alsa-record-example.c

/* 
  A Minimal Capture Program

  This program opens an audio interface for capture, configures it for
  stereo, 16 bit, 44.1kHz, interleaved conventional read/write
  access. Then its reads a chunk of random data from it, and exits. It
  isn't meant to be a real program.

  From on Paul David's tutorial : http://equalarea.com/paul/alsa-audio.html

  Fixes rate and buffer problems

  sudo apt-get install libasound2-dev
  gcc -o alsa-record-example -lasound alsa-record-example.c && ./alsa-record-example hw:0
*/

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
	      
main (int argc, char *argv[])
{
  int i;
  int err;
  char *buffer;
  int buffer_frames = 128;
  unsigned int rate = 44100;
  snd_pcm_t *capture_handle;
  snd_pcm_hw_params_t *hw_params;
	snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;

  if ((err = snd_pcm_open (&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 0)) < 0) {
    fprintf (stderr, "cannot open audio device %s (%s)\n", 
             argv[1],
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "audio interface opened\n");
		   
  if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
    fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params allocated\n");
				 
  if ((err = snd_pcm_hw_params_any (capture_handle, hw_params)) < 0) {
    fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params initialized\n");
	
  if ((err = snd_pcm_hw_params_set_access (capture_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
    fprintf (stderr, "cannot set access type (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params access setted\n");
	
  if ((err = snd_pcm_hw_params_set_format (capture_handle, hw_params, format)) < 0) {
    fprintf (stderr, "cannot set sample format (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params format setted\n");
	
  if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, hw_params, &rate, 0)) < 0) {
    fprintf (stderr, "cannot set sample rate (%s)\n",
             snd_strerror (err));
    exit (1);
  }
	
  fprintf(stdout, "hw_params rate setted\n");

  if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, 2)) < 0) {
    fprintf (stderr, "cannot set channel count (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params channels setted\n");
	
  if ((err = snd_pcm_hw_params (capture_handle, hw_params)) < 0) {
    fprintf (stderr, "cannot set parameters (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "hw_params setted\n");
	
  snd_pcm_hw_params_free (hw_params);

  fprintf(stdout, "hw_params freed\n");
	
  if ((err = snd_pcm_prepare (capture_handle)) < 0) {
    fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
             snd_strerror (err));
    exit (1);
  }

  fprintf(stdout, "audio interface prepared\n");

  buffer = malloc(128 * snd_pcm_format_width(format) / 8 * 2);

  fprintf(stdout, "buffer allocated\n");

  for (i = 0; i < 10; ++i) {
    if ((err = snd_pcm_readi (capture_handle, buffer, buffer_frames)) != buffer_frames) {
      fprintf (stderr, "read from audio interface failed (%s)\n",
               err, snd_strerror (err));
      exit (1);
    }
    fprintf(stdout, "read %d done\n", i);
  }

  free(buffer);

  fprintf(stdout, "buffer freed\n");
	
  snd_pcm_close (capture_handle);
  fprintf(stdout, "audio interface closed\n");

  exit (0);
}

In the following line:

buffer = malloc(128 * snd_pcm_format_width(format) / 8 * 2);

1. How did you decide the number 128 ?
2. Why did you choose a buffer size twice the length (you multiple by 2) compared to number of frames asked for ?

Thanks

@a2t2,

• "128": the frames count, declared at top of the program.
• "2" : one for each stereo channel.
• "snd_pcm_format_width(format)": bit width of format, in bits count.
• " / 8": to get bytes instead of bits (the result of snd_pcm_format_width).

I read the doc: about "snd_pcm_readi", it say it reads interleaved frames when non-blocking mode is set.

so im wondering:

• in which place do you set the non-blocking mode ?
• if this is only a audio channel, why it is "interleaved" ? In my poor understanding i can understand we need a interleaved chunk when we mix them with video frames, in other words we interleave a video chunk and a audio chunk togheter in single row, so i cant understand how you read a interleaved audio frame.

:)

The following line (in the loop) could lead to crash due to missing %d :)

fprintf (stderr, "read from audio interface failed (%s)\n",
                        err, snd_strerror (err));

@christiansalazar https://alsa-project.org/wiki/PCM_Ring_Buffer

Ring buffer organization
Interleaved access
Interleaved format is like this (C means channel sample):

C0 C1 C2 C3 C0 C1 C2 C3 ....
Non-Interleaved access
Non-Interleaved format is like this (C means channel sample):

C0 C0 C0 C0 ................ C1 C1 C1 C1 ............. C2 C2 C2 C2 ............... C3 C3 C3 C3 ...........
It's like separate mono buffers. Note that offsets of buffers might not be continuous (there might be a gap between channel areas).