Mercurial > mplayer.hg
view stream/audio_in.c @ 35006:733628daff6f
Missed documentation update.
author | reimar |
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date | Sat, 18 Aug 2012 15:56:18 +0000 |
parents | b39155e98ac3 |
children |
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/* * This file is part of MPlayer. * * MPlayer 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. * * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #include "audio_in.h" #include "mp_msg.h" #include "help_mp.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 CONFIG_ALSA case AUDIO_IN_ALSA: ai->alsa.handle = NULL; ai->alsa.log = NULL; ai->alsa.device = strdup("default"); return 0; #endif #ifdef CONFIG_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 CONFIG_ALSA case AUDIO_IN_ALSA: if (ai_alsa_init(ai) < 0) return -1; ai->setup = 1; return 0; #endif #ifdef CONFIG_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 CONFIG_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 #ifdef CONFIG_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 CONFIG_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 #ifdef CONFIG_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 CONFIG_ALSA int i; #endif if (ai->setup) return -1; switch (ai->type) { #ifdef CONFIG_ALSA case AUDIO_IN_ALSA: 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 CONFIG_OSS_AUDIO case AUDIO_IN_OSS: 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 CONFIG_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 #ifdef CONFIG_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 CONFIG_ALSA case AUDIO_IN_ALSA: return snd_pcm_start(ai->alsa.handle); #endif #ifdef CONFIG_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 CONFIG_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_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret)); if (ret == -EPIPE) { if (ai_alsa_xrun(ai) == 0) { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut); } else { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover); } } } else { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); } return -1; } return ret; #endif #ifdef CONFIG_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, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno)); } else { mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); } return -1; } return ret; #endif default: return -1; } }