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;
}

Thank you for this.

		if (pcm = snd_pcm_writei(pcm_handle, buff, frames) == -EPIPE) {
			printf("XRUN.\n");

In this code won't it always be a buffer overrun if you are trying to write frames out and can't?

The application can run in command line, but when added to startup script (eg. rc.local) so after power on it can run automatically, it crashes. The core dump file show it crashed at following:

root@linaro-ubuntu-desktop:/test# gdb ./playback /opt/core.playback.6238.1604316560.6
GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08
Copyright (C) 2011 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html>
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
and "show warranty" for details.
This GDB was configured as "arm-linux-gnueabi".
For bug reporting instructions, please see:
<http://bugs.launchpad.net/gdb-linaro/>...
Reading symbols from /test/playback...(no debugging symbols found)...done.
[New LWP 6238]
[Thread debugging using libthread_db enabled]
Core was generated by `/test/playback'.
Program terminated with signal 6, Aborted.
#0  0x2ad93ed6 in ?? () from /lib/arm-linux-gnueabi/libc.so.6
(gdb) bt full
#0  0x2ad93ed6 in ?? () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#1  0x2ada20da in raise () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#2  0x2ada4506 in abort () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#3  0x2ad9d1ec in __assert_fail () from /lib/arm-linux-gnueabi/libc.so.6
No symbol table info available.
#4  0x2ad1fb72 in snd_pcm_hw_refine () from /usr/lib/arm-linux-gnueabi/libasound.so.2
No symbol table info available.
#5  0x00008cd4 in main ()
No symbol table info available.

Note snd_pcm_hw_refine() is not directly called by this application.

I have hardcoded all parameters to 32000hz / 1chn / 30s and "/opt/test1.wav". The read changed to

	int fd = open("/opt/test1.wav", O_RDONLY);   ///added
	for (loops = (seconds * 1000000) / tmp; loops > 0; loops--) {

		if (pcm = read(fd, buff, buff_size) == 0) {   ///changed to fd from 0
			printf("Early end of file.\n");
			return 0;
		}

Further tracing found it crashed at snd_pcm_hw_params_any() after snd_pcm_open(), snd_pcm_hw_params_alloca().

Note: The application only crashes after auto-run from power on; if run the script in putty (command line), no problem.

Anyone encounter this problem? How have you solved it? Thank you.

Got another issue, XRUN in alternative cycle (this would happen after the playback from 2nd time onwards):
2020-11-02 21:03:42 [audio]XRUN:0,1 --- cycle 1, 0=previous is ok
2020-11-02 21:03:42 [audio]XRUN:0,3 --- cycle 3, 0=previous is ok
2020-11-02 21:03:42 [audio]XRUN:0,5 --- cycle 5, 0=previous is ok
2020-11-02 21:03:42 [audio]XRUN:0,7
2020-11-02 21:03:42 [audio]XRUN:0,9

The effect of XRUN is heard either partially or no sound.

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?