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Diffstat (limited to 'indra/llwebrtc/llwebrtc_impl.h')
| -rw-r--r-- | indra/llwebrtc/llwebrtc_impl.h | 444 |
1 files changed, 444 insertions, 0 deletions
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h new file mode 100644 index 0000000000..6543b2718c --- /dev/null +++ b/indra/llwebrtc/llwebrtc_impl.h @@ -0,0 +1,444 @@ +/** + * @file llwebrtc_impl.h + * @brief WebRTC dynamic library implementation header + * + * $LicenseInfo:firstyear=2023&license=viewerlgpl$ + * Second Life Viewer Source Code + * Copyright (C) 2023, Linden Research, Inc. + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public + * License as published by the Free Software Foundation; + * version 2.1 of the License only. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + * + * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA + * $/LicenseInfo$ + */ + +#ifndef LLWEBRTC_IMPL_H +#define LLWEBRTC_IMPL_H + +#define LL_MAKEDLL +#if defined(_WIN32) || defined(_WIN64) +#define WEBRTC_WIN 1 +#elif defined(__APPLE__) +#define WEBRTC_MAC 1 +#define WEBRTC_POSIX 1 +#elif __linux__ +#define WEBRTC_LINUX 1 +#define WEBRTC_POSIX 1 +#endif + +#include "llwebrtc.h" +// WebRTC Includes +#ifdef WEBRTC_WIN +#pragma warning(disable : 4996) // ignore 'deprecated.' We don't use the functions marked + // deprecated in the webrtc headers, but msvc complains anyway. + // Clang doesn't, and that's generally what webrtc uses. +#pragma warning(disable : 4068) // ignore 'invalid pragma.' There are clang pragma's in + // the webrtc headers, which msvc doesn't recognize. +#endif // WEBRTC_WIN + +#include "api/scoped_refptr.h" +#include "rtc_base/ref_count.h" +#include "rtc_base/ref_counted_object.h" +#include "rtc_base/ssl_adapter.h" +#include "rtc_base/thread.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" + + +namespace llwebrtc +{ + +class LLWebRTCPeerConnectionImpl; + +class LLWebRTCLogSink : public rtc::LogSink { +public: + LLWebRTCLogSink(LLWebRTCLogCallback* callback) : + mCallback(callback) + { + } + + // Destructor: close the log file + ~LLWebRTCLogSink() override + { + } + + void OnLogMessage(const std::string& msg, + rtc::LoggingSeverity severity) override + { + if (mCallback) + { + switch(severity) + { + case rtc::LS_VERBOSE: + mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); + break; + case rtc::LS_INFO: + mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); + break; + case rtc::LS_WARNING: + mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); + break; + case rtc::LS_ERROR: + mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, msg); + break; + default: + break; + } + } + } + + void OnLogMessage(const std::string& message) override + { + if (mCallback) + { + mCallback->LogMessage(LLWebRTCLogCallback::LOG_LEVEL_VERBOSE, message); + } + } + +private: + LLWebRTCLogCallback* mCallback; +}; + +// Implements a class allowing capture of audio data +// to determine audio level of the microphone. +class LLAudioDeviceObserver : public webrtc::AudioDeviceDataObserver +{ + 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; + + protected: + static const int NUM_PACKETS_TO_FILTER = 30; // 300 ms of smoothing (30 frames) + float mSumVector[NUM_PACKETS_TO_FILTER]; + float mMicrophoneEnergy; +}; + +// 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(); + ~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; + + // 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; + + float mSumVector[NUM_PACKETS_TO_FILTER]; + float mMicrophoneEnergy; + float mGain; +}; + + +// Primary singleton implementation for interfacing +// with the native webrtc library. +class LLWebRTCImpl : public LLWebRTCDeviceInterface +#if __x86_64__ && !