Mercurial > mplayer.hg
view libmpdemux/audio_in.c @ 8425:58063f77aa4d
10l - demux_avi handles AVI_NI and AVI_NINI too
patch by R¸«ädiger Kuhlmann ruediger@ruediger-kuhlmann.de
author | arpi |
---|---|
date | Wed, 11 Dec 2002 20:48:51 +0000 |
parents | 772d6d27fd66 |
children | 31f12f99118b |
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#include <stdio.h> #include <stdlib.h> #include <unistd.h> #include "config.h" #if defined(USE_TV) && defined(HAVE_TV_V4L) #include "audio_in.h" #include "mp_msg.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 HAVE_ALSA9 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) { #ifdef HAVE_ALSA9 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) { #ifdef HAVE_ALSA9 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) { #ifdef HAVE_ALSA9 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) { #ifdef HAVE_ALSA9 int i; #endif if (ai->setup) return -1; switch (ai->type) { #ifdef HAVE_ALSA9 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) { #ifdef HAVE_ALSA9 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) { #ifdef HAVE_ALSA9 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) { #ifdef HAVE_ALSA9 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, "\nerror reading audio: %s\n", snd_strerror(ret)); if (ret == -EPIPE) { if (ai_alsa_xrun(ai) == 0) { mp_msg(MSGT_TV, MSGL_ERR, "Recovered from cross-run, some frames may be left out!\n"); } else { mp_msg(MSGT_TV, MSGL_ERR, "Fatal error, cannot recover!\n"); } } } else { mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); } 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, "\nerror reading audio: %s\n", strerror(errno)); } else { mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); } return -1; } return ret; #endif default: return -1; } } #endif