Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1)

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

Issue 2055493003: Voice Engine: Remove RED support (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove-red0
Patch Set: Created 4 years, 6 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 | « no previous file | webrtc/voice_engine/channel.cc » ('j') | webrtc/voice_engine/channel.cc » ('J')
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 343 matching lines...) Expand 10 before | Expand all | Expand 10 after
354 unsigned short* fractionLost); 354 unsigned short* fractionLost);
355 int SendApplicationDefinedRTCPPacket(unsigned char subType, 355 int SendApplicationDefinedRTCPPacket(unsigned char subType,
356 unsigned int name, 356 unsigned int name,
357 const char* data, 357 const char* data,
358 unsigned short dataLengthInBytes); 358 unsigned short dataLengthInBytes);
359 int GetRTPStatistics(unsigned int& averageJitterMs, 359 int GetRTPStatistics(unsigned int& averageJitterMs,
360 unsigned int& maxJitterMs, 360 unsigned int& maxJitterMs,
361 unsigned int& discardedPackets); 361 unsigned int& discardedPackets);
362 int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks); 362 int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
363 int GetRTPStatistics(CallStatistics& stats); 363 int GetRTPStatistics(CallStatistics& stats);
364 int SetREDStatus(bool enable, int redPayloadtype); 364 //int SetREDStatus(bool enable, int redPayloadtype);
the sun 2016/06/13 08:17:39 remove
kwiberg-webrtc 2016/06/13 11:21:14 Thanks. Done.
365 int GetREDStatus(bool& enabled, int& redPayloadtype); 365 //int GetREDStatus(bool& enabled, int& redPayloadtype);
366 int SetCodecFECStatus(bool enable); 366 int SetCodecFECStatus(bool enable);
367 bool GetCodecFECStatus(); 367 bool GetCodecFECStatus();
368 void SetNACKStatus(bool enable, int maxNumberOfPackets); 368 void SetNACKStatus(bool enable, int maxNumberOfPackets);
369 369
370 // From AudioPacketizationCallback in the ACM 370 // From AudioPacketizationCallback in the ACM
371 int32_t SendData(FrameType frameType, 371 int32_t SendData(FrameType frameType,
372 uint8_t payloadType, 372 uint8_t payloadType,
373 uint32_t timeStamp, 373 uint32_t timeStamp,
374 const uint8_t* payloadData, 374 const uint8_t* payloadData,
375 size_t payloadSize, 375 size_t payloadSize,
(...skipping 207 matching lines...) Expand 10 before | Expand all | Expand 10 after
583 PacketRouter* packet_router_ = nullptr; 583 PacketRouter* packet_router_ = nullptr;
584 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_; 584 std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
585 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_; 585 std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
586 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_; 586 std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
587 }; 587 };
588 588
589 } // namespace voe 589 } // namespace voe
590 } // namespace webrtc 590 } // namespace webrtc
591 591
592 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ 592 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/voice_engine/channel.cc » ('j') | webrtc/voice_engine/channel.cc » ('J')

Powered by Google App Engine
This is Rietveld 408576698