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-rw-r--r--indra/llwebrtc/llwebrtc.cpp781
1 files changed, 388 insertions, 393 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index 194c7cd5d4..6eb3478b59 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -1,6 +1,6 @@
/**
- * @file llaccordionctrl.cpp
- * @brief Accordion panel implementation
+ * @file llwebrtc.cpp
+ * @brief WebRTC interface implementation
*
* $LicenseInfo:firstyear=2023&license=viewerlgpl$
* Second Life Viewer Source Code
@@ -38,7 +38,7 @@
namespace llwebrtc
{
-const float VOLUME_SCALE_WEBRTC = 3.0f;
+const float VOLUME_SCALE_WEBRTC = 100;
LLAudioDeviceObserver::LLAudioDeviceObserver() : mMicrophoneEnergy(0.0), mSumVector {0} {}
@@ -57,7 +57,7 @@ void LLAudioDeviceObserver::OnCaptureData(const void *audio_samples,
float sample = (static_cast<float>(samples[index]) / (float) 32768);
energy += sample * sample;
}
-
+
// smooth it.
size_t buffer_size = sizeof(mSumVector) / sizeof(mSumVector[0]);
float totalSum = 0;
@@ -82,12 +82,12 @@ void LLAudioDeviceObserver::OnRenderData(const void *audio_samples,
void LLWebRTCImpl::init()
{
+ RTC_DCHECK(mPeerConnectionFactory);
mPlayoutDevice = -1;
mRecordingDevice = -1;
- mAnswerReceived = false;
rtc::InitializeSSL();
mTaskQueueFactory = webrtc::CreateDefaultTaskQueueFactory();
-
+
mNetworkThread = rtc::Thread::CreateWithSocketServer();
mNetworkThread->SetName("WebRTCNetworkThread", nullptr);
mNetworkThread->Start();
@@ -97,61 +97,101 @@ void LLWebRTCImpl::init()
mSignalingThread = rtc::Thread::Create();
mSignalingThread->SetName("WebRTCSignalingThread", nullptr);
mSignalingThread->Start();
-
+
mWorkerThread->PostTask(
- [this]()
- {
- mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
- mTuningDeviceModule =
- webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
- mTaskQueueFactory.get(),
- std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
- mTuningDeviceModule->Init();
- mTuningDeviceModule->SetStereoRecording(false);
- mTuningDeviceModule->SetStereoPlayout(true);
- mTuningDeviceModule->EnableBuiltInAEC(false);
- updateDevices();
- });
+ [this]()
+ {
+ mTuningAudioDeviceObserver = new LLAudioDeviceObserver;
+ mTuningDeviceModule =
+ webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(),
+ std::unique_ptr<webrtc::AudioDeviceDataObserver>(mTuningAudioDeviceObserver));
+ mTuningDeviceModule->Init();
+ mTuningDeviceModule->SetStereoRecording(false);
+ mTuningDeviceModule->SetStereoPlayout(true);
+ mTuningDeviceModule->EnableBuiltInAEC(false);
+ updateDevices();
+ });
+
+ rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();
+ webrtc::AudioProcessing::Config apm_config;
+ apm_config.echo_canceller.enabled = false;
+ apm_config.echo_canceller.mobile_mode = false;
+ apm_config.gain_controller1.enabled = true;
+ apm_config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
+ apm_config.gain_controller2.enabled = true;
+ 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;
+
+ mWorkerThread->BlockingCall(
+ [this]()
+ {
+ mPeerAudioDeviceObserver = new LLAudioDeviceObserver;
+ mPeerDeviceModule =
+ webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
+ mTaskQueueFactory.get(),
+ std::unique_ptr<webrtc::AudioDeviceDataObserver>(mPeerAudioDeviceObserver));
+ mPeerDeviceModule->Init();
+ mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mPeerDeviceModule->SetStereoRecording(false);
+ mPeerDeviceModule->SetStereoPlayout(true);
+ mPeerDeviceModule->EnableBuiltInAEC(false);
+ mPeerDeviceModule->InitMicrophone();
+ mPeerDeviceModule->InitSpeaker();
+ mPeerDeviceModule->InitRecording();
+ mPeerDeviceModule->InitPlayout();
+ mPeerDeviceModule->StartPlayout();
+ mPeerDeviceModule->StartRecording();
+ });
+
+ 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 */,
