mirror of
https://github.com/FFmpeg/FFmpeg.git
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b6267901c4
The libav code depends on mp3 startcodes only occuring at the start of frames. But there is nothing in mp3 that prevents them occuring elsewhere by chance. Thus the code would fail randomly, the new code searches for 3 consecutive valid frames. If this turns out to be still insufficient the number can be raised further, or additional checks added. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
329 lines
9.3 KiB
C
329 lines
9.3 KiB
C
/*
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* MP3 demuxer
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* Copyright (c) 2003 Fabrice Bellard
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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 "libavutil/avstring.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/dict.h"
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#include "libavutil/mathematics.h"
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#include "avformat.h"
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#include "internal.h"
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#include "id3v2.h"
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#include "id3v1.h"
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#include "libavcodec/mpegaudiodecheader.h"
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#define XING_FLAG_FRAMES 0x01
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#define XING_FLAG_SIZE 0x02
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#define XING_FLAG_TOC 0x04
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#define XING_TOC_COUNT 100
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typedef struct {
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int64_t filesize;
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int xing_toc;
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int start_pad;
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int end_pad;
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} MP3Context;
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/* mp3 read */
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static int mp3_read_probe(AVProbeData *p)
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{
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int max_frames, first_frames = 0;
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int fsize, frames, sample_rate;
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uint32_t header;
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uint8_t *buf, *buf0, *buf2, *end;
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AVCodecContext avctx;
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buf0 = p->buf;
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end = p->buf + p->buf_size - sizeof(uint32_t);
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while(buf0 < end && !*buf0)
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buf0++;
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max_frames = 0;
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buf = buf0;
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for(; buf < end; buf= buf2+1) {
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buf2 = buf;
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for(frames = 0; buf2 < end; frames++) {
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header = AV_RB32(buf2);
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fsize = avpriv_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate);
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if(fsize < 0)
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break;
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buf2 += fsize;
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}
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max_frames = FFMAX(max_frames, frames);
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if(buf == buf0)
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first_frames= frames;
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}
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// keep this in sync with ac3 probe, both need to avoid
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// issues with MPEG-files!
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if (first_frames>=4) return AVPROBE_SCORE_MAX/2+1;
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else if(max_frames>200)return AVPROBE_SCORE_MAX/2;
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else if(max_frames>=4) return AVPROBE_SCORE_MAX/4;
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else if(ff_id3v2_match(buf0, ID3v2_DEFAULT_MAGIC) && 2*ff_id3v2_tag_len(buf0) >= p->buf_size)
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return AVPROBE_SCORE_MAX/8;
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else if(max_frames>=1) return 1;
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else return 0;
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//mpegps_mp3_unrecognized_format.mpg has max_frames=3
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}
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static void read_xing_toc(AVFormatContext *s, int64_t filesize, int64_t duration)
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{
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int i;
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MP3Context *mp3 = s->priv_data;
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if (!filesize &&
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!(filesize = avio_size(s->pb))) {
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av_log(s, AV_LOG_WARNING, "Cannot determine file size, skipping TOC table.\n");
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return;
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}
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for (i = 0; i < XING_TOC_COUNT; i++) {
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uint8_t b = avio_r8(s->pb);
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av_add_index_entry(s->streams[0],
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av_rescale(b, filesize, 256),
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av_rescale(i, duration, XING_TOC_COUNT),
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0, 0, AVINDEX_KEYFRAME);
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}
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mp3->xing_toc = 1;
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}
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/**
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* Try to find Xing/Info/VBRI tags and compute duration from info therein
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*/
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static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
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{
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MP3Context *mp3 = s->priv_data;
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uint32_t v, spf;
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unsigned frames = 0; /* Total number of frames in file */
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unsigned size = 0; /* Total number of bytes in the stream */
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const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}};
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MPADecodeHeader c;
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int vbrtag_size = 0;
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v = avio_rb32(s->pb);
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if(ff_mpa_check_header(v) < 0)
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return -1;
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if (avpriv_mpegaudio_decode_header(&c, v) == 0)
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vbrtag_size = c.frame_size;
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if(c.layer != 3)
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return -1;
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spf = c.lsf ? 576 : 1152; /* Samples per frame, layer 3 */
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/* Check for Xing / Info tag */
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avio_skip(s->pb, xing_offtbl[c.lsf == 1][c.nb_channels == 1]);
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v = avio_rb32(s->pb);
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if(v == MKBETAG('X', 'i', 'n', 'g') || v == MKBETAG('I', 'n', 'f', 'o')) {
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v = avio_rb32(s->pb);
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if(v & XING_FLAG_FRAMES)
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frames = avio_rb32(s->pb);
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if(v & XING_FLAG_SIZE)
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size = avio_rb32(s->pb);
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if (v & XING_FLAG_TOC && frames)
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read_xing_toc(s, size, av_rescale_q(frames, (AVRational){spf, c.sample_rate},
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st->time_base));
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if(v & 8)
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avio_skip(s->pb, 4);
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v = avio_rb32(s->pb);
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if(v == MKBETAG('L', 'A', 'M', 'E') || v == MKBETAG('L', 'a', 'v', 'f')) {
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avio_skip(s->pb, 21-4);
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v= avio_rb24(s->pb);
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mp3->start_pad = v>>12;
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mp3-> end_pad = v&4095;
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st->skip_samples = mp3->start_pad + 528 + 1;
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av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3-> end_pad);
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}
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}
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/* Check for VBRI tag (always 32 bytes after end of mpegaudio header) */
