FFmpeg/libavformat/oggparseopus.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

190 lines
6.0 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 "libavutil/mem.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_SEEK_PREROLL_MS 80
#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;
int ret;
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->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
st->codecpar->codec_id = AV_CODEC_ID_OPUS;
st->codecpar->ch_layout.nb_channels = AV_RL8(packet + 9);
priv->pre_skip = AV_RL16(packet + 10);
st->codecpar->initial_padding = priv->pre_skip;
os->start_trimming = priv->pre_skip;
/*orig_sample_rate = AV_RL32(packet + 12);*/
/*gain = AV_RL16(packet + 16);*/
/*channel_map = AV_RL8 (packet + 18);*/
if ((ret = ff_alloc_extradata(st->codecpar, os->psize)) < 0)
return ret;
memcpy(st->codecpar->extradata, packet, os->psize);
st->codecpar->sample_rate = 48000;
st->codecpar->seek_preroll = av_rescale(OPUS_SEEK_PREROLL_MS,
st->codecpar->sample_rate, 1000);
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_stream_comment(avf, st, 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->granule > (1LL << 62)) {
av_log(avf, AV_LOG_ERROR, "Unsupported huge granule pos %"PRId64 "\n", os->granule);
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,
};