Voice Engine: Remove RED support

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 343 matching lines @@
                         unsigned short* fractionLost);
   int SendApplicationDefinedRTCPPacket(unsigned char subType,
                                        unsigned int name,
                                        const char* data,
                                        unsigned short dataLengthInBytes);
   int GetRTPStatistics(unsigned int& averageJitterMs,
                        unsigned int& maxJitterMs,
                        unsigned int& discardedPackets);
   int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
   int GetRTPStatistics(CallStatistics& stats);
-  int SetREDStatus(bool enable, int redPayloadtype);
-  int GetREDStatus(bool& enabled, int& redPayloadtype);
   int SetCodecFECStatus(bool enable);
   bool GetCodecFECStatus();
   void SetNACKStatus(bool enable, int maxNumberOfPackets);
 
   // From AudioPacketizationCallback in the ACM
   int32_t SendData(FrameType frameType,
                    uint8_t payloadType,
                    uint32_t timeStamp,
                    const uint8_t* payloadData,
                    size_t payloadSize,
@@ skipping 84 matching lines @@
                        const RTPHeader& header);
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
   int ResendPackets(const uint16_t* sequence_numbers, int length);
   int32_t MixOrReplaceAudioWithFile(int mixingFrequency);
   int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency);
   void UpdatePlayoutTimestamp(bool rtcp);
   void UpdatePacketDelay(uint32_t timestamp, uint16_t sequenceNumber);
   void RegisterReceiveCodecsToRTPModule();
 
-  int SetRedPayloadType(int red_payload_type);
   int SetSendRtpHeaderExtension(bool enable,
                                 RTPExtensionType type,
                                 unsigned char id);
 
   int32_t GetPlayoutFrequency();
   int64_t GetRTT(bool allow_associate_channel) const;
 
   rtc::CriticalSection _fileCritSect;
   rtc::CriticalSection _callbackCritSect;
   rtc::CriticalSection volume_settings_critsect_;
@@ skipping 102 matching lines @@
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_