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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/intreadwrite.h"
#include "libavutil/dict.h"
#include "libavutil/mathematics.h"
#include "avformat.h"
#include "internal.h"
#include "avio_internal.h"
#include "demux.h"
#include "id3v2.h"
#include "id3v1.h"
#include "replaygain.h"
#include "libavcodec/codec_id.h"
#include "libavcodec/mpegaudio.h"
#include "libavcodec/mpegaudiodecheader.h"
#define XING_FLAG_FRAMES 0x01
#define XING_FLAG_SIZE 0x02
#define XING_FLAG_TOC 0x04
#define XING_FLAC_QSCALE 0x08
#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;
enum CheckRet {
CHECK_WRONG_HEADER = -1,
CHECK_SEEK_FAILED = -2,
};
static int check(AVIOContext *pb, int64_t pos, uint32_t *header);
/* mp3 read */
static int mp3_read_probe(const AVProbeData *p)
{
int max_frames, first_frames = 0;
int whole_used = 0;
int frames, ret;
int framesizes, max_framesizes;
uint32_t header;
const uint8_t *buf, *buf0, *buf2, *buf3, *end;
buf0 = p->buf;
end = p->buf + p->buf_size - sizeof(uint32_t);
while (buf0 < end && !*buf0)
buf0++;
max_frames = 0;
max_framesizes = 0;
buf = buf0;
for (; buf < end; buf = buf2+1) {
buf2 = buf;
for (framesizes = frames = 0; buf2 < end; frames++) {
MPADecodeHeader h;
int header_emu = 0;
int available;
header = AV_RB32(buf2);
ret = avpriv_mpegaudio_decode_header(&h, header);
if (ret != 0)
break;
available = FFMIN(h.frame_size, end - buf2);
for (buf3 = buf2 + 4; buf3 < buf2 + available; buf3++) {
uint32_t next_sync = AV_RB32(buf3);
header_emu += (next_sync & MP3_MASK) == (header & MP3_MASK);
}
if (header_emu > 2)
break;
framesizes += h.frame_size;
if (available < h.frame_size) {
frames++;
break;
}
buf2 += h.frame_size;
}
max_frames = FFMAX(max_frames, frames);
max_framesizes = FFMAX(max_framesizes, framesizes);
if (buf == buf0) {
first_frames= frames;
if (buf2 == end + sizeof(uint32_t))
whole_used = 1;
}
}
// keep this in sync with ac3 probe, both need to avoid
// issues with MPEG-files!
if (first_frames>=7) return AVPROBE_SCORE_EXTENSION + 1;
else if (max_frames>200 && p->buf_size < 2*max_framesizes)return AVPROBE_SCORE_EXTENSION;
else if (max_frames>=4 && p->buf_size < 2*max_framesizes) 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 (first_frames > 1 && whole_used) return 5;
else if (max_frames>=1 && p->buf_size < 10*max_framesizes) 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 fast_seek = s->flags & AVFMT_FLAG_FAST_SEEK;
int fill_index = (mp3->usetoc || fast_seek) && duration > 0;
if (!filesize &&
(filesize = avio_size(s->pb)) <= 0) {
av_log(s, AV_LOG_WARNING, "Cannot determine file size, skipping TOC table.\n");
fill_index = 0;
filesize = 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) + 1))
FFStream *const sti = ffstream(st);
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}};
uint64_t fsize = avio_size(s->pb);
int64_t pos = avio_tell(s->pb);
fsize = fsize >= pos ? fsize - pos : 0;
/* 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 (fsize && mp3->header_filesize) {
uint64_t min, delta;
min = FFMIN(fsize, mp3->header_filesize);
delta = FFMAX(fsize, mp3->header_filesize) - min;
if (fsize > mp3->header_filesize && delta > min >> 4) {
mp3->frames = 0;
av_log(s, AV_LOG_WARNING,
"invalid concatenated file detected - using bitrate for duration\n");
