using ALSA one call

using ALSA one call

[kallipygos]

Hi Developers.

My name is Alf  and i new in this list .
I physics student , and i write some sound programs under linux.
Big sorry for my english , in school i learn only deutsch.

So far i used old /dev/dsp open funntion also under ALSA.

/*--OSD--*/#define RATE 44100   /* the sampling rate */
/*--OSD--*/#define SIZE 16      /* sample size: 8 or 16 bits */
/*--OSD--*/#define CHANNELS 2   /* 1 = mono 2 = stereo */
/*--OSD--*/void open_sound_device_wr(void)
/*--OSD--*/{//  fdo = open("/dev/dsp", O_RDWR );
/*--OSD--*/  fdo = open("/dev/dsp", O_WRONLY );
/*--OSD--*/  if (fdo < 0) {       perror("open of /dev/dsp failed");    exit(1);  }
/*--OSD--*/
/*--OSD--*/  arg = SIZE;      //// sample size
/*--OSD--*/  status = ioctl(fdo, SOUND_PCM_WRITE_BITS, &arg);
/*--OSD--*/  if (status == -1)     perror("SOUND_PCM_WRITE_BITS ioctl failed");
/*--OSD--*/  if (arg != SIZE)      perror("unable to set sample size");
/*--OSD--*/
/*--OSD--*/  arg = CHANNELS;  //// mono or stereo
/*--OSD--*/  status = ioctl(fdo, SOUND_PCM_WRITE_CHANNELS, &arg);
/*--OSD--*/  if (status == -1)     perror("SOUND_PCM_WRITE_CHANNELS ioctl failed");
/*--OSD--*/  if (arg != CHANNELS)  perror("unable to set number of channels");
/*--OSD--*/
/*--OSD--*/  arg = RATE;      //// sampling rate
/*--OSD--*/  status = ioctl(fdo, SOUND_PCM_WRITE_RATE, &arg);
/*--OSD--*/  if (status == -1)    perror("SOUND_PCM_WRITE_WRITE ioctl failed");
/*--OSD--*/}
----

Now i wana try da same with ALSA ,
for example all parameters put in one struckture and then simple call 
one  universal alsa_open funktion.
Is it posssible with ALSA ? 
Is here copy-paste example/function for fast, comfortable and easy ALSA use ?

for example so :
====
#include "alsa/asoundlib.h"
...
alsa_parm_struckt.device    = "default";  // "hw:0,0"; "hw:1,0";
alsa_parm_struckt.play_capt = SND_PCM_STREAM_PLAYBACK;
alsa_parm_struckt.format    = SND_PCM_FORMAT_S16_LE;
alsa_parm_struckt.access    = SND_PCM_ACCESS_RW_INTERLEAVED;
//alsa_parm_struckt.samplerate = 48000;
alsa_parm_struckt.samplerate = 44100;
alsa_parm_struckt.channelz  =2;
alsa_parm_struckt.bytes_per_sample = 2;
alsa_parm_struckt.latency   = 0; // ??
alsa_parm_struckt.nonblock  = 1; // ??
... other parameters set ...
...
alsa_open_function( & handle , & alsa_parm_struckt [ , ...] );

  and after this write data blocks 
 frames_2_write = ...
 for(;;)
 {
  ...
  frames = snd_pcm_writei(handle, buffer, frames_2_write );
  ...
 }
====

OK , this example works 
http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm__min_8c-example.html
but only if  samplerate = 48000; 
if  samplerate = 44100 - then not :(
44k file play in 48k mode :D  it sounds funnee :D

I remixed this program so that it read data from wave file.

gcc  pcm_min01.c -lasound -O2  -o pcm_min01 
./pcm_min01  '/mnt/hda2/Booty Luv - Shine/Booty Luv - Shine (Ian Carey remix).wav'
i gotta this :
ALSA lib pcm.c:7160:(snd_pcm_set_params) Rate doesn't match (requested 44100Hz, get 0Hz)
Playback open error: Invalid argument

With aplay no problems, it is played in da correct speed.
--
 aplay '/mnt/hda2/Booty Luv - Shine/Booty Luv - Shine (Ian Carey remix).wav'
Playing WAVE '/mnt/hda2/Booty Luv - Shine/Booty Luv - Shine (Ian Carey remix).wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
--



Any ideas/examples/... welcome
Tnx in advance

Alf





====




----

Re: using ALSA one call

[Clemens Ladisch]

kallipygos@inbox.lv wrote:
> OK , this example works 
> http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm__min_8c-example.html
> but only if  samplerate = 48000; 
> if  samplerate = 44100 - then not :(
> 44k file play in 48k mode :D  it sounds funnee :D
> 
> I remixed this program so that it read data from wave file.
> 
> gcc  pcm_min01.c -lasound -O2  -o pcm_min01 
> ./pcm_min01  '/mnt/hda2/Booty Luv - Shine/Booty Luv - Shine (Ian Carey remix).wav'
> i gotta this :
> ALSA lib pcm.c:7160:(snd_pcm_set_params) Rate doesn't match (requested 44100Hz, get 0Hz)

Please show the source code of your program.


