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[alsa-devel] ASoC and a codec that can't be controlled
by Timur Tabi 31 May '07

31 May '07

I'm working on some ASoC drivers for a new board, and we're using a CS4270. The CS4270 is interesting in that if it is connected in stand-alone mode, there is no way to configure it. The board wirings determine all the parameters. Therefore, my CS4270 codec driver will probably be very skimpy. My question is: does this mean that my cs4270.c file will *never* call these functions: snd_ctl_add snd_soc_cnew snd_soc_dapm_new_control snd_soc_dapm_connect_input snd_soc_dapm_new_widgets These are the functions used to add new controls and widgets. So how does ALSA know that it needs to call my I2S driver *instead* of my codec driver to do stuff like change volume? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] Will the real alsa wiki please stand up?
by j t 31 May '07

31 May '07

There something at: http://bugtrack.alsa-project.org/main/index.php/Main_Page which thinks it's the alsa wiki, and there something at: http://alsa.opensrc.org/Main_Page which alsa thinks it's the alsa wiki. Well, devs, which is the imposter? And would it be a good idea to reduce the number of distinct alsa wikis to something smaller, perhaps "1"? Jaime :-) PS: apologies to "To Tell the Truth".

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[alsa-devel] Native Instruments - Audio Kontrol 1 - snd_usb_caiaq module
by Alex Pintilie 31 May '07

31 May '07

Hallo! I just registered to this mailing list because a developer kindly advised me to do so. I own the external sound card Audio Kontrol 1 from Native Instruments and I was happy to find that drivers for it are already in the making. I downloaded alsa-driver-1.0.14rc4 and compiled successfully with ./hgcompile, then I installed the module with: # install -m 644 snd-usb-caiaq.ko /lib/modules/`uname -r`/kernel/drivers/snd-usb-caiaq.ko # /sbin/depmod -a After the command: # modprobe snd-usb-caiaq I get the following error: FATAL: Error inserting snd_usb_caiaq (/lib/modules/2.6.20-15-lowlatency/kernel/drivers/snd-usb-caiaq.ko): Unknown symbol in module, or unknown parameter (see dmesg) dmesg shows the following: [10095.330000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_new [10095.330000] snd_usb_caiaq: Unknown symbol snd_pcm_new [10095.330000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_limit_hw_rates [10095.330000] snd_usb_caiaq: Unknown symbol snd_pcm_limit_hw_rates [10095.331000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_preallocate_pages_for_all [10095.331000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_preallocate_pages_for_all [10095.331000] snd_usb_caiaq: Unknown symbol snd_hidden_kfree [10095.331000] snd_usb_caiaq: Unknown symbol snd_verbose_printk [10095.331000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_malloc_pages [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_malloc_pages [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_ioctl [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_ioctl [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_lib_free_pages [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_lib_free_pages [10095.332000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_set_ops [10095.332000] snd_usb_caiaq: Unknown symbol snd_pcm_set_ops [10095.333000] snd_usb_caiaq: Unknown symbol snd_hidden_kmalloc [10095.333000] snd_usb_caiaq: disagrees about version of symbol snd_pcm_period_elapsed [10095.333000] snd_usb_caiaq: Unknown symbol snd_pcm_period_elapsed This may not look like much but it's the effort from a couple of days searching, reading and testing. I hope you can help me. -- Alex

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[alsa-devel] Hardware-accelerated OpenAL support with ALSA?
by Chris Rankin 31 May '07

