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[alsa-devel] Noise Problem with rate convert plugins
by Andreas Rumpler 20 Jul '07

20 Jul '07

Hello, I'm new to this list.I hope someone could help me with a problem concerning the samplerate converter plugins of the alsa driver. Firstly some facts about my application. - Envy24 (VT1270) pci audio chip on a custom mainboard - playback of sounds/music with different sample rates from 8kHz to 48kHz (sample rate of the files) - the hardware (Envy24 chip) must always run with 44.1kHz or 48kHz (selectable by user), because the audio is outputed by a DAC and SPDIF and the SPDIF clock must not be changed according to the sample rate of the files. - Kernel is 2.6.22 and alsa is 1.0.14 So I'm using the samplerate (libsample rate based) plugin from the alsa-plugin package. Generaly it works very good. My problem is: Every time a playback is started a short plop noise is heard. The noise is on analog and SPDIF output. If I don't use the samplerate converter the noise is not heard at all. So it comes definitely from the rate converter plugin, I think. Changing the quality level of the plugin don't solve the problem. I've also switched to the new Speex rate converter plugin. With this plugin there is no noise at the start of playback. But here I have trouble with awful noise at some rate conversions (6kHz(source) -> 48kHz(output); 11.025kHz -> 44.1kHz; 22.05kHz -> 44.1kHz). So it's even worse than using the libsamplerate plugin. Finaly I've tried the libavcodec plugin, which is the worst according to noise. Could any comment this problem, please? This is my asound.conf: ############################################## # Playback over Line-Out and SPDIF simultaniously as default pcm.!default { type plug slave.pcm { type multi slaves.a.pcm "plughw:0,0" slaves.a.channels 2 slaves.b.pcm "hw:0,1" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 } pcm_slave.s1 { pcm default rate 44100 } pcm_slave.s2 { pcm default rate 48000 } pcm.rate_convert_44100 { type rate slave s1 converter samplerate } pcm.rate_convert_48000 { type rate slave s2 converter samplerate } ##################################################### Regards Andreas Rumpler

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[alsa-devel] The users view of an audio card
by David Webb 20 Jul '07

20 Jul '07

I've been editing a couple of the unofficial wiki pages and a number of issues have arisen. My own system has the NVidia/Realtek motherboard chips so I may be in the dark about what is available on other chips. Some of the questions are background to ensure that what I write later in the wiki is correct. Takashi Iwai suggested I raise the issues here: 1. A large number of users seem to be reporting problems with Nvidia and Via soundcards. As far as I can tell Nvidia and Via do not produce soundcards but they do produce chipsets with southbridges which can connect to a motherboard audio chip. The when alsamixer (or Kmix) runs it reports something like: Card: NVidia CK804 Chip: Realtek AL650 rev 0 Presumably because the southbridge is the primary PCI device. So: 1.1 Is the above always correct (so I can add it to the wiki)? 1.2 Could alsamixer etc be modified to give a more sensible answer for non-experts. Something like: Card: Motherboard via NVidia CK804 Chip: Realtek AL650 rev 0 might work. I presume a look-up table is used to convert the PCI 2-byte integers to the above names so presumably the extra information needed could be obtained at this level. 2. On different motherboards a Realtek 650 may be connected using a NVidia CK804 with the intel8x0 alsa driver or using a VIA VT8233 with the vt82xx driver. 2.1 Given that the audio chip registers are the same in both cases why are two different drivers needed? (Are they?) 2.2 If it is something to do with the southbridges, where may the manuals that describe the differences be found? 2.3 What would happen if the motherboard manufacturers kept the same southbridges but started using an audio chip with a different set of registers? (2.2a Where can an AD'97 manual be found? - I could not find one on the intel site) 3. The Realtek 650 datasheet implies that independently of the mixing in the main input -> line-out pathway, the input to the ADC (i.e. on capture) may include more than one source (i.e. both audio and microphone). However alsamixer (and kmix) only allows me one input stream. So: 3.1 Is the manual misleading or have I misread it? 3.2 Is the manual correct but the driver is based on an earlier intel chip which did not have this facility? Related to this the manual seems to show a number of options which I cannot identify in alsamixer. For example the manual shows two surroundsound volume stages between which there are switched links, one to and two from the main stero path. Alsamixer shows a single amplifier. It has one link 'from' the main path but I could not find a link 'to' the main path. 3.3 Is the driver assuming a old layout which fits most of the new audio chip layout but not all? (This is really a question about what sort of warnings/advice should my write up include). 3.4 At which stage does alsa relate the registers to the descriptions, aux-volume, aux-mute etc? Again does it assume a layout which fits most aspects of the chips listed? Thats enough for the moment. I presume I could sit down for a year and read and understand the code - but in the meantime any help will be welcome. Regards, David Webb.

