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[alsa-devel] Tascam US122 on Arm
by ST 04 Oct '07

04 Oct '07

Hi I am a little irritated that there has been no replies to my questions to this list (http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003387.…, http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003083.…). Am i on the wrong list? Please tell me if so. Probably its also that i didn't make clear in my last post that the tascam driver works on arm with my US122 sound card, for ca. 51 seconds. After that i get the following error: Sequence Error!(hcd_frame=6 ep=8in;wait=65538,frame=2). Most propably some urb of usb-frame 65538 is still missing. Cause could be too long delays in usb-hcd interrupt handling. I also tried insmod ./2.6.18/kernel/sound/usb/usx2y/snd-usb-usx2y.ko nrpacks=X with X=[1,2,4,8]. I thought that if it is really an CPU performance problem than probably a bigger buffer would eliminate the problem. However the error persisted. Even commenting out /* #define USX2Y_NRPACKS_VARIABLE y */ and setting #define USX2Y_NRPACKS 1 didn't help. So my question is now, is the 250Mhz ARM really to slow or is there any chance that there is an error in either the usb stack or the driver exposed only on the ARM architecture? Things speaking for the latter are: *Error allways in the 65536-65540 value. *Error happens most of the times after about 50secs. Since i do not know much about USB any hint is apreciated. ST

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[alsa-devel] about sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?)
by Akio 04 Oct '07

04 Oct '07

Dear ALL: after long time amr-wb codec loopback test. (above 90 hours) I got the following kernel message. it comes from snd_pcm_lib_read1() function. ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) What should I do? My platform is ARM926, with Kernel 2.6.19.2 ALSA version is 1.0.13. Best Regards, Akio

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[alsa-devel] different user causes 'no such device'
by william estrada 03 Oct '07

03 Oct '07

Hi guys, I have what seams like a interesting problem. I have written an ALSA app to record wav files and it runs while I am logged in an terminal as the logged in Gnome user or root. But when I su to another user, it can't find the default sound card. I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did not work. Anyone know of this problem and a fix for it. Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum > MSGrec - Version 1.1, Mt-Umunhum-Wireless.net > WavMSG: Paused: enter 'q' to stop, <ENTER> to start. > ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device > ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device > ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device > ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device > ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default

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[alsa-devel] Snd problem ... ( ACCESS_TYPE )
by errordeveloper＠gmail.com 03 Oct '07

03 Oct '07

Hi list. I' m using alsa-lib v1.0.14rc2 on gentoo-linux (v2.6.20-gentoo-r8) my soundcard is RME Hammerfall HDSP9632 (snd-hdsp driver) i have been using it for a while - all fine i only run a few applications for my sound experiments. one of those is Snd (from CCRMA), it was okay until i have done a few updates lately - kernel from 2.6.17 to 2.6.20 and alsa-lib as well. i'm not sure about either of these but i check Snd src-tree and the mailing list - nothing was changed .. the error msg that i get is: :(play) ;status of hw:0 hw_params status of hw:0 ACCESS: MMAP_NONINTERLEAVED RW_NONINTERLEAVED FORMAT: S32_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: [128 384] CHANNELS: [4 12] RATE: [32000 192000] PERIOD_TIME: (333 256000] PERIOD_SIZE: [64 8192] PERIOD_BYTES: [2560 393216] PERIODS: 2 BUFFER_TIME: (666 512000] BUFFER_SIZE: [128 16384] BUFFER_BYTES: [2560 786432] TICK_TIME: 1000 sw_params status of hw:0 start_mode: DATA xrun_mode: STOP tstamp_mode: NONE period_step: 0 sleep_min: 0 avail_min: 0 xfer_align: 0 silence_threshold: 0 silence_size: 0 boundary: 0 ;Invalid argument: hw:0: access type RW_INTERLEAVED not available can't play : Invalid argument: hw:0: access type RW_INTERLEAVED not available #f : i had tried to modify audio.c in Snd's tree to use both ACCESS types given above , but it didn't help - showed device is busy error instead. i'm only studying C, so can't be sure .. anyway here is a problem which i could not solve yet and haven't seen anyone helpful around #alsa, neither i could google anything out. for a ref.: Cmdist mailing list Cmdist(a)ccrma.stanford.edu http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist and our discussion is here: http://ccrma-mail.stanford.edu/pipermail/cmdist/2007-July/004221.html could anyone have a look at this at somepoint, please . -- thank you, Ilya. D errordeveloper|*@*|gmail|*dot*|com

