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[alsa-devel] Question on snd_soc_dai_link
by Timur Tabi 29 May '07

29 May '07

I have a question about these DAI structs for ASoC. Let's take eti_b1_wm8731.c as an example. It contains this structure: static struct snd_soc_dai_link eti_b1_dai = { .name = "WM8731", .stream_name = "WM8731", .cpu_dai = &at91_i2s_dai[1], .codec_dai = &wm8731_dai, .init = eti_b1_wm8731_init, .ops = &eti_b1_ops, }; Notice that the .cpu_dai field is hard-coded to use the 2nd I2S device on the AT91. The problem I'm having with this approach is that everything is hard-coded. On PowerPC, this approach doesn't work well, because we have a "device tree" that dictates what devices are present on the SOC. For a Freescale SOC that has two I2S devices (for example), both devices would be listed in the device tree, with the base address of the memory-mapped registers. This means that a Freescale I2S driver would have its 'probe' function called (much like PCI does) each time the kernel processes an I2S node in the device tree. This means that when the machine driver (which has the snd_soc_dai_link structure) loads, there's no guarantee that the I2S driver has also loaded and the probe functions have been called. It gets even more complicated if only one I2S device has a codec connected to it, because then I need to kmalloc the snd_soc_cpu_dai structures and somehow tell the snd_soc_dai_link structure which one to use. Has any thought been given to making these structures more dynamic and handling interdependencies among the drivers? As it stands now, it looks like I'm going to have to hard-code everything in the drivers and break the device tree paradigm. -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [PATCH] Sigmatel STAC9221 - Add Configs for all Apple Intel Macs
by Ivan N. Zlatev 29 May '07

29 May '07

This patch: * adds the pinconfigs for all 5 Apple boards and 14 Subsystem IDs (support for possibly all iMac, Mac, MacMini etc etc) * adds "intel-mac-v1" to v5 models which replace the current * reflects changes in Alsa-Configuration.txt Please review. Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] [PATCH] Disable debugging output for the ALS300 driver
by Ash Willis 29 May '07

29 May '07

> ----- Original Message ----- > From: "Takashi Iwai" <tiwai(a)suse.de> > To: "Ash Willis" <ashwillis(a)programmer.net> > Subject: Re: [alsa-devel] [PATCH] Disable debugging output for the ALS300 driver > Date: Tue, 29 May 2007 15:08:30 +0200 > > > At Sat, 26 May 2007 13:26:35 +0000, > Ash Willis wrote: > > > > Disables debugging output in the ALS300 driver. > > Also contains a whitespace cleanup and a fix for a potential bug. > > > > Signed-off-by: Ash Willis <ashwillis(a)programmer.net> > > Thanks for the patch. It seems that your MUA breaks the embedded > patch, so I had to apply it manually. Could you check whether the > latest HG is OK for you? I'm away from home right now. I'll test it as soon as I have a chance but it looks fine. > > Also I changed "if ((err = ...))" to two individual lines, which is a > newer standard kernel coding style. > Yeah, I left that for now. I'll go through the tree and tidy up remaining occurences of this style when I have some time. It seems that when people add drivers to the tree, they generally copy already existing drivers. It doesn't make much sense to be copying code that doesn't fit CodingStyle. Ash = Sports Memorabilia License Plates Signs All license plates $6 all signs $10 all frames $8 - same day shipping - fun company and selection. http://a8-asy.a8ww.net/a8-ads/adftrclick?redirectid=a44f49e42c5a283d05f0c96…

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[alsa-devel] [PATCH] Disable debugging output for the ALS300 driver
by Ash Willis 29 May '07

29 May '07

Disables debugging output in the ALS300 driver. Also contains a whitespace cleanup and a fix for a potential bug. Signed-off-by: Ash Willis <ashwillis(a)programmer.net> ---------------------------------------------------- diff -r 337962e19de6 -r 8ea4fa014563 pci/als300.c --- a/pci/als300.c Fri May 25 12:18:49 2007 +0200 +++ b/pci/als300.c Sat May 26 14:04:14 2007 +0000 @@ -88,8 +88,8 @@ #define PLAYBACK_BLOCK_COUNTER 0x9A #define RECORD_BLOCK_COUNTER 0x9B -#define DEBUG_CALLS 1 -#define DEBUG_PLAY_REC 1 +#define DEBUG_CALLS 0 +#define DEBUG_PLAY_REC 0 #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) @@ -730,10 +730,9 @@ static int __devinit snd_als300_create(s } chip->irq = pci->irq; - snd_als300_init(chip); - if (snd_als300_ac97(chip) < 0) { + if ((err = snd_als300_ac97(chip)) < 0) { snd_printk(KERN_WARNING "Could not create ac97\n"); snd_als300_free(chip); return err;

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[alsa-devel] sound on smdk2410
by Jose Henrique Spahn Torres 29 May '07

