Refactor/clean-up WebRTC voice to handle multiple voice streams
This is useful for cross-region voice, quick voice switching, etc.meow-7.2.2
Roxie Linden
parent
c33ebcfab6
commit
ebfa44cdb7
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@ -216,6 +216,7 @@ void LLWebRTCImpl::setCaptureDevice(const std::string &id)
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mWorkerThread->PostTask(
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[this, id]()
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{
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int16_t tuningRecordingDevice = 0;
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int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
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for (int16_t i = 0; i < captureDeviceCount; i++)
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{
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@ -225,18 +226,32 @@ void LLWebRTCImpl::setCaptureDevice(const std::string &id)
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if (id == guid || id == "Default") // first one in list is default
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{
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RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
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mRecordingDevice = i;
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tuningRecordingDevice = i;
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break;
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}
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}
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mTuningDeviceModule->StopRecording();
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mTuningDeviceModule->SetRecordingDevice(mRecordingDevice);
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mTuningDeviceModule->SetRecordingDevice(tuningRecordingDevice);
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mTuningDeviceModule->InitMicrophone();
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mTuningDeviceModule->InitRecording();
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mTuningDeviceModule->StartRecording();
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bool was_peer_recording = false;
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if (mPeerDeviceModule)
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{
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int16_t captureDeviceCount = mTuningDeviceModule->RecordingDevices();
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for (int16_t i = 0; i < captureDeviceCount; i++)
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{
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char name[webrtc::kAdmMaxDeviceNameSize];
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char guid[webrtc::kAdmMaxGuidSize];
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mTuningDeviceModule->RecordingDeviceName(i, name, guid);
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if (id == guid || id == "Default") // first one in list is default
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{
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RTC_LOG(LS_INFO)
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<< __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
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mRecordingDevice = i;
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break;
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}
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}
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was_peer_recording = mPeerDeviceModule->Recording();
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if (was_peer_recording)
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{
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@ -259,6 +274,7 @@ void LLWebRTCImpl::setRenderDevice(const std::string &id)
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[this, id]()
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{
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int16_t renderDeviceCount = mTuningDeviceModule->PlayoutDevices();
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int16_t tuningPlayoutDevice = 0;
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for (int16_t i = 0; i < renderDeviceCount; i++)
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{
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char name[webrtc::kAdmMaxDeviceNameSize];
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@ -267,7 +283,7 @@ void LLWebRTCImpl::setRenderDevice(const std::string &id)
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if (id == guid || id == "Default")
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{
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RTC_LOG(LS_INFO) << __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
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mPlayoutDevice = i;
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tuningPlayoutDevice = i;
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break;
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}
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}
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@ -277,30 +293,40 @@ void LLWebRTCImpl::setRenderDevice(const std::string &id)
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{
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mTuningDeviceModule->StopPlayout();
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}
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bool was_peer_mute = false;
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if (mPeerDeviceModule)
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{
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mPeerDeviceModule->SpeakerMute(&was_peer_mute);
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if (!was_peer_mute)
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{
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mPeerDeviceModule->SetSpeakerMute(true);
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}
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mPeerDeviceModule->SetSpeakerMute(true);
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}
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mTuningDeviceModule->SetPlayoutDevice(mPlayoutDevice);
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mTuningDeviceModule->SetPlayoutDevice(tuningPlayoutDevice);
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mTuningDeviceModule->InitSpeaker();
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mTuningDeviceModule->InitPlayout();
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if (was_tuning_playing)
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{
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mTuningDeviceModule->StartPlayout();
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}
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renderDeviceCount = mPeerDeviceModule->PlayoutDevices();
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if (mPeerDeviceModule)
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{
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for (int16_t i = 0; i < renderDeviceCount; i++)
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{
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char name[webrtc::kAdmMaxDeviceNameSize];
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char guid[webrtc::kAdmMaxGuidSize];
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mPeerDeviceModule->PlayoutDeviceName(i, name, guid);
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if (id == guid || id == "Default")
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{
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RTC_LOG(LS_INFO)
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<< __FUNCTION__ << "Set recording device to " << name << " " << guid << " " << i;
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mPlayoutDevice = i;
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break;
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}
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}
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mPeerDeviceModule->SetPlayoutDevice(mPlayoutDevice);
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mPeerDeviceModule->InitSpeaker();