+ apm);
+ apm->ApplyConfig(apm_config);
}
void LLWebRTCImpl::terminate()
{
+ for (auto& connection : mPeerConnections)
+ {
+ connection->terminate();
+ }
+ mPeerConnections.clear();
+
mSignalingThread->BlockingCall(
- [this]()
- {
- if (mPeerConnection)
- {
- mPeerConnection->Close();
- mPeerConnection = nullptr;
- }
- mPeerConnectionFactory = nullptr;
- });
+ [this]()
+ {
+ mPeerConnectionFactory = nullptr;
+ });
mWorkerThread->BlockingCall(
- [this]()
- {
- if (mTuningDeviceModule)
- {
- mTuningDeviceModule->StopRecording();
- mTuningDeviceModule->Terminate();
- }
- if (mPeerDeviceModule)
- {
- mPeerDeviceModule->StopRecording();
- mPeerDeviceModule->Terminate();
- }
- mTuningDeviceModule = nullptr;
- mPeerDeviceModule = nullptr;
- mTaskQueueFactory = nullptr;
- });
- mNetworkThread->BlockingCall(
- [this]()
- {
- if (mDataChannel)
- {
- mDataChannel->Close();
- mDataChannel = nullptr;
- }
- });
+ [this]()
+ {
+ if (mTuningDeviceModule)
+ {
+ mTuningDeviceModule->StopRecording();
+ mTuningDeviceModule->Terminate();
+ }
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->StopRecording();
+ mPeerDeviceModule->Terminate();
+ }
+ mTuningDeviceModule = nullptr;
+ mPeerDeviceModule = nullptr;
+ mTaskQueueFactory = nullptr;
+ });
}
void LLWebRTCImpl::refreshDevices()
@@ -164,7 +204,7 @@ void LLWebRTCImpl::setDevicesObserver(LLWebRTCDevicesObserver *observer) { mVoic
void LLWebRTCImpl::unsetDevicesObserver(LLWebRTCDevicesObserver *observer)
{
std::vector<LLWebRTCDevicesObserver *>::iterator it =
- std::find(mVoiceDevicesObserverList.begin(), mVoiceDevicesObserverList.end(), observer);
+ std::find(mVoiceDevicesObserverList.begin(), mVoiceDevicesObserverList.end(), observer);
if (it != mVoiceDevicesObserverList.end())
{
mVoiceDevicesObserverList.erase(it);
@@ -174,97 +214,97 @@ void LLWebRTCImpl::unsetDevicesObserver(LLWebRTCDevicesObserver *observer)
void LLWebRTCImpl::setCaptureDevice(const std::string &id)
{
mWorkerThread->PostTask(
- [this, id]()
- {
- int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
- for (int16_t i = 0; i < captureDeviceCount; i++)
- {
- char name[webrtc::kAdmMaxDeviceNameSize];
- char guid[webrtc::kAdmMaxGuidSize];
- mTuningDeviceModule->RecordingDeviceName(i, name, guid);
- if (id == guid || id == "Default") // first one in list is default
- {
- RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
- mRecordingDevice = i;
- break;
- }
- }
- mTuningDeviceModule->StopRecording();
- mTuningDeviceModule->SetRecordingDevice(mRecordingDevice);
- mTuningDeviceModule->InitMicrophone();
- mTuningDeviceModule->InitRecording();
- mTuningDeviceModule->StartRecording();
- bool was_peer_recording = false;
- if (mPeerDeviceModule)
- {
- was_peer_recording = mPeerDeviceModule->Recording();
- if (was_peer_recording)
- {
- mPeerDeviceModule->StopRecording();
- }
- mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
- mPeerDeviceModule->InitMicrophone();
- mPeerDeviceModule->InitRecording();
- if (was_peer_recording)
- {
- mPeerDeviceModule->StartRecording();
- }
- }
- });
+ [this, id]()
+ {
+ int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
+ for (int16_t i = 0; i < captureDeviceCount; i++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mTuningDeviceModule->RecordingDeviceName(i, name, guid);
+ if (id == guid || id == "Default") // first one in list is default
+ {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
+ mRecordingDevice = i;
+ break;
+ }
+ }
+ mTuningDeviceModule->StopRecording();
+ mTuningDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mTuningDeviceModule->InitMicrophone();
+ mTuningDeviceModule->InitRecording();
+ mTuningDeviceModule->StartRecording();
+ bool was_peer_recording = false;
+ if (mPeerDeviceModule)
