ALSA Audio Capture - PCM Works - MPEG Fails

[LightOfSolomon]

I picked up the sample ALSA code below from Linux Journal and it works fine for outputting PCM audio format. However, I would like to change the capture format to SND_PCM_FORMAT_MPEG to make it easier to stream audio to remote clients without having to do any audio conversion. I have been unsuccessful in figuring out how to change the capture format to MPEG. Has anyone sucessfully used this mode in any ALSA development?

Compile: gcc alsatest.c -lasound

Execute: ./alsatest >sample.pcm

You can import and play the raw audio using audacity. Again, the PCM mode works great... I can't figure out how to write directly to MPEG. :confused:

/*

Source: Linux Journal - http://www.linuxjournal.com/node/6735/print
Example 4 - Simple sound recording

This example reads from the default PCM device
and writes to standard output for 5 seconds of data.

*/

/* Use the newer ALSA API */
#define ALSA_PCM_NEW_HW_PARAMS_API

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

int main() {
long loops;
int rc;
int size;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
int dir;
snd_pcm_uframes_t frames;
char *buffer;


/* Open PCM device for recording (capture). */
rc = snd_pcm_open(&handle, "default",
SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0) {
fprintf(stderr,
"unable to open pcm device: %s\n",
snd_strerror(rc));
exit(1);
}

/* Allocate a hardware parameters object. */
snd_pcm_hw_params_alloca(&params);

/* Fill it in with default values. */
snd_pcm_hw_params_any(handle, params);

/* Set the desired hardware parameters. */

/*
if( snd_pcm_hw_params_test_format( handle, params, SND_PCM_FORMAT_MPEG ) >= 0)
{
printf("Doh!");
}
*/

//exit(0);

/* Interleaved mode */
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED);



/* Signed 16-bit little-endian format */
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE);

/* Two channels (stereo) */
snd_pcm_hw_params_set_channels(handle, params, 2);

/* 44100 bits/second sampling rate (CD quality) */
val = 44100;
snd_pcm_hw_params_set_rate_near(handle, params,
&val, &dir);

/* Set period size to 32 frames. */
frames = 32;
snd_pcm_hw_params_set_period_size_near(handle,
params, &frames, &dir);

/* Write the parameters to the driver */
rc = snd_pcm_hw_params(handle, params);
if (rc < 0) {
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(rc));
exit(1);
}

/* Use a buffer large enough to hold one period */
snd_pcm_hw_params_get_period_size(params,
&frames, &dir);
size = frames * 4; /* 2 bytes/sample, 2 channels */
buffer = (char *) malloc(size);

/* We want to loop for 5 seconds */
snd_pcm_hw_params_get_period_time(params,
&val, &dir);
loops = 5000000 / val;

while (loops > 0) {
loops--;
rc = snd_pcm_readi(handle, buffer, frames);
if (rc == -EPIPE) {
/* EPIPE means overrun */
fprintf(stderr, "overrun occurred\n");
snd_pcm_prepare(handle);
} else if (rc < 0) {
fprintf(stderr,
"error from read: %s\n",
snd_strerror(rc));
} else if (rc != (int)frames) {
fprintf(stderr, "short read, read %d frames\n", rc);
}
rc = write(1, buffer, size);
if (rc != size)
fprintf(stderr,
"short write: wrote %d bytes\n", rc);
}

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

return 0;
}

[tgalati4]

I would use arecord and lame and pipe them together: (you would have to refine the syntax)

arecord myrecordedpcmfile.pcm | lame myrecordedpcmfile.mp3

The article is from 2004, so I would expect some changes to the ALSA framework, so you need to be mindful of that. As a general rule, you need to install lame and use the lame mp3 libraries to record to mp3. There are licensing issues with recording to mp3, so you need to be aware of that as well.

[LightOfSolomon]

Thanks for your help Tgalati4! That works for now. Eventually, I would like to have self-contained code to handle this task.

[tgalati4]

You can look at the source code for arecord and lame and cut-and-paste the interesting parts.