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FFmpeg/libavformat/oggparseopus.c
Mark Harris 262451878b avformat/oggparseopus: fix segmented timestamps
Fix timestamp calculation for code 3 Ogg Opus packets with less than
2 bytes in the last segment (e.g. packet length 255 or 256).
A sample that would seek incorrectly in ffplay can be created with:
  ffmpeg -i in.wav -b:a 34k -vbr off -frame_duration 60 out.opus
and libopus 1.1

Also do not read past the end of the buffer when a packet has length 0.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2014-01-04 22:39:07 +01:00

179 lines
5.6 KiB
C

/*
* Opus parser for Ogg
* Copyright (c) 2012 Nicolas George
*
* 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 <string.h>
#include "libavutil/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#include "oggdec.h"
struct oggopus_private {
int need_comments;
unsigned pre_skip;
int64_t cur_dts;
};
#define OPUS_HEAD_SIZE 19
static int opus_header(AVFormatContext *avf, int idx)
{
struct ogg *ogg = avf->priv_data;
struct ogg_stream *os = &ogg->streams[idx];
AVStream *st = avf->streams[idx];
struct oggopus_private *priv = os->private;
uint8_t *packet = os->buf + os->pstart;
if (!priv) {
priv = os->private = av_mallocz(sizeof(*priv));
if (!priv)
return AVERROR(ENOMEM);
}
if (os->flags & OGG_FLAG_BOS) {
if (os->psize < OPUS_HEAD_SIZE || (AV_RL8(packet + 8) & 0xF0) != 0)
return AVERROR_INVALIDDATA;
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = AV_CODEC_ID_OPUS;
st->codec->channels = AV_RL8 (packet + 9);
priv->pre_skip = AV_RL16(packet + 10);
st->codec->delay = priv->pre_skip;
/*orig_sample_rate = AV_RL32(packet + 12);*/
/*gain = AV_RL16(packet + 16);*/
/*channel_map = AV_RL8 (packet + 18);*/
if (ff_alloc_extradata(st->codec, os->psize))
return AVERROR(ENOMEM);
memcpy(st->codec->extradata, packet, os->psize);
st->codec->sample_rate = 48000;
avpriv_set_pts_info(st, 64, 1, 48000);
priv->need_comments = 1;
return 1;
}
if (priv->need_comments) {
if (os->psize < 8 || memcmp(packet, "OpusTags", 8))
return AVERROR_INVALIDDATA;
ff_vorbis_comment(avf, &st->metadata, packet + 8, os->psize - 8);
priv->need_comments--;
return 1;
}
return 0;
}
static int opus_duration(uint8_t *src, int size)
{
unsigned nb_frames = 1;
unsigned toc = src[0];
unsigned toc_config = toc >> 3;
unsigned toc_count = toc & 3;
unsigned frame_size = toc_config < 12 ? FFMAX(480, 960 * (toc_config & 3)) :
toc_config < 16 ? 480 << (toc_config & 1) :
120 << (toc_config & 3);
if (toc_count == 3) {
if (size<2)
return AVERROR_INVALIDDATA;
nb_frames = src[1] & 0x3F;
} else if (toc_count) {
nb_frames = 2;
}
return frame_size * nb_frames;
}
static int opus_packet(AVFormatContext *avf, int idx)
{
struct ogg *ogg = avf->priv_data;
struct ogg_stream *os = &ogg->streams[idx];
AVStream *st = avf->streams[idx];
struct oggopus_private *priv = os->private;
uint8_t *packet = os->buf + os->pstart;
int ret;
if (!os->psize)
return AVERROR_INVALIDDATA;
if ((!os->lastpts || os->lastpts == AV_NOPTS_VALUE) && !(os->flags & OGG_FLAG_EOS)) {
int seg, d;
int duration;
uint8_t *last_pkt = os->buf + os->pstart;
uint8_t *next_pkt = last_pkt;
duration = 0;
seg = os->segp;
d = opus_duration(last_pkt, os->psize);
if (d < 0) {
os->pflags |= AV_PKT_FLAG_CORRUPT;
return 0;
}
duration += d;
last_pkt = next_pkt = next_pkt + os->psize;
for (; seg < os->nsegs; seg++) {
next_pkt += os->segments[seg];
if (os->segments[seg] < 255 && next_pkt != last_pkt) {
int d = opus_duration(last_pkt, next_pkt - last_pkt);
if (d > 0)
duration += d;
last_pkt = next_pkt;
}
}
os->lastpts =
os->lastdts = os->granule - duration;
}
if ((ret = opus_duration(packet, os->psize)) < 0)
return ret;
os->pduration = ret;
if (os->lastpts != AV_NOPTS_VALUE) {
if (st->start_time == AV_NOPTS_VALUE)
st->start_time = os->lastpts;
priv->cur_dts = os->lastdts = os->lastpts -= priv->pre_skip;
}
priv->cur_dts += os->pduration;
if ((os->flags & OGG_FLAG_EOS)) {
int64_t skip = priv->cur_dts - os->granule + priv->pre_skip;
skip = FFMIN(skip, os->pduration);
if (skip > 0) {
os->pduration = skip < os->pduration ? os->pduration - skip : 1;
os->end_trimming = skip;
av_log(avf, AV_LOG_DEBUG,
"Last packet was truncated to %d due to end trimming.\n",
os->pduration);
}
}
return 0;
}
const struct ogg_codec ff_opus_codec = {
.name = "Opus",
.magic = "OpusHead",
.magicsize = 8,
.header = opus_header,
.packet = opus_packet,
.nb_header = 1,
};