+ {
+ was_peer_recording = mPeerDeviceModule->Recording();
+ if (was_peer_recording)
+ {
+ mPeerDeviceModule->StopRecording();
+ }
+ mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
+ mPeerDeviceModule->InitMicrophone();
+ mPeerDeviceModule->InitRecording();
+ if (was_peer_recording)
+ {
+ mPeerDeviceModule->StartRecording();
+ }
+ }
+ });
}
void LLWebRTCImpl::setRenderDevice(const std::string &id)
{
mWorkerThread->PostTask(
- [this, id]()
- {
- int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
- for (int16_t i = 0; i < renderDeviceCount; i++)
- {
- char name[webrtc::kAdmMaxDeviceNameSize];
- char guid[webrtc::kAdmMaxGuidSize];
- mTuningDeviceModule->PlayoutDeviceName(i, name, guid);
- if (id == guid || id == "Default")
- {
- RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
- mPlayoutDevice = i;
- break;
- }
- }
- mTuningDeviceModule->SetSpeakerMute(true);
- bool was_tuning_playing = mTuningDeviceModule->Playing();
- if (was_tuning_playing)
- {
- mTuningDeviceModule->StopPlayout();
- }
- bool was_peer_mute = false;
- if (mPeerDeviceModule)
- {
- mPeerDeviceModule->SpeakerMute(&was_peer_mute);
- if (!was_peer_mute)
- {
- mPeerDeviceModule->SetSpeakerMute(true);
- }
- }
-
- mTuningDeviceModule->SetPlayoutDevice(mPlayoutDevice);
- mTuningDeviceModule->InitSpeaker();
- mTuningDeviceModule->InitPlayout();
- if (was_tuning_playing)
- {
- mTuningDeviceModule->StartPlayout();
- }
- if (mPeerDeviceModule)
- {
- mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
- mPeerDeviceModule->InitSpeaker();
- mPeerDeviceModule->InitPlayout();
- mPeerDeviceModule->StartPlayout();
- mPeerDeviceModule->SetSpeakerMute(was_peer_mute);
-
- }
- mTuningDeviceModule->SetSpeakerMute(false);
- });
+ [this, id]()
+ {
+ int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
+ for (int16_t i = 0; i < renderDeviceCount; i++)
+ {
+ char name[webrtc::kAdmMaxDeviceNameSize];
+ char guid[webrtc::kAdmMaxGuidSize];
+ mTuningDeviceModule->PlayoutDeviceName(i, name, guid);
+ if (id == guid || id == "Default")
+ {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
+ mPlayoutDevice = i;
+ break;
+ }
+ }
+ mTuningDeviceModule->SetSpeakerMute(true);
+ bool was_tuning_playing = mTuningDeviceModule->Playing();
+ if (was_tuning_playing)
+ {
+ mTuningDeviceModule->StopPlayout();
+ }
+ bool was_peer_mute = false;
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->SpeakerMute(&was_peer_mute);
+ if (!was_peer_mute)
+ {
+ mPeerDeviceModule->SetSpeakerMute(true);
+ }
+ }
+
+ mTuningDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mTuningDeviceModule->InitSpeaker();
+ mTuningDeviceModule->InitPlayout();
+ if (was_tuning_playing)
+ {
+ mTuningDeviceModule->StartPlayout();
+ }
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
+ mPeerDeviceModule->InitSpeaker();
+ mPeerDeviceModule->InitPlayout();
+ mPeerDeviceModule->StartPlayout();
+ mPeerDeviceModule->SetSpeakerMute(was_peer_mute);
+
+ }
+ mTuningDeviceModule->SetSpeakerMute(false);
+ });
}
void LLWebRTCImpl::updateDevices()
@@ -278,7 +318,7 @@ void LLWebRTCImpl::updateDevices()
mTuningDeviceModule->PlayoutDeviceName(index, name, guid);
renderDeviceList.emplace_back(name, guid);
}
-
+
int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
LLWebRTCVoiceDeviceList captureDeviceList;
for (int16_t index = 0; index < captureDeviceCount; index++)
@@ -297,113 +337,100 @@ void LLWebRTCImpl::updateDevices()
void LLWebRTCImpl::setTuningMode(bool enable)
{
mWorkerThread->BlockingCall(
- [this, enable]()
- {
- if (enable)
- {
+ [this, enable]()
+ {
+ if (enable)
+ {
+
+ mTuningDeviceModule->StartRecording();
+ mTuningDeviceModule->SetMicrophoneMute(false);
+
+
+ mTuningDeviceModule->SetSpeakerMute(false);
+
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->StopRecording();
