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#include <stdio.h>
#include <stdlib.h>
#include "config.h"
#if defined(USE_TV) && defined(HAVE_TV_V4L)
#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
case AUDIO_IN_OSS:
ai->oss.audio_fd = -1;
ai->oss.device = strdup("/dev/dsp");
return 0;
default:
return -1;
}
}
int audio_in_setup(audio_in_t *ai)
{
int err;
switch (ai->type) {
#ifdef HAVE_ALSA9
case AUDIO_IN_ALSA:
if (ai_alsa_init(ai) < 0) return -1;
ai->setup = 1;
return 0;
#endif
case AUDIO_IN_OSS:
if (ai_oss_init(ai) < 0) return -1;
ai->setup = 1;
return 0;
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
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;
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
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;
default:
return -1;
}
}
int audio_in_set_device(audio_in_t *ai, char *device)
{
int i;
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 < strlen(ai->alsa.device); i++) {
if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':';
}
return 0;
#endif
case AUDIO_IN_OSS:
if (ai->oss.device) free(ai->oss.device);
ai->oss.device = strdup(device);
return 0;
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
case AUDIO_IN_OSS:
close(ai->oss.audio_fd);
ai->setup = 0;
return 0;
default:
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
case AUDIO_IN_OSS:
return 0;
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));
} else {
mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
}
return -1;
}
return ret;
#endif
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;
default:
return -1;
}
}
#endif
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