at91-ssc.c o Rename at91-i2s.h -> at91-ssc.h o Add DSP hardware formats. o Rename various I2S labels to SSC Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 2/2] ASoC WM8753 codec - build changes
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch adds the WM8753 codec driver to the kernel build system. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 3/3] ASoC AT91xxxx build
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch by Frank Madarino updates the AT91xxxx Makefile and Kconfig to build the renamed SSC files. Changes:- o Rename various i2s labels to ssc Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 2/3] ASoC AT91xxxx eti B1 machine SSC changes
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch by Frank Madarino updates the eti B1 machine to use the newer AT91xxxx SSC core with the DSP/PCM audio hardware changes. Changes:- o #include "at91-ssc.h" instead of "at91-i2s.h" o Rename various I2S labels to SSC Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH] ASoC DAPM switching for reentrant codec paths.
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch fixes an issue whereby power was applied to any inactive analog path that would leave and reenter a codec (e.g. ACOP -> ACIN on WM8753). This change now checks for such paths and DAPM will power them down when not in use. Signed-off-by: Seth Forshee <seth.forshee(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH] ASoC WM9712 kmemdup
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch creates the WM9712 codec register cache using kmemdup instead of doing a kzalloc followed by a memcpy. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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