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[alsa-devel] ASOC: Best way to get private data from machine driver to the other drivers?
by Timur Tabi 02 Oct '07

02 Oct '07

I'm writing an ASOC driver. When my machine driver loads, it determines some hardware information for the DMA controllers, some other hardware information for the I2S controller, and even more information for the codec. I want to be able to pass all this information to each of the three other drivers. Unfortunately, it doesn't appear that I have enough "private_data" pointers available to me. The current ASOC drivers all use statically defined global structures and indexes into arrays of these structures to identify individual devices, such associating I2S controller #2 with DMA channels 3 and 4. On PowerPC, this doesn't work. The actual hardware configuration is determined at runtime (see arch/powerpc/boot/dts/), and so the machine driver needs to collect all the information and make all the device-to-device associations. For example, the PCM driver has an open function: static struct snd_pcm_ops fsl_dma_ops = { .open = fsl_dma_open, ... static int fsl_dma_open(struct snd_pcm_substream *substream) { ... What I would like is for fsl_dma_open() to be able to obtain pointers to the DMA controller's registers (there are two controllers with 4 channels each) by looking inside 'substream'. How can I do that? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] some problem about alsa lib usage on arm926
by Akio 02 Oct '07

02 Oct '07

Dear All: right now, I'm using alsa lib interface to work with voip application. Our hardware platform is an ARM926 266MHz box with Linux kernel 2.6.19.2. We got some problems. would you like to do us favors? 1. what is the different between "plughw:0,0", "defalut", "hw:0.0"? after reading the document in http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html about "PCM naming conventions", we still not very sure the relationship between those device conventions. is there any more detail documents talk about on this? right now, we are using alsa lib 1.0.13 and the device is "plughw:0,0". 2. Is there any documents detail discuss the setting about sample rate, buffer size, period size. because, I have two different voice codec to handle, one is narrow band(8k Hz), the other is wide band(16K Hz). right now, I using the following functions and setting to make alsa work. snd_pcm_hw_params_set_rate_near sets sample rate, 8k or 16k. snd_pcm_hw_params_set_period_size_near sets period size, current is 128. snd_pcm_hw_params_set_buffer_size_near sets buffer size, 8k is 2048, 16k is 4096. the unit seems be samples not byte. 3. the code was based on the following URL. http://www.suse.de/~mana/alsa090_howto.html http://www.equalarea.com/paul/alsa-audio.html open capture and playback at one function, and provide one read and one write interface to up layer. here comes three problems: a. the channel opened is just only one, but If I set the snd_pcm_hw_params_set_access as SND_PCM_ACCESS_RW_NONINTERLEAVED. using snd_pcm_readn and snd_pcm_writen the code will not work. what is the wrong? but if set snd_pcm_hw_params_set_access as SND_PCM_ACCESS_RW_INTERLEAVED and using snd_pcm_readi and snd_pcm_writei. the code work very well. one channel also have data interleave problem? I don't understand why? b. in snd_pcm_open the block flag is set as 0. what is the mode of the operation? blocking or non-blocking? c. call the xrun_recovery() timing. currently the xrun_recovery only do on EPIPE. the following is the handling core. err = snd_pcm_prepare(handle); if(err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) { nanosleep(&retryTime, &tmpTime); LOG("audio device resume fail, %s\n", snd_strerror(err)); } if(err < 0) { err = snd_pcm_prepare(handle); if(err < 0) LOG("audio data IO fail, %s\n", snd_strerror(err)); } } else { if(err < 0) LOG("audio device prepare fail, %s\n", snd_strerror(err)); } Does the code have any missing point to handle? If any missing, what should I to add? 4. BTW, I call xrun_recovery both in capture and playback direction. After long time test. the test procedure is as the following steps: 1. capture voice data as PCM data. 2. encoded as amr-wb bit stream. 3. decoded as PCM data from encoded bit stream. 4. playback PCM data. after 16 hours continue tests, there are some delay, about 800 mili seconds. is there any way to reduce the delay after long time test. we want make the delay about < 400 mili seconds after one week continue test. BTW, is there any way to prevent xrun_recovery calling? is there any index to monitor ALSA device buffer information? 5. If we want using salsa lib to replace the usage of alsa lib, what do we have to take care? how about the latency and performance between salsa and alsa? how about the stability of salsa? There are so many problems after many tests. If you found any documents we need to study more detail. please tell us. Thanks in advance. Thank you. Best Regards, Akio

