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alsa.c
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alsa.c
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/* $Id$ */
/*
* Copyright (c) 2013 Dimitri Sokolyuk <[email protected]>
*
* Permission to use, copy, modify, and distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
#include <assert.h>
#include <err.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>
#include "sio.h"
#define STEREO 2
#define RATE 48000
#define FPS 25
static snd_pcm_t *hdl;
static snd_pcm_hw_params_t *par;
static int16_t *buffer;
static unsigned int samples;
struct data {
int16_t left;
int16_t right;
};
int
init_sio(void)
{
snd_pcm_uframes_t round;
unsigned int rate;
int rc;
rc = snd_pcm_open(&hdl, "default", SND_PCM_STREAM_CAPTURE, 0);
if (rc < 0)
errx(1, "unable to open pcm device: %s", snd_strerror(rc));
snd_pcm_hw_params_malloc(&par);
snd_pcm_hw_params_any(hdl, par);
snd_pcm_hw_params_set_access(hdl, par,
SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(hdl, par,
SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_channels(hdl, par, STEREO);
snd_pcm_hw_params_set_rate(hdl, par, RATE, 0);
rc = snd_pcm_hw_params(hdl, par);
if (rc < 0)
errx(1, "unable to set hw parameters: %s", snd_strerror(rc));
snd_pcm_hw_params_get_period_size(par, &round, NULL);
snd_pcm_hw_params_get_rate(par, &rate, 0);
snd_pcm_hw_params_free(par);
snd_pcm_prepare(hdl);
samples = rate / FPS;
samples -= samples % round - round;
buffer = calloc(samples * STEREO, sizeof(int16_t));
assert(buffer);
return 0;
}
unsigned int
max_samples_sio(void)
{
return samples;
}
void
read_sio(double *left, double *right, size_t n)
{
snd_pcm_sframes_t rc;
struct data *data;
int i;
if (n > samples)
n = samples;
rc = snd_pcm_readi(hdl, buffer, samples);
if (rc != samples) {
warnx("audio read error: %s", snd_strerror(rc));
if (rc == -EPIPE)
snd_pcm_prepare(hdl);
}
data = (struct data *)&buffer[samples - n];
for (i = 0; i < n; i++) {
left[i] = data[i].left;
right[i] = data[i].right;
}
}
void
free_sio(void)
{
snd_pcm_drain(hdl);
snd_pcm_close(hdl);
free(buffer);
}