ffmpeg/libavformat/mp3dec.c
Michael Niedermayer 0844630e6b avformat/mp3dec: Add usetoc option to allow dlsabling the use of the xing TOC
The toc is inexact and not using it can thus make sense.
Using it is faster though, thus the opposite can similarly makes sense

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2013-07-06 05:43:48 +02:00

350 lines
9.9 KiB
C

/*
* MP3 demuxer
* Copyright (c) 2003 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "libavutil/avstring.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/dict.h"
#include "libavutil/mathematics.h"
#include "avformat.h"
#include "internal.h"
#include "id3v2.h"
#include "id3v1.h"
#include "libavcodec/mpegaudiodecheader.h"
#define XING_FLAG_FRAMES 0x01
#define XING_FLAG_SIZE 0x02
#define XING_FLAG_TOC 0x04
#define XING_TOC_COUNT 100
typedef struct {
AVClass *class;
int64_t filesize;
int xing_toc;
int start_pad;
int end_pad;
int usetoc;
} MP3DecContext;
/* mp3 read */
static int mp3_read_probe(AVProbeData *p)
{
int max_frames, first_frames = 0;
int fsize, frames, sample_rate;
uint32_t header;
const uint8_t *buf, *buf0, *buf2, *end;
AVCodecContext avctx;
buf0 = p->buf;
end = p->buf + p->buf_size - sizeof(uint32_t);
while(buf0 < end && !*buf0)
buf0++;
max_frames = 0;
buf = buf0;
for(; buf < end; buf= buf2+1) {
buf2 = buf;
for(frames = 0; buf2 < end; frames++) {
header = AV_RB32(buf2);
fsize = avpriv_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate);
if(fsize < 0)
break;
buf2 += fsize;
}
max_frames = FFMAX(max_frames, frames);
if(buf == buf0)
first_frames= frames;
}
// keep this in sync with ac3 probe, both need to avoid
// issues with MPEG-files!
if (first_frames>=4) return AVPROBE_SCORE_EXTENSION + 1;
else if(max_frames>200)return AVPROBE_SCORE_EXTENSION;
else if(max_frames>=4) return AVPROBE_SCORE_EXTENSION / 2;
else if(ff_id3v2_match(buf0, ID3v2_DEFAULT_MAGIC) && 2*ff_id3v2_tag_len(buf0) >= p->buf_size)
return AVPROBE_SCORE_EXTENSION / 4;
else if(max_frames>=1) return 1;
else return 0;
//mpegps_mp3_unrecognized_format.mpg has max_frames=3
}
static void read_xing_toc(AVFormatContext *s, int64_t filesize, int64_t duration)
{
int i;
MP3DecContext *mp3 = s->priv_data;
if (!mp3->usetoc)
return;
if (!filesize &&
!(filesize = avio_size(s->pb))) {
av_log(s, AV_LOG_WARNING, "Cannot determine file size, skipping TOC table.\n");
return;
}
for (i = 0; i < XING_TOC_COUNT; i++) {
uint8_t b = avio_r8(s->pb);
av_add_index_entry(s->streams[0],
av_rescale(b, filesize, 256),
av_rescale(i, duration, XING_TOC_COUNT),
0, 0, AVINDEX_KEYFRAME);
}
mp3->xing_toc = 1;
}
/**
* Try to find Xing/Info/VBRI tags and compute duration from info therein
*/
static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
{
MP3DecContext *mp3 = s->priv_data;
uint32_t v, spf;
unsigned frames = 0; /* Total number of frames in file */
unsigned size = 0; /* Total number of bytes in the stream */
const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}};
MPADecodeHeader c;
int vbrtag_size = 0;
int is_cbr;
v = avio_rb32(s->pb);
if(ff_mpa_check_header(v) < 0)
return -1;
if (avpriv_mpegaudio_decode_header(&c, v) == 0)
vbrtag_size = c.frame_size;
if(c.layer != 3)
return -1;
spf = c.lsf ? 576 : 1152; /* Samples per frame, layer 3 */
/* Check for Xing / Info tag */
avio_skip(s->pb, xing_offtbl[c.lsf == 1][c.nb_channels == 1]);
v = avio_rb32(s->pb);
is_cbr = v == MKBETAG('I', 'n', 'f', 'o');
if (v == MKBETAG('X', 'i', 'n', 'g') || is_cbr) {
v = avio_rb32(s->pb);
if(v & XING_FLAG_FRAMES)
frames = avio_rb32(s->pb);
if(v & XING_FLAG_SIZE)
size = avio_rb32(s->pb);
if (v & XING_FLAG_TOC && frames)
read_xing_toc(s, size, av_rescale_q(frames, (AVRational){spf, c.sample_rate},
st->time_base));
if(v & 8)
avio_skip(s->pb, 4);
v = avio_rb32(s->pb);
if(v == MKBETAG('L', 'A', 'M', 'E') || v == MKBETAG('L', 'a', 'v', 'f')) {
avio_skip(s->pb, 21-4);
v= avio_rb24(s->pb);
mp3->start_pad = v>>12;
mp3-> end_pad = v&4095;
st->skip_samples = mp3->start_pad + 528 + 1;