} else if (delta > min >> 4) {
av_log(s, AV_LOG_WARNING,
"filesize and duration do not match (growing file?)\n");
}
}
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 & XING_FLAC_QSCALE)
avio_rb32(s->pb);
/* 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')
|| AV_RB32(version) == MKBETAG('L', 'a', 'v', 'c')
) {
mp3->start_pad = v>>12;
mp3-> end_pad = v&4095;
sti->start_skip_samples = mp3->start_pad + 528 + 1;
if (mp3->frames) {
sti->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * (int64_t)spf;
sti->last_discard_sample = mp3->frames * (int64_t)spf;
}
if (!st->start_time)
st->start_time = av_rescale_q(sti->start_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;
int ret;
ffio_init_checksum(s->pb, ff_crcA001_update, 0);
v = avio_rb32(s->pb);
ret = avpriv_mpegaudio_decode_header(&c, v);
if (ret < 0)
return ret;
else if (ret == 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->codecpar->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)
{
FFFormatContext *const si = ffformatcontext(s);
MP3DecContext *mp3 = s->priv_data;
AVStream *st;
FFStream *sti;
int64_t off;
int ret;
int i;
s->metadata = si->id3v2_meta;
si->id3v2_meta = NULL;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
sti = ffstream(st);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = AV_CODEC_ID_MP3;
sti->need_parsing = AVSTREAM_PARSE_FULL_RAW;
st->start_time = 0;
// lcm of all mp3 sample rates
avpriv_set_pts_info(st, 64, 1, 14112000);
ffiocontext(s->pb)->maxsize = -1;
off = avio_tell(s->pb);
if (!av_dict_count(s->metadata))
ff_id3v1_read(s);
if (s->pb->seekable & AVIO_SEEKABLE_NORMAL)
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;
ret = ffio_ensure_seekback(s->pb, 64 * 1024 + MPA_MAX_CODED_FRAME_SIZE + 4);
if (ret < 0)
return ret;
off = avio_tell(s->pb);
for (i = 0; i < 64 * 1024; i++) {
uint32_t header, header2;
int frame_size;
frame_size = check(s->pb, off + i, &header);
if (frame_size > 0) {
ret = check(s->pb, off + i + frame_size, &header2);
if (ret >= 0 && (header & MP3_MASK) == (header2 & MP3_MASK))
break;
} else if (frame_size == CHECK_SEEK_FAILED) {
av_log(s, AV_LOG_ERROR, "Failed to find two consecutive MPEG audio frames.\n");
return AVERROR_INVALIDDATA;
}
}
if (i == 64 * 1024) {
off = avio_seek(s->pb, off, SEEK_SET);
} else {
av_log(s, i > 0 ? AV_LOG_INFO : AV_LOG_VERBOSE, "Skipping %d bytes of junk at %"PRId64".\n", i, off);
off = avio_seek(s->pb, off + i, SEEK_SET);
}
if (off < 0)
return off;
// the seek index is relative to the end of the xing vbr headers
for (int i = 0; i < sti->nb_index_entries; i++)
sti->index_entries[i].pos += off;
/* 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;
return ret;
}
#define SEEK_WINDOW 4096
static int check(AVIOContext *pb, int64_t pos, uint32_t *ret_header)
{
int64_t ret = avio_seek(pb, pos, SEEK_SET);
uint8_t header_buf[4];
unsigned header;
MPADecodeHeader sd;
if (ret < 0)
return CHECK_SEEK_FAILED;
ret = avio_read(pb, &header_buf[0], 4);
/* We should always find four bytes for a valid mpa header. */
if (ret < 4)
return CHECK_SEEK_FAILED;
header = AV_RB32(&header_buf[0]);
if (ff_mpa_check_header(header) < 0)
return CHECK_WRONG_HEADER;
if (avpriv_mpegaudio_decode_header(&sd, header) == 1)
return CHECK_WRONG_HEADER;
if (ret_header)
*ret_header = header;
