Removing the RX processing APIs from VoEAudioProcessing:

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 65 matching lines @@
 class VoERtcpObserver;
 
 // Helper class to simplify locking scheme for members that are accessed from
 // multiple threads.
 // Example: a member can be set on thread T1 and read by an internal audio
 // thread T2. Accessing the member via this class ensures that we are
 // safe and also avoid TSan v2 warnings.
 class ChannelState {
  public:
   struct State {
-    State()
-        : rx_apm_is_enabled(false),
-          input_external_media(false),
-          output_file_playing(false),
-          input_file_playing(false),
-          playing(false),
-          sending(false),
-          receiving(false) {}
-
-    bool rx_apm_is_enabled;
-    bool input_external_media;
-    bool output_file_playing;
-    bool input_file_playing;
-    bool playing;
-    bool sending;
-    bool receiving;
+    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();
   }
 
   State Get() const {
     rtc::CritScope lock(&lock_);
     return state_;
   }
 
-  void SetRxApmIsEnabled(bool enable) {
-    rtc::CritScope lock(&lock_);
-    state_.rx_apm_is_enabled = enable;
-  }
-
   void SetInputExternalMedia(bool enable) {
     rtc::CritScope lock(&lock_);
     state_.input_external_media = enable;
   }
 
   void SetOutputFilePlaying(bool enable) {
     rtc::CritScope lock(&lock_);
     state_.output_file_playing = enable;
   }
 
@@ skipping 169 matching lines @@
   int SetInitSequenceNumber(short sequenceNumber);
 
   // VoEVideoSyncExtended
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
   // DTMF
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type);
 
   // VoEAudioProcessingImpl
-  int UpdateRxVadDetection(AudioFrame& audioFrame);
-  int RegisterRxVadObserver(VoERxVadCallback& observer);
-  int DeRegisterRxVadObserver();
   int VoiceActivityIndicator(int& activity);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
-  int SetRxAgcStatus(bool enable, AgcModes mode);
-  int GetRxAgcStatus(bool& enabled, AgcModes& mode);
-  int SetRxAgcConfig(AgcConfig config);
-  int GetRxAgcConfig(AgcConfig& config);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NR
-  int SetRxNsStatus(bool enable, NsModes mode);
-  int GetRxNsStatus(bool& enabled, NsModes& mode);
-#endif
 
   // VoERTP_RTCP
   int SetLocalSSRC(unsigned int ssrc);
   int GetLocalSSRC(unsigned int& ssrc);
   int GetRemoteSSRC(unsigned int& ssrc);
   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetReceiveAudioLevelIndicationStatus(bool enable, unsigned char id);
   int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
   int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
   void EnableSendTransportSequenceNumber(int id);
@@ skipping 33 matching lines @@
   int32_t SendData(FrameType frameType,
                    uint8_t payloadType,
                    uint32_t timeStamp,
                    const uint8_t* payloadData,
                    size_t payloadSize,
                    const RTPFragmentationHeader* fragmentation) override;
 
   // From ACMVADCallback in the ACM
   int32_t InFrameType(FrameType frame_type) override;
 
-  int32_t OnRxVadDetected(int vadDecision);
-
   // From RtpData in the RTP/RTCP module
   int32_t OnReceivedPayloadData(const uint8_t* payloadData,
                                 size_t payloadSize,
                                 const WebRtcRTPHeader* rtpHeader) override;
   bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
 
   // From RtpFeedback in the RTP/RTCP module
   int32_t OnInitializeDecoder(int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               int frequency,
@@ skipping 144 matching lines @@
   // uses
   Statistics* _engineStatisticsPtr;
   OutputMixer* _outputMixerPtr;
   TransmitMixer* _transmitMixerPtr;
   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_;
-  std::unique_ptr<AudioProcessing> rx_audioproc_;  // far end AudioProcessing
-  VoERxVadCallback* _rxVadObserverPtr;
-  int32_t _oldVadDecision;
   int32_t _sendFrameType;  // Send data is voice, 1-voice, 0-otherwise
   // VoEBase
   bool _externalMixing;
   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;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
   // VoEVideoSync
   rtc::CriticalSection video_sync_lock_;
   // VoEAudioProcessing
-  bool _RxVadDetection;
-  bool _rxAgcIsEnabled;
-  bool _rxNsIsEnabled;
   bool restored_packet_in_use_;
   // RtcpBandwidthObserver
   std::unique_ptr<VoERtcpObserver> rtcp_observer_;
   std::unique_ptr<NetworkPredictor> network_predictor_;
   // An associated send channel.
   rtc::CriticalSection assoc_send_channel_lock_;
   ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 
   bool pacing_enabled_;
   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_;
   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_