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

Issue 1782053002: 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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290 int LeastRequiredDelayMs() const; 290 int LeastRequiredDelayMs() const;
291 int SetMinimumPlayoutDelay(int delayMs); 291 int SetMinimumPlayoutDelay(int delayMs);
292 int GetPlayoutTimestamp(unsigned int& timestamp); 292 int GetPlayoutTimestamp(unsigned int& timestamp);
293 int SetInitTimestamp(unsigned int timestamp); 293 int SetInitTimestamp(unsigned int timestamp);
294 int SetInitSequenceNumber(short sequenceNumber); 294 int SetInitSequenceNumber(short sequenceNumber);
295 295
296 // VoEVideoSyncExtended 296 // VoEVideoSyncExtended
297 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const; 297 int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
298 298
299 // VoEDtmf 299 // VoEDtmf
300 int SendTelephoneEventOutband(unsigned char eventCode, 300 int SendTelephoneEventOutband(int event, int duration_ms);
301 int lengthMs,
302 int attenuationDb,
303 bool playDtmfEvent);
304 int SendTelephoneEventInband(unsigned char eventCode, 301 int SendTelephoneEventInband(unsigned char eventCode,
305 int lengthMs, 302 int lengthMs,
306 int attenuationDb, 303 int attenuationDb,
307 bool playDtmfEvent); 304 bool playDtmfEvent);
308 int SetSendTelephoneEventPayloadType(unsigned char type); 305 int SetSendTelephoneEventPayloadType(unsigned char type);
309 int GetSendTelephoneEventPayloadType(unsigned char& type); 306 int GetSendTelephoneEventPayloadType(unsigned char& type);
310 307
311 // VoEAudioProcessingImpl 308 // VoEAudioProcessingImpl
312 int UpdateRxVadDetection(AudioFrame& audioFrame); 309 int UpdateRxVadDetection(AudioFrame& audioFrame);
313 int RegisterRxVadObserver(VoERxVadCallback& observer); 310 int RegisterRxVadObserver(VoERxVadCallback& observer);
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590 PacketRouter* packet_router_ = nullptr; 587 PacketRouter* packet_router_ = nullptr;
591 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; 588 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
592 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; 589 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
593 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; 590 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
594 }; 591 };
595 592
596 } // namespace voe 593 } // namespace voe
597 } // namespace webrtc 594 } // namespace webrtc
598 595
599 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 596 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
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