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diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
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+/**
+ * @file llwebrtc.cpp
+ * @brief WebRTC interface implementation
+ *
+ * $LicenseInfo:firstyear=2023&license=viewerlgpl$
+ * 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
+ * $/LicenseInfo$
+ */
+
+#include "llwebrtc_impl.h"
+#include <algorithm>
+#include <string.h>
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/media_stream_interface.h"
+#include "api/media_stream_track.h"
+#include "modules/audio_processing/audio_buffer.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+
+namespace llwebrtc
+{
+
+static int16_t PLAYOUT_DEVICE_DEFAULT = -1;
+static int16_t PLAYOUT_DEVICE_BAD = -2;
+static int16_t RECORD_DEVICE_DEFAULT = -1;
+static int16_t RECORD_DEVICE_BAD = -2;
+
+LLAudioDeviceObserver::LLAudioDeviceObserver() : mSumVector {0}, mMicrophoneEnergy(0.0) {}
+
+float LLAudioDeviceObserver::getMicrophoneEnergy() { return mMicrophoneEnergy; }
+
+// TODO: Pull smoothing/filtering code into a common helper function
+// for LLAudioDeviceObserver and LLCustomProcessor
+
+void LLAudioDeviceObserver::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)
+{
+ // calculate the energy
+ float energy = 0;
+ const short *samples = (const short *) audio_samples;
+ for (size_t index = 0; index < num_samples * num_channels; index++)
+ {
+ float sample = (static_cast<float>(samples[index]) / (float) 32767);
+ energy += sample * sample;
+ }
+
+ // smooth it.
+ size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
+ float totalSum = 0;
+ int i;
+ for (i = 0; i < (buffer_size - 1); i++)
+ {
+ mSumVector[i] = mSumVector[i + 1];
+ totalSum += mSumVector[i];
+ }
+ mSumVector[i] = energy;
+ totalSum += energy;
+ mMicrophoneEnergy = std::sqrt(totalSum / (num_samples * buffer_size));
+}
+
+void LLAudioDeviceObserver::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)
+{
+}
+
+LLCustomProcessor::LLCustomProcessor() : mSampleRateHz(0), mNumChannels(0), mMicrophoneEnergy(0.0), mGain(1.0)
+{
+ memset(mSumVector, 0, sizeof(mSumVector));
+}
+
+void LLCustomProcessor::Initialize(int sample_rate_hz, int num_channels)
+{
+ mSampleRateHz = sample_rate_hz;
+ mNumChannels = num_channels;
+ memset(mSumVector, 0, sizeof(mSumVector));
+}
+
+void LLCustomProcessor::Process(webrtc::AudioBuffer *audio_in)
+{
+ webrtc::StreamConfig stream_config;
+ stream_config.set_sample_rate_hz(mSampleRateHz);
+ stream_config.set_num_channels(mNumChannels);
+ std::vector<float *> frame;
+ std::vector<float> frame_samples;
+
+ if (audio_in->num_channels() < 1 || audio_in->num_frames() < 480)
+ {
+ return;
+ }
+
+ // grab the input audio
+ frame_samples.resize(stream_config.num_samples());
+ frame.resize(stream_config.num_channels());
+ for (size_t ch = 0; ch < stream_config.num_channels(); ++ch)
+ {
+ frame[ch] = &(frame_samples)[ch * stream_config.num_frames()];
+ }
+
+ audio_in->CopyTo(stream_config, &frame[0]);
+
+ // calculate the energy
+ float energy = 0;
+ for (size_t index = 0; index < stream_config.num_samples(); index++)
+ {
+ float sample = frame_samples[index];
+ sample = sample * mGain; // apply gain
+ frame_samples[index] = sample; // write processed sample back to buffer.
+ energy += sample * sample;
+ }
+
+ audio_in->CopyFrom(&frame[0], stream_config);
+
+ // smooth it.
+ size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
+ float totalSum = 0;
+ int i;
+ for (i = 0; i < (buffer_size - 1); i++)
+ {
+ mSumVector[i] = mSumVector[i + 1];
+ totalSum += mSumVector[i];
+ }
+ mSumVector[i] = energy;
+ totalSum += energy;
+ mMicrophoneEnergy = std::sqrt(totalSum / (stream_config.num_samples() * buffer_size));
+}
+
+//
+// LLWebRTCImpl implementation
+//
+
+LLWebRTCImpl::LLWebRTCImpl() :
+ mPeerCustomProcessor(nullptr),
+ mMute(true),
+ mTuningMode(false),
+ mPlayoutDevice(0),
+ mRecordingDevice(0),
+ mTuningAudioDeviceObserver(nullptr)
+{
+}
+
+void LLWebRTCImpl::init()
+{
+ mPlayoutDevice = 0;
+ mRecordingDevice = 0;
+ rtc::InitializeSSL();
+
+ // Normal logging is rather spammy, so turn it off.
+ rtc::LogMessage::LogToDebug(rtc::LS_NONE);
+ rtc::LogMessage::SetLogToStderr(true);
+
+ mTaskQueueFactory = webrtc::CreateDefaultTaskQueueFactory();
+
+ // Create the native threads.
