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
,