Simple sound playback using ALSA API and libasound

sound_playback.c

/*
 * 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;
}

Looks easier than I thought

./play 44100 2 2 < hello.wav
Just gives out a very short click sound and then Im getting early end of file. Can anyone help ?

pi@raspberrypi:~/tinyalsa $ ./play 44100 2 2 < hello.wav 
PCM name: 'default'
PCM state: PREPARED
channels: 2 (stereo)
rate: 44100 bps
seconds: 2
Early end of file.

@rsingh2083 The click sound is probably because it plays the metadata of the wav file. In a real player you would need to skip the header of the wav file.

How to play 24-bit audio?

How to play 24-bit audio?

You can use SND_PCM_FORMAT_S24_LE instead of SND_PCM_FORMAT_S16_LE

It's nice to see an attempt at a minimal ALSA program. There are some bugs that should probably be cleaned up, though. Examples:

• &rate should be just rate,

• variable pcm should be called err as in the ALSA docs,

• the audio file should be played to the end instead of for a certain number of seconds,

• buff_size should not presume sample width is 2 bytes. For calculating buff_size, one can use _physical_width

    snd_pcm_format_t sampfmt;
    snd_pcm_hw_params_get_format(params, &sampfmt);
    printf("sample format: %s\n", snd_pcm_format_description(sampfmt));
  
    snd_pcm_format_t sampwidth = snd_pcm_format_physical_width(sampfmt);
    printf("sample width: %d bits\n", sampwidth);

Hey! Thanks for sharing. It does run flawless but i'm running into memory corruption when wrapping logic into functions. I probably am thinking about it wrong but i don't see a problem with my code, but i want to rule out ALSA is doing something odd. If anyone could take a look it would be greatly appreciated.

https://gist.github.com/dedobbin/29adb1dae10932b4a88721bb00a1fc45

edit: oh, already found it. very silly, i used snd_pcm_hw_params_alloca which allocated on the stack obviously
should have used snd_pcm_hw_params_malloc

I have modified the code to play wav files instead of stdin, I find that the wav files are played at a faster tempo. The problem is described here https://stackoverflow.com/questions/77920231/wav-files-are-played-by-alsa-at-a-faster-tempo-is-there-a-way-to-fix-the-tempo

Could anyone kindly point me my mistake?