Best regards,
Clemens

Re: using ALSA one call

[Alfs Kurmis]

Moin Meister

Quoting Clemens Ladisch <clemens@ladisch.de>:
> > but only if  samplerate = 48000;
> > if  samplerate = 44100 - then not :(
> > 44k file play in 48k mode :D  it sounds funnee :D
> >
> > i gotta this :
> > ALSA lib pcm.c:7160:(snd_pcm_set_params) Rate doesn't match (requested
> 44100Hz, get 0Hz)
>
> Please show the source code of your program.

see below


I have on my notebook 2 sound cards
* built in
$ lspci
...
00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02)
...
and
second on USB wire  - Edirol by Roland ...
I have switch em to 44,1k

As given here
http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#g6aa164ed37308d66bcc079f5cd265a09
latency         required overall latency in us (0 = optimum latency for players)
i have try to set  optimum latency = 0;

snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED,
                                channelz, samplerate ,  0, 0)


./pcm_min01  '/mnt/hda2/Booty Luv - Shine/Booty Luv - Boogie 2nite (DJ Teddy-o remix).wav'
Now playing to USB device "hw:1,0"
gotta this :
Short write (expected 4096, wrote 360)
Short write (expected 4096, wrote 360)
Short write (expected 4096, wrote 2427)
wery fragmentary sound

with latency = 500000 us USB play is OK.

But why i can not play my wave on notebook sound device "hw:0,0" ??


Tnx in advance

Alf

/*
 *  This extra small demo sends a random samples to your speakers.
 */

#include "alsa/asoundlib.h"

#include <stdio.h>
#include <fcntl.h>     //
#include <unistd.h>    //
#include <sys/types.h> //

//static char *device = "default";                        /* playback device */
//static char *device = "hw:0,0";
static char *device = "hw:1,0";

snd_output_t *output = NULL;
//unsigned char buffer[16*1024];                          /* some random data */
/*unsigned*/ short buffer[/*4*1024*/8192];

int main( int argc, char *argv[] )
{
  int err;    
  unsigned int i;   int channelz, bytes_per_sample, samplerate;
  snd_pcm_t *handle;
  snd_pcm_sframes_t frames , frames_2_write;
         int einfd ;  int  rdstatus ;

//  samplerate = 48000;
  samplerate = 44100;
  channelz =2;
  bytes_per_sample = 2;
  frames_2_write = sizeof(buffer) / (channelz * bytes_per_sample);
  //for (i = 0; i < (4*1024) ; i++){  buffer[i] = random() & 0xffff; }
      einfd = open( argv[1] , O_RDONLY );  lseek(einfd, 48L ,SEEK_SET);

  if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
          printf("Playback open error: %s\n", snd_strerror(err));  exit(EXIT_FAILURE);
  }
  if ((err = snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED,
                                channelz, samplerate ,  0, 500000)) < 0) {   /* 500000  0.5sec */
          printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE);
  }

  for (i = 0; i < 128; i++) {
                rdstatus  =  read(einfd,&buffer[0],sizeof(buffer));
          frames = snd_pcm_writei(handle, buffer, frames_2_write );
          if (frames < 0)
                  frames = snd_pcm_recover(handle, frames, 1);
          if (frames < 0) {
                  printf("snd_pcm_writei failed: %s\n", snd_strerror(err));
                  /*break;*/  goto yuck_off;  }
          if (frames > 0 && frames < (long) frames_2_write)
                  printf("Short write (expected %li, wrote %li)\n", (long) frames_2_write, frames);
  }


yuck_off:
  close(einfd);
  snd_pcm_close(handle);
  return 0;
}









----

Re: using ALSA one call

[Clemens Ladisch]

Alfs Kurmis wrote:
> As given here
> http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m.html#g6aa164ed37308d66bcc079f5cd265a09
> latency         required overall latency in us (0 = optimum latency for players)
> i have try to set  optimum latency = 0;

That documentation is wrong; 0 is not allowed.  Just use 0.5 s, or some
larger value.

> But why i can not play my wave on notebook sound device "hw:0,0" ??

Because the "hw" device goes straight to the hardware and disallows
any conversion of sample rate/format.

Use "default", or something like "default:0" to select a specific card.

> snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED,
>                    channelz, samplerate ,  0, 500000)

... and setting soft_resample to 0 disallows resampling in any case.


HTH
Clemens