Remove unused methods in WebRtcVoiceEngine and VoiceMediaChannel.

webrtc/media/base/fakemediaengine.h

 /*
  *  Copyright (c) 2004 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 286 matching lines @@
     DtmfInfo(uint32_t ssrc, int event_code, int duration)
         : ssrc(ssrc),
           event_code(event_code),
           duration(duration) {}
     uint32_t ssrc;
     int event_code;
     int duration;
   };
   explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
                                  const AudioOptions& options)
-      : engine_(engine), time_since_last_typing_(-1), max_bps_(-1) {
+      : engine_(engine), max_bps_(-1) {
     output_scalings_[0] = 1.0;  // For default channel.
     SetOptions(options);
   }
   ~FakeVoiceMediaChannel();
   const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
   const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
   const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
   const std::vector<DtmfInfo>& dtmf_info_queue() const {
     return dtmf_info_queue_;
   }
@@ skipping 41 matching lines @@
   }
   virtual bool RemoveRecvStream(uint32_t ssrc) {
     if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
       return false;
     output_scalings_.erase(ssrc);
     return true;
   }
 
   virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
   virtual int GetOutputLevel() { return 0; }
-  void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
-  virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
-  virtual void SetTypingDetectionParameters(
-      int time_window, int cost_per_typing, int reporting_threshold,
-      int penalty_decay, int type_event_delay) {}
 
   virtual bool CanInsertDtmf() {
     for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
          it != send_codecs_.end(); ++it) {
       // Find the DTMF telephone event "codec".
       if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
         return true;
       }
     }
     return false;
@@ skipping 97 matching lines @@
       }
     }
     return true;
   }
 
   FakeVoiceEngine* engine_;
   std::vector<AudioCodec> recv_codecs_;
   std::vector<AudioCodec> send_codecs_;
   std::map<uint32_t, double> output_scalings_;
   std::vector<DtmfInfo> dtmf_info_queue_;
-  int time_since_last_typing_;
   AudioOptions options_;
   std::map<uint32_t, VoiceChannelAudioSink*> local_sinks_;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
   int max_bps_;
 };
 
 // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
 inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
                             uint32_t ssrc,
                             int event_code,
@@ skipping 450 matching lines @@
   virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
 
  private:
   std::vector<FakeDataMediaChannel*> channels_;
   std::vector<DataCodec> data_codecs_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_