+ mPeerDeviceModule->SetSpeakerMute(true);
+ }
+ }
+ else
+ {
+ if (mPeerDeviceModule)
+ {
+ mPeerDeviceModule->StartRecording();
+ mPeerDeviceModule->SetSpeakerMute(false);
+ }
+ }
+ });
+ for (auto& connection : mPeerConnections)
+ {
+ connection->enableTracks(enable ? false : !mMute);
+ }
+}
- mTuningDeviceModule->StartRecording();
- mTuningDeviceModule->SetMicrophoneMute(false);
+float LLWebRTCImpl::getTuningAudioLevel() { return 20 * mTuningAudioDeviceObserver->getMicrophoneEnergy(); }
+float LLWebRTCImpl::getPeerAudioLevel() { return 20 * mPeerAudioDeviceObserver->getMicrophoneEnergy(); }
- mTuningDeviceModule->SetSpeakerMute(false);
+void LLWebRTCImpl::setSpeakerVolume(float volume) { mPeerDeviceModule->SetSpeakerVolume(volume * VOLUME_SCALE_WEBRTC);}
+void LLWebRTCImpl::setMicrophoneVolume(float volume) { mPeerDeviceModule->SetMicrophoneVolume(volume * VOLUME_SCALE_WEBRTC);}
- if (mPeerDeviceModule)
- {
- mPeerDeviceModule->StopRecording();
- mPeerDeviceModule->SetSpeakerMute(true);
- }
- }
- else
- {
- if (mPeerDeviceModule)
- {
- mPeerDeviceModule->StartRecording();
- mPeerDeviceModule->SetSpeakerMute(false);
- }
- }
- });
- // set_enabled shouldn't be done on the worker thread
- if (mPeerConnection)
+//
+// Helpers
+//
+
+LLWebRTCPeerConnection * LLWebRTCImpl::newPeerConnection()
+{
+ rtc::scoped_refptr<LLWebRTCPeerConnectionImpl> peerConnection = rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>(new rtc::RefCountedObject<LLWebRTCPeerConnectionImpl>());
+ peerConnection->init(this);
+
+ mPeerConnections.emplace_back(peerConnection);
+ peerConnection->enableTracks(!mMute);
+ return peerConnection.get();
+}
+void LLWebRTCImpl::freePeerConnection(LLWebRTCPeerConnection * peer_connection)
+{
+ std::vector<rtc::scoped_refptr<LLWebRTCPeerConnectionImpl>>::iterator it =
+ std::find(mPeerConnections.begin(), mPeerConnections.end(), peer_connection);
+ if (it != mPeerConnections.end())
{
- auto senders = mPeerConnection->GetSenders();
- for (auto &sender : senders)
- {
- sender->track()->set_enabled(enable ? false : !mMute);
- }
+ (*it)->terminate();
+ mPeerConnections.erase(it);
}
}
//
-// LLWebRTCSignalInterface
+// LLWebRTCPeerConnection interface
//
-void LLWebRTCImpl::setSignalingObserver(LLWebRTCSignalingObserver *observer) { mSignalingObserverList.emplace_back(observer); }
+void LLWebRTCPeerConnectionImpl::init(LLWebRTCImpl * webrtc_impl)
+{
+ mWebRTCImpl = webrtc_impl;
+ mPeerConnectionFactory = mWebRTCImpl->getPeerConnectionFactory();
+}
+void LLWebRTCPeerConnectionImpl::terminate()
+{
+ shutdownConnection();
+}
+
+void LLWebRTCPeerConnectionImpl::setSignalingObserver(LLWebRTCSignalingObserver *observer) { mSignalingObserverList.emplace_back(observer); }
-void LLWebRTCImpl::unsetSignalingObserver(LLWebRTCSignalingObserver *observer)
+void LLWebRTCPeerConnectionImpl::unsetSignalingObserver(LLWebRTCSignalingObserver *observer)
{
std::vector<LLWebRTCSignalingObserver *>::iterator it =
- std::find(mSignalingObserverList.begin(), mSignalingObserverList.end(), observer);
+ std::find(mSignalingObserverList.begin(), mSignalingObserverList.end(), observer);
if (it != mSignalingObserverList.end())
{
mSignalingObserverList.erase(it);
}
}
-bool LLWebRTCImpl::initializeConnection()
+bool LLWebRTCPeerConnectionImpl::initializeConnection()
{
RTC_DCHECK(!mPeerConnection);
- RTC_DCHECK(mPeerConnectionFactory);
mAnswerReceived = false;
- rtc::scoped_refptr<webrtc::AudioProcessing> apm = webrtc::AudioProcessingBuilder().Create();
- webrtc::AudioProcessing::Config apm_config;
- apm_config.echo_canceller.enabled = false;
- apm_config.echo_canceller.mobile_mode = false;
- apm_config.gain_controller1.enabled = true;
- apm_config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveAnalog;