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avio_seek(s->pb, base + 4 + 32, SEEK_SET);
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v = avio_rb32(s->pb);
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if(v == MKBETAG('V', 'B', 'R', 'I')) {
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/* Check tag version */
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if(avio_rb16(s->pb) == 1) {
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/* skip delay and quality */
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avio_skip(s->pb, 4);
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size = avio_rb32(s->pb);
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frames = avio_rb32(s->pb);
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}
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}
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if(!frames && !size)
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return -1;
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/* Skip the vbr tag frame */
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avio_seek(s->pb, base + vbrtag_size, SEEK_SET);
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if(frames)
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st->duration = av_rescale_q(frames, (AVRational){spf, c.sample_rate},
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st->time_base);
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if(size && frames)
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st->codec->bit_rate = av_rescale(size, 8 * c.sample_rate, frames * (int64_t)spf);
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return 0;
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}
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static int mp3_read_header(AVFormatContext *s)
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{
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MP3Context *mp3 = s->priv_data;
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AVStream *st;
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int64_t off;
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = AV_CODEC_ID_MP3;
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st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
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st->start_time = 0;
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// lcm of all mp3 sample rates
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avpriv_set_pts_info(st, 64, 1, 14112000);
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s->pb->maxsize = -1;
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off = avio_tell(s->pb);
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if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX))
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ff_id3v1_read(s);
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if(s->pb->seekable)
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mp3->filesize = avio_size(s->pb);
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if (mp3_parse_vbr_tags(s, st, off) < 0)
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avio_seek(s->pb, off, SEEK_SET);
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/* the parameters will be extracted from the compressed bitstream */
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return 0;
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}
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#define MP3_PACKET_SIZE 1024
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static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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MP3Context *mp3 = s->priv_data;
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int ret, size;
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int64_t pos;
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size= MP3_PACKET_SIZE;
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pos = avio_tell(s->pb);
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if(mp3->filesize > ID3v1_TAG_SIZE && pos < mp3->filesize)
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size= FFMIN(size, mp3->filesize - pos);
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ret= av_get_packet(s->pb, pkt, size);
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if (ret <= 0) {
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if(ret<0)
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return ret;
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return AVERROR_EOF;
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}
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pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
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pkt->stream_index = 0;
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if (ret >= ID3v1_TAG_SIZE &&
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memcmp(&pkt->data[ret - ID3v1_TAG_SIZE], "TAG", 3) == 0)
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ret -= ID3v1_TAG_SIZE;
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/* note: we need to modify the packet size here to handle the last
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packet */
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pkt->size = ret;
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return ret;
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}
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static int check(AVFormatContext *s, int64_t pos)
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{
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int64_t ret = avio_seek(s->pb, pos, SEEK_SET);
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unsigned header;
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MPADecodeHeader sd;
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if (ret < 0)
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return ret;
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header = avio_rb32(s->pb);
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if (ff_mpa_check_header(header) < 0)
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return -1;
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if (avpriv_mpegaudio_decode_header(&sd, header) == 1)
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return -1;
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return sd.frame_size;
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}
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static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,
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int flags)
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{
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MP3Context *mp3 = s->priv_data;
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AVIndexEntry *ie;
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AVStream *st = s->streams[0];
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int64_t ret = av_index_search_timestamp(st, timestamp, flags);
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uint32_t header = 0;
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int i, j;
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if (!mp3->xing_toc) {
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st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
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return -1;
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}
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if (ret < 0)
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return ret;
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ie = &st->index_entries[ret];
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ret = avio_seek(s->pb, ie->pos, SEEK_SET);
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if (ret < 0)
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return ret;
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#define MIN_VALID 3
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for(i=0; i<4096; i++) {
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int64_t pos = ie->pos + i;
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for(j=0; j<MIN_VALID; j++) {
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ret = check(s, pos);
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if(ret < 0)
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break;
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pos += ret;
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}
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if(j==MIN_VALID)
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break;
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}
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if(j!=MIN_VALID)
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i=0;
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ret = avio_seek(s->pb, ie->pos + i, SEEK_SET);
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if (ret < 0)
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return ret;
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ff_update_cur_dts(s, st, ie->timestamp);
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st->skip_samples = ie->timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
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return 0;
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}
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AVInputFormat ff_mp3_demuxer = {
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.name = "mp3",
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.long_name = NULL_IF_CONFIG_SMALL("MP2/3 (MPEG audio layer 2/3)"),
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.priv_data_size = sizeof(MP3Context),
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.read_probe = mp3_read_probe,
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.read_header = mp3_read_header,
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.read_packet = mp3_read_packet,
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.read_seek = mp3_seek,
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.flags = AVFMT_GENERIC_INDEX,
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.extensions = "mp2,mp3,m2a", /* XXX: use probe */
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};
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