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_HEAD_SIZE 19
37 static int opus_header(AVFormatContext *avf, int idx)
39 struct ogg *ogg = avf->priv_data;
40 struct ogg_stream *os = &ogg->streams[idx];
41 AVStream *st = avf->streams[idx];
42 struct oggopus_private *priv = os->private;
43 uint8_t *packet = os->buf + os->pstart;
46 priv = os->private = av_mallocz(sizeof(*priv));
48 return AVERROR(ENOMEM);
51 if (os->flags & OGG_FLAG_BOS) {
52 if (os->psize < OPUS_HEAD_SIZE || (AV_RL8(packet + 8) & 0xF0) != 0)
53 return AVERROR_INVALIDDATA;
54 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
55 st->codec->codec_id = AV_CODEC_ID_OPUS;
56 st->codec->channels = AV_RL8 (packet + 9);
57 priv->pre_skip = AV_RL16(packet + 10);
58 st->codec->delay = priv->pre_skip;
59 /*orig_sample_rate = AV_RL32(packet + 12);*/
60 /*gain = AV_RL16(packet + 16);*/
61 /*channel_map = AV_RL8 (packet + 18);*/
63 if (ff_alloc_extradata(st->codec, os->psize))
64 return AVERROR(ENOMEM);
66 memcpy(st->codec->extradata, packet, os->psize);
68 st->codec->sample_rate = 48000;
69 avpriv_set_pts_info(st, 64, 1, 48000);
70 priv->need_comments = 1;
74 if (priv->need_comments) {
75 if (os->psize < 8 || memcmp(packet, "OpusTags", 8))
76 return AVERROR_INVALIDDATA;
77 ff_vorbis_comment(avf, &st->metadata, packet + 8, os->psize - 8);
78 priv->need_comments--;
85 static int opus_duration(uint8_t *src, int size)
87 unsigned nb_frames = 1;
88 unsigned toc = src[0];
89 unsigned toc_config = toc >> 3;
90 unsigned toc_count = toc & 3;
91 unsigned frame_size = toc_config < 12 ? FFMAX(480, 960 * (toc_config & 3)) :
92 toc_config < 16 ? 480 << (toc_config & 1) :
93 120 << (toc_config & 3);
96 return AVERROR_INVALIDDATA;
97 nb_frames = src[1] & 0x3F;
98 } else if (toc_count) {
102 return frame_size * nb_frames;
105 static int opus_packet(AVFormatContext *avf, int idx)
107 struct ogg *ogg = avf->priv_data;
108 struct ogg_stream *os = &ogg->streams[idx];
109 AVStream *st = avf->streams[idx];
110 struct oggopus_private *priv = os->private;
111 uint8_t *packet = os->buf + os->pstart;
115 return AVERROR_INVALIDDATA;
117 if ((!os->lastpts || os->lastpts == AV_NOPTS_VALUE) && !(os->flags & OGG_FLAG_EOS)) {
120 uint8_t *last_pkt = os->buf + os->pstart;
121 uint8_t *next_pkt = last_pkt;
125 d = opus_duration(last_pkt, os->psize);
127 os->pflags |= AV_PKT_FLAG_CORRUPT;
131 last_pkt = next_pkt = next_pkt + os->psize;
132 for (; seg < os->nsegs; seg++) {
133 next_pkt += os->segments[seg];
134 if (os->segments[seg] < 255 && next_pkt != last_pkt) {
135 int d = opus_duration(last_pkt, next_pkt - last_pkt);
142 os->lastdts = os->granule - duration;
145 if ((ret = opus_duration(packet, os->psize)) < 0)
149 if (os->lastpts != AV_NOPTS_VALUE) {
150 if (st->start_time == AV_NOPTS_VALUE)
151 st->start_time = os->lastpts;
152 priv->cur_dts = os->lastdts = os->lastpts -= priv->pre_skip;
155 priv->cur_dts += os->pduration;
156 if ((os->flags & OGG_FLAG_EOS)) {
157 int64_t skip = priv->cur_dts - os->granule + priv->pre_skip;
158 skip = FFMIN(skip, os->pduration);
160 os->pduration = skip < os->pduration ? os->pduration - skip : 1;
161 os->end_trimming = skip;
162 av_log(avf, AV_LOG_DEBUG,
163 "Last packet was truncated to %d due to end trimming.\n",
171 const struct ogg_codec ff_opus_codec = {
175 .header = opus_header,
176 .packet = opus_packet,