- apm_config.gain_controller2.enabled = true;
- 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;
-
- mWorkerThread->BlockingCall(
- [this]()
- {
- mPeerAudioDeviceObserver = new LLAudioDeviceObserver;
- mPeerDeviceModule =
- webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio,
- mTaskQueueFactory.get(),
- std::unique_ptr<webrtc::AudioDeviceDataObserver>(mPeerAudioDeviceObserver));
- mPeerDeviceModule->Init();
- mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
- mPeerDeviceModule->SetRecordingDevice(mRecordingDevice);
- mPeerDeviceModule->SetStereoRecording(false);
- mPeerDeviceModule->SetStereoPlayout(true);
- mPeerDeviceModule->EnableBuiltInAEC(false);
- mPeerDeviceModule->InitMicrophone();
- mPeerDeviceModule->InitSpeaker();
- mPeerDeviceModule->InitRecording();
- mPeerDeviceModule->InitPlayout();
- mPeerDeviceModule->StopPlayout();
- mPeerDeviceModule->StopRecording();
- });
-
- 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 */,
- apm);
- apm->ApplyConfig(apm_config);
-
webrtc::PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
webrtc::PeerConnectionInterface::IceServer server;
@@ -419,7 +446,7 @@ bool LLWebRTCImpl::initializeConnection()
config.servers.push_back(server);
server.uri = "stun:stun4.l.google.com:19302";
config.servers.push_back(server);
-
+
webrtc::PeerConnectionDependencies pc_dependencies(this);
auto error_or_peer_connection = mPeerConnectionFactory->CreatePeerConnectionOrError(config, std::move(pc_dependencies));
if (error_or_peer_connection.ok())
@@ -432,35 +459,36 @@ bool LLWebRTCImpl::initializeConnection()
shutdownConnection();
return false;
}
-
+
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);
}
-
+
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
-
+
cricket::AudioOptions audioOptions;
audioOptions.auto_gain_control = true;
audioOptions.echo_cancellation = false; // incompatible with opus stereo
audioOptions.noise_suppression = true;
-
+
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
+
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(audioOptions).get()));
+ mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(audioOptions).get()));
audio_track->set_enabled(true);
stream->AddTrack(audio_track);
-
+
mPeerConnection->AddTrack(audio_track, {"SLStream"});
-
+
auto senders = mPeerConnection->GetSenders();
-
+
for (auto &sender : senders)
{
webrtc::RtpParameters params;
@@ -474,7 +502,7 @@ bool LLWebRTCImpl::initializeConnection()
params.codecs.push_back(codecparam);
sender->SetParameters(params);
}
-
+
auto receivers = mPeerConnection->GetReceivers();
for (auto &receiver : receivers)
{
@@ -491,102 +519,78 @@ bool LLWebRTCImpl::initializeConnection()
}
webrtc::PeerConnectionInterface::RTCOfferAnswerOptions offerOptions;
mPeerConnection->CreateOffer(this, offerOptions);
-
+
return true;
}
-void LLWebRTCImpl::shutdownConnection()
+void LLWebRTCPeerConnectionImpl::shutdownConnection()
{
- mSignalingThread->BlockingCall(
- [this]()
- {
- if (mPeerConnection)
- {
- mPeerConnection->Close();
- mPeerConnection = nullptr;
- }
- mPeerConnectionFactory = nullptr;
- });
- mWorkerThread->BlockingCall(
- [this]()
- {
- if (mPeerDeviceModule)
- {
- mPeerDeviceModule->StopRecording();
- mPeerDeviceModule->Terminate();
- }
- mPeerDeviceModule = nullptr;
- if (mPeerAudioDeviceObserver)
- {
- mPeerAudioDeviceObserver = nullptr;
- }
- });
- mNetworkThread->BlockingCall(
- [this]()
+ mWebRTCImpl->SignalingBlockingCall(
+ [this]()
+ {
+ if (mPeerConnection)
+ {
+ mPeerConnection->Close();
+ mPeerConnection = nullptr;
+ }
+ });
+
+ mWebRTCImpl->NetworkBlockingCall(
+ [this]()
+ {
+ if (mDataChannel)
+ {
+ mDataChannel->Close();
+ mDataChannel = nullptr;
+ }
+ });
+}
+