__FreeBSD__ + , public webrtc::AudioDeviceSink +#endif +{ + public: + LLWebRTCImpl(LLWebRTCLogCallback* logCallback); + ~LLWebRTCImpl() + { + delete mLogSink; + } + + void init(); + void terminate(); + + // + // LLWebRTCDeviceInterface + // + + void setAudioConfig(LLWebRTCDeviceInterface::AudioConfig config = LLWebRTCDeviceInterface::AudioConfig()) override; + + void refreshDevices() override; + + void setDevicesObserver(LLWebRTCDevicesObserver *observer) override; + void unsetDevicesObserver(LLWebRTCDevicesObserver *observer) override; + + void setCaptureDevice(const std::string& id) override; + void setRenderDevice(const std::string& id) override; + + void setTuningMode(bool enable) override; + float getTuningAudioLevel() override; + float getPeerConnectionAudioLevel() override; + + void setPeerConnectionGain(float gain) override; + + // + // AudioDeviceSink + // + void OnDevicesUpdated() +#if __x86_64__ && !__FreeBSD__ + override +#endif + ; + + // + // Helpers + // + + // The following thread helpers allow the + // LLWebRTCPeerConnectionImpl class to post + // tasks to the native webrtc threads. + void PostWorkerTask(absl::AnyInvocable<void() &&> task, + const webrtc::Location& location = webrtc::Location::Current()) + { + mWorkerThread->PostTask(std::move(task), location); + } + + void PostSignalingTask(absl::AnyInvocable<void() &&> task, + const webrtc::Location& location = webrtc::Location::Current()) + { + mSignalingThread->PostTask(std::move(task), location); + } + + void PostNetworkTask(absl::AnyInvocable<void() &&> task, + const webrtc::Location& location = webrtc::Location::Current()) + { + mNetworkThread->PostTask(std::move(task), location); + } + + void WorkerBlockingCall(rtc::FunctionView<void()> functor, + const webrtc::Location& location = webrtc::Location::Current()) + { + mWorkerThread->BlockingCall(std::move(functor), location); + } + + void SignalingBlockingCall(rtc::FunctionView<void()> functor, + const webrtc::Location& location = webrtc::Location::Current()) + { + mSignalingThread->BlockingCall(std::move(functor), location); + } + + void NetworkBlockingCall(rtc::FunctionView<void()> functor, + const webrtc::Location& location = webrtc::Location::Current()) + { + mNetworkThread->BlockingCall(std::move(functor), location); + } + + // Allows the LLWebRTCPeerConnectionImpl class to retrieve the + // native webrtc PeerConnectionFactory. + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> getPeerConnectionFactory() + { + return mPeerConnectionFactory; + } + + // create or destroy a peer connection. + LLWebRTCPeerConnectionInterface* newPeerConnection(); + void freePeerConnection(LLWebRTCPeerConnectionInterface* peer_connection); + + // enables/disables capture via the capture device + void setRecording(bool recording); + + void setPlayout(bool playing); + + protected: + LLWebRTCLogSink* mLogSink; + + // The native webrtc threads + std::unique_ptr<rtc::Thread> mNetworkThread; + std::unique_ptr<rtc::Thread> mWorkerThread; + std::unique_ptr<rtc::Thread> mSignalingThread; + + // The factory that allows creation of native webrtc PeerConnections. + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory; + + rtc::scoped_refptr<webrtc::AudioProcessing> mAudioProcessingModule; + + // more native webrtc stuff + std::unique_ptr<webrtc::TaskQueueFactory> mTaskQueueFactory; + + + // Devices + void updateDevices(); + rtc::scoped_refptr<webrtc::AudioDeviceModule> mTuningDeviceModule; + rtc::scoped_refptr<webrtc::AudioDeviceModule> mPeerDeviceModule; + std::vector<LLWebRTCDevicesObserver *> mVoiceDevicesObserverList; + + // accessors in native webrtc for devices aren't apparently implemented yet. + bool mTuningMode; + int32_t mRecordingDevice; + LLWebRTCVoiceDeviceList mRecordingDeviceList; + + int32_t mPlayoutDevice; + LLWebRTCVoiceDeviceList mPlayoutDeviceList; + + bool