+ mNetworkThread = rtc::Thread::CreateWithSocketServer();
+ mNetworkThread->SetName("WebRTCNetworkThread", nullptr);
+ mNetworkThread->Start();
+ mWorkerThread = rtc::Thread::Create();
+ mWorkerThread->SetName("WebRTCWorkerThread", nullptr);
+ mWorkerThread->Start();
+ mSignalingThread = rtc::Thread::Create();
+ mSignalingThread->SetName("WebRTCSignalingThread", nullptr);
+ mSignalingThread->Start();
+
+ mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
+ mWorkerThread->PostTask(
+ [this]()
+ {
+ // Initialize the audio devices on the Worker Thread
+ mTuningDeviceModule =
+ webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(),
+ std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
+
+ mTuningDeviceModule->Init();
+ mTuningDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mTuningDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mTuningDeviceModule->EnableBuiltInAEC(false);
+#if __x86_64__ && !__FreeBSD__
+ mTuningDeviceModule->SetAudioDeviceSink(this);
+#endif
+ mTuningDeviceModule->InitMicrophone();
+ mTuningDeviceModule->InitSpeaker();
+ mTuningDeviceModule->InitRecording();
+ mTuningDeviceModule->InitPlayout();
+ mTuningDeviceModule->SetStereoRecording(true);
+ mTuningDeviceModule->SetStereoPlayout(true);
+ updateDevices();
+ });
+
+ mWorkerThread->BlockingCall(
+ [this]()
+ {
+ // the peer device module doesn't need an observer
+ // as we pull peer data after audio processing.
+ mPeerDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(),
+ nullptr);
+ mPeerDeviceModule->Init();
+ mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mPeerDeviceModule->EnableBuiltInAEC(false);
+ mPeerDeviceModule->InitMicrophone();
+ mPeerDeviceModule->InitSpeaker();
+ mPeerDeviceModule->InitRecording();
+ mPeerDeviceModule->InitPlayout();
+ mPeerDeviceModule->SetStereoRecording(true);
+ mPeerDeviceModule->SetStereoPlayout(true);
+ });
+
+ // The custom processor allows us to retrieve audio data (and levels)
+ // from after other audio processing such as AEC, AGC, etc.
+ mPeerCustomProcessor = new LLCustomProcessor;
+ webrtc::AudioProcessingBuilder apb;
+ apb.SetCapturePostProcessing(std::unique_ptr<webrtc::CustomProcessing>(mPeerCustomProcessor));
+ mAudioProcessingModule = apb.Create();
+
+ webrtc::AudioProcessing::Config apm_config;
+ apm_config.echo_canceller.enabled = false;
+ apm_config.echo_canceller.mobile_mode = false;
+ apm_config.gain_controller1.enabled = false;
+ apm_config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+ apm_config.gain_controller2.enabled = false;
+ apm_config.high_pass_filter.enabled = true;
+ apm_config.noise_suppression.enabled = true;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh;
+ apm_config.transient_suppression.enabled = true;
+ apm_config.pipeline.multi_channel_render = true;
+ apm_config.pipeline.multi_channel_capture = true;
+ apm_config.pipeline.multi_channel_capture = true;
+
+ webrtc::ProcessingConfig processing_config;
+ processing_config.input_stream().set_num_channels(2);
+ processing_config.input_stream().set_sample_rate_hz(48000);
+ processing_config.output_stream().set_num_channels(2);
+ processing_config.output_stream().set_sample_rate_hz(48000);
+ processing_config.reverse_input_stream().set_num_channels(2);
+ processing_config.reverse_input_stream().set_sample_rate_hz(48000);
+ processing_config.reverse_output_stream().set_num_channels(2);
+ processing_config.reverse_output_stream().set_sample_rate_hz(48000);
+
+ mAudioProcessingModule->ApplyConfig(apm_config);
+ mAudioProcessingModule->Initialize(processing_config);
+
+
+ mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(),
+ mWorkerThread.get(),
+ mSignalingThread.get(),
+ mPeerDeviceModule,
+ webrtc::CreateBuiltinAudioEncoderFactory(),
+ webrtc::CreateBuiltinAudioDecoderFactory(),
+ nullptr /* video_encoder_factory */,
+ nullptr /* video_decoder_factory */,
+ nullptr /* audio_mixer */,
+ mAudioProcessingModule);
+
+ mWorkerThread->BlockingCall([this]() { mPeerDeviceModule->StartPlayout(); });
+}
+
+void LLWebRTCImpl::terminate()
+{
+ for (auto &connection : mPeerConnections)
+ {
+ connection->terminate();
+ }
+
+ // connection->terminate() above spawns a number of Signaling thread calls to
+ // shut down the connection. The following Blocking Call will wait
+ // until they're done before it's executed, allowing time to clean up.
+
+ mSignalingThread->BlockingCall([this]() { mPeerConnectionFactory = nullptr; });
+
+ mPeerConnections.clear();
+
+ mWorkerThread->BlockingCall(
+ [this]()
+ {
+ if (mTuningDeviceModule)
+ {
+ mTuningDeviceModule->StopRecording();
+ mTuningDeviceModule->Terminate();
+ }
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->StopRecording();
+ mPeerDeviceModule->Terminate();
+ }
+ mTuningDeviceModule = nullptr;
+ mPeerDeviceModule = nullptr;
+ mTaskQueueFactory = nullptr;
+ });
+}
+
+//
+// Devices functions
+//
+// Most device-related functionality needs to happen
+// on the worker thread (the audio thread,) so those calls will be
+// proxied over to that thread.
