diff options
Diffstat (limited to 'indra/llwebrtc/llwebrtc_impl.h')
| -rw-r--r-- | indra/llwebrtc/llwebrtc_impl.h | 414 |
1 files changed, 323 insertions, 91 deletions
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h index dfdb19e9be..7b23b11208 100644 --- a/indra/llwebrtc/llwebrtc_impl.h +++ b/indra/llwebrtc/llwebrtc_impl.h @@ -54,12 +54,12 @@ #include "rtc_base/ref_counted_object.h" #include "rtc_base/ssl_adapter.h" #include "rtc_base/thread.h" +#include "rtc_base/logging.h" #include "api/peer_connection_interface.h" #include "api/media_stream_interface.h" #include "api/create_peerconnection_factory.h" #include "modules/audio_device/include/audio_device.h" #include "modules/audio_device/include/audio_device_data_observer.h" -#include "rtc_base/task_queue.h" #include "api/task_queue/task_queue_factory.h" #include "api/task_queue/default_task_queue_factory.h" #include "modules/audio_device/include/audio_device_defines.h" @@ -69,35 +69,30 @@ namespace llwebrtc class LLWebRTCPeerConnectionImpl; -class LLWebRTCLogSink : public rtc::LogSink { +class LLWebRTCLogSink : public webrtc::LogSink +{ public: - LLWebRTCLogSink(LLWebRTCLogCallback* callback) : - mCallback(callback) - { - } + LLWebRTCLogSink(LLWebRTCLogCallback* callback) : mCallback(callback) {} // Destructor: close the log file - ~LLWebRTCLogSink() override - { - } + ~LLWebRTCLogSink() override {} - void OnLogMessage(const std::string& msg, - rtc::LoggingSeverity severity) override + void OnLogMessage(const std::string& msg, webrtc::LoggingSeverity severity) override { if (mCallback) { - switch(severity) + switch (severity) { - case rtc::LS_VERBOSE: + case webrtc::LS_VERBOSE: mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); break; - case rtc::LS_INFO: + case webrtc::LS_INFO: mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); break; - case rtc::LS_WARNING: + case webrtc::LS_WARNING: mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); break; - case rtc::LS_ERROR: + case webrtc::LS_ERROR: mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); break; default: @@ -118,67 +113,301 @@ private: LLWebRTCLogCallback* mCallback; }; -// Implements a class allowing capture of audio data -// to determine audio level of the microphone. -class LLAudioDeviceObserver : public webrtc::AudioDeviceDataObserver +// ----------------------------------------------------------------------------- +// A proxy transport that forwards capture data to two AudioTransport sinks: +// - the "engine" (libwebrtc's VoiceEngine) +// - the "user" (your app's listener) +// +// Playout (NeedMorePlayData) goes only to the engine by default to avoid +// double-writing into the output buffer. See notes below if you want a tap. +// ----------------------------------------------------------------------------- +class LLWebRTCAudioTransport : public webrtc::AudioTransport { - public: - LLAudioDeviceObserver(); - - // Retrieve the RMS audio loudness - float getMicrophoneEnergy(); - - // Data retrieved from the caputure device is - // passed in here for processing. - void OnCaptureData(const void *audio_samples, - const size_t num_samples, - const size_t bytes_per_sample, - const size_t num_channels, - const uint32_t samples_per_sec) override; - - // This is for data destined for the render device. - // not currently used. - void OnRenderData(const void *audio_samples, - const size_t num_samples, - const size_t bytes_per_sample, - const size_t num_channels, - const uint32_t samples_per_sec) override; +public: + LLWebRTCAudioTransport(); + + void SetEngineTransport(webrtc::AudioTransport* t); + + // -------- Capture path: fan out to both sinks -------- + int32_t RecordedDataIsAvailable(const void* audio_data, + size_t number_of_samples, + size_t bytes_per_sample, + size_t number_of_channels, + uint32_t samples_per_sec, + uint32_t total_delay_ms, + int32_t