Mercurial > mplayer.hg
view stream/audio_in.c @ 33556:520fb0f7544c
Rename GUI directory 'mplayer' and some files in it.
The directory 'mplayer' contains the files for the user interface and
has thus been renamed 'ui'.
Inside this directory the following files have been renamed to better
reflect their contents:
mw.c -> main.c
sw.c -> sub.c
pb.c -> playbar.c
gui_common.* -> render.*
play.* -> actions.*
author | ib |
---|---|
date | Sat, 18 Jun 2011 16:03:31 +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; } }