how to play sound using user sample application instead of aplay

Hi Thangameena,

Why don't you try like this (just using simple "system" command) ?

Or Is there any intention to use "ALSA 'C' app ?

int main() {
    system("aplay sound.wav");
    return 0;
}

You can also feed the ".wav" file while executing the binary through "argv", you have to modify the "argc and argv stuffs in 'C'' code.

this is my code.. 

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>

#define BUFF_SIZE 4096

int main (int argc, char *argv[])
{
int err;
short buf[BUFF_SIZE];
int rate = 44100; /* Sample rate */
unsigned int exact_rate; /* Sample rate returned by */
/* Handle for the PCM device */
snd_pcm_t *playback_handle;
/* Playback stream */
snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK;
/* This structure contains information about the hardware and can be used */
/*tospecify the configuration to be used for */
/* the PCM stream. */
snd_pcm_hw_params_t *hw_params;

static char *device = "default"; /* playback device */

/* Open PCM. The last parameter of this function is the mode. */
//if ((err = snd_pcm_open (&playback_handle, device, stream, 0)) < 0) {
if ((err = snd_pcm_open (&playback_handle,device, stream, 0)) < 0) {
fprintf (stderr, "cannot open audio device (%s)\n", snd_strerror (err));
exit (1);
}

/* Allocate the snd_pcm_hw_params_t structure on the stack. */
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
fprintf (stderr, "cannot allocate hardware parameters (%s)\n", snd_strerror (err));
exit (1);
}

/* Init hwparams with full configuration space */
if ((err = snd_pcm_hw_params_any (playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err));
exit (1);
}

/* Set access type. */
if ((err = snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err));
exit (1);
}

/* Set sample format */
if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err));
exit (1);
}

/* Set sample rate. If the exact rate is not supported by the hardware, use nearest possible rate. */
exact_rate = rate;
if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, &exact_rate, 0)) < 0) {
fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err));
exit (1);
}

if (rate != exact_rate) {
fprintf(stderr, "The rate %d Hz is not supported by your hardware.\n ==> Using %d Hz instead.\n", rate, exact_rate);
}

/* Set number of channels */
if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2)) < 0) {
fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err));
exit (1);
}

/* Apply HW parameter settings to PCM device and prepare device. */
if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) {
fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err));
exit (1);
}

snd_pcm_hw_params_free (hw_params);

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

/* Write some junk data to produce sound. */
if ((err = snd_pcm_writei (playback_handle, buf, BUFF_SIZE/2)) != BUFF_SIZE/2) {
fprintf (stderr, "write to audio interface failed (%s)\n", snd_strerror (err));
exit (1);
} else {
fprintf (stdout, "snd_pcm_writei successful\n");
}
snd_pcm_close (playback_handle);

exit (0);
}

gcc -o audio audio.c -lasound

./audio  -D 'hw:0,0' -r 48000 -f S16_LE -c 2  bird.wav

 but its not playing..   

ya i need to use alsa c app

This program is not playing anything because it never writes anything into buf.

What do you want to do in your program that aplay doesn't do?

 ya now i changed the code .. to write into buffer..

now compiled and run ./audio < bird.wav.. now its playing in my pc.. then i complied for my target board  using arm-arago..  but its not playing  rather i get some noise .. 

this is the changes i done

{

loops = 5000000 / val;

while (loops > 0) {
loops--;
rc = read(0, buffer, size);
if (rc == 0) {
fprintf(stderr, "end of file on input\n");
break;
} else if (rc != size) {
fprintf(stderr,
"short read: read %d bytes\n", rc);
}
rc = snd_pcm_writei(handle, buffer, frames);
if (rc == -EPIPE) {
/* EPIPE means underrun */
fprintf(stderr, "underrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0) {
fprintf(stderr,
"error from writei: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr,
"short write, write %d frames\n", rc);
}
}

snd_pcm_drain(handle);
snd_pcm_close(handle);
free(buffer);

}

Does aplay work with this file?

Does the program print anything?

ya aplay works.. 

in target i will give aplay bird.wav.. i can hear sound..

if i give my compiled binary  to the target

./audio  < bird.wav 

it will produce noise..

underrun occurred
underrun occurred
underrun occurred
underrun occurred
underrun occurred
underrun occurred
underrun occurred
short read: read 28 bytes
end of file on input

Hi Ranjana,

The "underrun" issue seems to be the code is only for loopback the data for 5 seconds.

So, you can record and play simultaneously through piping.

Change

size = frames * 4; /* 2 bytes/sample, 2 channels */

TO

size = frames * sizeof(double);

its still underrun.. i changed the size.. working in lcdk.. so according to audio ic .. did i need to change any files.. 

0272.example.c

compiled using ..arm-arago-linux-gnueabi-gcc -o example  example.c -lasound

run ./example < musi.wav

What board and processor are you using ?

custom or TI EVM ?

hi stalin,

i am using omap l138 lcdk evm..   

the same program playing musing in my pc.. 

my doubt is did i need to put any library files  in lcdk.??