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/**
* @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 <string>
#include <vector>
#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)
{
if (mID.empty())
{
mID = display_name;
}
};
};
typedef std::vector<LLWebRTCVoiceDevice> 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 a peer connection has shut down
virtual void OnPeerConnectionClosed() = 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<std::string> mUrls;
std::string mUserName;
std::string mPassword;
};
std::vector<IceServers> 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
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