Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(17)

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

Issue 2709723003: Initial implementation of RtpTransportControllerReceive and related interfaces.
Patch Set: Merge remote-tracking branch 'origin/master' into design-RtpTransportReceiveController Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/video/rtp_video_stream_receiver.cc ('k') | webrtc/voice_engine/channel.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 266 matching lines...) Expand 10 before | Expand all | Expand 10 after
277 277
278 // DTMF. 278 // DTMF.
279 int SendTelephoneEventOutband(int event, int duration_ms); 279 int SendTelephoneEventOutband(int event, int duration_ms);
280 int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency); 280 int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
281 281
282 // VoERTP_RTCP 282 // VoERTP_RTCP
283 int SetLocalSSRC(unsigned int ssrc); 283 int SetLocalSSRC(unsigned int ssrc);
284 int GetLocalSSRC(unsigned int& ssrc); 284 int GetLocalSSRC(unsigned int& ssrc);
285 int GetRemoteSSRC(unsigned int& ssrc); 285 int GetRemoteSSRC(unsigned int& ssrc);
286 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id); 286 int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
287 int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
288 void EnableSendTransportSequenceNumber(int id); 287 void EnableSendTransportSequenceNumber(int id);
289 void EnableReceiveTransportSequenceNumber(int id);
290 288
291 void RegisterSenderCongestionControlObjects( 289 void RegisterSenderCongestionControlObjects(
292 RtpTransportControllerSendInterface* transport, 290 RtpTransportControllerSendInterface* transport,
293 RtcpBandwidthObserver* bandwidth_observer); 291 RtcpBandwidthObserver* bandwidth_observer);
294 void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router); 292 void RegisterReceiverCongestionControlObjects(PacketRouter* packet_router);
295 void ResetSenderCongestionControlObjects(); 293 void ResetSenderCongestionControlObjects();
296 void ResetReceiverCongestionControlObjects(); 294 void ResetReceiverCongestionControlObjects();
297 void SetRTCPStatus(bool enable); 295 void SetRTCPStatus(bool enable);
298 int GetRTCPStatus(bool& enabled); 296 int GetRTCPStatus(bool& enabled);
299 int SetRTCP_CNAME(const char cName[256]); 297 int SetRTCP_CNAME(const char cName[256]);
(...skipping 251 matching lines...) Expand 10 before | Expand all | Expand 10 after
551 549
552 bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false; 550 bool encoder_queue_is_active_ GUARDED_BY(encoder_queue_lock_) = false;
553 551
554 rtc::TaskQueue* encoder_queue_ = nullptr; 552 rtc::TaskQueue* encoder_queue_ = nullptr;
555 }; 553 };
556 554
557 } // namespace voe 555 } // namespace voe
558 } // namespace webrtc 556 } // namespace webrtc
559 557
560 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 558 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
OLDNEW
« no previous file with comments | « webrtc/video/rtp_video_stream_receiver.cc ('k') | webrtc/voice_engine/channel.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698