+void LLWebRTCPeerConnectionImpl::enableTracks(bool enable)
+{
+ // set_enabled shouldn't be done on the worker thread
+ if (mPeerConnection)
+ {
+ auto senders = mPeerConnection->GetSenders();
+ for (auto &sender : senders)
{
- if (mDataChannel)
- {
- mDataChannel->Close();
- mDataChannel = nullptr;
- }
- });
+ sender->track()->set_enabled(enable);
+ }
+ }
}
-void LLWebRTCImpl::AnswerAvailable(const std::string &sdp)
+void LLWebRTCPeerConnectionImpl::AnswerAvailable(const std::string &sdp)
{
RTC_LOG(LS_INFO) << __FUNCTION__ << " Remote SDP: " << sdp;
-
- mSignalingThread->PostTask(
- [this, sdp]()
- {
- RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
- mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
- rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
- });
+
+ mWebRTCImpl->PostSignalingTask(
+ [this, sdp]()
+ {
+ RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
+ mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
+ rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
+ });
}
-void LLWebRTCImpl::setMute(bool mute)
+void LLWebRTCPeerConnectionImpl::setMute(bool mute)
{
mMute = mute;
auto senders = mPeerConnection->GetSenders();
-
+
RTC_LOG(LS_INFO) << __FUNCTION__ << (mute ? "disabling" : "enabling") << " streams count " << senders.size();
-
+
for (auto &sender : senders)
{
sender->track()->set_enabled(!mMute);
}
}
-void LLWebRTCImpl::setSpeakerVolume(float volume)
-{
- mSignalingThread->PostTask(
- [this, volume]()
- {
- auto receivers = mPeerConnection->GetReceivers();
-
- RTC_LOG(LS_INFO) << __FUNCTION__ << "Set volume" << receivers.size();
- for (auto &receiver : receivers)
- {
- webrtc::MediaStreamTrackInterface *track = receiver->track().get();
- if (track->kind() == webrtc::MediaStreamTrackInterface::kAudioKind)
- {
- webrtc::AudioTrackInterface *audio_track = static_cast<webrtc::AudioTrackInterface *>(track);
- webrtc::AudioSourceInterface *source = audio_track->GetSource();
- source->SetVolume(VOLUME_SCALE_WEBRTC * volume);
- }
- }
- });
-}
-
-float LLWebRTCImpl::getTuningAudioLevel() { return 20 * mTuningAudioDeviceObserver->getMicrophoneEnergy(); }
-
//
// PeerConnectionObserver implementation.
//
-void LLWebRTCImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
- const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>> &streams)
+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;
@@ -601,18 +605,18 @@ void LLWebRTCImpl::OnAddTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface>
receiver->SetParameters(params);
}
-void LLWebRTCImpl::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver)
+void LLWebRTCPeerConnectionImpl::OnRemoveTrack(rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver)
{
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << receiver->id();
}
-void LLWebRTCImpl::OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> channel)
+void LLWebRTCPeerConnectionImpl::OnDataChannel(rtc::scoped_refptr<webrtc::DataChannelInterface> channel)
{
mDataChannel = channel;
channel->RegisterObserver(this);
}
-void LLWebRTCImpl::OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)
+void LLWebRTCPeerConnectionImpl::OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGatheringState new_state)
{
LLWebRTCSignalingObserver::IceGatheringState webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_NEW;
switch (new_state)
@@ -631,7 +635,7 @@ void LLWebRTCImpl::OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGath
webrtc_new_state = LLWebRTCSignalingObserver::IceGatheringState::ICE_GATHERING_NEW;
return;
}
-
+
if (mAnswerReceived)
{
for (auto &observer : mSignalingObserverList)
@@ -642,19 +646,17 @@ void LLWebRTCImpl::OnIceGatheringChange(webrtc::PeerConnectionInterface::IceGath
}
// Called any time the PeerConnectionState changes.
