Remove usage of VoEAudioProcessing from WVoE/MC.

webrtc/voice_engine/transmit_mixer.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 129 matching lines @@
 
     int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
 
     virtual ~TransmitMixer();
 
     // MonitorObserver
     void OnPeriodicProcess();
 
 
     // FileCallback
-    void PlayNotification(int32_t id,
-                          uint32_t durationMs);
+    void PlayNotification(const int32_t id,

[hlundin-webrtc, 2017/02/13 21:03:57]

Why the consts?

[the sun, 2017/02/13 23:34:47]

Because that's how the FileCallback interface defines them. This is going away
"Real Soon"(tm), and changing the impl here was simply less work than changing
the interface.

[hlundin-webrtc, 2017/02/14 07:50:43]

Acknowledged.

+                          const uint32_t durationMs);
 
-    void RecordNotification(int32_t id,
-                            uint32_t durationMs);
+    void RecordNotification(const int32_t id,
+                            const uint32_t durationMs);
 
-    void PlayFileEnded(int32_t id);
+    void PlayFileEnded(const int32_t id);
 
-    void RecordFileEnded(int32_t id);
+    void RecordFileEnded(const int32_t id);
 
 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     // Typing detection
     int TimeSinceLastTyping(int &seconds);
     int SetTypingDetectionParameters(int timeWindow,
                                      int costPerTyping,
                                      int reportingThreshold,
                                      int penaltyDecay,
                                      int typeEventDelay);
 #endif
 
-  void EnableStereoChannelSwapping(bool enable);
+  // Virtual to allow mocking.
+  virtual void EnableStereoChannelSwapping(bool enable);
   bool IsStereoChannelSwappingEnabled();
 
+protected:
+    TransmitMixer() = default;
+
 private:
     TransmitMixer(uint32_t instanceId);
 
     // Gets the maximum sample rate and number of channels over all currently
     // sending codecs.
     void GetSendCodecInfo(int* max_sample_rate, size_t* max_channels);
 
     void GenerateAudioFrame(const int16_t audioSamples[],
                             size_t nSamples,
                             size_t nChannels,
                             int samplesPerSec);
     int32_t RecordAudioToFile(uint32_t mixingFrequency);
 
     int32_t MixOrReplaceAudioWithFile(
         int mixingFrequency);
 
     void ProcessAudio(int delay_ms, int clock_drift, int current_mic_level,
                       bool key_pressed);
 
 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     void TypingDetection(bool keyPressed);
 #endif
 
     // uses
-    Statistics* _engineStatisticsPtr;
-    ChannelManager* _channelManagerPtr;
-    AudioProcessing* audioproc_;
-    VoiceEngineObserver* _voiceEngineObserverPtr;
-    ProcessThread* _processThreadPtr;
+    Statistics* _engineStatisticsPtr = nullptr;
+    ChannelManager* _channelManagerPtr = nullptr;
+    AudioProcessing* audioproc_ = nullptr;
+    VoiceEngineObserver* _voiceEngineObserverPtr = nullptr;
+    ProcessThread* _processThreadPtr = nullptr;
 
     // owns
     MonitorModule _monitorModule;
     AudioFrame _audioFrame;
     PushResampler<int16_t> resampler_;  // ADM sample rate -> mixing rate
     std::unique_ptr<FilePlayer> file_player_;
     std::unique_ptr<FileRecorder> file_recorder_;
     std::unique_ptr<FileRecorder> file_call_recorder_;
-    int _filePlayerId;
-    int _fileRecorderId;
-    int _fileCallRecorderId;
-    bool _filePlaying;
-    bool _fileRecording;
-    bool _fileCallRecording;
+    int _filePlayerId = 0;
+    int _fileRecorderId = 0;
+    int _fileCallRecorderId = 0;
+    bool _filePlaying = false;
+    bool _fileRecording = false;
+    bool _fileCallRecording = false;
     voe::AudioLevel _audioLevel;
     // protect file instances and their variables in MixedParticipants()
     rtc::CriticalSection _critSect;
     rtc::CriticalSection _callbackCritSect;
 
 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     webrtc::TypingDetection _typingDetection;
-    bool _typingNoiseWarningPending;
-    bool _typingNoiseDetected;
+    bool _typingNoiseWarningPending = false;
+    bool _typingNoiseDetected = false;
 #endif
-    bool _saturationWarning;
+    bool _saturationWarning = false;
 
-    int _instanceId;
-    bool _mixFileWithMicrophone;
-    uint32_t _captureLevel;
-    VoEMediaProcess* external_postproc_ptr_;
-    VoEMediaProcess* external_preproc_ptr_;
-    bool _mute;
-    bool stereo_codec_;
-    bool swap_stereo_channels_;
+    int _instanceId = 0;
+    bool _mixFileWithMicrophone = false;
+    uint32_t _captureLevel = 0;
+    VoEMediaProcess* external_postproc_ptr_ = nullptr;
+    VoEMediaProcess* external_preproc_ptr_ = nullptr;
+    bool _mute = false;
+    bool stereo_codec_ = false;
+    bool swap_stereo_channels_ = false;
 };
-
 }  // namespace voe
-
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H