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ffmpeg/libavformat/mp3dec.c

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/*
* 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;
int64_t header_filesize;
int xing_toc;
int start_pad;
int end_pad;
int usetoc;
int is_cbr;
} 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;
int fill_index = mp3->usetoc && duration > 0;
if (!filesize &&
!(filesize = avio_size(s->pb))) {
av_log(s, AV_LOG_WARNING, "Cannot determine file size, skipping TOC table.\n");
fill_index = 0;
}
for (i = 0; i < XING_TOC_COUNT; i++) {
uint8_t b = avio_r8(s->pb);
if (fill_index)
av_add_index_entry(s->streams[0],
av_rescale(b, filesize, 256),
av_rescale(i, duration, XING_TOC_COUNT),
0, 0, AVINDEX_KEYFRAME);
}
if (fill_index)
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)
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);
mp3->is_cbr = is_cbr;
mp3->header_filesize = size;
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, ie1;
AVStream *st = s->streams[0];
int64_t ret = av_index_search_timestamp(st, timestamp, flags);
int i, j;
if (mp3->is_cbr && st->duration > 0 && mp3->header_filesize > s->data_offset) {
int64_t filesize = avio_size(s->pb);
int64_t duration;
if (filesize <= s->data_offset)
filesize = mp3->header_filesize;
filesize -= s->data_offset;
duration = av_rescale(st->duration, filesize, mp3->header_filesize - s->data_offset);
ie = &ie1;
timestamp = av_clip64(timestamp, 0, duration);
ie->timestamp = timestamp;
ie->pos = av_rescale(timestamp, filesize, duration) + s->data_offset;
} else if (mp3->xing_toc) {
if (ret < 0)
return ret;
ie = &st->index_entries[ret];
} else {
st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0;
return -1;
}
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,
};
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