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[alsa-devel] ASoC v2 on PowerPC
by Jon Smirl 05 Oct '07

05 Oct '07

I'm looking into write a ASoC v2 driver for the Efika. Efika uses a Freescale MPC5200B and a STAC9766. This is a practice driver before writing one for our target hardware (which doesn't work yet, same CPU but different codec). I've brought the changes from wolfson/asoc-v2-dev into my powerpc tree and got a couple of config errors with ARCH=powerpc. sound/soc/pxa/Kconfig:23:warning: 'select' used by config symbol 'SND_PXA2XX_SOC_SSP' refers to undefined symbol 'PXA_SSP' sound/soc/au1x/Kconfig:35:warning: 'select' used by config symbol 'SND_SOC_EXM1200_SSI' refers to undefined symbol 'SND_SOC_CS4251X' Suggestions on the best way to start? Target hardware also has a WM8804 on it. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Why was the built-in software volume control removed?
by kyle cronan 04 Oct '07

04 Oct '07

I think I may have posted this to the wrong list. There wasn't any response yet on alsa-user, so I'm reposting here. Summary: I'm seeing a lot of glitches ever since the built-in software volume was removed. What was the reason for this change? Softvol doesn't work nearly as well for me. ________________ Hi everyone, I recently updated my media center which is running Fedora devel. It pulled down a new kernel with updated ALSA drivers for my HDA audio chip (Intel 82801H + ALC883). I use IEC958 output to a pair of studio monitors (good sound with minimal fuss). The first thing I noticed about the update was that my software volume control was no longer saving state. It was resetting to something around 80% every time an app opened the audio device, a level which, in my setup, is quite loud! Then I saw that in alsamixer I had no master volume control, not even a PCM control! So I started experimented with my settings, and I found that after I selected "PCM" under "Default Mixer Tracks" in gnome-sound-properties I had a PCM volume control again (still no master), and apps using alsa output would once again remember the system volume. Don't know what that gnome setting actually did, but I've not been able to get it back to the original state, even when I ctrl-click to deselect "PCM" in the list. However, apps that use legacy OSS output still blast me with nearly full volume every time. I've also noticed another glitch: using the volume control applet to adjust my PCM volume, the audio sometimes skips. If I quickly drag the slider back and forth, it always does. I'm pretty sure I would have noticed this if it occurred with my previous kernel (haven't had a chance to test that yet). Anyway, looking into this problem, I found the following note in the changes document at http://www.alsa-project.org/main/index.php/Changes_v1.0.15rc2_v1.0.15rc3 Sound Core ... - Remove the built-in software volume control in favour of softvol. So, my (main) question is: is there a good reason that the built-in software volume was removed? I didn't find anything about it in the archives for this list or -devel. I'm really not in favor of this change, because It used to work flawlessly for me, whereas now I'm running into all these strange problems with my IEC958 output. Thanks, Kyle Cronan <kyle(a)pbx.org>

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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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