Mercurial > mplayer.hg
view libmpcodecs/ae_lavc.c @ 30088:4977e04f3a18
Add support for parsing audio streams (though should be easy to extend to video)
via libavcodec.
Parsing can be done at the demuxer stage (currently disabled) or at the decoder
(ad_ffmpeg, enabled).
Should allow using the libavcodec AAC, DTS, ... decoders independent of container
format.
author | reimar |
---|---|
date | Sun, 27 Dec 2009 15:28:01 +0000 |
parents | a3cc38ad5878 |
children | bbb6ebec87a0 |
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#include <stdio.h> #include <stdlib.h> #include <inttypes.h> #include <unistd.h> #include <string.h> #include <sys/types.h> #include "config.h" #include "m_option.h" #include "mp_msg.h" #include "libmpdemux/aviheader.h" #include "libmpdemux/ms_hdr.h" #include "stream/stream.h" #include "libmpdemux/muxer.h" #include "ae_lavc.h" #include "help_mp.h" #include "libaf/af_format.h" #include "libaf/reorder_ch.h" #include "libavcodec/avcodec.h" #include "libavutil/intreadwrite.h" static AVCodec *lavc_acodec; static AVCodecContext *lavc_actx; extern char *lavc_param_acodec; extern int lavc_param_abitrate; extern int lavc_param_atag; extern int lavc_param_audio_global_header; extern int avcodec_initialized; static int compressed_frame_size = 0; #ifdef CONFIG_LIBAVFORMAT #include "libavformat/avformat.h" extern const struct AVCodecTag *mp_wav_taglists[]; #endif static int bind_lavc(audio_encoder_t *encoder, muxer_stream_t *mux_a) { mux_a->wf = malloc(sizeof(WAVEFORMATEX)+lavc_actx->extradata_size+256); mux_a->wf->wFormatTag = lavc_param_atag; mux_a->wf->nChannels = lavc_actx->channels; mux_a->wf->nSamplesPerSec = lavc_actx->sample_rate; mux_a->wf->nAvgBytesPerSec = (lavc_actx->bit_rate / 8); mux_a->avg_rate= lavc_actx->bit_rate; mux_a->h.dwRate = mux_a->wf->nAvgBytesPerSec; if(lavc_actx->block_align) mux_a->h.dwSampleSize = mux_a->h.dwScale = lavc_actx->block_align; else { mux_a->h.dwScale = (mux_a->wf->nAvgBytesPerSec * lavc_actx->frame_size)/ mux_a->wf->nSamplesPerSec; /* for cbr */ if ((mux_a->wf->nAvgBytesPerSec * lavc_actx->frame_size) % mux_a->wf->nSamplesPerSec) { mux_a->h.dwScale = lavc_actx->frame_size; mux_a->h.dwRate = lavc_actx->sample_rate; mux_a->h.dwSampleSize = 0; // Blocksize not constant } else mux_a->h.dwSampleSize = 0; } if(mux_a->h.dwSampleSize) mux_a->wf->nBlockAlign = mux_a->h.dwSampleSize; else mux_a->wf->nBlockAlign = 1; mux_a->h.dwSuggestedBufferSize = (encoder->params.audio_preload*mux_a->wf->nAvgBytesPerSec)/1000; mux_a->h.dwSuggestedBufferSize -= mux_a->h.dwSuggestedBufferSize % mux_a->wf->nBlockAlign; switch(lavc_param_atag) { case 0x11: /* imaadpcm */ mux_a->wf->wBitsPerSample = 4; mux_a->wf->cbSize = 2; AV_WL16(mux_a->wf+1, lavc_actx->frame_size); break; case 0x55: /* mp3 */ mux_a->wf->cbSize = 12; mux_a->wf->wBitsPerSample = 0; /* does not apply */ ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->wID = 1; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->fdwFlags = 2; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nBlockSize = mux_a->wf->nBlockAlign; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nFramesPerBlock = 1; ((MPEGLAYER3WAVEFORMAT *) (mux_a->wf))->nCodecDelay = 0; break; default: mux_a->wf->wBitsPerSample = 0; /* Unknown */ if (lavc_actx->extradata && (lavc_actx->extradata_size > 0)) { memcpy(mux_a->wf+1, lavc_actx->extradata, lavc_actx->extradata_size); mux_a->wf->cbSize = lavc_actx->extradata_size; } else mux_a->wf->cbSize = 0; break; } // Fix allocation mux_a->wf = realloc(mux_a->wf, sizeof(WAVEFORMATEX)+mux_a->wf->cbSize); encoder->input_format = AF_FORMAT_S16_NE; encoder->min_buffer_size = mux_a->h.dwSuggestedBufferSize; encoder->max_buffer_size = mux_a->h.dwSuggestedBufferSize*2; return 1; } static int encode_lavc(audio_encoder_t *encoder, uint8_t *dest, void *src, int size, int max_size) { int n; if ((encoder->params.channels == 6 || encoder->params.channels == 5) && (!strcmp(lavc_acodec->name,"ac3") || !strcmp(lavc_acodec->name,"libfaac"))) { int isac3 = !strcmp(lavc_acodec->name,"ac3"); reorder_channel_nch(src, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, isac3 ? AF_CHANNEL_LAYOUT_LAVC_DEFAULT : AF_CHANNEL_LAYOUT_AAC_DEFAULT, encoder->params.channels, size / 2, 2); } n = avcodec_encode_audio(lavc_actx, dest, size, src); compressed_frame_size = n; return n; } static int close_lavc(audio_encoder_t *encoder) { compressed_frame_size = 0; return 1; } static int get_frame_size(audio_encoder_t *encoder) { int sz = compressed_frame_size; compressed_frame_size = 0; return sz; } #ifndef CONFIG_LIBAVFORMAT static uint32_t lavc_find_atag(char *codec) { if(codec == NULL) return 0; if(! strcasecmp(codec, "mp2")) return 0x50; if(! strcasecmp(codec, "mp3")) return 0x55; if(! strcasecmp(codec, "ac3")) return 0x2000; if(! strcasecmp(codec, "adpcm_ima_wav")) return 0x11; if(! strncasecmp(codec, "bonk", 4)) return 0x2048; return 0; } #endif int mpae_init_lavc(audio_encoder_t *encoder) { encoder->params.samples_per_frame = encoder->params.sample_rate; encoder->params.bitrate = encoder->params.sample_rate * encoder->params.channels * 2 * 8; if(!lavc_param_acodec) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_NoLavcAudioCodecName); return 0; } if(!avcodec_initialized){ avcodec_init(); avcodec_register_all(); avcodec_initialized=1; } lavc_acodec = avcodec_find_encoder_by_name(lavc_param_acodec); if (!lavc_acodec) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_LavcAudioCodecNotFound, lavc_param_acodec); return 0; } if(lavc_param_atag == 0) { #ifdef CONFIG_LIBAVFORMAT lavc_param_atag = av_codec_get_tag(mp_wav_taglists, lavc_acodec->id); #else lavc_param_atag = lavc_find_atag(lavc_param_acodec); #endif if(!lavc_param_atag) { mp_msg(MSGT_MENCODER, MSGL_FATAL, "Couldn't find wav tag for specified codec, exit\n"); return 0; } } lavc_actx = avcodec_alloc_context(); if(lavc_actx == NULL) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_CouldntAllocateLavcContext); return 0; } lavc_actx->codec_type = CODEC_TYPE_AUDIO; lavc_actx->codec_id = lavc_acodec->id; // put sample parameters lavc_actx->channels = encoder->params.channels; lavc_actx->sample_rate = encoder->params.sample_rate; lavc_actx->time_base.num = 1; lavc_actx->time_base.den = encoder->params.sample_rate; if(lavc_param_abitrate<1000) lavc_actx->bit_rate = encoder->params.bitrate = lavc_param_abitrate * 1000; else lavc_actx->bit_rate = encoder->params.bitrate = lavc_param_abitrate; /* * Special case for adpcm_ima_wav. * The bitrate is only dependent on samplerate. * We have to known frame_size and block_align in advance, * so I just copied the code from libavcodec/adpcm.c * * However, ms adpcm_ima_wav uses a block_align of 2048, * lavc defaults to 1024 */ if(lavc_param_atag == 0x11) { int blkalign = 2048; int framesize = (blkalign - 4 * lavc_actx->channels) * 8 / (4 * lavc_actx->channels) + 1; lavc_actx->bit_rate = lavc_actx->sample_rate*8*blkalign/framesize; } if((lavc_param_audio_global_header&1) /*|| (video_global_header==0 && (oc->oformat->flags & AVFMT_GLOBALHEADER))*/){ lavc_actx->flags |= CODEC_FLAG_GLOBAL_HEADER; } if(lavc_param_audio_global_header&2){ lavc_actx->flags2 |= CODEC_FLAG2_LOCAL_HEADER; } if(avcodec_open(lavc_actx, lavc_acodec) < 0) { mp_msg(MSGT_MENCODER, MSGL_FATAL, MSGTR_CouldntOpenCodec, lavc_param_acodec, lavc_param_abitrate); return 0; } if(lavc_param_atag == 0x11) { lavc_actx->block_align = 2048; lavc_actx->frame_size = (lavc_actx->block_align - 4 * lavc_actx->channels) * 8 / (4 * lavc_actx->channels) + 1; } encoder->decode_buffer_size = lavc_actx->frame_size * 2 * encoder->params.channels; while (encoder->decode_buffer_size < 1024) encoder->decode_buffer_size *= 2; encoder->bind = bind_lavc; encoder->get_frame_size = get_frame_size; encoder->encode = encode_lavc; encoder->close = close_lavc; return 1; }