return sd.frame_size;
}
static int64_t mp3_sync(AVFormatContext *s, int64_t target_pos, int flags)
{
int dir = (flags&AVSEEK_FLAG_BACKWARD) ? -1 : 1;
int64_t best_pos;
int best_score, i, j;
int64_t ret;
avio_seek(s->pb, FFMAX(target_pos - SEEK_WINDOW, 0), SEEK_SET);
ret = avio_seek(s->pb, target_pos, SEEK_SET);
if (ret < 0)
return ret;
#define MIN_VALID 3
best_pos = target_pos;
best_score = 999;
for (i = 0; i < SEEK_WINDOW; i++) {
int64_t pos = target_pos + (dir > 0 ? i - SEEK_WINDOW/4 : -i);
int64_t candidate = -1;
int score = 999;
if (pos < 0)
continue;
for (j = 0; j < MIN_VALID; j++) {
ret = check(s->pb, pos, NULL);
if (ret < 0) {
if (ret == CHECK_WRONG_HEADER) {
break;
} else if (ret == CHECK_SEEK_FAILED) {
av_log(s, AV_LOG_ERROR, "Could not seek to %"PRId64".\n", pos);
return AVERROR(EINVAL);
}
}
if ((target_pos - pos)*dir <= 0 && FFABS(MIN_VALID/2-j) < score) {
candidate = pos;
score = FFABS(MIN_VALID/2-j);
}
pos += ret;
}
if (best_score > score && j == MIN_VALID) {
best_pos = candidate;
best_score = score;
if(score == 0)
break;
}
}
return avio_seek(s->pb, best_pos, SEEK_SET);
}
static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp,
int flags)
{
FFFormatContext *const si = ffformatcontext(s);
MP3DecContext *mp3 = s->priv_data;
AVIndexEntry *ie, ie1;
AVStream *st = s->streams[0];
FFStream *const sti = ffstream(st);
int64_t best_pos;
int fast_seek = s->flags & AVFMT_FLAG_FAST_SEEK;
int64_t filesize = mp3->header_filesize;
if (filesize <= 0) {
int64_t size = avio_size(s->pb);
if (size > 0 && size > si->data_offset)
filesize = size - si->data_offset;
}
if (mp3->xing_toc && (mp3->usetoc || (fast_seek && !mp3->is_cbr))) {
int64_t ret = av_index_search_timestamp(st, timestamp, flags);
// NOTE: The MP3 TOC is not a precise lookup table. Accuracy is worse
// for bigger files.
av_log(s, AV_LOG_WARNING, "Using MP3 TOC to seek; may be imprecise.\n");
if (ret < 0)
return ret;
ie = &sti->index_entries[ret];
} else if (fast_seek && st->duration > 0 && filesize > 0) {
if (!mp3->is_cbr)
av_log(s, AV_LOG_WARNING, "Using scaling to seek VBR MP3; may be imprecise.\n");
ie = &ie1;
timestamp = av_clip64(timestamp, 0, st->duration);
ie->timestamp = timestamp;
ie->pos = av_rescale(timestamp, filesize, st->duration) + si->data_offset;
} else {
return -1; // generic index code
}
best_pos = mp3_sync(s, ie->pos, flags);
if (best_pos < 0)
return best_pos;
if (mp3->is_cbr && ie == &ie1 && mp3->frames && mp3->header_filesize > 0) {
int frame_duration = av_rescale(st->duration, 1, mp3->frames);
ie1.timestamp = frame_duration * av_rescale(best_pos - si->data_offset, mp3->frames, mp3->header_filesize);
}
avpriv_update_cur_dts(s, st, ie->timestamp);
return 0;
}
static const AVOption options[] = {
{ "usetoc", "use table of contents", offsetof(MP3DecContext, usetoc), AV_OPT_TYPE_BOOL, {.i64 = 0}, 0, 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,
};
const FFInputFormat ff_mp3_demuxer = {
.p.name = "mp3",
.p.long_name = NULL_IF_CONFIG_SMALL("MP2/3 (MPEG audio layer 2/3)"),
.p.flags = AVFMT_GENERIC_INDEX,
.p.extensions = "mp2,mp3,m2a,mpa", /* XXX: use probe */
.p.priv_class = &demuxer_class,
.read_probe = mp3_read_probe,
.read_header = mp3_read_header,
.read_packet = mp3_read_packet,
.read_seek = mp3_seek,
.priv_data_size = sizeof(MP3DecContext),
};