Remove the VoiceEngineObserver callback interface.

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.
  */
 
 #ifndef VOICE_ENGINE_TRANSMIT_MIXER_H_
 #define VOICE_ENGINE_TRANSMIT_MIXER_H_
 
 #include <memory>
 
 #include "common_audio/resampler/include/push_resampler.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/audio_processing/typing_detection.h"
 #include "modules/include/module_common_types.h"
 #include "rtc_base/criticalsection.h"
 #include "voice_engine/audio_level.h"
 #include "voice_engine/include/voe_base.h"
-#include "voice_engine/monitor_module.h"
 #include "voice_engine/voice_engine_defines.h"
 
 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1
 #else
 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0
 #endif
 
 namespace webrtc {
 class AudioProcessing;
 class ProcessThread;
 
 namespace voe {
 
 class ChannelManager;
 class MixedAudio;
 class Statistics;
 
 class TransmitMixer {
 public:
     static int32_t Create(TransmitMixer*& mixer, uint32_t instanceId);
 
     static void Destroy(TransmitMixer*& mixer);
 
-    int32_t SetEngineInformation(ProcessThread& processThread,
-                                 Statistics& engineStatistics,
-                                 ChannelManager& channelManager);
+    void SetEngineInformation(ChannelManager* channelManager);
 
-    int32_t SetAudioProcessingModule(
-        AudioProcessing* audioProcessingModule);
+    int32_t SetAudioProcessingModule(AudioProcessing* audioProcessingModule);
 
     int32_t PrepareDemux(const void* audioSamples,
                          size_t nSamples,
                          size_t nChannels,
                          uint32_t samplesPerSec,
                          uint16_t totalDelayMS,
                          int32_t  clockDrift,
                          uint16_t currentMicLevel,
                          bool keyPressed);
 
@@ skipping 11 matching lines @@
     virtual int16_t AudioLevelFullRange() const;
 
     // See description of "totalAudioEnergy" in the WebRTC stats spec:
     // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
     // 'virtual' to allow mocking.
     virtual double GetTotalInputEnergy() const;
 
     // 'virtual' to allow mocking.
     virtual double GetTotalInputDuration() const;
 
-    int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
-
     virtual ~TransmitMixer();
 
-#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    // Periodic callback from the MonitorModule.
-    void OnPeriodicProcess();
-#endif
-
   // Virtual to allow mocking.
   virtual void EnableStereoChannelSwapping(bool enable);
   bool IsStereoChannelSwappingEnabled();
 
+  // Virtual to allow mocking.
+  virtual bool typing_noise_detected() const;
+
 protected:
-#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    TransmitMixer() : _monitorModule(this) {}
-#else
     TransmitMixer() = default;
-#endif
 
 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);
 
     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);
+    void TypingDetection(bool key_pressed);
 #endif
 
     // uses
-    Statistics* _engineStatisticsPtr = nullptr;
     ChannelManager* _channelManagerPtr = nullptr;
     AudioProcessing* audioproc_ = nullptr;
-    VoiceEngineObserver* _voiceEngineObserverPtr = nullptr;
-    ProcessThread* _processThreadPtr = nullptr;
 
     // owns
     AudioFrame _audioFrame;
     PushResampler<int16_t> resampler_;  // ADM sample rate -> mixing rate
     voe::AudioLevel _audioLevel;
-    // protect file instances and their variables in MixedParticipants()
-    rtc::CriticalSection _critSect;
-    rtc::CriticalSection _callbackCritSect;
 
 #if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-    MonitorModule<TransmitMixer> _monitorModule;
-    webrtc::TypingDetection _typingDetection;
-    bool _typingNoiseWarningPending = false;
-    bool _typingNoiseDetected = false;
+    webrtc::TypingDetection typing_detection_;
 #endif
 
+    rtc::CriticalSection lock_;
+    bool typing_noise_detected_ RTC_GUARDED_BY(lock_) = false;
+
     int _instanceId = 0;
     uint32_t _captureLevel = 0;
     bool stereo_codec_ = false;
     bool swap_stereo_channels_ = false;
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // VOICE_ENGINE_TRANSMIT_MIXER_H_