clock_drift, + uint32_t current_mic_level, + bool key_pressed, + uint32_t& new_mic_level) override; + + // -------- Playout path: delegate to engine only -------- + int32_t NeedMorePlayData(size_t number_of_samples, + size_t bytes_per_sample, + size_t number_of_channels, + uint32_t samples_per_sec, + void* audio_data, + size_t& number_of_samples_out, + int64_t* elapsed_time_ms, + int64_t* ntp_time_ms) override; + + // Method to pull mixed render audio data from all active VoE channels. + // The data will not be passed as reference for audio processing internally. + void PullRenderData(int bits_per_sample, + int sample_rate, + size_t number_of_channels, + size_t number_of_frames, + void* audio_data, + int64_t* elapsed_time_ms, + int64_t* ntp_time_ms) override; + + float GetMicrophoneEnergy() { return mMicrophoneEnergy.load(std::memory_order_relaxed); } + void SetGain(float gain) { mGain.store(gain, std::memory_order_relaxed); } + +private: + std::atomic<webrtc::AudioTransport*> engine_{ nullptr }; + static const int NUM_PACKETS_TO_FILTER = 30; // 300 ms of smoothing (30 frames) + float mSumVector[NUM_PACKETS_TO_FILTER]; + std::atomic<float> mMicrophoneEnergy; + std::atomic<float> mGain{ 0.0f }; - protected: - static const int NUM_PACKETS_TO_FILTER = 30; // 300 ms of smoothing (30 frames) - float mSumVector[NUM_PACKETS_TO_FILTER]; - float mMicrophoneEnergy; }; + +// ----------------------------------------------------------------------------- +// LLWebRTCAudioDeviceModule +// - Wraps a real ADM to provide microphone energy for tuning +// ----------------------------------------------------------------------------- +class LLWebRTCAudioDeviceModule : public webrtc::AudioDeviceModule +{ +public: + explicit LLWebRTCAudioDeviceModule(webrtc::scoped_refptr<webrtc::AudioDeviceModule> inner) : inner_(std::move(inner)), tuning_(false) + { + RTC_CHECK(inner_); + } + + // ----- AudioDeviceModule interface: we mostly forward to |inner_| ----- + int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override { return inner_->ActiveAudioLayer(audioLayer); } + + int32_t RegisterAudioCallback(webrtc::AudioTransport* engine_transport) override + { + // The engine registers its transport here. We put our audio transport between engine and ADM. + audio_transport_.SetEngineTransport(engine_transport); + // Register our proxy with the real ADM. + return inner_->RegisterAudioCallback(&audio_transport_); + } + + int32_t Init() override { return inner_->Init(); } + int32_t Terminate() override { return inner_->Terminate(); } + bool Initialized() const override { return inner_->Initialized(); } + + // --- Device enumeration/selection (forward) --- + int16_t PlayoutDevices() override { return inner_->PlayoutDevices(); } + int16_t RecordingDevices() override { return inner_->RecordingDevices(); } + int32_t PlayoutDeviceName(uint16_t index, char name[webrtc::kAdmMaxDeviceNameSize], char guid[webrtc::kAdmMaxGuidSize]) override + { + return inner_->PlayoutDeviceName(index, name, guid); + } + int32_t RecordingDeviceName(uint16_t index, char name[webrtc::kAdmMaxDeviceNameSize], char guid[webrtc::kAdmMaxGuidSize]) override + { + return inner_->RecordingDeviceName(index, name, guid); + } + int32_t SetPlayoutDevice(uint16_t index) override { return inner_->SetPlayoutDevice(index); } + int32_t SetRecordingDevice(uint16_t index) override { return inner_->SetRecordingDevice(index); } + + // Windows default/communications selectors, if your branch exposes them: + int32_t SetPlayoutDevice(WindowsDeviceType type) override { return inner_->SetPlayoutDevice(type); } + int32_t SetRecordingDevice(WindowsDeviceType type) override { return inner_->SetRecordingDevice(type); } + + // --- Init/start/stop (forward) --- + int32_t InitPlayout() override { return inner_->InitPlayout(); } + bool PlayoutIsInitialized() const override { return inner_->PlayoutIsInitialized(); } + int32_t StartPlayout() override { + if (tuning_) return 0; // For tuning, don't allow playout + return inner_->StartPlayout(); + } + int32_t StopPlayout() override { return inner_->StopPlayout(); } + bool Playing() const override { return inner_->Playing(); } + + int32_t InitRecording() override { return inner_->InitRecording(); } + bool RecordingIsInitialized() const override { return inner_->RecordingIsInitialized(); } + int32_t StartRecording() override { + // ignore start recording as webrtc.lib will + // send one when streams first connect, resulting + // in an inadvertant 'recording' when mute is on. + // We take full control of StartRecording via + // ForceStartRecording below. + return 0; + } + int32_t StopRecording() override { + if (tuning_) return 0; // if we're tuning, disregard the StopRecording we get from disabling the streams + return inner_->StopRecording(); + } + int32_t ForceStartRecording() { return inner_->StartRecording(); } + int32_t ForceStopRecording() { return inner_->StopRecording(); } + bool Recording() const override { return inner_->Recording(); } + + // --- Stereo opts (forward if available on your branch) --- + int32_t SetStereoPlayout(bool enable) override { return inner_->SetStereoPlayout(enable); } + int32_t SetStereoRecording(bool enable) override { return inner_->SetStereoRecording(enable); } + int32_t PlayoutIsAvailable(bool* available) override { return inner_->PlayoutIsAvailable(available); } + int32_t RecordingIsAvailable(bool* available) override { return inner_->RecordingIsAvailable(available); } + + // --- AGC/Volume/Mute/etc. (forward) --- + int32_t SetMicrophoneVolume(uint32_t volume) override { return inner_->SetMicrophoneVolume(volume); } + int32_t MicrophoneVolume(uint32_t* volume) const override { return inner_->MicrophoneVolume(volume); } + + // --- Speaker/Microphone init (forward) --- + int32_t InitSpeaker() override { return inner_->InitSpeaker(); } + bool SpeakerIsInitialized() const override { return inner_->SpeakerIsInitialized(); } + int32_t InitMicrophone() override { return inner_->InitMicrophone(); } + bool MicrophoneIsInitialized() const override { return inner_->MicrophoneIsInitialized(); } + + // --- Speaker Volume (forward) --- + int32_t SpeakerVolumeIsAvailable(bool* available) override { return inner_->SpeakerVolumeIsAvailable(available); } + int32_t SetSpeakerVolume(uint32_t volume) override { return inner_->SetSpeakerVolume(volume); } + int32_t SpeakerVolume(uint32_t* volume) const override { return inner_->SpeakerVolume(volume); } + int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return inner_->MaxSpeakerVolume(maxVolume); } + int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return inner_->MinSpeakerVolume(minVolume); } + + // --- Microphone Volume (forward) --- + int32_t MicrophoneVolumeIsAvailable(bool* available) override { return inner_->MicrophoneVolumeIsAvailable(available); } + int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return inner_->MaxMicrophoneVolume(maxVolume); } + int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return inner_->MinMicrophoneVolume(minVolume); } + + // --- Speaker Mute (forward) --- + int32_t SpeakerMuteIsAvailable(bool* available) override { return inner_->SpeakerMuteIsAvailable(available); } + int32_t SetSpeakerMute(bool enable) override { return inner_->SetSpeakerMute(enable); } + int32_t SpeakerMute(bool* enabled) const override { return inner_->SpeakerMute(enabled); } + + // --- Microphone Mute (forward) --- + int32_t MicrophoneMuteIsAvailable(bool* available) override { return inner_->MicrophoneMuteIsAvailable(available); } + int32_t SetMicrophoneMute(bool enable) override { return inner_->SetMicrophoneMute(enable); } + int32_t MicrophoneMute(bool* enabled) const override { return inner_->MicrophoneMute(enabled); } + + // --- Stereo