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[alsa-devel] eliminating sequencer and midi from my alsa module dependencies
by John L. Utz III 20 Jul '07

20 Jul '07

Hello; What arguments should give to the current version of alsa to eliminate the aforementioned things? here is my current conf string from hgcompile, am i missing something? args="--with-debug=full --with-isapnp=no --with-sequencer=no --with-oss=no --with-cards=via82xx" tnx! johnu

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[alsa-devel] Problem and workaround for Intel HDA
by Yann Le Brech 19 Jul '07

19 Jul '07

The problem : missing controls of the sound cards for an Intel HDA, Toshiba Satellite Pro U200 Full description : I was using the kernel 2.6.19 for a while, and I got some interesting functions like controlling the sound level on headphones or speakers separately. Then I upgraded to the kernel 2.6.21.1 and suddently the level controls changed, to useless level controls, like Master and PCM only (and some other useless). I wanted to get back to the previous control set, and I've found a solution by checking what was done in debugging mode. The previous kernel was not detecting the model config, but the new one did. So I simply strapped the detection routine by a return -1 and it did the trick. I know its not the best thing to do, but it was the simplest thing I thought about. Outputs : 2.6.19's Alsa output : ALSA sound/pci/hda/hda_intel.c:695: codec_mask = 0x3 2.6.21.1 : ALSA sound/pci/hda/hda_intel.c:695: codec_mask = 0x3 ALSA sound/pci/hda/hda_codec.c:1757: hda_codec: model 'toshiba' is selected for config 1179:1 (Toshiba U205) How to see the problem : Just see the list of controls in any mixer software. Environment : Toshiba Satellite Pro U200 Software : KateOS 3.2 with Linux 2.6.19 (working) and 2.6.21.1 (not working) Workaround : int __devinit snd_hda_check_board_config(struct hda_codec *codec, int num_configs, const char **models, const struct snd_pci_quirk *tbl) { return -1; // WORKAROUND if (codec->bus->modelname && models) { int i; for (i = 0; i < num_configs; i++) { if (models[i] && -- a++ Y.

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[alsa-devel] [PATCH] Add quirk for Asus P5LD2
by Claudio Matsuoka 19 Jul '07

19 Jul '07

Another simple patch that adds a system to the ALC882 quirk table. Tested with a week-old HG version, buildsystem of the current alsa-driver seems to be broken.

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[alsa-devel] a new synthesizer driver
by Steve Longerbeam 19 Jul '07

19 Jul '07

Hi all, I'm embarking on a big task: writing a new synthesizer driver. My device has all the ingredients for a MIDI wavetable sythesizer: 64 voices, each voice with envelope, pitch, and LFO controls. Also global effects like reverb. It's similar to the emu10k1. So my first question would be, is the synth code under synth/emux/ general enough to use for a different synthesizer chip? Ie, can I implement my own operators in 'struct snd_emux_operators' and pass that to snd_emux_register() ? Would it be a simple matter to make synth/emux/ a generic synth layer, by moving all the source in synth/emux/ up one level to synth/, and then rename all functions and data structures starting with "snd_emux" to "snd_synth" ? For instance, snd_emux_operators becomes snd_synth_operators, snd_emux_control() becomes snd_synth_control(), etc. It seems this path is already partially completed, since synth/emux/soundfont.c is a generic soundfont loader. The thing I want to avoid doing is creating a synth/my_synth/ directory and duplicating lots of code from synth/emux/ (including just copying soundfont.c verbatim!). Steve

