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Diffstat (limited to 'indra/llwebrtc/llwebrtc.cpp')
-rw-r--r--indra/llwebrtc/llwebrtc.cpp203
1 files changed, 102 insertions, 101 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index e55d94d128..9b3ec2889b 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -86,7 +86,9 @@ void LLWebRTCImpl::init()
mPlayoutDevice = 0;
mRecordingDevice = 0;
rtc::InitializeSSL();
- rtc::LogMessage::LogToDebug(rtc::LS_WARNING);
+ rtc::LogMessage::LogToDebug(rtc::LS_NONE);
+ rtc::LogMessage::SetLogToStderr(true);
+
mTaskQueueFactory = webrtc::CreateDefaultTaskQueueFactory();
mNetworkThread = rtc::Thread::CreateWithSocketServer();
@@ -98,7 +100,7 @@ void LLWebRTCImpl::init()
mSignalingThread = rtc::Thread::Create();
mSignalingThread->SetName("WebRTCSignalingThread", nullptr);
mSignalingThread->Start();
-
+
mWorkerThread->PostTask(
[this]()
{
@@ -466,93 +468,98 @@ bool LLWebRTCPeerConnectionImpl::initializeConnection()
RTC_DCHECK(!mPeerConnection);
mAnswerReceived = false;
- webrtc::PeerConnectionInterface::RTCConfiguration config;
- 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);
-
- 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();
- 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;
-
- mLocalStream = mPeerConnectionFactory->CreateLocalMediaStream("SLStream");
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
- mPeerConnectionFactory->CreateAudioTrack("SLAudio", mPeerConnectionFactory->CreateAudioSource(audioOptions).get()));
- audio_track->set_enabled(true);
- 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);
+ mWebRTCImpl->PostSignalingTask(
+ [this]()
+ {
+ webrtc::PeerConnectionInterface::RTCConfiguration config;
+ 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);
+
+ 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();
+ 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 = false; // incompatible with opus stereo
+ 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(true);
+ 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;
}
@@ -839,7 +846,7 @@ void LLWebRTCPeerConnectionImpl::OnIceCandidate(const webrtc::IceCandidateInterf
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()));
}
}
@@ -878,21 +885,16 @@ void LLWebRTCPeerConnectionImpl::OnSuccess(webrtc::SessionDescriptionInterface *
sdp_mangled_stream << sdp_line << "\n";
}
}
-
- 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));
+
RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp_mangled_stream.str();
-
-
+
for (auto &observer : mSignalingObserverList)
{
observer->OnOfferAvailable(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));
}
void LLWebRTCPeerConnectionImpl::OnFailure(webrtc::RTCError error) { RTC_LOG(LS_ERROR) << ToString(error.type()) << ": " << error.message(); }
@@ -930,7 +932,6 @@ void LLWebRTCPeerConnectionImpl::OnSetRemoteDescriptionComplete(webrtc::RTCError
//
void LLWebRTCPeerConnectionImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error)
{
-
}
//
generated by cgit v1.2.3 (git 2.25.1) at 2024-09-20 04:03:24 +0000