Support (forward) --- + int32_t StereoPlayoutIsAvailable(bool* available) const override { return inner_->StereoPlayoutIsAvailable(available); } + int32_t StereoPlayout(bool* enabled) const override { return inner_->StereoPlayout(enabled); } + int32_t StereoRecordingIsAvailable(bool* available) const override { return inner_->StereoRecordingIsAvailable(available); } + int32_t StereoRecording(bool* enabled) const override { return inner_->StereoRecording(enabled); } + + // --- Delay/Timing (forward) --- + int32_t PlayoutDelay(uint16_t* delayMS) const override { return inner_->PlayoutDelay(delayMS); } + + // --- Built-in Audio Processing (forward) --- + bool BuiltInAECIsAvailable() const override { return inner_->BuiltInAECIsAvailable(); } + bool BuiltInAGCIsAvailable() const override { return inner_->BuiltInAGCIsAvailable(); } + bool BuiltInNSIsAvailable() const override { return inner_->BuiltInNSIsAvailable(); } + int32_t EnableBuiltInAEC(bool enable) override { return inner_->EnableBuiltInAEC(enable); } + int32_t EnableBuiltInAGC(bool enable) override { return inner_->EnableBuiltInAGC(enable); } + int32_t EnableBuiltInNS(bool enable) override { return inner_->EnableBuiltInNS(enable); } + + // --- Additional AudioDeviceModule methods (forward) --- + int32_t GetPlayoutUnderrunCount() const override { return inner_->GetPlayoutUnderrunCount(); } + + // Used to generate RTC stats. If not implemented, RTCAudioPlayoutStats will + // not be present in the stats. + std::optional<Stats> GetStats() const override { return inner_->GetStats(); } + +// Only supported on iOS. +#if defined(WEBRTC_IOS) + virtual int GetPlayoutAudioParameters(AudioParameters* params) const override { return inner_->GetPlayoutAudioParameters(params); } + virtual int GetRecordAudioParameters(AudioParameters* params) override { return inner_->GetRecordAudioParameters(params); } +#endif // WEBRTC_IOS + + virtual int32_t GetPlayoutDevice() const override { return inner_->GetPlayoutDevice(); } + virtual int32_t GetRecordingDevice() const override { return inner_->GetRecordingDevice(); } + virtual int32_t SetObserver(webrtc::AudioDeviceObserver* observer) override { return inner_->SetObserver(observer); } + + // tuning microphone energy calculations + float GetMicrophoneEnergy() { return audio_transport_.GetMicrophoneEnergy(); } + void SetTuningMicGain(float gain) { audio_transport_.SetGain(gain); } + void SetTuning(bool tuning, bool mute) + { + tuning_ = tuning; + if (tuning) + { + inner_->InitRecording(); + inner_->StartRecording(); + inner_->StopPlayout(); + } + else + { + if (mute) + { + inner_->StopRecording(); + } + else + { + inner_->InitRecording(); + inner_->StartRecording(); + } + inner_->StartPlayout(); + } + } + +protected: + ~LLWebRTCAudioDeviceModule() override = default; + +private: + webrtc::scoped_refptr<webrtc::AudioDeviceModule> inner_; + LLWebRTCAudioTransport audio_transport_; + + bool tuning_; +}; + +class LLCustomProcessorState +{ + +public: + float getMicrophoneEnergy() { return mMicrophoneEnergy.load(std::memory_order_relaxed); } + void setMicrophoneEnergy(float energy) { mMicrophoneEnergy.store(energy, std::memory_order_relaxed); } + + void setGain(float gain) + { + mGain.store(gain, std::memory_order_relaxed); + mDirty.store(true, std::memory_order_relaxed); + } + + float getGain() { return mGain.load(std::memory_order_relaxed); } + + bool getDirty() { return mDirty.exchange(false, std::memory_order_relaxed); } + + protected: + std::atomic<bool> mDirty{ true }; + std::atomic<float> mMicrophoneEnergy{ 0.0f }; + std::atomic<float> mGain{ 0.0f }; +}; + +using LLCustomProcessorStatePtr = std::shared_ptr<LLCustomProcessorState>; + // Used to process/retrieve audio levels after // all of the processing (AGC, AEC, etc.) for display in-world to the user. class LLCustomProcessor : public webrtc::CustomProcessing { - public: - LLCustomProcessor(); +public: + LLCustomProcessor(LLCustomProcessorStatePtr state); ~LLCustomProcessor() override {} // (Re-) Initializes the submodule. void Initialize(int sample_rate_hz, int num_channels) override; // Analyzes the given capture or render signal. - void Process(webrtc::AudioBuffer *audio) override; + void Process(webrtc::AudioBuffer* audio) override; // Returns a string representation of the module state. std::string ToString() const override { return ""; } - float getMicrophoneEnergy() { return mMicrophoneEnergy; } - - void setGain(float gain) { mGain = gain; } - - protected: - static const int NUM_PACKETS_TO_FILTER = 30; // 300 ms of smoothing - int mSampleRateHz; - int mNumChannels; +protected: + static const int NUM_PACKETS_TO_FILTER = 30; // 300 ms of smoothing + int mSampleRateHz{ 48000 }; + int mNumChannels{ 2 }; + int mRampFrames{ 2 }; + float mCurrentGain{ 0.0f }; + float mGainStep{ 0.0f }; float mSumVector[NUM_PACKETS_TO_FILTER]; - float mMicrophoneEnergy; - float mGain; + friend LLCustomProcessorState; + LLCustomProcessorStatePtr mState; }; @@ -187,7 +416,7 @@ class LLCustomProcessor : public webrtc::CustomProcessing #if CM_WEBRTC class LLWebRTCImpl : public LLWebRTCDeviceInterface #else -class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceSink +class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceObserver #endif { public: @@ -218,10 +447,15 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS float getTuningAudioLevel() override; float getPeerConnectionAudioLevel() override; - void setPeerConnectionGain(float gain) override; + void setMicGain(float gain) override; + void setTuningMicGain(float gain) override; + + void setMute(bool mute, int delay_ms = 20) override; + + void intSetMute(bool mute, int delay_ms = 20); // - // AudioDeviceSink + // AudioDeviceObserver // #if CM_WEBRTC void OnDevicesUpdated(); @@ -254,19 +488,19 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS mNetworkThread->PostTask(std::move(task), location); } - void WorkerBlockingCall(rtc::FunctionView<void()> functor, + void WorkerBlockingCall(webrtc::FunctionView<void()> functor, const webrtc::Location& location = webrtc::Location::Current()) { mWorkerThread->BlockingCall(std::move(functor), location); } - void SignalingBlockingCall(rtc::FunctionView<void()> functor, + void SignalingBlockingCall(webrtc::FunctionView<void()> functor, const webrtc::Location& location = webrtc::Location::Current()) { mSignalingThread->BlockingCall(std::move(functor), location); } - void NetworkBlockingCall(rtc::FunctionView<void()> functor, + void NetworkBlockingCall(webrtc::FunctionView<void()> functor, const webrtc::Location& location = webrtc::Location::Current()) { mNetworkThread->BlockingCall(std::move(functor), location); @@ -274,7 +508,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS // Allows the LLWebRTCPeerConnectionImpl class to retrieve the // native webrtc PeerConnectionFactory. - rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> getPeerConnectionFactory() + webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> getPeerConnectionFactory() { return mPeerConnectionFactory; } @@ -283,49 +517,47 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceS LLWebRTCPeerConnectionInterface* newPeerConnection(); void freePeerConnection(LLWebRTCPeerConnectionInterface* peer_connection); - // enables/disables capture via the capture device - void setRecording(bool recording); - - void setPlayout(bool playing); - protected: + + void workerDeployDevices(); LLWebRTCLogSink* mLogSink; // The native webrtc threads - std::unique_ptr<rtc::Thread> mNetworkThread; - std::unique_ptr<rtc::Thread> mWorkerThread; - std::unique_ptr<rtc::Thread> mSignalingThread; + std::unique_ptr<webrtc::Thread> mNetworkThread; + std::unique_ptr<webrtc::Thread> mWorkerThread; + std::unique_ptr<webrtc::Thread> mSignalingThread; // The