mirror of https://github.com/mpv-player/mpv
193 lines
3.8 KiB
C
193 lines
3.8 KiB
C
#include "config.h"
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#include "audio_in.h"
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#include "mp_msg.h"
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <errno.h>
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// sanitizes ai structure before calling other functions
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int audio_in_init(audio_in_t *ai, int type)
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{
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ai->type = type;
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ai->setup = 0;
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ai->channels = -1;
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ai->samplerate = -1;
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ai->blocksize = -1;
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ai->bytes_per_sample = -1;
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ai->samplesize = -1;
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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ai->alsa.handle = NULL;
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ai->alsa.log = NULL;
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ai->alsa.device = strdup("default");
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return 0;
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#endif
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case AUDIO_IN_OSS:
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ai->oss.audio_fd = -1;
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ai->oss.device = strdup("/dev/dsp");
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return 0;
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default:
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return -1;
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}
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}
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int audio_in_setup(audio_in_t *ai)
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{
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int err;
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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if (ai_alsa_init(ai) < 0) return -1;
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ai->setup = 1;
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return 0;
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#endif
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case AUDIO_IN_OSS:
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if (ai_oss_init(ai) < 0) return -1;
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ai->setup = 1;
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return 0;
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default:
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return -1;
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}
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}
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int audio_in_set_samplerate(audio_in_t *ai, int rate)
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{
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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ai->req_samplerate = rate;
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if (!ai->setup) return 0;
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if (ai_alsa_setup(ai) < 0) return -1;
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return ai->samplerate;
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#endif
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case AUDIO_IN_OSS:
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ai->req_samplerate = rate;
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if (!ai->setup) return 0;
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if (ai_oss_set_samplerate(ai) < 0) return -1;
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return ai->samplerate;
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default:
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return -1;
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}
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}
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int audio_in_set_channels(audio_in_t *ai, int channels)
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{
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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ai->req_channels = channels;
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if (!ai->setup) return 0;
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if (ai_alsa_setup(ai) < 0) return -1;
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return ai->channels;
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#endif
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case AUDIO_IN_OSS:
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ai->req_channels = channels;
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if (!ai->setup) return 0;
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if (ai_oss_set_channels(ai) < 0) return -1;
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return ai->channels;
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default:
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return -1;
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}
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}
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int audio_in_set_device(audio_in_t *ai, char *device)
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{
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int i;
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if (ai->setup) return -1;
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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if (ai->alsa.device) free(ai->alsa.device);
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ai->alsa.device = strdup(device);
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/* mplayer cannot handle colons in arguments */
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for (i = 0; i < strlen(ai->alsa.device); i++) {
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if (ai->alsa.device[i] == ',') ai->alsa.device[i] = ':';
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}
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return 0;
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#endif
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case AUDIO_IN_OSS:
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if (ai->oss.device) free(ai->oss.device);
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ai->oss.device = strdup(device);
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return 0;
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default:
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return -1;
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}
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}
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int audio_in_uninit(audio_in_t *ai)
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{
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if (ai->setup) {
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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if (ai->alsa.log)
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snd_output_close(ai->alsa.log);
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if (ai->alsa.handle) {
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snd_pcm_close(ai->alsa.handle);
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}
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ai->setup = 0;
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return 0;
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#endif
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case AUDIO_IN_OSS:
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close(ai->oss.audio_fd);
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ai->setup = 0;
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return 0;
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default:
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return -1;
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}
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}
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}
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int audio_in_start_capture(audio_in_t *ai)
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{
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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return snd_pcm_start(ai->alsa.handle);
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#endif
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case AUDIO_IN_OSS:
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return 0;
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default:
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return -1;
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}
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}
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int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer)
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{
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int ret;
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switch (ai->type) {
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#ifdef HAVE_ALSA9
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case AUDIO_IN_ALSA:
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ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size);
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if (ret != ai->alsa.chunk_size) {
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if (ret < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret));
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} else {
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mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
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}
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return -1;
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}
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return ret;
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#endif
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case AUDIO_IN_OSS:
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ret = read(ai->oss.audio_fd, buffer, ai->blocksize);
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if (ret != ai->blocksize) {
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if (ret < 0) {
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mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno));
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} else {
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mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n");
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}
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return -1;
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}
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return ret;
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default:
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return -1;
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}
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}
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