29 May '07

Hi, I am trying recent Zoltan Devai's patchs for kernel 2.6.22-rc2 on a smdk2410 board. I reached the point where the kernel recognized the uda1341 sound chip, however, when I try to play something, I get the output below. Can anyone, please, show me the path to follow to get sound working? Thanks, Henrique /home $ l drwxr-xr-x 3 root root 0 Jan 1 00:00 . drwxr-xr-x 18 root root 1024 May 18 2007 .. -r--r--r-- 1 root root 3419671 Jan 1 1970 jj.mp3 /home $ mpg123 jj.mp3 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 anasoc: UDA1341 <-> s3c24xx-i2s info: asoc: rate mask 0xfe asoc: min ch 2 max ch 2 asoc: min rate 8000 max rate 48000 version 0.65; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes asoc: machine hw_params failed ... (lots of asoc: messages) ... asoc: machine hw_params failed pop wq checking: Playback status: inactive waiting: yes pop wq D3 UDA1341 Playback Playing MPEG stream 1 of 1: jj.mp3 ... Title: Times Like These Artist: Jack Johnson Comment: Album: On And On Year: 2003 Genre: Acoustic MPEG 1.0 layer III, 192 kbits/s, 44100 Hz stereo [audio.c:264] error: No supported rate found! /home $

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[alsa-devel] HDA Codec: Add support for MSI K9N Ultra
by Tobin Davis 29 May '07

29 May '07

This patch adds the MSI K9N Ultra system to the realtek patch. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] ICE1724 alsa driver
by Rossco 29 May '07

29 May '07

Hi, I hope someone can help me. (tried the user list but no response) I have a hitpoint 7.1ch Envy24ht-s based card with the usual vt1616 multichannel codec and the wolfson WM8728. The card is recognised as a chaintech av710 with the only difference being that the card has spdif in as well as out. It appears to be physically similar to the maddog entertainer (would be good to know if this is also recognised as an AV710) By default 2 channel stereo in played through the "front" channel of the card through the vt1616 codec and I appear to be able to set spdif out through the mixer (but dont have another device to test at the moment). I have been able to change default output to go through the wolfson dac through an asound.state file posted on the web. I have to admit I do not quite understand what the file has done as it only appears to have removed some of the controls rather than change or add anything. Only two devices are detected for the card the vt1616 and IEC958, neither are shown as having any subdevices, therefore I am unsure how the asound.state file was able to address the wolfson chip which requires i2s input rather ac-link. This seems to indicate i2s and ac-link modes of the envy24ht-s chip are active. Most AV710 users seem happy enough to be able to route to toslink out and/or the higher quality wolfson chip, therefore there is little discussion of addressing the other channels of the card. I really want to be able to address each of the channels of the card so that I can use the convolution programme Brutefir with the card to create a multiway crossover for my loudspeakers. Is there anyway I can define each of the cards outputs so that I can select to send audio to any one of them? I am also keen to get the spdif in working and particularly the syncing to spdif in of the clock that the ice1724 chip supports. Can these features be added to the driver? Please advise of any required info, I have all the datasheets and can trace chip pins also. Thanks in advance. Ross

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[alsa-devel] [PATCH] Add quirk for Supermicro PDSBA to alc883_cfg_tbl[]
by Daniel T. Chen 29 May '07

29 May '07

Attached is a diff against current alsa-kernel hg tip. Thanks, Daniel Chen

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[alsa-devel] hardware not asking for more data using asyn call back
by Ashlesha Shintre 29 May '07

29 May '07

Hi, In my asynchronous playback program, the hardware stops asking for any data and the program gets stuck after one call to the callback function. I have pasted the program below -- please let me know if something is wrong - Thanks, Ashlesha. /* 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 > > short int *buffer; > int ind=10, fd1; > 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,length,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); > > buflen = 4*SIZE; > buffer = (short int *) malloc (buflen*sizeof(short int)); > > wto = NULL; > rfrom = &buffer[0]; > > count=0; > a=1; > while(count<buflen && a>0) > a=read(fd1,&buffer[count++],sizeof(short int)); > if(a<0) > printf("error in reading from wav file\n"); > > 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; > } > else > { > printf("rfm at the end, rfrom -buffer[buflen-1] > =%d\n",(*rfrom - buffer[buflen-1])); > wto = rfrom; > rfrom = &buffer[0]; > } > for(k=0; k<SIZE; k++) > { > a=read(fd1,(wto+k),sizeof(short int)); > if(a<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,buffer); > 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; > sleep(1); > } > > err = snd_pcm_close(handle); > if(err<0) > printf("error in closing pcm device: %s\n",snd_strerror(err)); > close(fd1); > 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; > //short int *bufferin = > snd_async_handler_get_callback_private(pcm_callback); > > while((avail=snd_pcm_avail_update(pcm_handle)) >= period_size) > { > printf("available frames = %d\n",avail); > frames = snd_pcm_writei(pcm_handle, rfrom, SIZE); > 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"); > // break; > } > } > > if(frames >0 && frames < SIZE) > printf("expected to write %d, wrote %d\n",SIZE,frames); > count = 0; > a=1; > while(count<SIZE && a>0){ > a=read(fd1,(wto+count),sizeof(short int)); //write to > the buffer from file > ++count; > } > printf("count = %d\n",count); > if(a <0){ //EOF > wto = NULL; > printf("end of file reached\n"); > break; > } > else{ > if(rfrom < &buffer[buflen-1]) > { > wto = wto+count; > rfrom = rfrom + SIZE; > } > else > { > wto = wto + count; > rfrom = &buffer[0]; > } //location of where to start > writing from next > } > ++ind; > } > > printf("going out of callback, ind = %d\n",ind); > } > > >

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[alsa-devel] async playback works despite error in snd_pcm_start.
by Ashlesha Shintre 28 May '07

28 May '07

Hi, I managed to write code to playback sound asynchronously -- however, this code works (i.e data is read from .wav file and output to the playback device) despite the snd_pcm_start giving a "file descriptor in bad state" error.. can someone explain this please? Thanks, Ashlesha.

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