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Simple sound playback using ALSA API and libasound
/*
* Simple sound playback using ALSA API and libasound.
*
* Compile:
* $ cc -o play sound_playback.c -lasound
*
* Usage:
* $ ./play <sample_rate> <channels> <seconds> < <file>
*
* Examples:
* $ ./play 44100 2 5 < /dev/urandom
* $ ./play 22050 1 8 < /path/to/file.wav
*
* Copyright (C) 2009 Alessandro Ghedini <alessandro@ghedini.me>
* --------------------------------------------------------------
* "THE BEER-WARE LICENSE" (Revision 42):
* Alessandro Ghedini wrote this file. As long as you retain this
* notice you can do whatever you want with this stuff. If we
* meet some day, and you think this stuff is worth it, you can
* buy me a beer in return.
* --------------------------------------------------------------
*/
#include <alsa/asoundlib.h>
#include <stdio.h>
#define PCM_DEVICE "default"
int main(int argc, char **argv) {
unsigned int pcm, tmp, dir;
int rate, channels, seconds;
snd_pcm_t *pcm_handle;
snd_pcm_hw_params_t *params;
snd_pcm_uframes_t frames;
char *buff;
int buff_size, loops;
if (argc < 4) {
printf("Usage: %s <sample_rate> <channels> <seconds>\n",
argv[0]);
return -1;
}
rate = atoi(argv[1]);
channels = atoi(argv[2]);
seconds = atoi(argv[3]);
/* Open the PCM device in playback mode */
if (pcm = snd_pcm_open(&pcm_handle, PCM_DEVICE,
SND_PCM_STREAM_PLAYBACK, 0) < 0)
printf("ERROR: Can't open \"%s\" PCM device. %s\n",
PCM_DEVICE, snd_strerror(pcm));
/* Allocate parameters object and fill it with default values*/
snd_pcm_hw_params_alloca(&params);
snd_pcm_hw_params_any(pcm_handle, params);
/* Set parameters */
if (pcm = snd_pcm_hw_params_set_access(pcm_handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED) < 0)
printf("ERROR: Can't set interleaved mode. %s\n", snd_strerror(pcm));
if (pcm = snd_pcm_hw_params_set_format(pcm_handle, params,
SND_PCM_FORMAT_S16_LE) < 0)
printf("ERROR: Can't set format. %s\n", snd_strerror(pcm));
if (pcm = snd_pcm_hw_params_set_channels(pcm_handle, params, channels) < 0)
printf("ERROR: Can't set channels number. %s\n", snd_strerror(pcm));
if (pcm = snd_pcm_hw_params_set_rate_near(pcm_handle, params, &rate, 0) < 0)
printf("ERROR: Can't set rate. %s\n", snd_strerror(pcm));
/* Write parameters */
if (pcm = snd_pcm_hw_params(pcm_handle, params) < 0)
printf("ERROR: Can't set harware parameters. %s\n", snd_strerror(pcm));
/* Resume information */
printf("PCM name: '%s'\n", snd_pcm_name(pcm_handle));
printf("PCM state: %s\n", snd_pcm_state_name(snd_pcm_state(pcm_handle)));
snd_pcm_hw_params_get_channels(params, &tmp);
printf("channels: %i ", tmp);
if (tmp == 1)
printf("(mono)\n");
else if (tmp == 2)
printf("(stereo)\n");
snd_pcm_hw_params_get_rate(params, &tmp, 0);
printf("rate: %d bps\n", tmp);
printf("seconds: %d\n", seconds);
/* Allocate buffer to hold single period */
snd_pcm_hw_params_get_period_size(params, &frames, 0);
buff_size = frames * channels * 2 /* 2 -> sample size */;
buff = (char *) malloc(buff_size);
snd_pcm_hw_params_get_period_time(params, &tmp, NULL);
for (loops = (seconds * 1000000) / tmp; loops > 0; loops--) {
if (pcm = read(0, buff, buff_size) == 0) {
printf("Early end of file.\n");
return 0;
}
if (pcm = snd_pcm_writei(pcm_handle, buff, frames) == -EPIPE) {
printf("XRUN.\n");
snd_pcm_prepare(pcm_handle);
} else if (pcm < 0) {
printf("ERROR. Can't write to PCM device. %s\n", snd_strerror(pcm));
}
}
snd_pcm_drain(pcm_handle);
snd_pcm_close(pcm_handle);
free(buff);
return 0;
}
@mark222

Hi, just came across this as I am trying to learn ALSA programming. This all makes sense to me except the reading of the WAV file input... it does not seem to read the WAV headers, it just reads starting from the first byte and interprets them as sound samples. But a WAV file has metadata before the sample data starts (e.g. "RIFF...WAVE..., etc).

@AttufliX

Yeah, WAV-file has a header which contains metadata, the player should skip this section by fseek(fp,44, SEEK_SET);

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