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mPeerDeviceModule->InitPlayout();
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mPeerDeviceModule->StartPlayout();
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mPeerDeviceModule->SetSpeakerMute(was_peer_mute);
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mPeerDeviceModule->SetSpeakerMute(false);
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}
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mTuningDeviceModule->SetSpeakerMute(false);
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@ -376,6 +402,8 @@ float LLWebRTCImpl::getPeerAudioLevel() { return 20 * mPeerAudioDeviceObserver->
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void LLWebRTCImpl::setSpeakerVolume(float volume) { mPeerDeviceModule->SetSpeakerVolume( (uint32_t)(volume * VOLUME_SCALE_WEBRTC));}
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void LLWebRTCImpl::setMicrophoneVolume(float volume) { mPeerDeviceModule->SetMicrophoneVolume((uint32_t)(volume * VOLUME_SCALE_WEBRTC));}
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void LLWebRTCImpl::setMute(bool mute) { mPeerDeviceModule->SetMicrophoneMute(mute); }
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//
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// Helpers
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//
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@ -566,9 +594,12 @@ void LLWebRTCPeerConnectionImpl::AnswerAvailable(const std::string &sdp)
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mWebRTCImpl->PostSignalingTask(
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[this, sdp]()
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{
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RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
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mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
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rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
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if (mPeerConnection)
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{
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RTC_LOG(LS_INFO) << __FUNCTION__ << " " << mPeerConnection->peer_connection_state();
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mPeerConnection->SetRemoteDescription(webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp),
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rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface>(this));
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}
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});
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}
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@ -844,17 +875,6 @@ void LLWebRTCPeerConnectionImpl::OnSetLocalDescriptionComplete(webrtc::RTCError
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}
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void LLWebRTCPeerConnectionImpl::setAudioObserver(LLWebRTCAudioObserver *observer) { mAudioObserverList.emplace_back(observer); }
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void LLWebRTCPeerConnectionImpl::unsetAudioObserver(LLWebRTCAudioObserver *observer)
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{
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std::vector<LLWebRTCAudioObserver *>::iterator it = std::find(mAudioObserverList.begin(), mAudioObserverList.end(), observer);
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if (it != mAudioObserverList.end())
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{
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mAudioObserverList.erase(it);
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}
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}
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//
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// DataChannelObserver implementation
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//
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@ -897,9 +917,12 @@ void LLWebRTCPeerConnectionImpl::OnMessage(const webrtc::DataBuffer& buffer)
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void LLWebRTCPeerConnectionImpl::sendData(const std::string& data, bool binary)
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{
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rtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
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webrtc::DataBuffer buffer(cowBuffer, binary);
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mDataChannel->Send(buffer);
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if (mDataChannel)
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{
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rtc::CopyOnWriteBuffer cowBuffer(data.data(), data.length());
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webrtc::DataBuffer buffer(cowBuffer, binary);
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mDataChannel->Send(buffer);
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}
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}
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void LLWebRTCPeerConnectionImpl::setDataObserver(LLWebRTCDataObserver* observer) { mDataObserverList.emplace_back(observer); }
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@ -88,19 +88,13 @@ class LLWebRTCDeviceInterface
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virtual void setSpeakerVolume(float volume) = 0; // volume between 0.0 and 1.0
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virtual void setMicrophoneVolume(float volume) = 0; // volume between 0.0 and 1.0
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virtual void setMute(bool mute) = 0;
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virtual float getPeerAudioLevel() = 0;
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};
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class LLWebRTCAudioObserver
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{
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public:
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};
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class LLWebRTCAudioInterface
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{
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public:
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virtual void setAudioObserver(LLWebRTCAudioObserver *observer) = 0;
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virtual void unsetAudioObserver(LLWebRTCAudioObserver *observer) = 0;
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virtual void setMute(bool mute) = 0;
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};
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@ -118,7 +118,8 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface
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float getPeerAudioLevel() override;
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void setSpeakerVolume(float volume) override; // range 0.0-1.0
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void setMicrophoneVolume(float volume) override; // range 0.0-1.0
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void setMicrophoneVolume(float volume) override; // range 0.0-1.0
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void setMute(bool mute) override;
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//
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// Helpers
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@ -227,8 +228,6 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnection,
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//
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// LLWebRTCAudioInterface
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//
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void setAudioObserver(LLWebRTCAudioObserver *observer) override;
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void unsetAudioObserver(LLWebRTCAudioObserver *observer) override;
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void setMute(bool mute) override;
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//
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@ -292,8 +291,6 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnection,
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bool mAnswerReceived;