31 May '07

Hi, I have been looking at getting OpenAL working on my machine, since OpenAL is the only way that games under Windows Vista can now provide hardware-accelerated sound effects. This suggests that a lot of PC games could be adding OpenAL support soon, which means that it would be handy if Linux had an efficient native OpenAL implementation that Wine could wrap. Fedora Core does provide an OpenAL library from www.openal.org, but it looks as if this library's ALSA backend does not provide any hardware acceleration. (I have an "Audigy 2 ZS" sound card, which can at least support hardware mixing.) Various alternative ALSA-based solutions do seem to exist on the Web, e.g. http://dino.e4a.it/openal-alsa/ and http://www.lost.org.uk/openal.html However, it seems to me that OpenAL support would be better served as an actual ALSA package that distros would be able to ship. Are there any plans to provide official OpenAL support in ALSA, please? Thanks for any information, Chris P.S. My Audigy 2 also seems to have EAX support. Does the ALSA driver support that functionality too? I can't say that "World of Warcraft" seems to use it, although Wine's DirectSound3D support still has lots of gaps. _________________________________________________________ Inbox cluttering up with junk? Clean up with Yahoo! Mail. http://uk.docs.yahoo.com/mail/isp_targeting.html

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[alsa-devel] Userspace Device Driver
by Klaus Zimmermann 31 May '07

31 May '07

Greetings List, on various places on the alsa site I read about the possibility of writing a userspace device driver. I am interested thatin, since I have a nice Edirol FA-66 ieee1394 interface which works nicely under jack with the ffado (http://ffado.org , former freebob) driver. >From what I read it seems I could either write an external plugin to the alsa-lib, which would be truly userspace and fine except I it would always have to be configured from .asoundrc and would probably not be found by legacy programs like ekiga. Or I could get really messy and have a userspace daemon relay audio data between the ffado library, i.e. librawieee1394, and a yet to be written alsa-driver module. This would at least yield the advantage, that the driver was indistinguishable from a true hardware driver. So: Is it possible to write a userspace driver that is indistinguishable from a hardware driver? TIA Klaus Zimmermann P.S.: I am aware of the jackplug plugin. However even with very conservative buffer settings I get underruns and x-runs quickly leading to a failing jack :(

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Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support forSMDK2440boards
by Jose Henrique Spahn Torres 30 May '07

30 May '07

-----Original Message----- From: Zoltan Devai [mailto:zdevai@gmail.com] Sent: sexta-feira, 25 de Maio de 2007 14:40 To: Jose Henrique Spahn Torres Subject: Re: [alsa-devel] [RFC] [PATCH 0/2] ASoC sound support forSMDK2440boards 2007/5/25, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > > 2007/5/24, Jose Henrique Spahn Torres <jose.torres(a)eldorado.org.br>: > usbcore: registered new interface driver hci_usb Advanced Linux Sound > Architecture Driver Version 1.0.14rc4 (Wed May 16 > 09:45:46 2007 UTC). > ASoC version 0.13.1 > SMDK2440 ALSA SoC Audio driver > UDA1341 SoC Audio Codec > L3 write > L3 transfer > L3 sendbyte: 16 > L3 sendbyte: 40 > asoc: UDA1341 <-> s3c24xx-i2s mapping ok ALSA device list: > #0: SMDK2440 (UDA1341) > TCP cubic registered > ... Fine. > However, when triyng to play something I get this: > > asoc: machine hw_params failed > > This is in sound/soc/soc-core.c, line 494. > I couldn't go any futher then this. > Any hint? This combination of processor, crystal and codec only supports the following rates: 11.025 Khz, 22.050Khz and 44.100 Khz Please try to play a file with these sample rates. (If you have alsa-tools, you could try 'speaker-test -r 44.100'). BTW, could you tell me the freq of the crystal on the processor board ? (Is it 16.934 Mhz?) Zoltan Hi, I used speaker-test as indicated above and the result is: pendrive $ ./speaker-test -r 44100 speaker-test 1.0.11 Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Using 16 octaves of pink noise ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default Playback open error: -2,No such file or directory I have also tried: /pendrive $ ./aplay -f cd 01-jack_johnson-times_like_these-rns.wav ALSA lib pcm.c:2146:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:547: audio open error: No such file or directory /pendrive $ so I don't know what is happening here. Does anyone have any idea? Thanks, Henrique

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[alsa-devel] Fw: hardware not asking for more data using asyn call back
by stan 30 May '07

