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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
#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; }
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;
};
// Primary singleton implementation for interfacing
// with the native webrtc library.
class LLWebRTCImpl : public LLWebRTCDeviceInterface, public webrtc::AudioDeviceSink
{
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;
//
// AudioDeviceSink
//
void OnDevicesUpdated() override;
//
// 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 mPlayoutDevice;
int32_t mRecordingDevice;
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(InitOptions options = InitOptions()) 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;
bool mClosing;
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
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