mMute; + + LLAudioDeviceObserver * mTuningAudioDeviceObserver; + LLCustomProcessor * mPeerCustomProcessor; + + // peer connections + std::vector<rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>> mPeerConnections; +}; + + +// The implementation of a peer connection, which contains +// the various interfaces used by the viewer to interact with +// the webrtc connection. +class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnectionInterface, + public LLWebRTCAudioInterface, + public LLWebRTCDataInterface, + public webrtc::PeerConnectionObserver, + public webrtc::CreateSessionDescriptionObserver, + public webrtc::SetRemoteDescriptionObserverInterface, + public webrtc::SetLocalDescriptionObserverInterface, + public webrtc::DataChannelObserver + +{ + public: + LLWebRTCPeerConnectionImpl(); + ~LLWebRTCPeerConnectionImpl(); + + void init(LLWebRTCImpl * webrtc_impl); + void terminate(); + + virtual void AddRef() const override = 0; + virtual rtc::RefCountReleaseStatus Release() const override = 0; + + // + // LLWebRTCPeerConnection + // + bool initializeConnection(const InitOptions& options) override; + bool shutdownConnection() override; + + void setSignalingObserver(LLWebRTCSignalingObserver *observer) override; + void unsetSignalingObserver(LLWebRTCSignalingObserver *observer) override; + void AnswerAvailable(const std::string &sdp) override; + + // + // LLWebRTCAudioInterface + // + void setMute(bool mute) override; + void setReceiveVolume(float volume) override; // volume between 0.0 and 1.0 + void setSendVolume(float volume) override; // volume between 0.0 and 1.0 + + // + // LLWebRTCDataInterface + // + void sendData(const std::string& data, bool binary=false) override; + void setDataObserver(LLWebRTCDataObserver *observer) override; + void unsetDataObserver(LLWebRTCDataObserver *observer) override; + + // + // PeerConnectionObserver implementation. + // + + 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 OnRenegotiationNeeded() override {} + void OnIceConnectionChange(webrtc::PeerConnectionInterface::IceConnectionState new_state) override {}; + void OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state) override; + void OnIceCandidate(const webrtc::IceCandidateInterface *candidate) override; + void OnIceConnectionReceivingChange(bool receiving) override {} + void OnConnectionChange(webrtc::PeerConnectionInterface::PeerConnectionState new_state) override; + + // + // CreateSessionDescriptionObserver implementation. + // + void OnSuccess(webrtc::SessionDescriptionInterface *desc) override; + void OnFailure(webrtc::RTCError error) override; + + // + // SetRemoteDescriptionObserverInterface implementation. + // + void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override; + + // + // SetLocalDescriptionObserverInterface implementation. + // + void OnSetLocalDescriptionComplete(webrtc::RTCError error) override; + + // + // DataChannelObserver implementation. + // + void OnStateChange() override; + void OnMessage(const webrtc::DataBuffer& buffer) override; + + // Helpers + void resetMute(); + void enableSenderTracks(bool enable); + void enableReceiverTracks(bool enable); + + protected: + + LLWebRTCImpl * mWebRTCImpl; + + rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> mPeerConnectionFactory; + + bool mMute; + + // signaling + std::vector<LLWebRTCSignalingObserver *> mSignalingObserverList; + std::vector<std::unique_ptr<webrtc::IceCandidateInterface>> mCachedIceCandidates; + bool mAnswerReceived; + + rtc::scoped_refptr<webrtc::PeerConnectionInterface> mPeerConnection; + rtc::scoped_refptr<webrtc::MediaStreamInterface> mLocalStream; + + // data + std::vector<LLWebRTCDataObserver *> mDataObserverList; + rtc::scoped_refptr<webrtc::DataChannelInterface> mDataChannel; +}; + +} + +#endif // LLWEBRTC_IMPL_H |