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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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 |
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69 uint32_t CaptureLevel() const; | 69 uint32_t CaptureLevel() const; |
70 | 70 |
71 int32_t StopSend(); | 71 int32_t StopSend(); |
72 | 72 |
73 // TODO(solenberg): Remove, once AudioMonitor is gone. | 73 // TODO(solenberg): Remove, once AudioMonitor is gone. |
74 int8_t AudioLevel() const; | 74 int8_t AudioLevel() const; |
75 | 75 |
76 // 'virtual' to allow mocking. | 76 // 'virtual' to allow mocking. |
77 virtual int16_t AudioLevelFullRange() const; | 77 virtual int16_t AudioLevelFullRange() const; |
78 | 78 |
| 79 // See description of "totalAudioEnergy" in the WebRTC stats spec: |
| 80 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaud
ioenergy |
| 81 // 'virtual' to allow mocking. |
| 82 virtual double GetTotalInputEnergy() const; |
| 83 |
| 84 // 'virtual' to allow mocking. |
| 85 virtual double GetTotalInputDuration() const; |
| 86 |
79 bool IsRecordingCall(); | 87 bool IsRecordingCall(); |
80 | 88 |
81 bool IsRecordingMic(); | 89 bool IsRecordingMic(); |
82 | 90 |
83 int StartPlayingFileAsMicrophone(const char* fileName, | 91 int StartPlayingFileAsMicrophone(const char* fileName, |
84 bool loop, | 92 bool loop, |
85 FileFormats format, | 93 FileFormats format, |
86 int startPosition, | 94 int startPosition, |
87 float volumeScaling, | 95 float volumeScaling, |
88 int stopPosition, | 96 int stopPosition, |
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182 std::unique_ptr<FilePlayer> file_player_; | 190 std::unique_ptr<FilePlayer> file_player_; |
183 std::unique_ptr<FileRecorder> file_recorder_; | 191 std::unique_ptr<FileRecorder> file_recorder_; |
184 std::unique_ptr<FileRecorder> file_call_recorder_; | 192 std::unique_ptr<FileRecorder> file_call_recorder_; |
185 int _filePlayerId = 0; | 193 int _filePlayerId = 0; |
186 int _fileRecorderId = 0; | 194 int _fileRecorderId = 0; |
187 int _fileCallRecorderId = 0; | 195 int _fileCallRecorderId = 0; |
188 bool _filePlaying = false; | 196 bool _filePlaying = false; |
189 bool _fileRecording = false; | 197 bool _fileRecording = false; |
190 bool _fileCallRecording = false; | 198 bool _fileCallRecording = false; |
191 voe::AudioLevel _audioLevel; | 199 voe::AudioLevel _audioLevel; |
| 200 double totalInputEnergy_ = 0.0; |
| 201 double totalInputDuration_ = 0.0; |
192 // protect file instances and their variables in MixedParticipants() | 202 // protect file instances and their variables in MixedParticipants() |
193 rtc::CriticalSection _critSect; | 203 rtc::CriticalSection _critSect; |
194 rtc::CriticalSection _callbackCritSect; | 204 rtc::CriticalSection _callbackCritSect; |
195 | 205 |
196 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION | 206 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION |
197 MonitorModule<TransmitMixer> _monitorModule; | 207 MonitorModule<TransmitMixer> _monitorModule; |
198 webrtc::TypingDetection _typingDetection; | 208 webrtc::TypingDetection _typingDetection; |
199 bool _typingNoiseWarningPending = false; | 209 bool _typingNoiseWarningPending = false; |
200 bool _typingNoiseDetected = false; | 210 bool _typingNoiseDetected = false; |
201 #endif | 211 #endif |
202 | 212 |
203 int _instanceId = 0; | 213 int _instanceId = 0; |
204 bool _mixFileWithMicrophone = false; | 214 bool _mixFileWithMicrophone = false; |
205 uint32_t _captureLevel = 0; | 215 uint32_t _captureLevel = 0; |
206 bool stereo_codec_ = false; | 216 bool stereo_codec_ = false; |
207 bool swap_stereo_channels_ = false; | 217 bool swap_stereo_channels_ = false; |
208 }; | 218 }; |
209 } // namespace voe | 219 } // namespace voe |
210 } // namespace webrtc | 220 } // namespace webrtc |
211 | 221 |
212 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H | 222 #endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H |
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