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rtc::scoped_refptr<webrtc::PeerConnectionInterface> mPeerConnection;
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std::vector<LLWebRTCAudioObserver *> mAudioObserverList;
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std::vector<LLWebRTCDataObserver *> mDataObserverList;
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rtc::scoped_refptr<webrtc::DataChannelInterface> mDataChannel;
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File diff suppressed because it is too large
Load Diff
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@ -39,6 +39,7 @@ class LLWebRTCProtocolParser;
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#include "llcallingcard.h" // for LLFriendObserver
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#include "lleventcoro.h"
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#include "llcoros.h"
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#include "llparcel.h"
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#include <queue>
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#include "json/reader.h"
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@ -53,14 +54,13 @@ class LLWebRTCProtocolParser;
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#include <llwebrtc.h>
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class LLAvatarName;
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class LLVoiceWebRTCConnection;
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typedef boost::shared_ptr<LLVoiceWebRTCConnection> connectionPtr_t;
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class LLWebRTCVoiceClient : public LLSingleton<LLWebRTCVoiceClient>,
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virtual public LLVoiceModuleInterface,
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virtual public LLVoiceEffectInterface,
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public llwebrtc::LLWebRTCDevicesObserver,
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public llwebrtc::LLWebRTCSignalingObserver,
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public llwebrtc::LLWebRTCAudioObserver,
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public llwebrtc::LLWebRTCDataObserver
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public llwebrtc::LLWebRTCDevicesObserver
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{
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LLSINGLETON_C11(LLWebRTCVoiceClient);
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LOG_CLASS(LLWebRTCVoiceClient);
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@ -72,6 +72,8 @@ public:
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//@{
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void init(LLPumpIO *pump) override; // Call this once at application startup (creates connector)
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void terminate() override; // Call this to clean up during shutdown
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static bool isShuttingDown() { return sShuttingDown; }
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const LLVoiceVersionInfo& getVersion() override;
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@ -145,8 +147,12 @@ public:
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void setNonSpatialChannel(const std::string &uri,
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const std::string &credentials) override;
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bool setSpatialChannel(const std::string &uri,
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const std::string &credentials) override;
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bool setSpatialChannel(const std::string &uri, const std::string &credentials) override
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{
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return setSpatialChannel(uri, credentials, INVALID_PARCEL_ID);
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}
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bool setSpatialChannel(const std::string &uri, const std::string &credentials, S32 localParcelID);
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void leaveNonSpatialChannel() override;
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@ -163,9 +169,9 @@ public:
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//@{
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// start a voice channel with the specified user
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void callUser(const LLUUID &uuid) override;
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bool isValidChannel(std::string &channelHandle) override;
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bool answerInvite(std::string &channelHandle) override;
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void declineInvite(std::string &channelHandle) override;
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bool isValidChannel(std::string &channelID) override;
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bool answerInvite(std::string &channelID) override;
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void declineInvite(std::string &channelID) override;
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//@}
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/////////////////////////
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@ -232,10 +238,14 @@ public:
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bool isPreviewPlaying() override { return false; }
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//@}
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// authorize the user
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void userAuthorized(const std::string& user_id,
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const LLUUID &agentID) override;
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void userAuthorized(const std::string &user_id, const LLUUID &agentID) override {};
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void OnConnectionEstablished(const std::string& channelID);
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void OnConnectionFailure(const std::string &channelID);
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void sendPositionAndVolumeUpdate(bool force);
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void updateOwnVolume();
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//////////////////////////////
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/// @name Status notification
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@ -247,10 +257,6 @@ public:
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void addObserver(LLVoiceClientParticipantObserver* observer) override;
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void removeObserver(LLVoiceClientParticipantObserver* observer) override;
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//@}
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//@}
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//////////////////////////////
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/// @name Devices change notification
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@ -260,64 +266,17 @@ public:
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const llwebrtc::LLWebRTCVoiceDeviceList &capture_devices) override;
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//@}
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//////////////////////////////
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/// @name Signaling notification
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// LLWebRTCSignalingObserver
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//@{
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void OnIceGatheringState(IceGatheringState state) override;
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void OnIceCandidate(const llwebrtc::LLWebRTCIceCandidate &candidate) override;
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void OnOfferAvailable(const std::string &sdp) override;
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void OnRenegotiationNeeded() override;
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void OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface *audio_interface) override;
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//@}
