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

Issue 1804523002: Relanding https://codereview.webrtc.org/1715883002/ in pieces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 4 years, 9 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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539 int32_t _oldVadDecision; 539 int32_t _oldVadDecision;
540 int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise 540 int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
541 // VoEBase 541 // VoEBase
542 bool _externalMixing; 542 bool _externalMixing;
543 bool _mixFileWithMicrophone; 543 bool _mixFileWithMicrophone;
544 // VoEVolumeControl 544 // VoEVolumeControl
545 bool _mute; 545 bool _mute;
546 float _panLeft; 546 float _panLeft;
547 float _panRight; 547 float _panRight;
548 float _outputGain; 548 float _outputGain;
549 // VoEDtmf
550 bool _playOutbandDtmfEvent;
551 // VoeRTP_RTCP 549 // VoeRTP_RTCP
552 uint32_t _lastLocalTimeStamp; 550 uint32_t _lastLocalTimeStamp;
553 int8_t _lastPayloadType; 551 int8_t _lastPayloadType;
554 bool _includeAudioLevelIndication; 552 bool _includeAudioLevelIndication;
555 // VoENetwork 553 // VoENetwork
556 AudioFrame::SpeechType _outputSpeechType; 554 AudioFrame::SpeechType _outputSpeechType;
557 // VoEVideoSync 555 // VoEVideoSync
558 rtc::CriticalSection video_sync_lock_; 556 rtc::CriticalSection video_sync_lock_;
559 uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_); 557 uint32_t _average_jitter_buffer_delay_us GUARDED_BY(video_sync_lock_);
560 uint32_t _previousTimestamp; 558 uint32_t _previousTimestamp;
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575 PacketRouter* packet_router_ = nullptr; 573 PacketRouter* packet_router_ = nullptr;
576 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; 574 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
577 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; 575 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
578 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; 576 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
579 }; 577 };
580 578
581 } // namespace voe 579 } // namespace voe
582 } // namespace webrtc 580 } // namespace webrtc
583 581
584 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 582 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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