+//
+void LLWebRTCImpl::setRecording(bool recording)
+{
+ mWorkerThread->PostTask(
+ [this, recording]()
+ {
+ if (recording)
+ {
+ mPeerDeviceModule->StartRecording();
+ }
+ else
+ {
+ mPeerDeviceModule->StopRecording();
+ }
+ });
+}
+
+void LLWebRTCImpl::setAudioConfig(LLWebRTCDeviceInterface::AudioConfig config)
+{
+ webrtc::AudioProcessing::Config apm_config;
+ apm_config.echo_canceller.enabled = config.mEchoCancellation;
+ apm_config.echo_canceller.mobile_mode = false;
+ apm_config.gain_controller1.enabled = config.mAGC;
+ apm_config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+ apm_config.gain_controller2.enabled = false;
+ apm_config.high_pass_filter.enabled = true;
+ apm_config.transient_suppression.enabled = true;
+ apm_config.pipeline.multi_channel_render = true;
+ apm_config.pipeline.multi_channel_capture = true;
+ apm_config.pipeline.multi_channel_capture = true;
+
+ switch (config.mNoiseSuppressionLevel)
+ {
+ case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_NONE:
+ apm_config.noise_suppression.enabled = false;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kLow;
+ break;
+ case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_LOW:
+ apm_config.noise_suppression.enabled = true;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kLow;
+ break;
+ case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_MODERATE:
+ apm_config.noise_suppression.enabled = true;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kModerate;
+ break;
+ case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_HIGH:
+ apm_config.noise_suppression.enabled = true;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kHigh;
+ break;
+ case LLWebRTCDeviceInterface::AudioConfig::NOISE_SUPPRESSION_LEVEL_VERY_HIGH:
+ apm_config.noise_suppression.enabled = true;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kVeryHigh;
+ break;
+ default:
+ apm_config.noise_suppression.enabled = false;
+ apm_config.noise_suppression.level = webrtc::AudioProcessing::Config::NoiseSuppression::kLow;
+ }
+ mAudioProcessingModule->ApplyConfig(apm_config);
+}
+
+void LLWebRTCImpl::refreshDevices()
+{
+ mWorkerThread->PostTask([this]() { updateDevices(); });
+}
+
+void LLWebRTCImpl::setDevicesObserver(LLWebRTCDevicesObserver *observer) { mVoiceDevicesObserverList.emplace_back(observer); }
+
+void LLWebRTCImpl::unsetDevicesObserver(LLWebRTCDevicesObserver *observer)
+{
+ std::vector<LLWebRTCDevicesObserver *>::iterator it =
+ std::find(mVoiceDevicesObserverList.begin(), mVoiceDevicesObserverList.end(), observer);
+ if (it != mVoiceDevicesObserverList.end())
+ {
+ mVoiceDevicesObserverList.erase(it);
+ }
+}
+
+void ll_set_device_module_capture_device(rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module, int16_t device)
+{
+ device_module->StopRecording();
+#if WEBRTC_WIN
+ if (device < 0)
+ {
+ device_module->SetRecordingDevice(webrtc::AudioDeviceModule::kDefaultDevice);
+ }
+ else
+ {
+ device_module->SetRecordingDevice(device);
+ }
+#else
+ // passed in default is -1, but the device list
+ // has it at 0
+ device_module->SetRecordingDevice(device + 1);
+#endif
+ device_module->InitMicrophone();
+ device_module->InitRecording();
+ device_module->SetStereoRecording(false);
+ device_module->StartRecording();
+}
+
+void LLWebRTCImpl::setCaptureDevice(const std::string &id)
+{
+ int16_t recordingDevice = RECORD_DEVICE_DEFAULT;
+ if (id != "Default")
+ {
+ for (int16_t i = 0; i < mRecordingDeviceList.size(); i++)
+ {
+ if (mRecordingDeviceList[i].mID == id)
+ {
+ recordingDevice = i;
+ break;
+ }
+ }
+ }
+ if (recordingDevice == mRecordingDevice)
+ {
+ return;
+ }
+ mRecordingDevice = recordingDevice;
+ if (mTuningMode)
+ {
+ mWorkerThread->PostTask([this, recordingDevice]() { ll_set_device_module_capture_device(mTuningDeviceModule, recordingDevice); });
+ }
+ else
+ {
+ mWorkerThread->PostTask([this, recordingDevice]() { ll_set_device_module_capture_device(mPeerDeviceModule, recordingDevice); });
+ }
+}
+
+
+void ll_set_device_module_render_device(rtc::scoped_refptr<webrtc::AudioDeviceModule> device_module, int16_t device)
+{
+ device_module->StopPlayout();
+#if WEBRTC_WIN
+ if (device < 0)
+ {
+ device_module->SetPlayoutDevice(webrtc::AudioDeviceModule::kDefaultDevice);
+ }
+ else
+ {
+ device_module->SetPlayoutDevice(device);
+ }
+#else
+ device_module->SetPlayoutDevice(device + 1);
+#endif
+ device_module->InitSpeaker();
+ device_module->InitPlayout();
+ device_module->SetStereoPlayout(true);
+}
+
+void LLWebRTCImpl::setRenderDevice(const std::string &id)
+{
+ int16_t playoutDevice = PLAYOUT_DEVICE_DEFAULT;
+ if (id != "Default")
+ {
+ for (int16_t i = 0; i < mPlayoutDeviceList.size(); i++)
+ {
+ if (mPlayoutDeviceList[i].mID == id)
+ {
+ playoutDevice = i;
+ break;
+ }
+ }
+ }
+ if (playoutDevice == mPlayoutDevice)
+ {
+ return;
+ }
+ mPlayoutDevice = playoutDevice;
+
+ if (mTuningMode)
+ {
+ mWorkerThread->PostTask(
+ [this, playoutDevice]()
+ {
+ ll_set_device_module_render_device(mTuningDeviceModule, playoutDevice);
+ });
+ }
+ else
+ {
+ mWorkerThread->PostTask(
+ [this, playoutDevice]()
+ {
+ ll_set_device_module_render_device(mPeerDeviceModule, playoutDevice);
+ mPeerDeviceModule->StartPlayout();
+ });
+ }
+}
+
+// updateDevices needs to happen on the worker thread.
