1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
|
/*
* Opus decoder/demuxer common functions
* Copyright (c) 2012 Andrew D'Addesio
* Copyright (c) 2013-2014 Mozilla Corporation
*
* 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
*/
#ifndef AVCODEC_OPUS_H
#define AVCODEC_OPUS_H
#include <stdint.h>
#include "libavutil/audio_fifo.h"
#include "libavutil/float_dsp.h"
#include "libavutil/frame.h"
#include "libswresample/swresample.h"
#include "avcodec.h"
#include "opus_rc.h"
#define MAX_FRAME_SIZE 1275
#define MAX_FRAMES 48
#define MAX_PACKET_DUR 5760
#define CELT_SHORT_BLOCKSIZE 120
#define CELT_OVERLAP CELT_SHORT_BLOCKSIZE
#define CELT_MAX_LOG_BLOCKS 3
#define CELT_MAX_FRAME_SIZE (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS))
#define CELT_MAX_BANDS 21
#define SILK_HISTORY 322
#define SILK_MAX_LPC 16
#define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1)
#define ROUND_MUL16(a,b) ((MUL16(a, b) + 16384) >> 15)
#define OPUS_TS_HEADER 0x7FE0 // 0x3ff (11 bits)
#define OPUS_TS_MASK 0xFFE0 // top 11 bits
static const uint8_t opus_default_extradata[30] = {
'O', 'p', 'u', 's', 'H', 'e', 'a', 'd',
1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
};
enum OpusMode {
OPUS_MODE_SILK,
OPUS_MODE_HYBRID,
OPUS_MODE_CELT,
OPUS_MODE_NB
};
enum OpusBandwidth {
OPUS_BANDWIDTH_NARROWBAND,
OPUS_BANDWIDTH_MEDIUMBAND,
OPUS_BANDWIDTH_WIDEBAND,
OPUS_BANDWIDTH_SUPERWIDEBAND,
OPUS_BANDWIDTH_FULLBAND,
OPUS_BANDWITH_NB
};
typedef struct SilkContext SilkContext;
typedef struct CeltFrame CeltFrame;
typedef struct OpusPacket {
int packet_size; /**< packet size */
int data_size; /**< size of the useful data -- packet size - padding */
int code; /**< packet code: specifies the frame layout */
int stereo; /**< whether this packet is mono or stereo */
int vbr; /**< vbr flag */
int config; /**< configuration: tells the audio mode,
** bandwidth, and frame duration */
int frame_count; /**< frame count */
int frame_offset[MAX_FRAMES]; /**< frame offsets */
int frame_size[MAX_FRAMES]; /**< frame sizes */
int frame_duration; /**< frame duration, in samples @ 48kHz */
enum OpusMode mode; /**< mode */
enum OpusBandwidth bandwidth; /**< bandwidth */
} OpusPacket;
typedef struct OpusStreamContext {
AVCodecContext *avctx;
int output_channels;
/* number of decoded samples for this stream */
int decoded_samples;
/* current output buffers for this stream */
float *out[2];
int out_size;
/* Buffer with samples from this stream for synchronizing
* the streams when they have different resampling delays */
AVAudioFifo *sync_buffer;
OpusRangeCoder rc;
OpusRangeCoder redundancy_rc;
SilkContext *silk;
CeltFrame *celt;
AVFloatDSPContext *fdsp;
float silk_buf[2][960];
float *silk_output[2];
DECLARE_ALIGNED(32, float, celt_buf)[2][960];
float *celt_output[2];
DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
float *redundancy_output[2];
/* buffers for the next samples to be decoded */
float *cur_out[2];
int remaining_out_size;
float *out_dummy;
int out_dummy_allocated_size;
SwrContext *swr;
AVAudioFifo *celt_delay;
int silk_samplerate;
/* number of samples we still want to get from the resampler */
int delayed_samples;
OpusPacket packet;
int redundancy_idx;
} OpusStreamContext;
// a mapping between an opus stream and an output channel
typedef struct ChannelMap {
int stream_idx;
int channel_idx;
// when a single decoded channel is mapped to multiple output channels, we
// write to the first output directly and copy from it to the others
// this field is set to 1 for those copied output channels
int copy;
// this is the index of the output channel to copy from
int copy_idx;
// this channel is silent
int silence;
} ChannelMap;
typedef struct OpusContext {
AVClass *av_class;
OpusStreamContext *streams;
int apply_phase_inv;
int nb_streams;
int nb_stereo_streams;
AVFloatDSPContext *fdsp;
int16_t gain_i;
float gain;
ChannelMap *channel_maps;
} OpusContext;
int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
int self_delimited);
int ff_opus_parse_extradata(AVCodecContext *avctx, OpusContext *s);
int ff_silk_init(AVCodecContext *avctx, SilkContext **ps, int output_channels);
void ff_silk_free(SilkContext **ps);
void ff_silk_flush(SilkContext *s);
/**
* Decode the LP layer of one Opus frame (which may correspond to several SILK
* frames).
*/
int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
float *output[2],
enum OpusBandwidth bandwidth, int coded_channels,
int duration_ms);
/* Encode or decode CELT bands */
void ff_celt_quant_bands(CeltFrame *f, OpusRangeCoder *rc);
/* Encode or decode CELT bitallocation */
void ff_celt_bitalloc(CeltFrame *f, OpusRangeCoder *rc, int encode);
#endif /* AVCODEC_OPUS_H */
|