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//////////////////////////////
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/// @name Signaling notification
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// LLWebRTCAudioObserver
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//@{
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//@}
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/////////////////////////
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/// @name Data Notification
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/// LLWebRTCDataObserver
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//@{
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void OnDataReceived(const std::string& data, bool binary) override;
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void OnDataChannelReady(llwebrtc::LLWebRTCDataInterface *data_interface) override;
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//@}
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void processIceUpdates();
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void onIceUpdateComplete(bool ice_completed, const LLSD& result);
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void onIceUpdateError(int retries, std::string url, LLSD body, bool ice_completed, const LLSD& result);
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//@}
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protected:
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//////////////////////
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// WebRTC Specific definitions
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enum streamState
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{
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streamStateUnknown = 0,
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streamStateIdle = 1,
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streamStateConnected = 2,
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streamStateRinging = 3,
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streamStateConnecting = 6, // same as WebRTC session_media_connecting enum
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streamStateDisconnecting = 7, //Same as WebRTC session_media_disconnecting enum
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};
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struct participantState
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{
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public:
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participantState(const LLUUID& agent_id);
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bool updateMuteState(); // true if mute state has changed
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bool updateMuteState(); // true if mute state has changed
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bool isAvatar();
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std::string mURI;
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LLUUID mAvatarID;
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std::string mAccountName;
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std::string mDisplayName;
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LLFrameTimer mSpeakingTimeout;
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F32 mLastSpokeTimestamp;
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@ -337,6 +296,12 @@ protected:
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typedef boost::shared_ptr<participantState> participantStatePtr_t;
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typedef boost::weak_ptr<participantState> participantStateWptr_t;
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participantStatePtr_t findParticipantByID(const std::string &channelID, const LLUUID &id);
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participantStatePtr_t addParticipantByID(const std::string& channelID, const LLUUID &id);
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void removeParticipantByID(const std::string& channelID, const LLUUID &id);
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protected:
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typedef std::map<const std::string, participantStatePtr_t> participantMap;
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typedef std::map<const LLUUID, participantStatePtr_t> participantUUIDMap;
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@ -348,7 +313,7 @@ protected:
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typedef boost::function<void(const ptr_t &)> sessionFunc_t;
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static ptr_t createSession();
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static ptr_t createSession(const std::string& channelID, S32 parcel_local_id);
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~sessionState();
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participantStatePtr_t addParticipant(const LLUUID& agent_id);
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@ -358,28 +323,37 @@ protected:
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participantStatePtr_t findParticipant(const std::string &uri);
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participantStatePtr_t findParticipantByID(const LLUUID& id);
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static ptr_t matchSessionByHandle(const std::string &handle);
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static ptr_t matchSessionByURI(const std::string &uri);
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static ptr_t matchSessionByParticipant(const LLUUID &participant_id);
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static ptr_t matchSessionByChannelID(const std::string& channel_id);
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void shutdownAllConnections();
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bool isCallBackPossible();
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bool isTextIMPossible();
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void processSessionStates();
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void OnConnectionEstablished(const std::string &channelID);
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void OnConnectionFailure(const std::string &channelID);
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void sendData(const std::string &data);
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static void for_each(sessionFunc_t func);
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static void reapEmptySessions();
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bool isEmpty() { return mWebRTCConnections.empty(); }
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std::string mHandle;
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std::string mGroupHandle;
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std::string mSIPURI;
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std::string mChannelID;
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std::string mAlias;
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std::string mName;
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std::string mAlternateSIPURI;
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std::string mErrorStatusString;
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std::queue<std::string> mTextMsgQueue;
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LLUUID mIMSessionID;
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LLUUID mCallerID;
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int mErrorStatusCode;
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int mMediaStreamState;
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bool mIsChannel; // True for both group and spatial channels (false for p2p, PSTN)
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bool mIsSpatial; // True for spatial channels
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bool mIsP2P;
|
||||
|
|
@ -399,18 +373,20 @@ protected:
|
|||
LLUUID mVoiceFontID;
|
||||
|
||||
static void VerifySessions();
|
||||
static void deleteAllSessions();
|
||||
|
||||
private:
|
||||
|
||||
std::map<std::string, connectionPtr_t> mWebRTCConnections;
|
||||
std::string mPrimaryConnectionID;
|
||||
|
||||
sessionState();
|
||||
|
||||
static std::set<wptr_t> mSession; // canonical list of outstanding sessions.