+void LLWebRTCImpl::updateDevices()
+{
+ int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
+
+ mPlayoutDeviceList.clear();
+#if WEBRTC_WIN
+ int16_t index = 0;
+#else
+ // index zero is always "Default" for darwin/linux,
+ // which is a special case, so skip it.
+ int16_t index = 1;
+#endif
+ for (; index < renderDeviceCount; index++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mTuningDeviceModule->PlayoutDeviceName(index, name, guid);
+ mPlayoutDeviceList.emplace_back(name, guid);
+ }
+
+ int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
+
+ mRecordingDeviceList.clear();
+#if WEBRTC_WIN
+ index = 0;
+#else
+ // index zero is always "Default" for darwin/linux,
+ // which is a special case, so skip it.
+ index = 1;
+#endif
+ for (; index < captureDeviceCount; index++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mTuningDeviceModule->RecordingDeviceName(index, name, guid);
+ mRecordingDeviceList.emplace_back(name, guid);
+ }
+
+ for (auto &observer : mVoiceDevicesObserverList)
+ {
+ observer->OnDevicesChanged(mPlayoutDeviceList, mRecordingDeviceList);
+ }
+}
+
+void LLWebRTCImpl::OnDevicesUpdated()
+{
+ // reset these to a bad value so an update is forced
+ mRecordingDevice = RECORD_DEVICE_BAD;
+ mPlayoutDevice = PLAYOUT_DEVICE_BAD;
+
+ updateDevices();
+}
+
+
+void LLWebRTCImpl::setTuningMode(bool enable)
+{
+ mTuningMode = enable;
+ mWorkerThread->PostTask(
+ [this, enable] {
+ if (enable)
+ {
+ mPeerDeviceModule->StopRecording();
+ mPeerDeviceModule->StopPlayout();
+ ll_set_device_module_render_device(mTuningDeviceModule, mPlayoutDevice);
+ ll_set_device_module_capture_device(mTuningDeviceModule, mRecordingDevice);
+ mTuningDeviceModule->InitPlayout();
+ mTuningDeviceModule->InitRecording();
+ mTuningDeviceModule->StartRecording();
+ // TODO: Starting Playout on the TDM appears to create an audio artifact (click)
+ // in this case, so disabling it for now. We may have to do something different
+ // if we enable 'echo playback' via the TDM when tuning.
+ //mTuningDeviceModule->StartPlayout();
+ }
+ else
+ {
+ mTuningDeviceModule->StopRecording();
+ //mTuningDeviceModule->StopPlayout();
+ ll_set_device_module_render_device(mPeerDeviceModule, mPlayoutDevice);
+ ll_set_device_module_capture_device(mPeerDeviceModule, mRecordingDevice);
+ mPeerDeviceModule->InitPlayout();
+ mPeerDeviceModule->InitRecording();
+ mPeerDeviceModule->StartPlayout();
+ mPeerDeviceModule->StartRecording();
+ }
+ }
+ );
+ mSignalingThread->PostTask(
+ [this, enable]
+ {
+ for (auto &connection : mPeerConnections)
+ {
+ if (enable)
+ {
+ connection->enableSenderTracks(false);
+ }
+ else
+ {
+ connection->resetMute();
+ }
+ connection->enableReceiverTracks(!enable);
+ }
+ });
+}
+
+float LLWebRTCImpl::getTuningAudioLevel() { return -20 * log10f(mTuningAudioDeviceObserver->getMicrophoneEnergy()); }
+
+float LLWebRTCImpl::getPeerConnectionAudioLevel() { return -20 * log10f(mPeerCustomProcessor->getMicrophoneEnergy()); }
+
+void LLWebRTCImpl::setPeerConnectionGain(float gain) { mPeerCustomProcessor->setGain(gain); }
+
+
+//
+// Peer Connection Helpers
+//
+
+LLWebRTCPeerConnectionInterface *LLWebRTCImpl::newPeerConnection()
+{
+ rtc::scoped_refptr<LLWebRTCPeerConnectionImpl> peerConnection = rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>(new rtc::RefCountedObject<LLWebRTCPeerConnectionImpl>());
+ peerConnection->init(this);
+
+ mPeerConnections.emplace_back(peerConnection);
+ peerConnection->enableSenderTracks(!mMute);
+ return peerConnection.get();
+}
+
+void LLWebRTCImpl::freePeerConnection(LLWebRTCPeerConnectionInterface* peer_connection)
+{
+ std::vector<rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>>::iterator it =
+ std::find(mPeerConnections.begin(), mPeerConnections.end(), peer_connection);
+ if (it != mPeerConnections.end())
+ {
+ mPeerConnections.erase(it);
+ }
+ if (mPeerConnections.empty())
+ {
+ setRecording(false);
+ }
+}
+
+
+//
+// LLWebRTCPeerConnectionImpl implementation.
