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
|
/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
namespace webrtc {
struct CodecInst;
class AudioEncoderOpus final : public AudioEncoder {
public:
enum ApplicationMode {
kVoip = 0,
kAudio = 1,
};
struct Config {
bool IsOk() const;
int frame_size_ms = 20;
int num_channels = 1;
int payload_type = 120;
ApplicationMode application = kVoip;
int bitrate_bps = 64000;
bool fec_enabled = false;
int max_playback_rate_hz = 48000;
int complexity = kDefaultComplexity;
bool dtx_enabled = false;
private:
#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
// If we are on Android, iOS and/or ARM, use a lower complexity setting as
// default, to save encoder complexity.
static const int kDefaultComplexity = 5;
#else
static const int kDefaultComplexity = 9;
#endif
};
explicit AudioEncoderOpus(const Config& config);
explicit AudioEncoderOpus(const CodecInst& codec_inst);
~AudioEncoderOpus() override;
size_t MaxEncodedBytes() const override;
int SampleRateHz() const override;
int NumChannels() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
size_t max_encoded_bytes,
uint8_t* encoded) override;
void Reset() override;
bool SetFec(bool enable) override;
// Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
// being inactive. During that, it still sends 2 packets (one for content, one
// for signaling) about every 400 ms.
bool SetDtx(bool enable) override;
bool SetApplication(Application application) override;
void SetMaxPlaybackRate(int frequency_hz) override;
void SetProjectedPacketLossRate(double fraction) override;
void SetTargetBitrate(int target_bps) override;
// Getters for testing.
double packet_loss_rate() const { return packet_loss_rate_; }
ApplicationMode application() const { return config_.application; }
bool dtx_enabled() const { return config_.dtx_enabled; }
private:
int Num10msFramesPerPacket() const;
int SamplesPer10msFrame() const;
bool RecreateEncoderInstance(const Config& config);
Config config_;
double packet_loss_rate_;
std::vector<int16_t> input_buffer_;
OpusEncInst* inst_;
uint32_t first_timestamp_in_buffer_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INCLUDE_AUDIO_ENCODER_OPUS_H_
|
