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ffmpeg/libavformat/mp3dec.c
Dale Curtis 460132c998 lavf/mp3dec: don't adjust start time; packets are not adjusted. 
7546ac2fee made it so that the start_time for mp3 files is
adjusted for skip_samples. However, this appears incorrect because
subsequent packet timestamps are not adjusted and skip_samples are
applied by deleting data from a packet without changing the timestamp.

E.g., we are told the start_time is ~25ms and we get a packet with a
timestamp of 0 that has had the skip_samples discarded from it. As such
rendering engines may incorrectly discard everything prior to the
25ms thinking that is where playback should officially start. Since the
samples were deleted without adjusting timestamps though, the true
start_time is still 0.

Other formats like MP4 with edit lists will adjust both the start
time and the timestamps of subsequent packets to avoid this issue.

Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2020-05-27 10:22:17 +02:00

617 lines
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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/avcodec.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))) {
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) + 1))
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);
fsize = fsize >= avio_tell(s->pb) ? fsize - avio_tell(s->pb) : 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;
st->start_skip_samples = mp3->start_pad + 528 + 1;
if (mp3->frames) {
st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * (int64_t)spf;
st->last_discard_sample = mp3->frames * (int64_t)spf;
}
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)
{
MP3DecContext *mp3 = s->priv_data;
AVStream *st;
int64_t off;
int ret;
int i;
s->metadata = s->internal->id3v2_meta;
s->internal->id3v2_meta = NULL;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->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 & 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;
off = avio_tell(s->pb);
for (i = 0; i < 64 * 1024; i++) {
uint32_t header, header2;
int frame_size;
if (!(i&1023))
ffio_ensure_seekback(s->pb, i + 1024 + 4);
frame_size = check(s->pb, off + i, &header);
if (frame_size > 0) {
ret = avio_seek(s->pb, off, SEEK_SET);
if (ret < 0)
return ret;
ffio_ensure_seekback(s->pb, i + 1024 + frame_size + 4);
ret = check(s->pb, off + i + frame_size, &header2);
if (ret >= 0 &&
(header & MP3_MASK) == (header2 & MP3_MASK))
{
av_log(s, i > 0 ? AV_LOG_INFO : AV_LOG_VERBOSE, "Skipping %d bytes of junk at %"PRId64".\n", i, off);
ret = avio_seek(s->pb, off + i, SEEK_SET);
if (ret < 0)
return ret;
break;
} else if (ret == CHECK_SEEK_FAILED) {
av_log(s, AV_LOG_ERROR, "Invalid frame size (%d): Could not seek to %"PRId64".\n", frame_size, off + i + frame_size);
return AVERROR(EINVAL);
}
} else if (frame_size == CHECK_SEEK_FAILED) {
av_log(s, AV_LOG_ERROR, "Failed to read frame size: Could not seek to %"PRId64".\n", (int64_t) (i + 1024 + frame_size + 4));
return AVERROR(EINVAL);
}
ret = avio_seek(s->pb, off, SEEK_SET);
if (ret < 0)
return ret;
}
// the seek index is relative to the end of the xing vbr headers
for (i = 0; i < st->nb_index_entries; i++)
st->index_entries[i].pos += avio_tell(s->pb);
/* 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)
{
MP3DecContext *mp3 = s->priv_data;
AVIndexEntry *ie, ie1;
AVStream *st = s->streams[0];
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 > s->internal->data_offset)
filesize = size - s->internal->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 = &st->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) + s->internal->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) {
int frame_duration = av_rescale(st->duration, 1, mp3->frames);
ie1.timestamp = frame_duration * av_rescale(best_pos - s->internal->data_offset, mp3->frames, mp3->header_filesize);
}
ff_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,
};
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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