30 May '07

Begin forwarded message: Date: Tue, 29 May 2007 15:19:12 -0400 From: "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> To: stanl(a)cox.net, alsa-devel(a)alsa-project.org Subject: Re: hardware not asking for more data using asyn call back Hi, Thanks Stan, for your response - Instead of copying pcm data from a wav file, i decided to copy all the data to a master buffer first and then see if the circular buffer implementation works - so now instead of copying from the wave file, i m copying data from the master buffer to the circular buffer.. however, as per your suggestion, if i copy the data in the while loop in main, then, it might not always be copied between 2 consecutive callbacks, but maybe more, as the callbacks are asynchronous. however, the hardware still does not ask for more data after executing the callback function about twice -- is there a way to flush the hardware buffer before beginning playback? I have pasted my code below Regards, Ashlesha. <snip> I've been doing some thinking about what you are trying to accomplish and have a suggestion. I think you would be better off letting jack handle the back end/sound portion. Run a daemon/service listening on an arbitrary tcp/ip port. On startup it initializes alsa and jack. When it receives a request, it forks a process which opens a new channel on jack and creates a buffer for it. It then plays the sound it receives on that link via jack, silence when the buffer is empty. You would have to write client software for the user on the network to contact your daemon/service. You might even find something like this already out there.

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[alsa-devel] How the dma interrupt handlers have been registered in ALSA?
by Pharaoh . 30 May '07

30 May '07

Hi I am referring omap-alsa-dma.c and the related codec files. We usually register the IRQ handler by calling request_irq() function but in this mentioned files the IRQ handler etc is defined but it is not registered anywhere i.e. request_irq is not used anywhere. Is there any other way of registering irq handlers? The irq handlers address is passed to omap_request_dma, but there also it is not registered. I am missing the entire thing?

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[alsa-devel] [1/4] 2.6.22-rc3: known regressions
by Michal Piotrowski 30 May '07

30 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc3. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Unclassified Subject : long freezes on thinkpad t60 References : http://lkml.org/lkml/2007/5/24/100 Submitter : Miklos Szeredi <miklos(a)szeredi.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Status : problem is being debugged ACPI Subject : unable to shutdown on kernel 2.6.22-rc2 References : http://bugzilla.kernel.org/show_bug.cgi?id=8516 Submitter : Thierry Volpiatto <tvolpiatt(a)neuf.fr> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Status : Unknown File systems Subject : Oops in dentry_iput with 2.6.22-rc2 on AMD64 References : http://lkml.org/lkml/2007/5/22/4 Submitter : Florin Iucha <florin(a)iucha.net> Status : Unknown Kbuild Subject : make M=$PWD modules_install does nothing References : http://lkml.org/lkml/2007/5/27/190 Submitter : Andrey Borzenkov <arvidjaar(a)mail.ru> Status : Unknown Regards, Michal -- "Najbardziej brakowało mi twojego milczenia." -- Andrzej Sapkowski "Coś więcej"

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Re: [alsa-devel] hardware not asking for more data using asyn call back
by stan 30 May '07