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[alsa-devel] issue with ADI A1986A
by l1983＠bankerinter.net 01 Oct '07

01 Oct '07

Hello, I need some help from you guys. I,m a new linux user and totally unfamiliar with all this coding... I have an issue with my motherboard "ASSUS M2N-MX" and it's onboard sound chip :"ADI AD1986A" witch supports 6 channel CODEC , Jack-Sensing & Enumeration technology , and with S/PDIF OUT interface. Can you please somehow, (or somebody you know) , make for me some installable .deb archive that will help me to set up my sound, since i can't do it on my own. :( Or help me some how else, (either sending me some address's or by giving me some other advice. HELP ME PLEASE!!!!!! l1983(a)bankerinter.net

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[alsa-devel] short writes?
by william estrada 30 Sep '07

30 Sep '07

Hi group, I have written my own ALSA recorder and player. Along with that I have several bugs. The bug I am working on right now is 'short writes' when I use the pipe command into my player. My player runs and asks for the file to play, so that I may play incoming files as they are sent in. I have a program that uses 'inotify' to watch for incoming file and then pipe the file name into my player. Now with all of that, is there a API call that will put the writei into blocking mode? Or an API call that will flush the output from writei? Or any other suggestions? Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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[alsa-devel] nforce ac97 plays too fast
by Stas Sergeev 30 Sep '07

30 Sep '07

Hello. Trying to get a software amplification to work with NForce CK804 AC'97, I've found out that whatever the asound.conf I write, the sound starts playing much faster than it should. After some debugging, I've found out the following: - The card (or the driver) only supports 48000KHz. - The program (any, because libao and artsd suffer from that) is trying to set some rate with snd_pcm_hw_params_set_rate_near(), and that succeeds, but the returned rate is 48000, while the requested is, say, 22050. No wonder it then plays with the wrong speed. Of course the program can check the returned rate, but it sets the "direction" parameter to 0, which should mean "take exactly that value". I've looked into an alsa-tutorial, and the examples there do not check the returned rate too - dir==0 should be enough. Looking into the sources, it seems the culprit is the snd_interval_refine_min/max(), which do not care about the (i->min > min && !openmin) case at all. The question is: is setting dir=0 (or even NULL, since it is a pointer) enough to set the exact rate? And if so - how snd_interval_refine_min() should handle the (i->min > min && !openmin) case? Or if not - what does the dir==0 mean and why it is not adjusted to the proper value on return, when the rate is changed? (i.e. the rate parameter changes from 22050 to 48000, but the dir parameter remains 0, doesn't change) With the default configuration the rate is proper, but I find it annoying that altering asound.conf in pretty much any way gets the wrong results.

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[alsa-devel] Asus M2NPV-VM snd_hda_intel problem - also with Alsa 1.0.15rc3
by Ma Begaj 29 Sep '07

29 Sep '07

Hi, for the sweet love of ... does anyone have an idea how to make this sound chip working... I have Asus M2NPV-VM with AD1986A sound chip. I tried "model=3stack position_fix=1", but it does not help. surround does not work and sound mutes whenever I change volume. when I use model=ultra or eapd-laptop, stereo is at stable and I can change volume. but surround I tried it with vanilla kernel 2.6.21.5/26.22.6 and snd_hda_intel from the kernel without any success. I even compiled Alsa 1.0.15rc3, but situation did no change. anybody ... with any idea ... I would appreciate any help. any help at all. thanks. M.

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