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Side by Side Diff: webrtc/voice_engine/channel.h

Issue 3011623002: Add new ANA stats to GetStats() to count the number of actions taken by each controller. (Closed)
Patch Set: Fix for failing test. Created 3 years, 3 months ago
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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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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);
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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_
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