av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3-> end_pad);
}
}
/* Check for VBRI tag (always 32 bytes after end of mpegaudio header) */
avio_seek(s->pb, base + 4 + 32, SEEK_SET);
v = avio_rb32(s->pb);
if(v == MKBETAG('V', 'B', 'R', 'I')) {
/* Check tag version */
if(avio_rb16(s->pb) == 1) {
/* skip delay and quality */
avio_skip(s->pb, 4);
size = avio_rb32(s->pb);
frames = avio_rb32(s->pb);
}
}
if(!frames && !size)
return -1;
/* Skip the vbr tag frame */
avio_seek(s->pb, base + vbrtag_size, SEEK_SET);
if(frames)
st->duration = av_rescale_q(frames, (AVRational){spf, c.sample_rate},
st->time_base);
if (size && frames && !is_cbr)
st->codec->bit_rate = av_rescale(size, 8 * c.sample_rate, frames * (int64_t)spf);
return 0;
}
static int mp3_read_header(AVFormatContext *s)
{
MP3DecContext *mp3 = s->priv_data;
AVStream *st;
int64_t off;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = AV_CODEC_ID_MP3;
st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
st->start_time = 0;
// lcm of all mp3 sample rates
avpriv_set_pts_info(st, 64, 1, 14112000);
s->pb->maxsize = -1;
off = avio_tell(s->pb);
if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX))
ff_id3v1_read(s);
if(s->pb->seekable)
mp3->filesize = avio_size(s->pb);
if (mp3_parse_vbr_tags(s, st, off) < 0)
avio_seek(s->pb, off, SEEK_SET);
/* the parameters will be extracted from the compressed bitstream */
return 0;
}
#define MP3_PACKET_SIZE 1024
static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
{
MP3DecContext *mp3 = s->priv_data;
int ret, size;
int64_t pos;
size= MP3_PACKET_SIZE;
pos = avio_tell(s->pb);
if(mp3->filesize > ID3v1_TAG_SIZE && pos < mp3->filesize)
size= FFMIN(size, mp3->filesize - pos);
ret= av_get_packet(s->pb, pkt, size);
if (ret <= 0) {
if(ret<0)
return ret;
return AVERROR_EOF;
}
pkt->flags &= ~AV_PKT_FLAG_CORRUPT;
pkt->stream_index = 0;
if (ret >= ID3v1_TAG_SIZE &&
memcmp(&pkt->data[ret - ID3v1_TAG_SIZE], "TAG", 3) == 0)
ret -= ID3v1_TAG_SIZE;
/* note: we need to modify the packet size here to handle the last
packet */
pkt->size = ret;
return ret;
}
static int check(AVFormatContext *s, int64_t pos)
{
int64_t ret = avio_seek(s->pb, pos, SEEK_SET);
unsigned header;
MPADecodeHeader sd;
if (ret < 0)
return ret;
header = avio_rb32(s->pb);
if (ff_mpa_check_header(header) < 0)
return -1;
if (avpriv_mpegaudio_decode_header(&sd, header) == 1)
return -1;
return sd.frame_size;
}
static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,
int flags)
{
MP3DecContext *mp3 = s->priv_data;
AVIndexEntry *ie;
AVStream *st = s->streams[0];
int64_t ret = av_index_search_timestamp(st, timestamp, flags);
int i, j;
if (!mp3->xing_toc) {
st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
return -1;
}
if (ret < 0)
return ret;
ie = &st->index_entries[ret];
ret = avio_seek(s->pb, ie->pos, SEEK_SET);
if (ret < 0)
return ret;
#define MIN_VALID 3
for(i=0; i<4096; i++) {
int64_t pos = ie->pos + i;
for(j=0; j<MIN_VALID; j++) {
ret = check(s, pos);
if(ret < 0)
break;
pos += ret;
}
if(j==MIN_VALID)
break;
}
if(j!=MIN_VALID)
i=0;
ret = avio_seek(s->pb, ie->pos + i, SEEK_SET);
if (ret < 0)
return ret;
ff_update_cur_dts(s, st, ie->timestamp);
st->skip_samples = ie->timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
return 0;
}
static const AVOption options[] = {
{ "usetoc", "use table of contents", offsetof(MP3DecContext, usetoc), AV_OPT_TYPE_INT, {.i64 = -1}, -1, 1, AV_OPT_FLAG_DECODING_PARAM},
{ NULL },
};
static const AVClass demuxer_class = {
.class_name = "mp3",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
.category = AV_CLASS_CATEGORY_DEMUXER,
};
AVInputFormat ff_mp3_demuxer = {
.name = "mp3",
.long_name = NULL_IF_CONFIG_SMALL("MP2/3 (MPEG audio layer 2/3)"),
.read_probe = mp3_read_probe,
.read_header = mp3_read_header,
.read_packet = mp3_read_packet,
.read_seek = mp3_seek,
.priv_data_size = sizeof(MP3DecContext),
.flags = AVFMT_GENERIC_INDEX,
.extensions = "mp2,mp3,m2a", /* XXX: use probe */
.priv_class = &demuxer_class,
};