+//
+// Most peer connection (signaling) happens on
+// the signaling thread.
+
+LLWebRTCPeerConnectionImpl::LLWebRTCPeerConnectionImpl() :
+ mWebRTCImpl(nullptr),
+ mPeerConnection(nullptr),
+ mMute(false),
+ mAnswerReceived(false)
+{
+}
+
+LLWebRTCPeerConnectionImpl::~LLWebRTCPeerConnectionImpl()
+{
+ mSignalingObserverList.clear();
+ mDataObserverList.clear();
+}
+
+//
+// LLWebRTCPeerConnection interface
+//
+
+void LLWebRTCPeerConnectionImpl::init(LLWebRTCImpl * webrtc_impl)
+{
+ mWebRTCImpl = webrtc_impl;
+ mPeerConnectionFactory = mWebRTCImpl->getPeerConnectionFactory();
+}
+void LLWebRTCPeerConnectionImpl::terminate()
+{
+ mWebRTCImpl->PostSignalingTask(
+ [=]()
+ {
+ if (mPeerConnection)
+ {
+ if (mDataChannel)
+ {
+ {
+ mDataChannel->Close();
+ mDataChannel = nullptr;
+ }
+ }
+
+ mPeerConnection->Close();
+ if (mLocalStream)
+ {
+ auto tracks = mLocalStream->GetAudioTracks();
+ for (auto& track : tracks)
+ {
+ mLocalStream->RemoveTrack(track);
+ }
+ mLocalStream = nullptr;
+ }
+ mPeerConnection = nullptr;
+
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnPeerConnectionClosed();
+ }
+ }
+ });
+}
+
+void LLWebRTCPeerConnectionImpl::setSignalingObserver(LLWebRTCSignalingObserver *observer) { mSignalingObserverList.emplace_back(observer); }
+
+void LLWebRTCPeerConnectionImpl::unsetSignalingObserver(LLWebRTCSignalingObserver *observer)
+{
+ std::vector<LLWebRTCSignalingObserver *>::iterator it =
+ std::find(mSignalingObserverList.begin(), mSignalingObserverList.end(), observer);
+ if (it != mSignalingObserverList.end())
+ {
+ mSignalingObserverList.erase(it);
+ }
+}
+
+
+bool LLWebRTCPeerConnectionImpl::initializeConnection(const LLWebRTCPeerConnectionInterface::InitOptions& options)
+{
+ RTC_DCHECK(!mPeerConnection);
+ mAnswerReceived = false;
+
+ mWebRTCImpl->PostSignalingTask(
+ [this,options]()
+ {
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
+ for (auto server : options.mServers)
+ {
+ webrtc::PeerConnectionInterface::IceServer ice_server;
+ for (auto url : server.mUrls)
+ {
+ ice_server.urls.push_back(url);
+ }
+ ice_server.username = server.mUserName;
+ ice_server.password = server.mPassword;
+ config.servers.push_back(ice_server);
+ }
+ config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+
+ config.set_min_port(60000);
+ config.set_max_port(60100);
+
+ webrtc::PeerConnectionDependencies pc_dependencies(this);
+ auto error_or_peer_connection = mPeerConnectionFactory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
+ if (error_or_peer_connection.ok())
+ {
+ mPeerConnection = std::move(error_or_peer_connection.value());
+ }
+ else
+ {
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << "Error creating peer connection: " << error_or_peer_connection.error().message();
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnRenegotiationNeeded();
+ }
+ return;
+ }
+
+ webrtc::DataChannelInit init;
+ init.ordered = true;
+
+ auto data_channel_or_error = mPeerConnection->CreateDataChannelOrError("SLData", &init);
+ if (data_channel_or_error.ok())
+ {
+ mDataChannel = std::move(data_channel_or_error.value());
+
+ mDataChannel->RegisterObserver(this);
+ }
+
+ cricket::AudioOptions audioOptions;
+ audioOptions.auto_gain_control = true;
+ audioOptions.echo_cancellation = true;
+ audioOptions.noise_suppression = true;
+
+ mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
+
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
+ mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(audioOptions).get()));
+ audio_track->set_enabled(false);
+ mLocalStream->AddTrack(audio_track);
+
+ mPeerConnection->AddTrack(audio_track, {"SLStream"});
+
+ auto senders = mPeerConnection->GetSenders();
+
+ for (auto &sender : senders)
+ {
+ webrtc::RtpParameters params;
+ webrtc::RtpCodecParameters codecparam;
+ codecparam.name = "opus";
+ codecparam.kind = cricket::MEDIA_TYPE_AUDIO;
+ codecparam.clock_rate = 48000;
+ codecparam.num_channels = 2;
+ codecparam.parameters["stereo"] = "1";
+ codecparam.parameters["sprop-stereo"] = "1";
+ params.codecs.push_back(codecparam);
+ sender->SetParameters(params);
+ }
+
+ auto receivers = mPeerConnection->GetReceivers();
+ for (auto &receiver : receivers)
+ {
+ webrtc::RtpParameters params;
+ webrtc::RtpCodecParameters codecparam;
+ codecparam.name = "opus";
+ codecparam.kind = cricket::MEDIA_TYPE_AUDIO;
+ codecparam.clock_rate = 48000;
+ codecparam.num_channels = 2;
+ codecparam.parameters["stereo"] = "1";
+ codecparam.parameters["sprop-stereo"] = "1";
+ params.codecs.push_back(codecparam);
+ receiver->SetParameters(params);
+ }
+
+ webrtc::PeerConnectionInterface::RTCOfferAnswerOptions offerOptions;
+ mPeerConnection->CreateOffer(this, offerOptions);
+ });
+
+ return true;
+}
+
+bool LLWebRTCPeerConnectionImpl::shutdownConnection()
+{
+ terminate();
+ return true;
+}
+
+void LLWebRTCPeerConnectionImpl::enableSenderTracks(bool enable)
+{
+ // set_enabled shouldn't be done on the worker thread.