factory that allows creation of native webrtc PeerConnections. - rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory; + webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory; - rtc::scoped_refptr<webrtc::AudioProcessing> mAudioProcessingModule; + webrtc::scoped_refptr<webrtc::AudioProcessing> mAudioProcessingModule; // more native webrtc stuff - std::unique_ptr<webrtc::TaskQueueFactory> mTaskQueueFactory; + std::unique_ptr<webrtc::TaskQueueFactory> mTaskQueueFactory; // Devices void updateDevices(); - rtc::scoped_refptr<webrtc::AudioDeviceModule> mTuningDeviceModule; - rtc::scoped_refptr<webrtc::AudioDeviceModule> mPeerDeviceModule; + void deployDevices(); + std::atomic<int> mDevicesDeploying; + webrtc::scoped_refptr<LLWebRTCAudioDeviceModule> mDeviceModule; std::vector<LLWebRTCDevicesObserver *> mVoiceDevicesObserverList; // accessors in native webrtc for devices aren't apparently implemented yet. bool mTuningMode; - int32_t mRecordingDevice; + std::string mRecordingDevice; LLWebRTCVoiceDeviceList mRecordingDeviceList; - int32_t mPlayoutDevice; + std::string mPlayoutDevice; LLWebRTCVoiceDeviceList mPlayoutDeviceList; bool mMute; + float mGain; - LLAudioDeviceObserver * mTuningAudioDeviceObserver; - LLCustomProcessor * mPeerCustomProcessor; + LLCustomProcessorStatePtr mPeerCustomProcessor; // peer connections - std::vector<rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>> mPeerConnections; + std::vector<webrtc::scoped_refptr<LLWebRTCPeerConnectionImpl>> mPeerConnections; }; @@ -350,7 +582,7 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnectionInterface, void terminate(); virtual void AddRef() const override = 0; - virtual rtc::RefCountReleaseStatus Release() const override = 0; + virtual webrtc::RefCountReleaseStatus Release() const override = 0; // // LLWebRTCPeerConnection @@ -381,10 +613,10 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnectionInterface, // void OnSignalingChange(webrtc::PeerConnectionInterface::SignalingState new_state) override {} - void OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, - const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams) override; - void OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override; - void OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override; + void OnAddTrack(webrtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver, + const std::vector<webrtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams) override; + void OnRemoveTrack(webrtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override; + void OnDataChannel(webrtc::scoped_refptr<webrtc::DataChannelInterface> channel) override; void OnRenegotiationNeeded() override {} void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}; void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state) override; @@ -423,7 +655,7 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnectionInterface, LLWebRTCImpl * mWebRTCImpl; - rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory; + webrtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory; typedef enum { MUTE_INITIAL, @@ -437,12 +669,12 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnectionInterface, std::vector<std::unique_ptr<webrtc::IceCandidateInterface>> mCachedIceCandidates; bool mAnswerReceived; - rtc::scoped_refptr<webrtc::PeerConnectionInterface> mPeerConnection; - rtc::scoped_refptr<webrtc::MediaStreamInterface> mLocalStream; + webrtc::scoped_refptr<webrtc::PeerConnectionInterface> mPeerConnection; + webrtc::scoped_refptr<webrtc::MediaStreamInterface> mLocalStream; // data std::vector<LLWebRTCDataObserver *> mDataObserverList; - rtc::scoped_refptr<webrtc::DataChannelInterface> mDataChannel; + webrtc::scoped_refptr<webrtc::DataChannelInterface> mDataChannel; }; } |