mirror of
https://github.com/mpv-player/mpv.git
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78128bddda
I hate tabs. This replaces all tabs in all source files with spaces. The only exception is old-makefile. The replacement was made by running the GNU coreutils "expand" command on every file. Since the replacement was automatic, it's possible that some formatting was destroyed (but perhaps only if it was assuming that the end of a tab does not correspond to aligning the end to multiples of 8 spaces).
297 lines
7.1 KiB
C
297 lines
7.1 KiB
C
/*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include "config.h"
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#include "audio_in.h"
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#include "common/msg.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, struct mp_log *log, 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->log = log;
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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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#if HAVE_ALSA
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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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#if HAVE_OSS_AUDIO
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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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#endif
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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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ai->sndio.hdl = NULL;
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ai->sndio.device = strdup("default");
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return 0;
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#endif
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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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switch (ai->type) {
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#if HAVE_ALSA
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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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#if HAVE_OSS_AUDIO
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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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#endif
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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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if (ai_sndio_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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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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#if HAVE_ALSA
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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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#if HAVE_OSS_AUDIO
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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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#endif
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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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ai->req_samplerate = rate;
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if (!ai->setup) return 0;
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if (ai_sndio_setup(ai) < 0) return -1;
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return ai->samplerate;
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#endif
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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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#if HAVE_ALSA
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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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#if HAVE_OSS_AUDIO
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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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#endif
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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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ai->req_channels = channels;
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if (!ai->setup) return 0;
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if (ai_sndio_setup(ai) < 0) return -1;
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return ai->channels;
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#endif
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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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#if HAVE_ALSA
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int i;
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#endif
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if (ai->setup) return -1;
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switch (ai->type) {
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#if HAVE_ALSA
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case AUDIO_IN_ALSA:
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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 < (int)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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#if HAVE_OSS_AUDIO
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case AUDIO_IN_OSS:
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free(ai->oss.device);
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ai->oss.device = strdup(device);
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return 0;
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#endif
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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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if (ai->sndio.device) free(ai->sndio.device);
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ai->sndio.device = strdup(device);
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return 0;
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#endif
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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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#if HAVE_ALSA
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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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#if HAVE_OSS_AUDIO
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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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#endif
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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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if (ai->sndio.hdl)
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sio_close(ai->sndio.hdl);
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ai->setup = 0;
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return 0;
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#endif
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}
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}
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return -1;
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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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#if HAVE_ALSA
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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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#if HAVE_OSS_AUDIO
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case AUDIO_IN_OSS:
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return 0;
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#endif
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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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if (!sio_start(ai->sndio.hdl))
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return -1;
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return 0;
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#endif
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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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#if HAVE_ALSA
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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_ERR(ai, "\nError reading audio: %s\n", snd_strerror(ret));
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if (ret == -EPIPE) {
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if (ai_alsa_xrun(ai) == 0) {
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MP_ERR(ai, "Recovered from cross-run, some frames may be left out!\n");
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} else {
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MP_ERR(ai, "Fatal error, cannot recover!\n");
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}
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}
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} else {
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MP_ERR(ai, "\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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#if HAVE_OSS_AUDIO
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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_ERR(ai, "\nError reading audio: %s\n", strerror(errno));
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} else {
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MP_ERR(ai, "\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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#if HAVE_SNDIO
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case AUDIO_IN_SNDIO:
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ret = sio_read(ai->sndio.hdl, buffer, ai->blocksize);
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if (ret != ai->blocksize) {
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if (ret < 0) {
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MP_ERR(ai, "\nError reading audio: %s\n", strerror(errno));
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} else {
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MP_ERR(ai, "\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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default:
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return -1;
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}
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}
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