OLD | NEW |
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 |
(...skipping 250 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
261 int GetSpeechOutputLevel() const; | 261 int GetSpeechOutputLevel() const; |
262 int GetSpeechOutputLevelFullRange() const; | 262 int GetSpeechOutputLevelFullRange() const; |
263 // See description of "totalAudioEnergy" in the WebRTC stats spec: | 263 // See description of "totalAudioEnergy" in the WebRTC stats spec: |
264 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudio
energy | 264 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudio
energy |
265 double GetTotalOutputEnergy() const; | 265 double GetTotalOutputEnergy() const; |
266 double GetTotalOutputDuration() const; | 266 double GetTotalOutputDuration() const; |
267 | 267 |
268 // Stats. | 268 // Stats. |
269 int GetNetworkStatistics(NetworkStatistics& stats); | 269 int GetNetworkStatistics(NetworkStatistics& stats); |
270 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; | 270 void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const; |
| 271 ANAStats GetANAStatistics() const; |
271 | 272 |
272 // Audio+Video Sync. | 273 // Audio+Video Sync. |
273 uint32_t GetDelayEstimate() const; | 274 uint32_t GetDelayEstimate() const; |
274 int SetMinimumPlayoutDelay(int delayMs); | 275 int SetMinimumPlayoutDelay(int delayMs); |
275 int GetPlayoutTimestamp(unsigned int& timestamp); | 276 int GetPlayoutTimestamp(unsigned int& timestamp); |
276 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; | 277 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; |
277 | 278 |
278 // DTMF. | 279 // DTMF. |
279 int SendTelephoneEventOutband(int event, int duration_ms); | 280 int SendTelephoneEventOutband(int event, int duration_ms); |
280 int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency); | 281 int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency); |
(...skipping 270 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
551 | 552 |
552 bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false; | 553 bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false; |
553 | 554 |
554 rtc::TaskQueue* encoder_queue_ = nullptr; | 555 rtc::TaskQueue* encoder_queue_ = nullptr; |
555 }; | 556 }; |
556 | 557 |
557 } // namespace voe | 558 } // namespace voe |
558 } // namespace webrtc | 559 } // namespace webrtc |
559 | 560 |
560 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 561 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
OLD | NEW |