| Index: voice_engine/transmit_mixer.h
|
| diff --git a/voice_engine/transmit_mixer.h b/voice_engine/transmit_mixer.h
|
| index a04f92b6bfc6428ffea511842ae6918092d7c68d..f26359553784f71010b521f7d07473d899216275 100644
|
| --- a/voice_engine/transmit_mixer.h
|
| +++ b/voice_engine/transmit_mixer.h
|
| @@ -20,7 +20,6 @@
|
| #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)
|
| @@ -45,12 +44,9 @@ public:
|
|
|
| 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,
|
| @@ -82,25 +78,17 @@ public:
|
| // '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);
|
| @@ -118,31 +106,25 @@ private:
|
| 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;
|
|
|