diff options
Diffstat (limited to 'indra/llwebrtc/llwebrtc.cpp')
| -rw-r--r-- | indra/llwebrtc/llwebrtc.cpp | 98 |
1 files changed, 71 insertions, 27 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp index 34d950b804..c51bcfcdd5 100644 --- a/indra/llwebrtc/llwebrtc.cpp +++ b/indra/llwebrtc/llwebrtc.cpp @@ -157,7 +157,6 @@ LLWebRTCImpl::LLWebRTCImpl() : void LLWebRTCImpl::init() { - RTC_DCHECK(mPeerConnectionFactory); mPlayoutDevice = 0; mRecordingDevice = 0; rtc::InitializeSSL(); @@ -222,12 +221,10 @@ void LLWebRTCImpl::init() mPeerCustomProcessor = new LLCustomProcessor; webrtc::AudioProcessingBuilder apb; apb.SetCapturePostProcessing(std::unique_ptr<webrtc::CustomProcessing>(mPeerCustomProcessor)); - rtc::scoped_refptr<webrtc::AudioProcessing> apm = apb.Create(); + mAudioProcessingModule = apb.Create(); - // TODO: wire some of these to the primary interface and ultimately - // to the UI to allow user config. webrtc::AudioProcessing::Config apm_config; - apm_config.echo_canceller.enabled = true; + 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; @@ -250,8 +247,8 @@ void LLWebRTCImpl::init() processing_config.reverse_output_stream().set_num_channels(2); processing_config.reverse_output_stream().set_sample_rate_hz(48000); - apm->Initialize(processing_config); - apm->ApplyConfig(apm_config); + mAudioProcessingModule->Initialize(processing_config); + mAudioProcessingModule->ApplyConfig(apm_config); mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(), mWorkerThread.get(), @@ -262,7 +259,7 @@ void LLWebRTCImpl::init() nullptr /* video_encoder_factory */, nullptr /* video_decoder_factory */, nullptr /* audio_mixer */, - apm); + mAudioProcessingModule); mWorkerThread->BlockingCall([this]() { mPeerDeviceModule->StartPlayout(); }); } @@ -318,6 +315,49 @@ void LLWebRTCImpl::setRecording(bool recording) }); } +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 = 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.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(); }); @@ -616,32 +656,36 @@ void LLWebRTCPeerConnectionImpl::unsetSignalingObserver(LLWebRTCSignalingObserve } } -// TODO: Add initialization structure through which -// stun and turn servers may be passed in from -// the sim or login. -bool LLWebRTCPeerConnectionImpl::initializeConnection() +bool LLWebRTCPeerConnectionImpl::initializeConnection(LLWebRTCPeerConnectionInterface::InitOptions options) { RTC_DCHECK(!mPeerConnection); mAnswerReceived = false; mWebRTCImpl->PostSignalingTask( - [this]() + [this, options]() { + std::vector<LLWebRTCPeerConnectionInterface::InitOptions::IceServers> servers = options.mServers; + if(servers.empty()) + { + LLWebRTCPeerConnectionInterface::InitOptions::IceServers ice_servers; + ice_servers.mUrls.push_back("stun:stun.l.google.com:19302"); + ice_servers.mUrls.push_back("stun1:stun.l.google.com:19302"); + ice_servers.mUrls.push_back("stun2:stun.l.google.com:19302"); + ice_servers.mUrls.push_back("stun3:stun.l.google.com:19302"); + ice_servers.mUrls.push_back("stun4:stun.l.google.com:19302"); + } + webrtc::PeerConnectionInterface::RTCConfiguration config; + for (auto server : servers) + { + webrtc::PeerConnectionInterface::IceServer ice_server; + ice_server.urls = server.mUrls; + ice_server.username = server.mUserName; + ice_server.password = server.mPassword; + config.servers.push_back(ice_server); + } + config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; - webrtc::PeerConnectionInterface::IceServer server; - server.uri = "stun:roxie-turn.staging.secondlife.io:3478"; - config.servers.push_back(server); - server.uri = "stun:stun.l.google.com:19302"; - config.servers.push_back(server); - server.uri = "stun:stun1.l.google.com:19302"; - config.servers.push_back(server); - server.uri = "stun:stun2.l.google.com:19302"; - config.servers.push_back(server); - server.uri = "stun:stun3.l.google.com:19302"; - config.servers.push_back(server); - server.uri = "stun:stun4.l.google.com:19302"; - config.servers.push_back(server); config.set_min_port(60000); config.set_max_port(60100); @@ -671,7 +715,7 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection() cricket::AudioOptions audioOptions; audioOptions.auto_gain_control = true; - audioOptions.echo_cancellation = true; // incompatible with opus stereo + audioOptions.echo_cancellation = false; // incompatible with opus stereo audioOptions.noise_suppression = true; mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream"); |