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https://github.com/xenia-project/FFmpeg.git
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988f585fcb
* qatar/master: (44 commits) replacement Indeo 3 decoder gsm demuxer: do not allocate packet twice. flvenc: use first packet delay as global delay. ac3enc: doxygen update. imc: return error codes instead of 0 for error conditions. imc: return meaningful error codes instead of -1 imc: do not set channel layout for stereo imc: validate channel count imc: check for ff_fft_init() failure imc: check output buffer size before decoding imc: use DSPContext.bswap16_buf() to byte-swap packet data rtsp: add allowed_media_types option libgsm: add flush function to reset the decoder state when seeking libgsm: simplify decoding by using a loop gsm: log error message when packet is too small libgsmdec: do not needlessly set *data_size to 0 gsmdec: do not needlessly set *data_size to 0 gsmdec: add flush function to reset the decoder state when seeking libgsmdec: check output buffer size before decoding gsmdec: log error message when output buffer is too small. ... Conflicts: Changelog ffplay.c libavcodec/indeo3.c libavcodec/mjpeg_parser.c libavcodec/vp3.c libavformat/cutils.c libavformat/id3v2.c libavutil/parseutils.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
109 lines
3.3 KiB
C
109 lines
3.3 KiB
C
/*
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* MPEG Audio parser
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* Copyright (c) 2003 Fabrice Bellard
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* Copyright (c) 2003 Michael Niedermayer
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "parser.h"
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#include "mpegaudiodecheader.h"
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typedef struct MpegAudioParseContext {
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ParseContext pc;
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int frame_size;
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uint32_t header;
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int header_count;
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} MpegAudioParseContext;
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#define MPA_HEADER_SIZE 4
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/* header + layer + bitrate + freq + lsf/mpeg25 */
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#define SAME_HEADER_MASK \
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(0xffe00000 | (3 << 17) | (3 << 10) | (3 << 19))
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static int mpegaudio_parse(AVCodecParserContext *s1,
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AVCodecContext *avctx,
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const uint8_t **poutbuf, int *poutbuf_size,
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const uint8_t *buf, int buf_size)
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{
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MpegAudioParseContext *s = s1->priv_data;
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ParseContext *pc = &s->pc;
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uint32_t state= pc->state;
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int i;
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int next= END_NOT_FOUND;
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for(i=0; i<buf_size; ){
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if(s->frame_size){
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int inc= FFMIN(buf_size - i, s->frame_size);
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i += inc;
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s->frame_size -= inc;
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if(!s->frame_size){
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next= i;
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break;
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}
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}else{
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while(i<buf_size){
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int ret, sr, channels, bit_rate, frame_size;
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state= (state<<8) + buf[i++];
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ret = avpriv_mpa_decode_header(avctx, state, &sr, &channels, &frame_size, &bit_rate);
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if (ret < 4) {
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if(i > 4)
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s->header_count= -2;
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} else {
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if((state&SAME_HEADER_MASK) != (s->header&SAME_HEADER_MASK) && s->header)
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s->header_count= -3;
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s->header= state;
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s->header_count++;
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s->frame_size = ret-4;
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if(s->header_count > 1){
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avctx->sample_rate= sr;
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avctx->channels = channels;
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avctx->frame_size = frame_size;
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avctx->bit_rate = bit_rate;
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}
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break;
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}
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}
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}
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}
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pc->state= state;
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if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
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*poutbuf = NULL;
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*poutbuf_size = 0;
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return buf_size;
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}
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*poutbuf = buf;
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*poutbuf_size = buf_size;
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return next;
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}
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AVCodecParser ff_mpegaudio_parser = {
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.codec_ids = { CODEC_ID_MP1, CODEC_ID_MP2, CODEC_ID_MP3 },
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.priv_data_size = sizeof(MpegAudioParseContext),
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.parser_parse = mpegaudio_parse,
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.parser_close = ff_parse_close,
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};
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