Mercurial > mplayer.hg
view libmpdemux/audio_in.c @ 18150:710d4bc5f8c9
Using channel count, samplerate and input bps values from the container
instead of the decoder breaks some DTS samples where the container says
the audio has 6 channels but the decoder gives 2. In this case take the
number of channels from the decoder instead, the output will almost
certainly be badly garbled anyway if the number of channels is wrong.
patch by Uoti Urpala, uoti <<.>> urpala <<@>> pp1 <<.>> inet <<.>> fi
author | diego |
---|---|
date | Wed, 19 Apr 2006 20:12:01 +0000 |
parents | dfbe8cd0e081 |
children | d2d9d011203f |
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line source
#include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #if defined(USE_TV) && (defined(HAVE_TV_V4L) || defined(HAVE_TV_V4L2)) #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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) case AUDIO_IN_ALSA: ai->alsa.handle = NULL; ai->alsa.log = NULL; ai->alsa.device = strdup("default"); return 0; #endif #ifdef USE_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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) case AUDIO_IN_ALSA: if (ai_alsa_init(ai) < 0) return -1; ai->setup = 1; return 0; #endif #ifdef USE_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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) 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 USE_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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) 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 USE_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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) int i; #endif if (ai->setup) return -1; switch (ai->type) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) 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 < (int)strlen(ai->alsa.device); i++) { if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; } return 0; #endif #ifdef USE_OSS_AUDIO case AUDIO_IN_OSS: if (ai->oss.device) 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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) 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 USE_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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) case AUDIO_IN_ALSA: return snd_pcm_start(ai->alsa.handle); #endif #ifdef USE_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) { #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) 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 USE_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; } } #endif