Remove saturation warning support from TransmitMixer.

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.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 #define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
 
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_processing/typing_detection.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/voice_engine/file_player.h"
 #include "webrtc/voice_engine/file_recorder.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/level_indicator.h"
 #include "webrtc/voice_engine/monitor_module.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
+// Should this be turned on in Chromium as well?

[hlundin-webrtc, 2017/02/28 12:56:47]

I believe it is.

 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 1
 #else
 #define WEBRTC_VOICE_ENGINE_TYPING_DETECTION 0
 #endif
 
 namespace webrtc {
 
 class AudioProcessing;
 class ProcessThread;
 
@@ skipping 86 matching lines @@
     int StartRecordingCall(OutStream* stream, const CodecInst* codecInst);
 
     int StopRecordingCall();
 
     void SetMixWithMicStatus(bool mix);
 
     int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
 
     virtual ~TransmitMixer();
 
+#if WEBRTC_VOICE_ENGINE_TYPING_DETECTION
     // Periodic callback from the MonitorModule.
     void OnPeriodicProcess();
+#endif
 
     // FileCallback
     void PlayNotification(const int32_t id,
                           const uint32_t durationMs);
 
     void RecordNotification(const int32_t id,
                             const uint32_t durationMs);
 
     void PlayFileEnded(const 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
 
   // Virtual to allow mocking.
   virtual void EnableStereoChannelSwapping(bool enable);
   bool IsStereoChannelSwappingEnabled();
 
 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,
@@ skipping 12 matching lines @@
 #endif
 
     // uses
     Statistics* _engineStatisticsPtr = nullptr;
     ChannelManager* _channelManagerPtr = nullptr;
     AudioProcessing* audioproc_ = nullptr;
     VoiceEngineObserver* _voiceEngineObserverPtr = nullptr;
     ProcessThread* _processThreadPtr = nullptr;
 
     // owns
-    MonitorModule<TransmitMixer> _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 = 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
+    MonitorModule<TransmitMixer> _monitorModule;
     webrtc::TypingDetection _typingDetection;
     bool _typingNoiseWarningPending = false;
     bool _typingNoiseDetected = false;
 #endif
-    bool _saturationWarning = false;
 
     int _instanceId = 0;
     bool _mixFileWithMicrophone = false;
     uint32_t _captureLevel = 0;
     bool _mute = false;
     bool stereo_codec_ = false;
     bool swap_stereo_channels_ = false;
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H