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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_ |
13 | 13 |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
| 16 #include "webrtc/base/checks.h" |
16 #include "webrtc/base/scoped_ptr.h" | 17 #include "webrtc/base/scoped_ptr.h" |
17 #include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h" | |
18 #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" | 18 #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h" |
19 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | 19 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
20 | 20 |
21 namespace webrtc { | 21 namespace webrtc { |
22 | 22 |
23 // NOTE: This class has neither ThreadChecker, nor locks. The owner of an | 23 struct CodecInst; |
24 // AudioEncoderOpus object must ensure that it is not accessed concurrently. | |
25 | 24 |
26 class AudioEncoderOpus final : public AudioEncoder { | 25 class AudioEncoderOpus final : public AudioEncoder { |
27 public: | 26 public: |
28 enum ApplicationMode { | 27 enum ApplicationMode { |
29 kVoip = 0, | 28 kVoip = 0, |
30 kAudio = 1, | 29 kAudio = 1, |
31 }; | 30 }; |
32 | 31 |
33 struct Config { | 32 struct Config { |
34 Config(); | |
35 bool IsOk() const; | 33 bool IsOk() const; |
36 int frame_size_ms; | 34 int frame_size_ms = 20; |
37 int num_channels; | 35 int num_channels = 1; |
38 int payload_type; | 36 int payload_type = 120; |
39 ApplicationMode application; | 37 ApplicationMode application = kVoip; |
40 int bitrate_bps; | 38 int bitrate_bps = 64000; |
41 bool fec_enabled; | 39 bool fec_enabled = false; |
42 int max_playback_rate_hz; | 40 int max_playback_rate_hz = 48000; |
43 int complexity; | 41 int complexity = kDefaultComplexity; |
44 bool dtx_enabled; | 42 bool dtx_enabled = false; |
| 43 |
| 44 private: |
| 45 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM) |
| 46 // If we are on Android, iOS and/or ARM, use a lower complexity setting as |
| 47 // default, to save encoder complexity. |
| 48 static const int kDefaultComplexity = 5; |
| 49 #else |
| 50 static const int kDefaultComplexity = 9; |
| 51 #endif |
45 }; | 52 }; |
46 | 53 |
47 explicit AudioEncoderOpus(const Config& config); | 54 explicit AudioEncoderOpus(const Config& config); |
| 55 explicit AudioEncoderOpus(const CodecInst& codec_inst); |
48 ~AudioEncoderOpus() override; | 56 ~AudioEncoderOpus() override; |
49 | 57 |
| 58 size_t MaxEncodedBytes() const override; |
50 int SampleRateHz() const override; | 59 int SampleRateHz() const override; |
51 int NumChannels() const override; | 60 int NumChannels() const override; |
52 size_t MaxEncodedBytes() const override; | |
53 size_t Num10MsFramesInNextPacket() const override; | 61 size_t Num10MsFramesInNextPacket() const override; |
54 size_t Max10MsFramesInAPacket() const override; | 62 size_t Max10MsFramesInAPacket() const override; |
55 int GetTargetBitrate() const override; | 63 int GetTargetBitrate() const override; |
56 void SetTargetBitrate(int bits_per_second) override; | |
57 void SetProjectedPacketLossRate(double fraction) override; | |
58 | |
59 double packet_loss_rate() const { return packet_loss_rate_; } | |
60 ApplicationMode application() const { return application_; } | |
61 bool dtx_enabled() const { return dtx_enabled_; } | |
62 | 64 |
63 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, | 65 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, |
64 const int16_t* audio, | 66 const int16_t* audio, |
65 size_t max_encoded_bytes, | 67 size_t max_encoded_bytes, |
66 uint8_t* encoded) override; | 68 uint8_t* encoded) override; |
67 | 69 |
68 private: | 70 void Reset() override; |
69 const size_t num_10ms_frames_per_packet_; | |
70 const int num_channels_; | |
71 const int payload_type_; | |
72 const ApplicationMode application_; | |
73 int bitrate_bps_; | |
74 const bool dtx_enabled_; | |
75 const size_t samples_per_10ms_frame_; | |
76 std::vector<int16_t> input_buffer_; | |
77 OpusEncInst* inst_; | |
78 uint32_t first_timestamp_in_buffer_; | |
79 double packet_loss_rate_; | |
80 }; | |
81 | |
82 struct CodecInst; | |
83 | |
84 class AudioEncoderMutableOpus | |
85 : public AudioEncoderMutableImpl<AudioEncoderOpus> { | |
86 public: | |
87 explicit AudioEncoderMutableOpus(const CodecInst& codec_inst); | |
88 bool SetFec(bool enable) override; | 71 bool SetFec(bool enable) override; |
89 | 72 |
90 // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice | 73 // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice |
91 // being inactive. During that, it still sends 2 packets (one for content, one | 74 // being inactive. During that, it still sends 2 packets (one for content, one |
92 // for signaling) about every 400 ms. | 75 // for signaling) about every 400 ms. |
93 bool SetDtx(bool enable) override; | 76 bool SetDtx(bool enable) override; |
94 | 77 |
95 bool SetApplication(Application application) override; | 78 bool SetApplication(Application application) override; |
96 bool SetMaxPlaybackRate(int frequency_hz) override; | 79 bool SetMaxPlaybackRate(int frequency_hz) override; |
97 AudioEncoderOpus::ApplicationMode application() const { | 80 void SetProjectedPacketLossRate(double fraction) override; |
98 CriticalSectionScoped cs(encoder_lock_.get()); | 81 void SetTargetBitrate(int target_bps) override; |
99 return encoder()->application(); | 82 |
100 } | 83 // Getters for testing. |
101 double packet_loss_rate() const { | 84 double packet_loss_rate() const { return packet_loss_rate_; } |
102 CriticalSectionScoped cs(encoder_lock_.get()); | 85 ApplicationMode application() const { return config_.application; } |
103 return encoder()->packet_loss_rate(); | 86 bool dtx_enabled() const { return config_.dtx_enabled; } |
104 } | 87 |
105 bool dtx_enabled() const { | 88 private: |
106 CriticalSectionScoped cs(encoder_lock_.get()); | 89 int Num10msFramesPerPacket() const; |
107 return encoder()->dtx_enabled(); | 90 int SamplesPer10msFrame() const; |
108 } | 91 bool RecreateEncoderInstance(const Config& config); |
| 92 |
| 93 Config config_; |
| 94 double packet_loss_rate_; |
| 95 std::vector<int16_t> input_buffer_; |
| 96 OpusEncInst* inst_; |
| 97 uint32_t first_timestamp_in_buffer_; |
109 }; | 98 }; |
110 | 99 |
111 } // namespace webrtc | 100 } // namespace webrtc |
112 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_
H_ | 101 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_
H_ |
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