+ if (mPeerConnection)
+ {
+ auto senders = mPeerConnection->GetSenders();
+ for (auto &sender : senders)
+ {
+ sender->track()->set_enabled(enable);
+ }
+ }
+}
+
+void LLWebRTCPeerConnectionImpl::enableReceiverTracks(bool enable)
+{
+ // set_enabled shouldn't be done on the worker thread
+ if (mPeerConnection)
+ {
+ auto receivers = mPeerConnection->GetReceivers();
+ for (auto &receiver : receivers)
+ {
+ receiver->track()->set_enabled(enable);
+ }
+ }
+}
+
+// Tell the peer connection that we've received a SDP answer from the sim.
+void LLWebRTCPeerConnectionImpl::AnswerAvailable(const std::string &sdp)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " Remote SDP: " << sdp;
+
+ mWebRTCImpl->PostSignalingTask(
+ [this, sdp]()
+ {
+ if (mPeerConnection)
+ {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
+ mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
+ rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
+ }
+ });
+}
+
+
+//
+// LLWebRTCAudioInterface implementation
+//
+
+void LLWebRTCPeerConnectionImpl::setMute(bool mute)
+{
+ mMute = mute;
+ mWebRTCImpl->PostSignalingTask(
+ [this]()
+ {
+ if (mPeerConnection)
+ {
+ auto senders = mPeerConnection->GetSenders();
+
+ RTC_LOG(LS_INFO) << __FUNCTION__ << (mMute ? "disabling" : "enabling") << " streams count " << senders.size();
+ for (auto &sender : senders)
+ {
+ auto track = sender->track();
+ if (track)
+ {
+ track->set_enabled(!mMute);
+ }
+ }
+ }
+ });
+}
+
+void LLWebRTCPeerConnectionImpl::resetMute()
+{
+ setMute(mMute);
+}
+
+void LLWebRTCPeerConnectionImpl::setReceiveVolume(float volume)
+{
+ mWebRTCImpl->PostSignalingTask(
+ [this, volume]()
+ {
+ if (mPeerConnection)
+ {
+ auto receivers = mPeerConnection->GetReceivers();
+
+ for (auto &receiver : receivers)
+ {
+ for (auto &stream : receiver->streams())
+ {
+ for (auto &track : stream->GetAudioTracks())
+ {
+ track->GetSource()->SetVolume(volume);
+ }
+ }
+ }
+ }
+ });
+}
+
+void LLWebRTCPeerConnectionImpl::setSendVolume(float volume)
+{
+ mWebRTCImpl->PostSignalingTask(
+ [this, volume]()
+ {
+ if (mLocalStream)
+ {
+ for (auto &track : mLocalStream->GetAudioTracks())
+ {
+ track->GetSource()->SetVolume(volume*5.0);
+ }
+ }
+ });
+}
+
+//
+// PeerConnectionObserver implementation.
+//
+
+void LLWebRTCPeerConnectionImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
+ const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
+ webrtc::RtpParameters params;
+ webrtc::RtpCodecParameters codecparam;
+ codecparam.name = "opus";
+ codecparam.kind = cricket::MEDIA_TYPE_AUDIO;
+ codecparam.clock_rate = 48000;
+ codecparam.num_channels = 2;
+ codecparam.parameters["stereo"] = "1";
+ codecparam.parameters["sprop-stereo"] = "1";
+ params.codecs.push_back(codecparam);
+ receiver->SetParameters(params);
+}
+
+void LLWebRTCPeerConnectionImpl::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
+}
+
+void LLWebRTCPeerConnectionImpl::OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> channel)
+{
+ if (mDataChannel)
+ {
+ mDataChannel->UnregisterObserver();
+ }
+ mDataChannel = channel;
+ channel->RegisterObserver(this);
+}
+
+void LLWebRTCPeerConnectionImpl::OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)
+{
+ LLWebRTCSignalingObserver::EIceGatheringState webrtc_new_state = LLWebRTCSignalingObserver::EIceGatheringState::ICE_GATHERING_NEW;
+ switch (new_state)
+ {
+ case webrtc::PeerConnectionInterface::IceGatheringState::kIceGatheringNew:
+ webrtc_new_state = LLWebRTCSignalingObserver::EIceGatheringState::ICE_GATHERING_NEW;
+ break;
+ case webrtc::PeerConnectionInterface::IceGatheringState::kIceGatheringGathering:
+ webrtc_new_state = LLWebRTCSignalingObserver::EIceGatheringState::ICE_GATHERING_GATHERING;
+ break;
+ case webrtc::PeerConnectionInterface::IceGatheringState::kIceGatheringComplete:
+ webrtc_new_state = LLWebRTCSignalingObserver::EIceGatheringState::ICE_GATHERING_COMPLETE;
+ break;
+ default:
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << " Bad Ice Gathering State" << new_state;
+ webrtc_new_state = LLWebRTCSignalingObserver::EIceGatheringState::ICE_GATHERING_NEW;
+ return;
+ }
+
+ if (mAnswerReceived)
+ {
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnIceGatheringState(webrtc_new_state);
+ }
+ }
+}
+
+// Called any time the PeerConnectionState changes.
