mirror of
https://github.com/mpv-player/mpv
synced 2024-12-20 22:02:59 +00:00
7e38b1aefb
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@10538 b3059339-0415-0410-9bf9-f77b7e298cf2
223 lines
4.4 KiB
C
223 lines
4.4 KiB
C
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
#include <unistd.h>
|
|
|
|
#include "config.h"
|
|
|
|
#if defined(USE_TV) && (defined(HAVE_TV_V4L) || defined(HAVE_TV_V4L2))
|
|
|
|
#include "audio_in.h"
|
|
#include "mp_msg.h"
|
|
#include <string.h>
|
|
#include <errno.h>
|
|
|
|
// sanitizes ai structure before calling other functions
|
|
int audio_in_init(audio_in_t *ai, int type)
|
|
{
|
|
ai->type = type;
|
|
ai->setup = 0;
|
|
|
|
ai->channels = -1;
|
|
ai->samplerate = -1;
|
|
ai->blocksize = -1;
|
|
ai->bytes_per_sample = -1;
|
|
ai->samplesize = -1;
|
|
|
|
switch (ai->type) {
|
|
#ifdef HAVE_ALSA9
|
|
case AUDIO_IN_ALSA:
|
|
ai->alsa.handle = NULL;
|
|
ai->alsa.log = NULL;
|
|
ai->alsa.device = strdup("default");
|
|
return 0;
|
|
#endif
|
|
#ifdef USE_OSS_AUDIO
|
|
case AUDIO_IN_OSS:
|
|
ai->oss.audio_fd = -1;
|
|
ai->oss.device = strdup("/dev/dsp");
|
|
return 0;
|
|
#endif
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
int audio_in_setup(audio_in_t *ai)
|
|
{
|
|
|
|
switch (ai->type) {
|
|
#ifdef HAVE_ALSA9
|
|
case AUDIO_IN_ALSA:
|
|
if (ai_alsa_init(ai) < 0) return -1;
|
|
ai->setup = 1;
|
|
return 0;
|
|
#endif
|
|
#ifdef USE_OSS_AUDIO
|
|
case AUDIO_IN_OSS:
|
|
if (ai_oss_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) {
|
|
#ifdef HAVE_ALSA9
|
|
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
|
|
#ifdef USE_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
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
int audio_in_set_channels(audio_in_t *ai, int channels)
|
|
{
|
|
switch (ai->type) {
|
|
#ifdef HAVE_ALSA9
|
|
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
|
|
#ifdef USE_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
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
int audio_in_set_device(audio_in_t *ai, char *device)
|
|
{
|
|
#ifdef HAVE_ALSA9
|
|
int i;
|
|
#endif
|
|
if (ai->setup) return -1;
|
|
switch (ai->type) {
|
|
#ifdef HAVE_ALSA9
|
|
case AUDIO_IN_ALSA:
|
|
if (ai->alsa.device) free(ai->alsa.device);
|
|
ai->alsa.device = strdup(device);
|
|
/* mplayer cannot 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
|
|
#ifdef USE_OSS_AUDIO
|
|
case AUDIO_IN_OSS:
|
|
if (ai->oss.device) free(ai->oss.device);
|
|
ai->oss.device = strdup(device);
|
|
return 0;
|
|
#endif
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
int audio_in_uninit(audio_in_t *ai)
|
|
{
|
|
if (ai->setup) {
|
|
switch (ai->type) {
|
|
#ifdef HAVE_ALSA9
|
|
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
|
|
#ifdef USE_OSS_AUDIO
|
|
case AUDIO_IN_OSS:
|
|
close(ai->oss.audio_fd);
|
|
ai->setup = 0;
|
|
return 0;
|
|
#endif
|
|
}
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
int audio_in_start_capture(audio_in_t *ai)
|
|
{
|
|
switch (ai->type) {
|
|
#ifdef HAVE_ALSA9
|
|
case AUDIO_IN_ALSA:
|
|
return snd_pcm_start(ai->alsa.handle);
|
|
#endif
|
|
#ifdef USE_OSS_AUDIO
|
|
case AUDIO_IN_OSS:
|
|
return 0;
|
|
#endif
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer)
|
|
{
|
|
int ret;
|
|
|
|
switch (ai->type) {
|
|
#ifdef HAVE_ALSA9
|
|
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_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret));
|
|
if (ret == -EPIPE) {
|
|
if (ai_alsa_xrun(ai) == 0) {
|
|
mp_msg(MSGT_TV, MSGL_ERR, "Recovered from cross-run, some frames may be left out!\n");
|
|
} else {
|
|
mp_msg(MSGT_TV, MSGL_ERR, "Fatal error, cannot recover!\n");
|
|
}
|
|
}
|
|
} else {
|
|
mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
|
|
}
|
|
return -1;
|
|
}
|
|
return ret;
|
|
#endif
|
|
#ifdef USE_OSS_AUDIO
|
|
case AUDIO_IN_OSS:
|
|
ret = read(ai->oss.audio_fd, buffer, ai->blocksize);
|
|
if (ret != ai->blocksize) {
|
|
if (ret < 0) {
|
|
mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno));
|
|
} else {
|
|
mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
|
|
}
|
|
return -1;
|
|
}
|
|
return ret;
|
|
#endif
|
|
default:
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
#endif
|