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mpv/stream/audio_in.c

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/*
* This file is part of mpv.
*
* mpv is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* mpv is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "audio_in.h"
#include "common/common.h"
#include "common/msg.h"
#include <string.h>
#include <errno.h>
// sanitizes ai structure before calling other functions
int audio_in_init(audio_in_t *ai, struct mp_log *log, int type)
{
ai->type = type;
ai->setup = 0;
ai->log = log;
ai->channels = -1;
ai->samplerate = -1;
ai->blocksize = -1;
ai->bytes_per_sample = -1;
ai->samplesize = -1;
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
ai->alsa.handle = NULL;
ai->alsa.log = NULL;
ai->alsa.device = NULL;
return 0;
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
ai->oss.audio_fd = -1;
ai->oss.device = NULL;
return 0;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
ai->sndio.hdl = NULL;
ai->sndio.device = NULL;
return 0;
#endif
default:
return -1;
}
}
int audio_in_setup(audio_in_t *ai)
{
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
if (ai_alsa_init(ai) < 0) return -1;
ai->setup = 1;
return 0;
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
if (ai_oss_init(ai) < 0) return -1;
ai->setup = 1;
return 0;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
if (ai_sndio_init(ai) < 0) return -1;
ai->setup = 1;
return 0;
#endif
default:
return -1;
}
}
int audio_in_set_samplerate(audio_in_t *ai, int rate)
{
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
ai->req_samplerate = rate;
if (!ai->setup) return 0;
if (ai_alsa_setup(ai) < 0) return -1;
return ai->samplerate;
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
ai->req_samplerate = rate;
if (!ai->setup) return 0;
if (ai_oss_set_samplerate(ai) < 0) return -1;
return ai->samplerate;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
ai->req_samplerate = rate;
if (!ai->setup) return 0;
if (ai_sndio_setup(ai) < 0) return -1;
return ai->samplerate;
#endif
default:
return -1;
}
}
int audio_in_set_channels(audio_in_t *ai, int channels)
{
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
ai->req_channels = channels;
if (!ai->setup) return 0;
if (ai_alsa_setup(ai) < 0) return -1;
return ai->channels;
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
ai->req_channels = channels;
if (!ai->setup) return 0;
if (ai_oss_set_channels(ai) < 0) return -1;
return ai->channels;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
ai->req_channels = channels;
if (!ai->setup) return 0;
if (ai_sndio_setup(ai) < 0) return -1;
return ai->channels;
#endif
default:
return -1;
}
}
int audio_in_set_device(audio_in_t *ai, char *device)
{
#if HAVE_ALSA
int i;
#endif
if (ai->setup) return -1;
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
free(ai->alsa.device);
ai->alsa.device = strdup(device);
if (ai->alsa.device) {
/* mplayer could not handle colons in arguments */
for (i = 0; i < (int)strlen(ai->alsa.device); i++) {
if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':';
}
}
return 0;
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
free(ai->oss.device);
ai->oss.device = strdup(device);
return 0;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
if (ai->sndio.device) free(ai->sndio.device);
ai->sndio.device = strdup(device);
return 0;
#endif
default:
return -1;
}
}
int audio_in_uninit(audio_in_t *ai)
{
if (ai->setup) {
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
if (ai->alsa.log)
snd_output_close(ai->alsa.log);
if (ai->alsa.handle) {
snd_pcm_close(ai->alsa.handle);
}
ai->setup = 0;
return 0;
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
close(ai->oss.audio_fd);
ai->setup = 0;
return 0;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
if (ai->sndio.hdl)
sio_close(ai->sndio.hdl);
ai->setup = 0;
return 0;
#endif
}
}
return -1;
}
int audio_in_start_capture(audio_in_t *ai)
{
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
return snd_pcm_start(ai->alsa.handle);
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
return 0;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
if (!sio_start(ai->sndio.hdl))
return -1;
return 0;
#endif
default:
return -1;
}
}
int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer)
{
int ret;
switch (ai->type) {
#if HAVE_ALSA
case AUDIO_IN_ALSA:
ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size);
if (ret != ai->alsa.chunk_size) {
if (ret < 0) {
MP_ERR(ai, "\nError reading audio: %s\n", snd_strerror(ret));
if (ret == -EPIPE) {
if (ai_alsa_xrun(ai) == 0) {
MP_ERR(ai, "Recovered from cross-run, some frames may be left out!\n");
} else {
MP_ERR(ai, "Fatal error, cannot recover!\n");
}
}
} else {
MP_ERR(ai, "\nNot enough audio samples!\n");
}
return -1;
}
return ret;
#endif
#if HAVE_OSS_AUDIO
case AUDIO_IN_OSS:
ret = read(ai->oss.audio_fd, buffer, ai->blocksize);
if (ret != ai->blocksize) {
if (ret < 0) {
MP_ERR(ai, "\nError reading audio: %s\n", mp_strerror(errno));
} else {
MP_ERR(ai, "\nNot enough audio samples!\n");
}
return -1;
}
return ret;
#endif
#if HAVE_SNDIO
case AUDIO_IN_SNDIO:
ret = sio_read(ai->sndio.hdl, buffer, ai->blocksize);
if (ret != ai->blocksize) {
if (ret < 0) {
MP_ERR(ai, "\nError reading audio: %s\n", mp_strerror(errno));
} else {
MP_ERR(ai, "\nNot enough audio samples!\n");
}
return -1;
}
return ret;
#endif
default:
return -1;
}
}
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