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https://gitee.com/openharmony/third_party_ffmpeg
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15c6be8c7d
* qatar/master: tiertexseq: set correct block_align for audio tiertexseq: set audio stream start time to 0 voc/avs: Do not change the sample rate mid-stream. segafilm: use the sample rate as the time base for audio streams ea: fix audio pts psx-str: fix audio pts vqf: set packet duration tta demuxer: set packet duration mpegaudio_parser: do not ignore information from the first parsed frame mpegaudio_parser: be less picky about the start position thp: set audio packet durations avcodec: add a Vorbis parser to get packet duration vorbisdec: read the previous window flag for long windows lavc: free the output packet when encoding failed or produced no output. lavc: preserve avpkt->destruct in ff_alloc_packet(). lavc: clarify the meaning of AVCodecContext.frame_number. mpegts: Pad the packet buffer in handle_packet(). mpegts: Do not call read_sl_header() when no bytes remain in the buffer. Conflicts: libavcodec/mpegaudio_parser.c libavcodec/version.h libavformat/mpegts.c tests/ref/fate/pva-demux 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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s1->duration = 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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