+void LLWebRTCPeerConnectionImpl::OnConnectionChange(webrtc::PeerConnectionInterface::PeerConnectionState new_state)
+{
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << " Peer Connection State Change " << new_state;
+
+ switch (new_state)
+ {
+ case webrtc::PeerConnectionInterface::PeerConnectionState::kConnected:
+ {
+ mWebRTCImpl->PostWorkerTask([this]() {
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnAudioEstablished(this);
+ }
+ });
+ break;
+ }
+ case webrtc::PeerConnectionInterface::PeerConnectionState::kFailed:
+ case webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected:
+ {
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnRenegotiationNeeded();
+ }
+
+ break;
+ }
+ default:
+ {
+ break;
+ }
+ }
+}
+
+// Convert an ICE candidate into a string appropriate for trickling
+// to the Secondlife WebRTC server via the sim.
+static std::string iceCandidateToTrickleString(const webrtc::IceCandidateInterface *candidate)
+{
+ std::ostringstream candidate_stream;
+
+ candidate_stream <<
+ candidate->candidate().foundation() << " " <<
+ std::to_string(candidate->candidate().component()) << " " <<
+ candidate->candidate().protocol() << " " <<
+ std::to_string(candidate->candidate().priority()) << " " <<
+ candidate->candidate().address().ipaddr().ToString() << " " <<
+ candidate->candidate().address().PortAsString() << " typ ";
+
+ if (candidate->candidate().type() == cricket::LOCAL_PORT_TYPE)
+ {
+ candidate_stream << "host";
+ }
+ else if (candidate->candidate().type() == cricket::STUN_PORT_TYPE)
+ {
+ candidate_stream << "srflx " <<
+ "raddr " << candidate->candidate().related_address().ipaddr().ToString() << " " <<
+ "rport " << candidate->candidate().related_address().PortAsString();
+ }
+ else if (candidate->candidate().type() == cricket::RELAY_PORT_TYPE)
+ {
+ candidate_stream << "relay " <<
+ "raddr " << candidate->candidate().related_address().ipaddr().ToString() << " " <<
+ "rport " << candidate->candidate().related_address().PortAsString();
+ }
+ else if (candidate->candidate().type() == cricket::PRFLX_PORT_TYPE)
+ {
+ candidate_stream << "prflx " <<
+ "raddr " << candidate->candidate().related_address().ipaddr().ToString() << " " <<
+ "rport " << candidate->candidate().related_address().PortAsString();
+ }
+ else {
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << " Unknown candidate type " << candidate->candidate().type();
+ }
+ if (candidate->candidate().protocol() == "tcp")
+ {
+ candidate_stream << " tcptype " << candidate->candidate().tcptype();
+ }
+
+ return candidate_stream.str();
+}
+
+// The webrtc library has a new ice candidate.
+void LLWebRTCPeerConnectionImpl::OnIceCandidate(const webrtc::IceCandidateInterface *candidate)
+{
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << candidate->sdp_mline_index();
+
+ if (!candidate)
+ {
+ RTC_LOG(LS_ERROR) << __FUNCTION__ << " No Ice Candidate Given";
+ return;
+ }
+ if (mAnswerReceived)
+ {
+ // We've already received an answer SDP from the Secondlife WebRTC server
+ // so simply tell observers about our new ice candidate.
+ for (auto &observer : mSignalingObserverList)
+ {
+ LLWebRTCIceCandidate ice_candidate;
+ ice_candidate.mCandidate = iceCandidateToTrickleString(candidate);
+ ice_candidate.mMLineIndex = candidate->sdp_mline_index();
+ ice_candidate.mSdpMid = candidate->sdp_mid();
+ observer->OnIceCandidate(ice_candidate);
+ }
+ }
+ else
+ {
+ // As we've not yet received our answer, cache the candidate.
+ mCachedIceCandidates.push_back(
+ webrtc::CreateIceCandidate(candidate->sdp_mid(),
+ candidate->sdp_mline_index(),
+ candidate->candidate()));
+ }
+}
+
+//
+// CreateSessionDescriptionObserver implementation.
+//
+void LLWebRTCPeerConnectionImpl::OnSuccess(webrtc::SessionDescriptionInterface *desc)
+{
+ std::string sdp;
+ desc->ToString(&sdp);
+ RTC_LOG(LS_INFO) << sdp;
+ ;
+ // mangle the sdp as this is the only way currently to bump up
+ // the send audio rate to 48k
+ std::istringstream sdp_stream(sdp);
+ std::ostringstream sdp_mangled_stream;
+ std::string sdp_line;
+ std::string opus_payload;
+ while (std::getline(sdp_stream, sdp_line))
+ {
+ int bandwidth = 0;
+ int payload_id = 0;
+ // force mono down, stereo up
+ if (std::sscanf(sdp_line.c_str(), "a=rtpmap:%i opus/%i/2", &payload_id, &bandwidth) == 2)
+ {
+ opus_payload = std::to_string(payload_id);
+ sdp_mangled_stream << "a=rtpmap:" << opus_payload << " opus/48000/2" << "\n";
+ }
+ else if (sdp_line.find("a=fmtp:" + opus_payload) == 0)
+ {
+ sdp_mangled_stream << sdp_line << "a=fmtp:" << opus_payload
+ << " minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1;maxplaybackrate=48000;sprop-maxplaybackrate=48000;sprop-maxcapturerate=48000\n";
+ }
+ else
+ {
+ sdp_mangled_stream << sdp_line << "\n";
+ }
+ }
+
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp_mangled_stream.str();
+ std::string mangled_sdp = sdp_mangled_stream.str();
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnOfferAvailable(mangled_sdp);
+ }
+
+ mPeerConnection->SetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>(
+ webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, mangled_sdp)),
+ rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
+
+}
+
+void LLWebRTCPeerConnectionImpl::OnFailure(webrtc::RTCError error)
+{
+ RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnRenegotiationNeeded();
+ }
+}
+
+//
+// SetRemoteDescriptionObserverInterface implementation.