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[alsa-devel] Re-uploaded patch for bug #2708
by Nicola Fagnani 19 Jul '07

19 Jul '07

I re-uploaded the patch with the requested changes some time ago, and I was wondering if there is still something wrong with it. Thanks Nicola Fagnani

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[alsa-devel] underrun observed when aplay is used with mmap
by Pharaoh . 19 Jul '07

19 Jul '07

Hi , Finally I have got my driver working and now testing it using aplay. I am able to play a song without getting any underruns when I use read/write mode but when I use mmapped mode I get underrun warnings. After this if I use read/write mode again then I observe 1 warning of underrun. Any idea what could be wrong? I know why underruns occur etc. Does it have to do with mmapped mode? I dont think so... Here is the log. # aplay -Dplughw:0,0 --rate=44100 --format=s16 --channels=2 --mmap 44.wav Playing WAVE '44.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 0.384 ms long) underrun!!! (at least 0.118 ms long) underrun!!! (at least 0.124 ms long) underrun!!! (at least 0.117 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.128 ms long) underrun!!! (at least 0.116 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.125 ms long) ....... and it goes on till the song finishes. And after this I play without using mmap and log says.. # aplay -Dplughw:0,0 --rate=44100 --format=s16 --channels=2 44.wav Playing WAVE '44.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 516.424 ms long) and then the song plays without any underrun. -pharaoh.

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[alsa-devel] cs46xx thinkpad resume fix - #bug1946
by Pavel Kysilka 19 Jul '07

19 Jul '07

Hi, attached patch fix resume on thinkpad notebooks with cs46xx chip. Tested with Thinkpad A21m. Patch is opposite latest linus git 2.6.22. Bug #1946 may be closed. Signed-off-by: Pavel Kysilka <pavelk(a)bsys.cz> Pavel Kysilka ________ Information from NOD32 ________ This message was checked by NOD32 Antivirus System for Linux Mail Servers. http://www.eset.com

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[alsa-devel] PS3 - Surround sound on SPDIF set-up
by gt bradley 19 Jul '07

19 Jul '07

Hi, I've been beating my had against this for several weeks now. (started with fedora 6, tried adding in the plugings etc (version 1.0.12) -- also tried upgrading to the current 1.0.14 version, without success (couldn't produce surround-sound on teh spdif output). I have now switched to fedorea 7 and was able to have rythmbox play surround sound via spdif, but have since lost that ability. (but I'm convinced the hardware and low-level drivers support it). In order to limit the number of moving parts, I have reduced to just trying to get speaker-test to play on all 6 speakers. from pirut & I have alsa-lib 1.0.14-0.4.rc3.fc7.ppc(64) and alsa-utils 1.0.14-0.7.rc2.fc7.ppc The default install has no asound.conf file. (attempts to create a "better" one didn't improve the situation). I do not see a file matching the ps3 device in etc/alsa/cards nor etc/alsa/pcm, but I have not found any documentation on adding entries to the above directories. [root@ps3 pcm0p]# cat /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: SPDIF name: SPDIF subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 [root@ps3 ~]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PS3 [PS3], device 0: SPDIF [SPDIF] Subdevices: 1/1 Subdevice #0: subdevice #0 [root@ps3 ~]# aplay -L null Discard all samples (playback) or generate zero samples (capture) [root@ps3 ~]# amixer [root@ps3 ~]# amixer scontrols [root@ps3 ~]# speaker-test -c 6 speaker-test 1.0.14rc2 Playback device is default Stream parameters are 48000Hz, S16_BE, 6 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 4096 to 8192 Period size range from 256 to 256 Using max buffer size 8192 Periods = 4 was set period_size = 256 was set buffer_size = 8192 0 - Front Left 4 - Center 1 - Front Right 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 17.821324 * above test only produces sound on channels 0 & 1. xmms produces no sound and emits the following: ** WARNING **: alsa_setup_mixer(): Failed to find mixer element: PCM Is it possible to get some help/pointers in the right direction? thanks GT* *

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