-void LLWebRTCImpl::OnConnectionChange(webrtc::PeerConnectionInterface::PeerConnectionState new_state)
+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:
{
if (new_state == webrtc::PeerConnectionInterface::PeerConnectionState::kConnected)
{
- mWorkerThread->PostTask([this]() {
- mPeerDeviceModule->StartRecording();
- mPeerDeviceModule->StartPlayout();
+ mWebRTCImpl->PostWorkerTask([this]() {
for (auto &observer : mSignalingObserverList)
{
observer->OnAudioEstablished(this);
@@ -670,7 +672,7 @@ void LLWebRTCImpl::OnConnectionChange(webrtc::PeerConnectionInterface::PeerConne
{
observer->OnRenegotiationNeeded();
}
-
+
break;
}
default:
@@ -683,15 +685,15 @@ void LLWebRTCImpl::OnConnectionChange(webrtc::PeerConnectionInterface::PeerConne
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 ";
-
+ 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";
@@ -699,25 +701,25 @@ static std::string iceCandidateToTrickleString(const webrtc::IceCandidateInterfa
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();
+ "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();
+ "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();
+ "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")
+ if (candidate->candidate().protocol() == "tcp")
{
candidate_stream << " tcptype " << candidate->candidate().tcptype();
}
@@ -725,16 +727,16 @@ static std::string iceCandidateToTrickleString(const webrtc::IceCandidateInterfa
return candidate_stream.str();
}
-void LLWebRTCImpl::OnIceCandidate(const webrtc::IceCandidateInterface *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)
+ if (mAnswerReceived)
{
for (auto &observer : mSignalingObserverList)
{
@@ -745,22 +747,22 @@ void LLWebRTCImpl::OnIceCandidate(const webrtc::IceCandidateInterface *candidate
observer->OnIceCandidate(ice_candidate);
}
}
- else
+ else
{
mCachedIceCandidates.push_back(
- webrtc::CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(), candidate->candidate()));
+ webrtc::CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(), candidate->candidate()));
}
}
//
// CreateSessionDescriptionObserver implementation.
//
-void LLWebRTCImpl::OnSuccess(webrtc::SessionDescriptionInterface *desc)
+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);
@@ -780,7 +782,7 @@ void LLWebRTCImpl::OnSuccess(webrtc::SessionDescriptionInterface *desc)
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\n";
+ << " minptime=10;useinbandfec=1;stereo=1;sprop-stereo=1;maxplaybackrate=48000\n";
}
else
{
@@ -789,27 +791,27 @@ void LLWebRTCImpl::OnSuccess(webrtc::SessionDescriptionInterface *desc)
}
webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str());
-
-
-
+
+
+
mPeerConnection->SetLocalDescription(std::unique_ptr<webrtc::SessionDescriptionInterface>(
- webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str())),
- rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
+ webrtc::CreateSessionDescription(webrtc::SdpType::kOffer, sdp_mangled_stream.str())),
+ rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this));
RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp_mangled_stream.str();
-
-
+
+
for (auto &observer : mSignalingObserverList)
{
observer->OnOfferAvailable(sdp_mangled_stream.str());
}
}
-void LLWebRTCImpl::OnFailure(webrtc::RTCError error) { RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); }
+void LLWebRTCPeerConnectionImpl::OnFailure(webrtc::RTCError error) { RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); }
//
// SetRemoteDescriptionObserverInterface implementation.
//
-void LLWebRTCImpl::OnSetRemoteDescriptionComplete(webrtc::RTCError error)
+void LLWebRTCPeerConnectionImpl::OnSetRemoteDescriptionComplete(webrtc::RTCError error)
{
RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
if (!error.ok())
@@ -831,37 +833,20 @@ void LLWebRTCImpl::OnSetRemoteDescriptionComplete(webrtc::RTCError error)
}
mCachedIceCandidates.clear();
OnIceGatheringChange(mPeerConnection->ice_gathering_state());
-
+
}
//
// SetLocalDescriptionObserverInterface implementation.