+//
+void LLWebRTCPeerConnectionImpl::OnSetRemoteDescriptionComplete(webrtc::RTCError error)
+{
+ // we've received an answer SDP from the sim.
+
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
+ if (!error.ok())
+ {
+ RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnRenegotiationNeeded();
+ }
+ return;
+ }
+ mAnswerReceived = true;
+
+ // tell the observers about any cached ICE candidates.
+ for (auto &observer : mSignalingObserverList)
+ {
+ for (auto &candidate : mCachedIceCandidates)
+ {
+ LLWebRTCIceCandidate ice_candidate;
+ ice_candidate.mCandidate = iceCandidateToTrickleString(candidate.get());
+ ice_candidate.mMLineIndex = candidate->sdp_mline_index();
+ ice_candidate.mSdpMid = candidate->sdp_mid();
+ observer->OnIceCandidate(ice_candidate);
+ }
+ }
+ mCachedIceCandidates.clear();
+ if (mPeerConnection)
+ {
+ OnIceGatheringChange(mPeerConnection->ice_gathering_state());
+ }
+
+}
+
+//
+// SetLocalDescriptionObserverInterface implementation.
+//
+void LLWebRTCPeerConnectionImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error)
+{
+}
+
+//
+// DataChannelObserver implementation
+//
+
+void LLWebRTCPeerConnectionImpl::OnStateChange()
+{
+ if (!mDataChannel)
+ {
+ return;
+ }
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " Data Channel State: " << webrtc::DataChannelInterface::DataStateString(mDataChannel->state());
+ switch (mDataChannel->state())
+ {
+ case webrtc::DataChannelInterface::kOpen:
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " Data Channel State Open";
+ for (auto &observer : mSignalingObserverList)
+ {
+ observer->OnDataChannelReady(this);
+ }
+ break;
+ case webrtc::DataChannelInterface::kConnecting:
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " Data Channel State Connecting";
+ break;
+ case webrtc::DataChannelInterface::kClosing:
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " Data Channel State closing";
+ break;
+ case webrtc::DataChannelInterface::kClosed:
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " Data Channel State closed";
+ break;
+ default:
+ break;
+ }
+}
+
+void LLWebRTCPeerConnectionImpl::OnMessage(const webrtc::DataBuffer& buffer)
+{
+ std::string data((const char*)buffer.data.cdata(), buffer.size());
+ for (auto &observer : mDataObserverList)
+ {
+ observer->OnDataReceived(data, buffer.binary);
+ }
+}
+
+//
+// LLWebRTCDataInterface
+//
+
+void LLWebRTCPeerConnectionImpl::sendData(const std::string& data, bool binary)
+{
+ if (mDataChannel)
+ {
+ rtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
+ webrtc::DataBuffer buffer(cowBuffer, binary);
+ mWebRTCImpl->PostNetworkTask([this, buffer]() {
+ if (mDataChannel)
+ {
+ mDataChannel->Send(buffer);
+ }
+ });
+ }
+}
+
+void LLWebRTCPeerConnectionImpl::setDataObserver(LLWebRTCDataObserver* observer)
+{
+ mDataObserverList.emplace_back(observer);
+}
+
+void LLWebRTCPeerConnectionImpl::unsetDataObserver(LLWebRTCDataObserver* observer)
+{
+ std::vector<LLWebRTCDataObserver *>::iterator it =
+ std::find(mDataObserverList.begin(), mDataObserverList.end(), observer);
+ if (it != mDataObserverList.end())
+ {
+ mDataObserverList.erase(it);
+ }
+}
+
+LLWebRTCImpl * gWebRTCImpl = nullptr;
+LLWebRTCDeviceInterface * getDeviceInterface()
+{
+ return gWebRTCImpl;
+}
+
+LLWebRTCPeerConnectionInterface* newPeerConnection()
+{
+ return gWebRTCImpl->newPeerConnection();
+}
+
+void freePeerConnection(LLWebRTCPeerConnectionInterface* peer_connection)
+{
+ gWebRTCImpl->freePeerConnection(peer_connection);
+}
+
+
+void init()
+{
+ gWebRTCImpl = new LLWebRTCImpl();
+ gWebRTCImpl->init();
+}
+
+void terminate()
+{
+ if (gWebRTCImpl)
+ {
+ gWebRTCImpl->terminate();
+ gWebRTCImpl = nullptr;
+ }
+}
+
+} // namespace llwebrtc
generated by cgit v1.2.3 (git 2.25.1) at 2024-12-28 10:59:39 +0000