Instantly share code, notes, and snippets.

Embed
What would you like to do?
/*
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);
}
@heatblazer

This comment has been minimized.

heatblazer commented Dec 4, 2016

Thanks, a good starter guide. I am ref-ing to it right now.

@heatblazer

This comment has been minimized.

heatblazer commented Dec 5, 2016

Hello, I am expiriencing a metallic sound when I setup the recording system that way. Can you please point me a note I am making a misstake:
a link to my repo: https://github.com/heatblazer/recd2/blob/master/alsa-rec/alsarec.cpp

@hosseindehghanipour1998

This comment has been minimized.

hosseindehghanipour1998 commented May 15, 2017

Hi.thanks for the code but i copied it and pasted it into my IDE to test the code but it does not recognize the library. where can i download that too?

@a2t2

This comment has been minimized.

a2t2 commented Mar 5, 2018

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

@christiansalazar

This comment has been minimized.

christiansalazar commented Jul 28, 2018

@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).
@christiansalazar

This comment has been minimized.

christiansalazar commented Jul 28, 2018

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.

:)

Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment