diff options
Diffstat (limited to 'indra/llwebrtc/llwebrtc.cpp')
| -rw-r--r-- | indra/llwebrtc/llwebrtc.cpp | 94 |
1 files changed, 73 insertions, 21 deletions
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp index edf941436e..96be2c1f0b 100644 --- a/indra/llwebrtc/llwebrtc.cpp +++ b/indra/llwebrtc/llwebrtc.cpp @@ -55,7 +55,7 @@ void LLWebRTCImpl::init() mSignalingThread->SetName("WebRTCSignalingThread", nullptr); mSignalingThread->Start(); - mSignalingThread->PostTask( + mWorkerThread->PostTask( [this]() { mDeviceModule = webrtc::CreateAudioDeviceWithDataObserver(webrtc::AudioDeviceModule::AudioLayer::kPlatformDefaultAudio, @@ -68,7 +68,7 @@ void LLWebRTCImpl::init() void LLWebRTCImpl::refreshDevices() { - mSignalingThread->PostTask([this]() { updateDevices(); }); + mWorkerThread->PostTask([this]() { updateDevices(); }); } void LLWebRTCImpl::setDevicesObserver(LLWebRTCDevicesObserver *observer) { mVoiceDevicesObserverList.emplace_back(observer); } @@ -85,41 +85,48 @@ void LLWebRTCImpl::unsetDevicesObserver(LLWebRTCDevicesObserver *observer) void LLWebRTCImpl::setCaptureDevice(const std::string &id) { - mSignalingThread->PostTask( + mWorkerThread->PostTask( [this, id]() { + mDeviceModule->StopRecording(); int16_t captureDeviceCount = mDeviceModule->RecordingDevices(); for (int16_t index = 0; index < captureDeviceCount; index++) { char name[webrtc::kAdmMaxDeviceNameSize]; char guid[webrtc::kAdmMaxGuidSize]; mDeviceModule->RecordingDeviceName(index, name, guid); - if (id == guid || id == name) + if (id == guid || id == "Default") { + RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << index; mDeviceModule->SetRecordingDevice(index); break; } } + mDeviceModule->InitRecording(); + mDeviceModule->StartRecording(); }); } void LLWebRTCImpl::setRenderDevice(const std::string &id) { - mSignalingThread->PostTask( + mWorkerThread->PostTask( [this, id]() { + mDeviceModule->StopPlayout(); int16_t renderDeviceCount = mDeviceModule->RecordingDevices(); for (int16_t index = 0; index < renderDeviceCount; index++) { char name[webrtc::kAdmMaxDeviceNameSize]; char guid[webrtc::kAdmMaxGuidSize]; mDeviceModule->PlayoutDeviceName(index, name, guid); - if (id == guid || id == name) + if (id == guid || id == "Default") { mDeviceModule->SetPlayoutDevice(index); break; } } + mDeviceModule->InitPlayout(); + mDeviceModule->StartPlayout(); }); } @@ -156,7 +163,7 @@ void LLWebRTCImpl::updateDevices() void LLWebRTCImpl::setTuningMode(bool enable) { - mSignalingThread->PostTask( + mWorkerThread->PostTask( [this, enable]() { if (enable) @@ -181,7 +188,7 @@ void LLWebRTCImpl::OnCaptureData(const void *audio_samples, const size_t num_channels, const uint32_t samples_per_sec) { - if (bytes_per_sample != 4) + if (bytes_per_sample != 2) { return; } @@ -220,28 +227,31 @@ void LLWebRTCImpl::unsetSignalingObserver(LLWebRTCSignalingObserver *observer) } } + bool LLWebRTCImpl::initializeConnection() { RTC_DCHECK(!mPeerConnection); - RTC_DCHECK(!mPeerConnectionFactory); + RTC_DCHECK(mPeerConnectionFactory); mAnswerReceived = false; + + mSignalingThread->PostTask([this]() { initializeConnectionThreaded(); }); + return true; +} + + + +bool LLWebRTCImpl::initializeConnectionThreaded() +{ mPeerConnectionFactory = webrtc::CreatePeerConnectionFactory(mNetworkThread.get(), mWorkerThread.get(), mSignalingThread.get(), - nullptr /* default_adm */, + mDeviceModule, webrtc::CreateBuiltinAudioEncoderFactory(), webrtc::CreateBuiltinAudioDecoderFactory(), nullptr /* video_encoder_factory */, nullptr /* video_decoder_factory */, nullptr /* audio_mixer */, nullptr /* audio_processing */); - - if (!mPeerConnectionFactory) - { - shutdownConnection(); - return false; - } - webrtc::PeerConnectionInterface::RTCConfiguration config; config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan; webrtc::PeerConnectionInterface::IceServer server; @@ -272,16 +282,34 @@ bool LLWebRTCImpl::initializeConnection() cricket::AudioOptions audioOptions; audioOptions.auto_gain_control = true; - audioOptions.echo_cancellation = 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(cricket::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; + webrtc::RtpCodecParameters codecparam; + codecparam.name = "opus"; + codecparam.kind = cricket::MEDIA_TYPE_AUDIO; + codecparam.clock_rate = 48000; + codecparam.num_channels = 1; + codecparam.parameters["stereo"] = "0"; + codecparam.parameters["sprop-stereo"] = "0"; + + params.codecs.push_back(codecparam); + sender->SetParameters(params); + } + mPeerConnection->SetLocalDescription(rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface>(this)); RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->signaling_state(); @@ -297,6 +325,12 @@ void LLWebRTCImpl::shutdownConnection() void LLWebRTCImpl::AnswerAvailable(const std::string &sdp) { + std::istringstream sdp_stream(sdp); + std::string sdp_line; + while (std::getline(sdp_stream, sdp_line)) + { + RTC_LOG(LS_INFO) << __FUNCTION__ << " Remote SDP: " << sdp_line; + } mSignalingThread->PostTask( [this, sdp]() { @@ -515,10 +549,28 @@ void LLWebRTCImpl::OnSetLocalDescriptionComplete(webrtc::RTCError error) auto desc = mPeerConnection->pending_local_description(); std::string sdp; desc->ToString(&sdp); - RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << 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; + while (std::getline(sdp_stream, sdp_line)) { + int bandwidth = 0; + int payload_id = 0; + RTC_LOG(LS_INFO) << __FUNCTION__ << " Local SDP: " << sdp_line; + // force mono + if (std::sscanf(sdp_line.c_str(), "a=rtpmap:%i opus/%i/2", &payload_id, &bandwidth) == 2) + { + sdp_mangled_stream << sdp_line << "\n"; + } + else + { + sdp_mangled_stream << sdp_line << "\n"; + } + } for (auto &observer : mSignalingObserverList) { - observer->OnOfferAvailable(sdp); + observer->OnOfferAvailable(sdp_mangled_stream.str()); } } |