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FFmpeg/libavformat/mp3dec.c
Michael Niedermayer cdc166acd8 Merge commit '32d05934abc7427bb90380a4c1ab20a15fd7d821' 
* commit '32d05934abc7427bb90380a4c1ab20a15fd7d821':
  mp3dec: decode more data from Info header

Conflicts:
	libavformat/mp3dec.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2014-04-18 00:38:07 +02:00

478 lines
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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/crc.h"
#include "libavutil/dict.h"
#include "libavutil/mathematics.h"
#include "avformat.h"
#include "internal.h"
#include "avio_internal.h"
#include "id3v2.h"
#include "id3v1.h"
#include "replaygain.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;
unsigned frames; /* Total number of frames in file */
unsigned header_filesize; /* Total number of bytes in the stream */
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;
if(ff_mpa_check_header(AV_RB32(buf2)))
continue;
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 p->buf_size < PROBE_BUF_MAX ? AVPROBE_SCORE_EXTENSION / 4 : AVPROBE_SCORE_EXTENSION - 2;
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;
}
static void mp3_parse_info_tag(AVFormatContext *s, AVStream *st,
MPADecodeHeader *c, uint32_t spf)
{
#define LAST_BITS(k, n) ((k) & ((1 << (n)) - 1))
#define MIDDLE_BITS(k, m, n) LAST_BITS((k) >> (m), ((n) - (m)))
uint16_t crc;
uint32_t v;
char version[10];
uint32_t peak = 0;
int32_t r_gain = INT32_MIN, a_gain = INT32_MIN;
MP3DecContext *mp3 = s->priv_data;
static const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}};
/* Check for Xing / Info tag */
avio_skip(s->pb, xing_offtbl[c->lsf == 1][c->nb_channels == 1]);
v = avio_rb32(s->pb);
mp3->is_cbr = v == MKBETAG('I', 'n', 'f', 'o');
if (v != MKBETAG('X', 'i', 'n', 'g') && !mp3->is_cbr)
return;
v = avio_rb32(s->pb);
if (v & XING_FLAG_FRAMES)
mp3->frames = avio_rb32(s->pb);
if (v & XING_FLAG_SIZE)
mp3->header_filesize = avio_rb32(s->pb);
if (v & XING_FLAG_TOC)
read_xing_toc(s, mp3->header_filesize, av_rescale_q(mp3->frames,
(AVRational){spf, c->sample_rate},
st->time_base));
/* VBR quality */
if(v & 8)
avio_skip(s->pb, 4);
/* Encoder short version string */
memset(version, 0, sizeof(version));
avio_read(s->pb, version, 9);
/* Info Tag revision + VBR method */
avio_r8(s->pb);
/* Lowpass filter value */
avio_r8(s->pb);
/* ReplayGain peak */
v = avio_rb32(s->pb);
peak = av_rescale(v, 100000, 1 << 23);
/* Radio ReplayGain */
v = avio_rb16(s->pb);
if (MIDDLE_BITS(v, 13, 15) == 1) {
r_gain = MIDDLE_BITS(v, 0, 8) * 10000;
if (v & (1 << 9))
r_gain *= -1;
}
/* Audiophile ReplayGain */
v = avio_rb16(s->pb);
if (MIDDLE_BITS(v, 13, 15) == 2) {
a_gain = MIDDLE_BITS(v, 0, 8) * 10000;
if (v & (1 << 9))
a_gain *= -1;
}
/* Encoding flags + ATH Type */
avio_r8(s->pb);
/* if ABR {specified bitrate} else {minimal bitrate} */
avio_r8(s->pb);
/* Encoder delays */
v= avio_rb24(s->pb);
if(AV_RB32(version) == MKBETAG('L', 'A', 'M', 'E')
|| AV_RB32(version) == MKBETAG('L', 'a', 'v', 'f')) {
mp3->start_pad = v>>12;
mp3-> end_pad = v&4095;
st->skip_samples = mp3->start_pad + 528 + 1;
if (!st->start_time)
st->start_time = av_rescale_q(st->skip_samples,
(AVRational){1, c->sample_rate},
st->time_base);
av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3-> end_pad);
}
/* Misc */
avio_r8(s->pb);
/* MP3 gain */
avio_r8(s->pb);
/* Preset and surround info */
avio_rb16(s->pb);
/* Music length */
avio_rb32(s->pb);
/* Music CRC */
avio_rb16(s->pb);
/* Info Tag CRC */
crc = ffio_get_checksum(s->pb);
v = avio_rb16(s->pb);
if (v == crc) {
ff_replaygain_export_raw(st, r_gain, peak, a_gain, 0);
av_dict_set(&st->metadata, "encoder", version, 0);
}
}
static void mp3_parse_vbri_tag(AVFormatContext *s, AVStream *st, int64_t base)
{
uint32_t v;
MP3DecContext *mp3 = s->priv_data;
/* 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);
mp3->header_filesize = avio_rb32(s->pb);
mp3->frames = avio_rb32(s->pb);
}
}
}
/**
* 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)
{
uint32_t v, spf;
MPADecodeHeader c;
int vbrtag_size = 0;
MP3DecContext *mp3 = s->priv_data;
ffio_init_checksum(s->pb, ff_crcA001_update, 0);
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 */
mp3->frames = 0;
mp3->header_filesize = 0;
mp3_parse_info_tag(s, st, &c, spf);
mp3_parse_vbri_tag(s, st, base);
if (!mp3->frames && !mp3->header_filesize)
return -1;
/* Skip the vbr tag frame */
avio_seek(s->pb, base + vbrtag_size, SEEK_SET);
if (mp3->frames)
st->duration = av_rescale_q(mp3->frames, (AVRational){spf, c.sample_rate},
st->time_base);
if (mp3->header_filesize && mp3->frames && !mp3->is_cbr)
st->codec->bit_rate = av_rescale(mp3->header_filesize, 8 * c.sample_rate, mp3->frames * (int64_t)spf);
return 0;
}
static int mp3_read_header(AVFormatContext *s)
{
MP3DecContext *mp3 = s->priv_data;
AVStream *st;
int64_t off;
int ret;
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);
ret = ff_replaygain_export(st, s->metadata);
if (ret < 0)
return ret;
/* 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;
int dir = (flags&AVSEEK_FLAG_BACKWARD) ? -1 : 1;
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;
}
if (dir < 0)
avio_seek(s->pb, FFMAX(ie->pos - 4096, 0), SEEK_SET);
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*dir;
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*dir, 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,mpa", /* XXX: use probe */
.priv_class = &demuxer_class,
};
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