forked from KolibriOS/kolibrios
367 lines
11 KiB
C
367 lines
11 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/opt.h"
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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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AVClass *class;
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int64_t filesize;
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int64_t header_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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int usetoc;
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int is_cbr;
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} MP3DecContext;
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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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const 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_EXTENSION + 1;
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else if(max_frames>200)return AVPROBE_SCORE_EXTENSION;
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else if(max_frames>=4) return AVPROBE_SCORE_EXTENSION / 2;
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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 p->buf_size < PROBE_BUF_MAX ? AVPROBE_SCORE_EXTENSION / 4 : AVPROBE_SCORE_EXTENSION - 2;
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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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MP3DecContext *mp3 = s->priv_data;
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int fill_index = mp3->usetoc && duration > 0;
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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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fill_index = 0;
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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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if (fill_index)
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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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if (fill_index)
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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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MP3DecContext *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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static 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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int is_cbr;
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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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is_cbr = v == MKBETAG('I', 'n', 'f', 'o');
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if (v == MKBETAG('X', 'i', 'n', 'g') || is_cbr) {
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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)
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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 && !is_cbr)
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st->codec->bit_rate = av_rescale(size, 8 * c.sample_rate, frames * (int64_t)spf);
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mp3->is_cbr = is_cbr;
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mp3->header_filesize = size;
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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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MP3DecContext *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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MP3DecContext *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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MP3DecContext *mp3 = s->priv_data;
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AVIndexEntry *ie, ie1;
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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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int i, j;
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int dir = (flags&AVSEEK_FLAG_BACKWARD) ? -1 : 1;
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if (mp3->is_cbr && st->duration > 0 && mp3->header_filesize > s->data_offset) {
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int64_t filesize = avio_size(s->pb);
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int64_t duration;
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if (filesize <= s->data_offset)
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filesize = mp3->header_filesize;
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filesize -= s->data_offset;
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duration = av_rescale(st->duration, filesize, mp3->header_filesize - s->data_offset);
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ie = &ie1;
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timestamp = av_clip64(timestamp, 0, duration);
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ie->timestamp = timestamp;
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ie->pos = av_rescale(timestamp, filesize, duration) + s->data_offset;
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} else if (mp3->xing_toc) {
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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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} else {
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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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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*dir;
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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*dir, 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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static const AVOption options[] = {
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{ "usetoc", "use table of contents", offsetof(MP3DecContext, usetoc), AV_OPT_TYPE_INT, {.i64 = -1}, -1, 1, AV_OPT_FLAG_DECODING_PARAM},
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{ NULL },
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};
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static const AVClass demuxer_class = {
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.class_name = "mp3",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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.category = AV_CLASS_CATEGORY_DEMUXER,
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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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.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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.priv_data_size = sizeof(MP3DecContext),
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.flags = AVFMT_GENERIC_INDEX,
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.extensions = "mp2,mp3,m2a,mpa", /* XXX: use probe */
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.priv_class = &demuxer_class,
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};
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