3 * Copyright (c) 2012 Nicolas George
5 * This file is part of FFmpeg.
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 #include "libavutil/intreadwrite.h"
29 struct oggopus_private {
35 #define OPUS_SEEK_PREROLL_MS 80
36 #define OPUS_HEAD_SIZE 19
38 static int opus_header(AVFormatContext *avf, int idx)
40 struct ogg *ogg = avf->priv_data;
41 struct ogg_stream *os = &ogg->streams[idx];
42 AVStream *st = avf->streams[idx];
43 struct oggopus_private *priv = os->private;
44 uint8_t *packet = os->buf + os->pstart;
47 priv = os->private = av_mallocz(sizeof(*priv));
49 return AVERROR(ENOMEM);
52 if (os->flags & OGG_FLAG_BOS) {
53 if (os->psize < OPUS_HEAD_SIZE || (AV_RL8(packet + 8) & 0xF0) != 0)
54 return AVERROR_INVALIDDATA;
55 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
56 st->codec->codec_id = AV_CODEC_ID_OPUS;
57 st->codec->channels = AV_RL8 (packet + 9);
58 priv->pre_skip = AV_RL16(packet + 10);
59 st->codec->delay = priv->pre_skip;
60 /*orig_sample_rate = AV_RL32(packet + 12);*/
61 /*gain = AV_RL16(packet + 16);*/
62 /*channel_map = AV_RL8 (packet + 18);*/
64 if (ff_alloc_extradata(st->codec, os->psize))
65 return AVERROR(ENOMEM);
67 memcpy(st->codec->extradata, packet, os->psize);
69 st->codec->sample_rate = 48000;
70 av_codec_set_seek_preroll(st->codec,
71 av_rescale(OPUS_SEEK_PREROLL_MS,
72 st->codec->sample_rate, 1000));
73 avpriv_set_pts_info(st, 64, 1, 48000);
74 priv->need_comments = 1;
78 if (priv->need_comments) {
79 if (os->psize < 8 || memcmp(packet, "OpusTags", 8))
80 return AVERROR_INVALIDDATA;
81 ff_vorbis_stream_comment(avf, st, packet + 8, os->psize - 8);
82 priv->need_comments--;
89 static int opus_duration(uint8_t *src, int size)
91 unsigned nb_frames = 1;
92 unsigned toc = src[0];
93 unsigned toc_config = toc >> 3;
94 unsigned toc_count = toc & 3;
95 unsigned frame_size = toc_config < 12 ? FFMAX(480, 960 * (toc_config & 3)) :
96 toc_config < 16 ? 480 << (toc_config & 1) :
97 120 << (toc_config & 3);
100 return AVERROR_INVALIDDATA;
101 nb_frames = src[1] & 0x3F;
102 } else if (toc_count) {
106 return frame_size * nb_frames;
109 static int opus_packet(AVFormatContext *avf, int idx)
111 struct ogg *ogg = avf->priv_data;
112 struct ogg_stream *os = &ogg->streams[idx];
113 AVStream *st = avf->streams[idx];
114 struct oggopus_private *priv = os->private;
115 uint8_t *packet = os->buf + os->pstart;
119 return AVERROR_INVALIDDATA;
121 if ((!os->lastpts || os->lastpts == AV_NOPTS_VALUE) && !(os->flags & OGG_FLAG_EOS)) {
124 uint8_t *last_pkt = os->buf + os->pstart;
125 uint8_t *next_pkt = last_pkt;
129 d = opus_duration(last_pkt, os->psize);
131 os->pflags |= AV_PKT_FLAG_CORRUPT;
135 last_pkt = next_pkt = next_pkt + os->psize;
136 for (; seg < os->nsegs; seg++) {
137 next_pkt += os->segments[seg];
138 if (os->segments[seg] < 255 && next_pkt != last_pkt) {
139 int d = opus_duration(last_pkt, next_pkt - last_pkt);
146 os->lastdts = os->granule - duration;
149 if ((ret = opus_duration(packet, os->psize)) < 0)
153 if (os->lastpts != AV_NOPTS_VALUE) {
154 if (st->start_time == AV_NOPTS_VALUE)
155 st->start_time = os->lastpts;
156 priv->cur_dts = os->lastdts = os->lastpts -= priv->pre_skip;
159 priv->cur_dts += os->pduration;
160 if ((os->flags & OGG_FLAG_EOS)) {
161 int64_t skip = priv->cur_dts - os->granule + priv->pre_skip;
162 skip = FFMIN(skip, os->pduration);
164 os->pduration = skip < os->pduration ? os->pduration - skip : 1;
165 os->end_trimming = skip;
166 av_log(avf, AV_LOG_DEBUG,
167 "Last packet was truncated to %d due to end trimming.\n",
175 const struct ogg_codec ff_opus_codec = {
179 .header = opus_header,
180 .packet = opus_packet,