Remove RED support from WebRtcVoiceEngine/MediaChannel.

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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 116 matching lines @@
   struct Channel {
     Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
     bool playout = false;
     float volume_scale = 1.0f;
     bool vad = false;
     bool codec_fec = false;
     int max_encoding_bandwidth = 0;
     bool opus_dtx = false;
-    bool red = false;
     bool nack = false;
     int cn8_type = 13;
     int cn16_type = 105;
-    int red_type = 117;
     int nack_max_packets = 0;
     uint32_t send_ssrc = 0;
     int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     int neteq_capacity = -1;
     bool neteq_fast_accelerate = false;
   };
 
   FakeWebRtcVoiceEngine() {
@@ skipping 13 matching lines @@
   }
   bool GetPlayout(int channel) {
     return channels_[channel]->playout;
   }
   bool GetVAD(int channel) {
     return channels_[channel]->vad;
   }
   bool GetOpusDtx(int channel) {
     return channels_[channel]->opus_dtx;
   }
-  bool GetRED(int channel) {
-    return channels_[channel]->red;
-  }
   bool GetCodecFEC(int channel) {
     return channels_[channel]->codec_fec;
   }
   int GetMaxEncodingBandwidth(int channel) {
     return channels_[channel]->max_encoding_bandwidth;
   }
   bool GetNACK(int channel) {
     return channels_[channel]->nack;
   }
   int GetNACKMaxPackets(int channel) {
     return channels_[channel]->nack_max_packets;
   }
   int GetSendCNPayloadType(int channel, bool wideband) {
     return (wideband) ?
         channels_[channel]->cn16_type :
         channels_[channel]->cn8_type;
   }
-  int GetSendREDPayloadType(int channel) {
-    return channels_[channel]->red_type;
-  }
   void set_playout_fail_channel(int channel) {
     playout_fail_channel_ = channel;
   }
   void set_fail_create_channel(bool fail_create_channel) {
     fail_create_channel_ = fail_create_channel;
   }
   int AddChannel(const webrtc::Config& config) {
     if (fail_create_channel_) {
       return -1;
     }
@@ skipping 89 matching lines @@
 
   // webrtc::VoECodec
   WEBRTC_STUB(NumOfCodecs, ());
   WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
   WEBRTC_FUNC(SetSendCodec, (int channel, const webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
     // To match the behavior of the real implementation.
     if (_stricmp(codec.plname, "telephone-event") == 0 ||
         _stricmp(codec.plname, "audio/telephone-event") == 0 ||
         _stricmp(codec.plname, "CN") == 0 ||
-        _stricmp(codec.plname, "red") == 0 ) {
+        _stricmp(codec.plname, "red") == 0) {
       return -1;
     }
     channels_[channel]->send_codec = codec;
     ++num_set_send_codecs_;
     return 0;
   }
   WEBRTC_FUNC(GetSendCodec, (int channel, webrtc::CodecInst& codec)) {
     WEBRTC_CHECK_CHANNEL(channel);
     codec = channels_[channel]->send_codec;
     return 0;
@@ skipping 158 matching lines @@
                                   unsigned int& timestamp,
                                   unsigned int& playoutTimestamp,
                                   unsigned int* jitter,
                                   unsigned short* fractionLost));
   WEBRTC_STUB(GetRemoteRTCPReportBlocks,
               (int channel, std::vector<webrtc::ReportBlock>* receive_blocks));
   WEBRTC_STUB(GetRTPStatistics, (int channel, unsigned int& averageJitterMs,
                                  unsigned int& maxJitterMs,
                                  unsigned int& discardedPackets));
   WEBRTC_STUB(GetRTCPStatistics, (int channel, webrtc::CallStatistics& stats));
-  WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->red = enable;
-    channels_[channel]->red_type = redPayloadtype;
-    return 0;
-  }
-  WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    enable = channels_[channel]->red;
-    redPayloadtype = channels_[channel]->red_type;
-    return 0;
-  }
+  WEBRTC_STUB(SetREDStatus, (int channel, bool enable, int redPayloadtype));
+  WEBRTC_STUB(GetREDStatus, (int channel, bool& enable, int& redPayloadtype));
   WEBRTC_FUNC(SetNACKStatus, (int channel, bool enable, int maxNoPackets)) {
     WEBRTC_CHECK_CHANNEL(channel);
     channels_[channel]->nack = enable;
     channels_[channel]->nack_max_packets = maxNoPackets;
     return 0;
   }
 
   // webrtc::VoEVolumeControl
   WEBRTC_STUB(SetSpeakerVolume, (unsigned int));
   WEBRTC_STUB(GetSpeakerVolume, (unsigned int&));
@@ skipping 157 matching lines @@
   webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
   webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
   int playout_fail_channel_ = -1;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_