Mercurial > mplayer.hg
view libmpcodecs/ad_ffmpeg.c @ 32334:541a774ce8e4
Drop removal of AAC encoder from list of FFmpeg encoders.
The original reason to drop this particular encoder, that it
failed to build on many platforms, should no longer be the case.
author | diego |
---|---|
date | Fri, 01 Oct 2010 16:04:43 +0000 |
parents | c08363dc5320 |
children | 8317ee61e0b0 |
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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 "mp_msg.h" #include "help_mp.h" #include "ad_internal.h" #include "dec_audio.h" #include "vd_ffmpeg.h" #include "libaf/reorder_ch.h" #include "mpbswap.h" static const ad_info_t info = { "FFmpeg/libavcodec audio decoders", "ffmpeg", "Nick Kurshev", "ffmpeg.sf.net", "" }; LIBAD_EXTERN(ffmpeg) #define assert(x) #include "libavcodec/avcodec.h" static int preinit(sh_audio_t *sh) { sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE; return 1; } static int setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context) { int samplerate = lavc_context->sample_rate; int sample_format = sh_audio->sample_format; switch (lavc_context->sample_fmt) { case SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break; case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break; case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break; case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break; default: mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n"); } if(sh_audio->wf){ // If the decoder uses the wrong number of channels all is lost anyway. // sh_audio->channels=sh_audio->wf->nChannels; if (lavc_context->codec_id == CODEC_ID_AAC && samplerate == 2*sh_audio->wf->nSamplesPerSec) { mp_msg(MSGT_DECAUDIO, MSGL_WARN, "Ignoring broken container sample rate for ACC with SBR\n"); } else if (sh_audio->wf->nSamplesPerSec) samplerate=sh_audio->wf->nSamplesPerSec; } if (lavc_context->channels != sh_audio->channels || samplerate != sh_audio->samplerate || sample_format != sh_audio->sample_format) { sh_audio->channels=lavc_context->channels; sh_audio->samplerate=samplerate; sh_audio->sample_format = sample_format; sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8; return 1; } return 0; } static int init(sh_audio_t *sh_audio) { int tries = 0; int x; AVCodecContext *lavc_context; AVCodec *lavc_codec; mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n"); init_avcodec(); lavc_codec = avcodec_find_decoder_by_name(sh_audio->codec->dll); if(!lavc_codec){ mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll); return 0; } lavc_context = avcodec_alloc_context(); sh_audio->context=lavc_context; lavc_context->drc_scale = drc_level; lavc_context->sample_rate = sh_audio->samplerate; lavc_context->bit_rate = sh_audio->i_bps * 8; if(sh_audio->wf){ lavc_context->channels = sh_audio->wf->nChannels; lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec; lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8; lavc_context->block_align = sh_audio->wf->nBlockAlign; lavc_context->bits_per_coded_sample = sh_audio->wf->wBitsPerSample; } lavc_context->request_channels = audio_output_channels; lavc_context->codec_tag = sh_audio->format; //FOURCC lavc_context->codec_type = CODEC_TYPE_AUDIO; lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi /* alloc extra data */ if (sh_audio->wf && sh_audio->wf->cbSize > 0) { lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE); lavc_context->extradata_size = sh_audio->wf->cbSize; memcpy(lavc_context->extradata, sh_audio->wf + 1, lavc_context->extradata_size); } // for QDM2 if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata) { lavc_context->extradata = av_malloc(sh_audio->codecdata_len); lavc_context->extradata_size = sh_audio->codecdata_len; memcpy(lavc_context->extradata, (char *)sh_audio->codecdata, lavc_context->extradata_size); } /* open it */ if (avcodec_open(lavc_context, lavc_codec) < 0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec); return 0; } mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name); // printf("\nFOURCC: 0x%X\n",sh_audio->format); if(sh_audio->format==0x3343414D){ // MACE 3:1 sh_audio->ds->ss_div = 2*3; // 1 samples/packet sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet } else if(sh_audio->format==0x3643414D){ // MACE 6:1 sh_audio->ds->ss_div = 2*6; // 1 samples/packet sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet } // Decode at least 1 byte: (to get header filled) do { x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size); } while (x <= 0 && tries++ < 5); if(x>0) sh_audio->a_buffer_len=x; sh_audio->i_bps=lavc_context->bit_rate/8; if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec) sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec; switch (lavc_context->sample_fmt) { case SAMPLE_FMT_U8: case SAMPLE_FMT_S16: case SAMPLE_FMT_S32: case SAMPLE_FMT_FLT: break; default: return 0; } return 1; } static void uninit(sh_audio_t *sh) { AVCodecContext *lavc_context = sh->context; if (avcodec_close(lavc_context) < 0) mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec); av_freep(&lavc_context->extradata); av_freep(&lavc_context); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { AVCodecContext *lavc_context = sh->context; switch(cmd){ case ADCTRL_RESYNC_STREAM: avcodec_flush_buffers(lavc_context); ds_clear_parser(sh->ds); return CONTROL_TRUE; } return CONTROL_UNKNOWN; } static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen) { unsigned char *start=NULL; int y,len=-1; while(len<minlen){ AVPacket pkt; int len2=maxlen; double pts; int x=ds_get_packet_pts(sh_audio->ds,&start, &pts); if(x<=0) { start = NULL; x = 0; ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0); if (x <= 0) break; // error } else { int in_size = x; int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0); sh_audio->ds->buffer_pos -= in_size - consumed; } av_init_packet(&pkt); pkt.data = start; pkt.size = x; if (pts != MP_NOPTS_VALUE) { sh_audio->pts = pts; sh_audio->pts_bytes = 0; } y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt); //printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout); if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; } if(!sh_audio->parser && y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!) if(len2>0){ if (((AVCodecContext *)sh_audio->context)->channels >= 5) { int samplesize = av_get_bits_per_sample_format(((AVCodecContext *) sh_audio->context)->sample_fmt) / 8; reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT, AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT, ((AVCodecContext *)sh_audio->context)->channels, len2 / samplesize, samplesize); } //len=len2;break; if(len<0) len=len2; else len+=len2; buf+=len2; maxlen -= len2; sh_audio->pts_bytes += len2; } mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2); if (setup_format(sh_audio, sh_audio->context)) break; } return len; }