/** * @file llwebrtc.h * @brief WebRTC interface * * $LicenseInfo:firstyear=2023&license=viewerlgpl$ * 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 tSoftware * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA * * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA * $/LicenseInfo$ */ /* * llwebrtc wraps the native webrtc c++ library in a dynamic library with a simlified interface * so that the viewer can use it. This is done because native webrtc has a different * overall threading model than the viewer. * The native webrtc library is also compiled with clang, and has memory management * functions that conflict namespace-wise with those in the viewer. * * Due to these differences, code from the viewer cannot be pulled in to this * dynamic library, so it remains very simple. */ #ifndef LLWEBRTC_H #define LLWEBRTC_H #include #include #ifdef LL_MAKEDLL #ifdef WEBRTC_WIN #define LLSYMEXPORT __declspec(dllexport) #elif WEBRTC_LINUX #define LLSYMEXPORT __attribute__((visibility("default"))) #else #define LLSYMEXPORT /**/ #endif #else #define LLSYMEXPORT /**/ #endif // LL_MAKEDLL namespace llwebrtc { // LLWebRTCVoiceDevice is a simple representation of the // components of a device, used to communicate this // information to the viewer. // A note on threading. // Native WebRTC has it's own threading model. Some discussion // can be found here (https://webrtc.github.io/webrtc-org/native-code/native-apis/) // // Note that all callbacks to observers will occurr on one of the WebRTC native threads // (signaling, worker, etc.) Care should be taken to assure there are not // bad interactions with the viewer threads. class LLWebRTCVoiceDevice { public: std::string mDisplayName; // friendly name for user interface purposes std::string mID; // internal value for selection LLWebRTCVoiceDevice(const std::string &display_name, const std::string &id) : mDisplayName(display_name), mID(id) {}; }; typedef std::vector LLWebRTCVoiceDeviceList; // The LLWebRTCDeviceObserver should be implemented by the viewer // webrtc module, which will receive notifications when devices // change (are unplugged, etc.) class LLWebRTCDevicesObserver { public: virtual void OnDevicesChanged(const LLWebRTCVoiceDeviceList &render_devices, const LLWebRTCVoiceDeviceList &capture_devices) = 0; }; // The LLWebRTCDeviceInterface provides a way for the viewer // to enumerate, set, and get notifications of changes // for both capture (microphone) and render (speaker) // devices. class LLWebRTCDeviceInterface { public: struct AudioConfig { bool mAGC { true }; bool mEchoCancellation { true }; // TODO: The various levels of noise suppression are configured // on the APM which would require setting config on the APM. // We should pipe the various values through // later. typedef enum { NOISE_SUPPRESSION_LEVEL_NONE = 0, NOISE_SUPPRESSION_LEVEL_LOW, NOISE_SUPPRESSION_LEVEL_MODERATE, NOISE_SUPPRESSION_LEVEL_HIGH, NOISE_SUPPRESSION_LEVEL_VERY_HIGH } ENoiseSuppressionLevel; ENoiseSuppressionLevel mNoiseSuppressionLevel { NOISE_SUPPRESSION_LEVEL_VERY_HIGH }; }; virtual void setAudioConfig(AudioConfig config) = 0; // instructs webrtc to refresh the device list. virtual void refreshDevices() = 0; // set the capture and render devices using the unique identifier for the device virtual void setCaptureDevice(const std::string& id) = 0; virtual void setRenderDevice(const std::string& id) = 0; // Device observers for device change callbacks. virtual void setDevicesObserver(LLWebRTCDevicesObserver *observer) = 0; virtual void unsetDevicesObserver(LLWebRTCDevicesObserver *observer) = 0; // tuning and audio levels virtual void setTuningMode(bool enable) = 0; virtual float getTuningAudioLevel() = 0; // for use during tuning virtual float getPeerConnectionAudioLevel() = 0; // for use when not tuning }; // LLWebRTCAudioInterface provides the viewer with a way // to set audio characteristics (mute, send and receive volume) class LLWebRTCAudioInterface { public: virtual void setMute(bool mute) = 0; virtual void setReceiveVolume(float volume) = 0; // volume between 0.0 and 1.0 virtual void setSendVolume(float volume) = 0; // volume between 0.0 and 1.0 }; // LLWebRTCDataObserver allows the viewer voice module to be notified when // data is received over the data channel. class LLWebRTCDataObserver { public: virtual void OnDataReceived(const std::string& data, bool binary) = 0; }; // LLWebRTCDataInterface allows the viewer to send data over the data channel. class LLWebRTCDataInterface { public: virtual void sendData(const std::string& data, bool binary=false) = 0; virtual void setDataObserver(LLWebRTCDataObserver *observer) = 0; virtual void unsetDataObserver(LLWebRTCDataObserver *observer) = 0; }; // LLWebRTCIceCandidate is a basic structure containing // information needed for ICE trickling. struct LLWebRTCIceCandidate { std::string mCandidate; std::string mSdpMid; int mMLineIndex; }; // LLWebRTCSignalingObserver provides a way for the native // webrtc library to notify the viewer voice module of // various state changes. class LLWebRTCSignalingObserver { public: typedef enum e_ice_gathering_state { ICE_GATHERING_NEW, ICE_GATHERING_GATHERING, ICE_GATHERING_COMPLETE } EIceGatheringState; // Called when ICE gathering states have changed. // This may be called at any time, as ICE gathering // can be redone while a connection is up. virtual void OnIceGatheringState(EIceGatheringState state) = 0; // Called when a new ice candidate is available. virtual void OnIceCandidate(const LLWebRTCIceCandidate& candidate) = 0; // Called when an offer is available after a connection is requested. virtual void OnOfferAvailable(const std::string& sdp) = 0; // Called when a connection enters a failure state and renegotiation is needed. virtual void OnRenegotiationNeeded() = 0; // Called when the audio channel has been established and audio // can begin. virtual void OnAudioEstablished(LLWebRTCAudioInterface *audio_interface) = 0; // Called when the data channel has been established and data // transfer can begin. virtual void OnDataChannelReady(LLWebRTCDataInterface *data_interface) = 0; }; // LLWebRTCPeerConnectionInterface representsd a connection to a peer, // in most cases a Secondlife WebRTC server. This interface // allows for management of this peer connection. class LLWebRTCPeerConnectionInterface { public: struct InitOptions { // equivalent of PeerConnectionInterface::IceServer struct IceServers { // Valid formats are described in RFC7064 and RFC7065. // Urls should containe dns hostnames (not IP addresses) // as the TLS certificate policy is 'secure.' // and we do not currentply support TLS extensions. std::vector mUrls; std::string mUserName; std::string mPassword; }; std::vector mServers; }; virtual bool initializeConnection(const InitOptions& options) = 0; virtual bool shutdownConnection() = 0; virtual void setSignalingObserver(LLWebRTCSignalingObserver* observer) = 0; virtual void unsetSignalingObserver(LLWebRTCSignalingObserver* observer) = 0; virtual void AnswerAvailable(const std::string &sdp) = 0; }; // The following define the dynamic linked library // exports. // This library must be initialized before use. LLSYMEXPORT void init(); // And should be terminated as part of shutdown. LLSYMEXPORT void terminate(); // Return an interface for device management. LLSYMEXPORT LLWebRTCDeviceInterface* getDeviceInterface(); // Allocate and free peer connections. LLSYMEXPORT LLWebRTCPeerConnectionInterface* newPeerConnection(); LLSYMEXPORT void freePeerConnection(LLWebRTCPeerConnectionInterface *connection); } #endif // LLWEBRTC_H