30 May '07

Somehow I replied to the wrong message and it went to a personal email address. Putting it on the list so searches can find it (for what that is worth :-) ) Begin forwarded message: Date: Tue, 29 May 2007 18:18:22 -0700 From: stan <stanl(a)cox.net> To: "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> Subject: Re: hardware not asking for more data using asyn call back On Tue, 29 May 2007 15:19:12 -0400 "Ashlesha Shintre" <ashlesha.shintre(a)gmail.com> wrote: <snip> I didn't understand what you were asking before. It sounds to me like you want to have a thread running that accepts input from arbitrary processes and puts it in the main buffer. Then when the alsa callback occurs you want to take any new sound samples and send them to alsa. If that is the case, it makes more sense for the main buffer to be the circular buffer and the alsa buffer to be just a regular buffer. That's just personal preference though. As far as the handle error, I don't know what it means beyond what it says. You would have to look in the sources for the alsa library to see what causes that error to be generated. I don't see anything wrong in the logic of the code below, but the output indicates that you are not keeping up with the play rate. This is astonishing. At each play you can see that the available room in the buffer grows larger until larger than 1000, the buffer you have allocated. I'm not sure what infinite loop means and how your program is recovering from it. I presume that is where the time is being eaten up that isn't allowing you to keep up with the requests for data. I don't think you need to check if there is room when you get a callback. You should be able to just write the data to alsa using the writei call. i.e. eliminate the snd_pcm_avail_update call. I'm not familiar enough with alsa to help you further. Perhaps someone else can give you a hint or point you to the right place to look. > > /* Reading wav file into a buffer and then trying to play it */ > > > > #include <stdio.h> > > #include </usr/include/alsa/asoundlib.h> > > #include </usr/include/alsa/pcm.h> > > #include <sys/types.h> > > #include <unistd.h> > > #include <fcntl.h> > > > > #define SIZE 128 > > //mpos is the position in the master buffer to write from > > short int *buffer, *mbuf, *mpos; > > int ind=10, fd1, length; > > static snd_pcm_t *handle; > > static char *device = "default"; /* playback > > device */ > > snd_output_t *output = NULL; > > static short int *wto, *rfrom, buflen; > > > > void MyCallback(snd_async_handler_t *pcm_callback); > > > > int main() > > { > > int i,j,n,count,a,err,k=0; > > char c; > > snd_pcm_sframes_t frames; > > snd_async_handler_t *pcm_callback; > > > > > > > > if((fd1=open("SA2.WAV",O_RDONLY,0))==-1) > > printf("error opening wav file\n"); > > > > count=0; > > while(count++<40) > > a=read(fd1,&c,sizeof(char)); > > > > a=read(fd1,&length,sizeof(int)); > > > > n=length/SIZE; > > > > printf("length = %d\n",length); > > mbuf = (short int *) malloc (length * sizeof(short int)); > > mpos = &mbuf[0]; > > > > buflen = 4*SIZE; > > buffer = (short int *) malloc (buflen*sizeof(short int)); > > > > wto = NULL; > > rfrom = &buffer[0]; > > > > /* reading to master buffer */ > > count=0; > > a=1; > > while(count<length && a>0) > > a=read(fd1,&mbuf[count++],sizeof(short int)); > > if(a<0){ > > printf("error in reading from wav file\n"); > > exit(1); > > } > > close (fd1); > > > > for(count =0; count < buflen; count++) > > { > > buffer[count]=mbuf[count]; > > } > > /******************************/ > > > > /*ALSA DEV INIT*/ > > if((err = snd_pcm_open(&handle, device, > > SND_PCM_STREAM_PLAYBACK, 0)) < 0) { > > printf("error opening device: %s\n",snd_strerror(err)); > > return -1; > > } > > > > if((err = snd_pcm_set_params(handle, > > SND_PCM_FORMAT_S16_LE, > > SND_PCM_ACCESS_RW_INTERLEAVED, > > 1, > > 16000, > > 1, > > 1000)) < 0) { /* 2 sec */ > > printf("Playback open error: %s\n", snd_strerror(err)); > > return -2; > > } > > /*****************************/ > > > > printf("n=%d\n",n); > > for (i=0; i<50; i++) > > { > > printf("i=%d\n",i); > > frames = snd_pcm_writei(handle, rfrom, SIZE); > > if(frames < 0) // underrun > > { > > printf("underrun recovery\n"); > > frames = snd_pcm_recover(handle, frames,0); > > if(frames < 0){ > > printf("error in recovery\n"); > > return -3; > > } > > } > > if(frames >0 && frames < SIZE) > > printf("expected to write %d, wrote %d\n",SIZE,frames); > > > > printf("distance between ptrs before reinit is %d\n",rfrom > > - wto); if(rfrom < &buffer[buflen-1]) > > { > > wto = rfrom; > > rfrom = rfrom + SIZE; > > } > > if(rfrom >= &buffer[buflen-1]) > > { > > printf("rfm at the end, rfrom -buffer[buflen-1] > > =%d\n",(*rfrom - buffer[buflen-1])); > > //wto = rfrom; > > rfrom = &buffer[0]; > > } > > > > if((a = writetobuf())<0) > > printf("error in reading from wav file k = %d\n"); > > > > printf("buffer[i*SIZE] - rfrom = %d\n",(buffer[(i+1)*SIZE] - > > *rfrom)); > > > > } > > > > /* Async Handler */ > > > > > > err = > > snd_async_add_pcm_handler(&pcm_callback,handle,MyCallback,NULL); > > if(err<0) printf("add pcm handler error = > > %d\n%s\n",err,snd_strerror(err)); > > > > err = snd_pcm_start(handle); > > if(err<0) > > printf("error in starting snd pcm start err > > :%s\n",snd_strerror(err)); > > /******************************/ > > > > while (1) { > > > > if(wto == NULL) > > break; // redundancy -- already checking > > for this condition in MyCallback > > printf("infinite loop\n"); > > sleep(1); > > } > > > > err = snd_pcm_close(handle); > > if(err<0) > > printf("error in closing pcm device: > > %s\n",snd_strerror(err)); > > > > return 0; > > } > > > > void MyCallback(snd_async_handler_t *pcm_callback) > > { > > > > snd_pcm_t *pcm_handle = snd_async_handler_get_pcm(pcm_callback); > > snd_pcm_sframes_t avail; > > int count,a; > > snd_pcm_uframes_t period_size = 64; > > snd_pcm_sframes_t frames; > > > > if((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) ^ From the alsa lib api docs. | | Using of this function is useless for the standard read/write operations. Use it only for mmap access. See to snd_pcm_delay. Obtain delay for a running PCM handle. Parameters: pcm PCM handle delayp Returned delay in frames Returns: 0 on success otherwise a negative error code Delay is distance between current application frame position and sound frame position. It's positive and less than buffer size in normal situation, negative on playback underrun and greater than buffer size on capture overrun. > > { > > printf("available frames = %d\n",avail); > > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); Using rfrom is going to be problematic if you get to a > > if(frames < 0) // underrun > > { > > printf("underrun recovery\n"); > > frames = snd_pcm_prepare(pcm_handle); > > printf("error from snd_pcm_prepare is: > > %d\n",frames); if(frames < 0){ > > printf("error in recovery\n"); > > > > } > > } > > > > if(frames >0 && frames < SIZE) > > printf("expected to write %d, wrote > > %d\n",SIZE,frames); > > > > > > if(rfrom < &buffer[buflen-1]) > > { > > wto = rfrom; > > rfrom = rfrom + SIZE; > > } > > if(rfrom >= &buffer[buflen-1]) > > { > > // wto = rfrom; > > rfrom = &buffer[0]; > > } //location of where to start > > writing from next > > > > if(writetobuf()<0){ > > wto=NULL; > > exit(0); // returns 0 to the operating system -- so > > it ends the program! > > } > > ++ind; > > } > > > > printf("going out of callback, ind = %d\n",ind); > > } > > > > /*Function to copy data from master buffer (mbuf) to circular buffer > > (buffer)*/ > > int writetobuf() > > { > > int count=-1; > > > > while((++count < SIZE) && (mpos < &mbuf[length])) > > *(mpos++) = *(wto+count); > > > > if(mpos >= &mbuf[length-1]) > > return -1; > > else > > return 0; > > } > > i=49 error in starting snd pcm start err :File descriptor in bad state infinite loop inside callback, available frames = 364 going out of callback, ind = 1 infinite loop inside callback, available frames = 577 going out of callback, ind = 2 infinite loop inside callback, available frames = 790 underrun recovery error from snd_pcm_prepare is: 0 going out of callback, ind = 3 infinite loop infinite loop infinite loop

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