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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Wrong "From" email address selected earlier. Oops.

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

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[alsa-devel] ICE 1724 MIDI support still broken?
by Daniel James 24 Apr '07

24 Apr '07

Hello ALSA team, We have a user with an ESI Juli@ whose Linux box hangs when any MIDI application starts up. I read that MIDI support is broken generally on ICE 1724 cards; is this still the case in recent driver releases? If so, is there any interest in fixing this long-standing issue? The user has indicated that his box would be available by ssh for any tests required. Cheers! Daniel

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[alsa-devel] Standard config and plugin paths
by Takashi Iwai 24 Apr '07

24 Apr '07

Hi, while working on openSUSE 10.3 ALSA packaging, I had a discussion regarding the config and plugin paths. Basically, the current configs (typically in /usr/share/alsa or /etc/alsa) and binary plugins (/usr/lib/alsa-lib) are fairly strictly bound to the specific library, libasound.so.2. That is, if we publish a newer, incompatible version, libasound.so.3, these files will be very unlikely unusable, too. The suggestion was that we should keep these config and plugin files in the version-dependent paths so that no conflictions would occur at update. For example, instead of /usr/share/alsa, /usr/share/alsa2 or whatever. The fix would be easy -- just add some options to configure script of alsa-lib and alsa-plugins. So, the question is what path names are most suitable as the standard. For example, I find the following OK: - $DATADIR/alsa-1.0/ - $LIBDIR/alsa-lib-1.0/ Or, the deeper paths: - $DATADIR/alsa/1.0/ - $LIBDIR/alsa-lib/1.0/ For keeping the backward compatibility, we may implement fallback to read the old default path in addition, if the new path doesn't exist. In that case, the former suggestion would be cleaner. Any comments or suggestions? Takashi

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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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