Remove voe::Channel::StopReceive() and associated logic.

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 70 matching lines @@
 // thread T2. Accessing the member via this class ensures that we are
 // safe and also avoid TSan v2 warnings.
 class ChannelState {
  public:
   struct State {
     bool input_external_media = false;
     bool output_file_playing = false;
     bool input_file_playing = false;
     bool playing = false;
     bool sending = false;
-    bool receiving = false;
   };
 
   ChannelState() {}
   virtual ~ChannelState() {}
 
   void Reset() {
     rtc::CritScope lock(&lock_);
     state_ = State();
   }
 
@@ skipping 20 matching lines @@
   void SetPlaying(bool enable) {
     rtc::CritScope lock(&lock_);
     state_.playing = enable;
   }
 
   void SetSending(bool enable) {
     rtc::CritScope lock(&lock_);
     state_.sending = enable;
   }
 
-  void SetReceiving(bool enable) {
-    rtc::CritScope lock(&lock_);
-    state_.receiving = enable;
-  }
-
  private:
   rtc::CriticalSection lock_;
   State state_;
 };
 
 class Channel
     : public RtpData,
       public RtpFeedback,
       public FileCallback,  // receiving notification from file player &
                             // recorder
@@ skipping 35 matching lines @@
   // go.
   const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const;
 
   // API methods
 
   // VoEBase
   int32_t StartPlayout();
   int32_t StopPlayout();
   int32_t StartSend();
   int32_t StopSend();
-  int32_t StartReceiving();
-  int32_t StopReceiving();
-
+  void ResetDiscardedPacketCount();
   int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
   int32_t DeRegisterVoiceEngineObserver();
 
   // VoECodec
   int32_t GetSendCodec(CodecInst& codec);
   int32_t GetRecCodec(CodecInst& codec);
   int32_t SetSendCodec(const CodecInst& codec);
   void SetBitRate(int bitrate_bps);
   int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
   int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
@@ skipping 181 matching lines @@
   // From FileCallback
   void PlayNotification(int32_t id, uint32_t durationMs) override;
   void RecordNotification(int32_t id, uint32_t durationMs) override;
   void PlayFileEnded(int32_t id) override;
   void RecordFileEnded(int32_t id) override;
 
   uint32_t InstanceId() const { return _instanceId; }
   int32_t ChannelId() const { return _channelId; }
   bool Playing() const { return channel_state_.Get().playing; }
   bool Sending() const { return channel_state_.Get().sending; }
-  bool Receiving() const { return channel_state_.Get().receiving; }
   bool ExternalTransport() const {
     rtc::CritScope cs(&_callbackCritSect);
     return _externalTransport;
   }
   bool ExternalMixing() const { return _externalMixing; }
   RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
   int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
   uint32_t Demultiplex(const AudioFrame& audioFrame);
   // Demultiplex the data to the channel's |_audioFrame|. The difference
   // between this method and the overloaded method above is that |audio_data|
@@ skipping 147 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_