virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-12 19:18:44 +02:00
Code Issues Releases Activity
f524d2e47c
FFmpeg/libavformat/mp3dec.c
wm4 6c7f1155bb avformat/mp3dec: avoid early EOF with concatenated gapless mp3s 
Consider a file created with something like:

    cat file1.mp3 file2.mp3 > result.mp3

Then if file2.mp3 has gapless information, result.mp3 would stop playing
something in the middle. This happens because the gapless info directs
the decoder to discard all samples after a certain position. To make
matters worse, the gapless info of file2.mp3 will be used when playing
the file1.mp3 part, because the gapless info is located at the end of
the file.

While handling concatenated gapless files correctly would be insane and
a lot of effort (especially without scanning the whole file on opening),
it's easy to prevent at least early EOF. Playback will happen to work,
even if it's slightly broken.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-09-21 14:57:12 +02:00

508 lines
15 KiB
C
Raw Blame History

/*
* 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 && max_frames >= p->buf_size/10000) 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 && max_frames >= p->buf_size/10000) 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}};
uint64_t fsize = avio_size(s->pb);
/* 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;
} 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 & 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 (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;
}
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;
int64_t best_pos;
int best_score;
if ( mp3->is_cbr
&& st->duration > 0
&& mp3->header_filesize > s->data_offset
&& mp3->frames) {
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;
}
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
best_pos = ie->pos;
best_score = 999;
for(i=0; i<4096; i++) {
int64_t pos = ie->pos + (dir > 0 ? i - 1024 : -i);
int64_t candidate = -1;
int score = 999;
for(j=0; j<MIN_VALID; j++) {
ret = check(s, pos);
if(ret < 0)
break;
if ((ie->pos - pos)*dir <= 0 && abs(MIN_VALID/2-j) < score) {
candidate = pos;
score = abs(MIN_VALID/2-j);
}
pos += ret;
}
if (best_score > score && j == MIN_VALID) {
best_pos = candidate;
best_score = score;
if(score == 0)
break;
}
}
ret = avio_seek(s->pb, best_pos, 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,
};
Reference in New Issue View Git Blame Copy Permalink