Remove VoEVolumeControl interface.

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 227 matching lines @@
                                    int stopPosition,
                                    const CodecInst* codecInst);
   int StopPlayingFileAsMicrophone();
   int IsPlayingFileAsMicrophone() const;
   int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
   int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
   int StopRecordingPlayout();
 
   void SetMixWithMicStatus(bool mix);
 
-  // VoEVolumeControl
-  int GetSpeechOutputLevel(uint32_t& level) const;
-  int GetSpeechOutputLevelFullRange(uint32_t& level) const;
-  int SetInputMute(bool enable);
-  bool InputMute() const;
-  int SetOutputVolumePan(float left, float right);
-  int GetOutputVolumePan(float& left, float& right) const;
-  int SetChannelOutputVolumeScaling(float scaling);
-  int GetChannelOutputVolumeScaling(float& scaling) const;
+  // Muting, Volume and Level.
+  void SetInputMute(bool enable);
+  void SetChannelOutputVolumeScaling(float scaling);
+  int GetSpeechOutputLevel() const;
+  int GetSpeechOutputLevelFullRange() const;
 
   // VoENetEqStats
   int GetNetworkStatistics(NetworkStatistics& stats);
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
   // Audio+Video Sync
   uint32_t GetDelayEstimate() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
@@ skipping 118 matching lines @@
   void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
   void SetTransportOverhead(size_t transport_overhead_per_packet);
 
   // From OverheadObserver in the RTP/RTCP module
   void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
 
  protected:
   void OnIncomingFractionLoss(int fraction_lost);
 
  private:
+  bool InputMute() const;
   bool OnRtpPacketWithHeader(const uint8_t* received_packet,
                              size_t length,
                              RTPHeader *header);
   bool ReceivePacket(const uint8_t* packet,
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
   bool HandleRtxPacket(const uint8_t* packet,
                        size_t packet_length,
                        const RTPHeader& header);
@@ skipping 71 matching lines @@
   // uses
   Statistics* _engineStatisticsPtr;
   OutputMixer* _outputMixerPtr;
   ProcessThread* _moduleProcessThreadPtr;
   AudioDeviceModule* _audioDeviceModulePtr;
   VoiceEngineObserver* _voiceEngineObserverPtr;  // owned by base
   rtc::CriticalSection* _callbackCritSectPtr;    // owned by base
   Transport* _transportPtr;  // WebRtc socket or external transport
   RmsLevel rms_level_;
   int32_t _sendFrameType;  // Send data is voice, 1-voice, 0-otherwise
+  bool input_mute_ GUARDED_BY(volume_settings_critsect_);
+  bool previous_frame_muted_;  // Only accessed from PrepareEncodeAndSend().
+  float _outputGain GUARDED_BY(volume_settings_critsect_);
   // VoEBase
   bool _mixFileWithMicrophone;
-  // VoEVolumeControl
-  bool input_mute_ GUARDED_BY(volume_settings_critsect_);
-  bool previous_frame_muted_;  // Only accessed from PrepareEncodeAndSend().
-  float _panLeft GUARDED_BY(volume_settings_critsect_);
-  float _panRight GUARDED_BY(volume_settings_critsect_);
-  float _outputGain GUARDED_BY(volume_settings_critsect_);
   // VoeRTP_RTCP
   uint32_t _lastLocalTimeStamp;
   int8_t _lastPayloadType;
   bool _includeAudioLevelIndication;
   size_t transport_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
   size_t rtp_overhead_per_packet_ GUARDED_BY(overhead_per_packet_lock_);
   rtc::CriticalSection overhead_per_packet_lock_;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
   // VoEVideoSync
@@ skipping 14 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_