//
-void LLWebRTCImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error)
+void LLWebRTCPeerConnectionImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error)
{
-#if 0
- RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state();
- if (!error.ok())
- {
- RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message();
- return;
- }
- auto desc = mPeerConnection->pending_local_description();
- std::string sdp;
- desc->ToString(&sdp);
-
- RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp;
- ;
- for (auto &observer : mSignalingObserverList)
- {
- observer->OnOfferAvailable(sdp);
- }
-#endif
+
}
-void LLWebRTCImpl::setAudioObserver(LLWebRTCAudioObserver *observer) { mAudioObserverList.emplace_back(observer); }
+void LLWebRTCPeerConnectionImpl::setAudioObserver(LLWebRTCAudioObserver *observer) { mAudioObserverList.emplace_back(observer); }
-void LLWebRTCImpl::unsetAudioObserver(LLWebRTCAudioObserver *observer)
+void LLWebRTCPeerConnectionImpl::unsetAudioObserver(LLWebRTCAudioObserver *observer)
{
std::vector<LLWebRTCAudioObserver *>::iterator it = std::find(mAudioObserverList.begin(), mAudioObserverList.end(), observer);
if (it != mAudioObserverList.end())
@@ -870,22 +855,20 @@ void LLWebRTCImpl::unsetAudioObserver(LLWebRTCAudioObserver *observer)
}
}
-float LLWebRTCImpl::getAudioLevel() { return 20 * mPeerAudioDeviceObserver->getMicrophoneEnergy(); }
-
//
// DataChannelObserver implementation
//
-void LLWebRTCImpl::OnStateChange()
-{
+void LLWebRTCPeerConnectionImpl::OnStateChange()
+{
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 : mDataObserverList)
+ for (auto &observer : mSignalingObserverList)
{
- observer->OnDataChannelReady();
+ observer->OnDataChannelReady(this);
}
break;
case webrtc::DataChannelInterface::kConnecting:
@@ -903,7 +886,7 @@ void LLWebRTCImpl::OnStateChange()
}
-void LLWebRTCImpl::OnMessage(const webrtc::DataBuffer& buffer)
+void LLWebRTCPeerConnectionImpl::OnMessage(const webrtc::DataBuffer& buffer)
{
std::string data((const char*)buffer.data.cdata(), buffer.size());
for (auto &observer : mDataObserverList)
@@ -912,43 +895,55 @@ void LLWebRTCImpl::OnMessage(const webrtc::DataBuffer& buffer)
}
}
-void LLWebRTCImpl::sendData(const std::string& data, bool binary)
+void LLWebRTCPeerConnectionImpl::sendData(const std::string& data, bool binary)
{
rtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
webrtc::DataBuffer buffer(cowBuffer, binary);
mDataChannel->Send(buffer);
}
-void LLWebRTCImpl::setDataObserver(LLWebRTCDataObserver* observer) { mDataObserverList.emplace_back(observer); }
+void LLWebRTCPeerConnectionImpl::setDataObserver(LLWebRTCDataObserver* observer) { mDataObserverList.emplace_back(observer); }
-void LLWebRTCImpl::unsetDataObserver(LLWebRTCDataObserver* observer)
+void LLWebRTCPeerConnectionImpl::unsetDataObserver(LLWebRTCDataObserver* observer)
{
std::vector<LLWebRTCDataObserver *>::iterator it =
- std::find(mDataObserverList.begin(), mDataObserverList.end(), observer);
+ std::find(mDataObserverList.begin(), mDataObserverList.end(), observer);
if (it != mDataObserverList.end())
{
mDataObserverList.erase(it);
}
}
-rtc::RefCountedObject<LLWebRTCImpl> *gWebRTCImpl = nullptr;
-LLWebRTCDeviceInterface *getDeviceInterface() { return gWebRTCImpl; }
-LLWebRTCSignalInterface *getSignalingInterface() { return gWebRTCImpl; }
-LLWebRTCDataInterface *getDataInterface() { return gWebRTCImpl; }
+LLWebRTCImpl * gWebRTCImpl = nullptr;
+LLWebRTCDeviceInterface * getDeviceInterface()
+{
+ return gWebRTCImpl;
+}
+
+LLWebRTCPeerConnection * newPeerConnection()
+{
+ return gWebRTCImpl->newPeerConnection();
+}
+
+void freePeerConnection(LLWebRTCPeerConnection *peer_connection)
+{
+ gWebRTCImpl->freePeerConnection(peer_connection);
+}
void init()
{
- gWebRTCImpl = new rtc::RefCountedObject<LLWebRTCImpl>();
- gWebRTCImpl->AddRef();
+ gWebRTCImpl = new LLWebRTCImpl();
gWebRTCImpl->init();
}
void terminate()
-{
- gWebRTCImpl->terminate();
- gWebRTCImpl->Release();
- gWebRTCImpl = nullptr;
+{
+ if (gWebRTCImpl)
+ {
+ gWebRTCImpl->terminate();
+ gWebRTCImpl = nullptr;
+ }
}
} // namespace llwebrtc
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-10 17:50:02 +0000