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+/**
+ * @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;
+
+
+// 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();
+ ~LLWebRTCImpl() {}
+
+ 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);
+
+ protected:
+ // 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