|
||||
std::set<wptr_t>::iterator mMyIterator; // used for delete
|
||||
static std::map<std::string, ptr_t> mSessions; // canonical list of outstanding sessions.
|
||||
|
||||
static void for_eachPredicate(const wptr_t &a, sessionFunc_t func);
|
||||
static void for_eachPredicate(const std::pair<std::string, LLWebRTCVoiceClient::sessionState::wptr_t> &a, sessionFunc_t func);
|
||||
|
||||
static bool testByHandle(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string handle);
|
||||
static bool testByCreatingURI(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string uri);
|
||||
static bool testBySIPOrAlterateURI(const LLWebRTCVoiceClient::sessionState::wptr_t &a, std::string uri);
|
||||
static bool testByCallerId(const LLWebRTCVoiceClient::sessionState::wptr_t &a, LLUUID participantId);
|
||||
|
||||
};
|
||||
|
|
@ -422,28 +398,20 @@ protected:
|
|||
// Private Member Functions
|
||||
//////////////////////////////////////////////////////
|
||||
|
||||
|
||||
|
||||
static void predProcessSessionStates(const LLWebRTCVoiceClient::sessionStatePtr_t &session);
|
||||
static void predOnConnectionEstablished(const LLWebRTCVoiceClient::sessionStatePtr_t &session, std::string channelID);
|
||||
static void predOnConnectionFailure(const LLWebRTCVoiceClient::sessionStatePtr_t &session, std::string channelID);
|
||||
static void predSendData(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const std::string& spatial_data, const std::string& volume_data);
|
||||
static void predUpdateOwnVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 audio_level);
|
||||
static void predSetMuteMic(const LLWebRTCVoiceClient::sessionStatePtr_t &session, bool mute);
|
||||
//////////////////////////////
|
||||
/// @name TVC/Server management and communication
|
||||
//@{
|
||||
// Call this if the connection to the daemon terminates unexpectedly. It will attempt to reset everything and relaunch.
|
||||
void daemonDied();
|
||||
|
||||
// Call this if we're just giving up on voice (can't provision an account, etc.). It will clean up and go away.
|
||||
void giveUp();
|
||||
|
||||
|
||||
void connectorCreate();
|
||||
void connectorShutdown();
|
||||
void closeSocket(void);
|
||||
|
||||
// void requestVoiceAccountProvision(S32 retries = 3);
|
||||
void setLoginInfo(
|
||||
const std::string& account_name,
|
||||
const std::string& password,
|
||||
const std::string& channel_sdp);
|
||||
void logout();
|
||||
|
||||
|
||||
//@}
|
||||
|
|
@ -486,30 +454,12 @@ protected:
|
|||
/////////////////////////////
|
||||
BOOL getAreaVoiceDisabled(); // returns true if the area the avatar is in is speech-disabled.
|
||||
// Use this to determine whether to show a "no speech" icon in the menu bar.
|
||||
|
||||
participantStatePtr_t findParticipantByID(const LLUUID& id);
|
||||
participantStatePtr_t addParticipantByID(const LLUUID &id);
|
||||
void removeParticipantByID(const LLUUID &id);
|
||||
|
||||
#if 0
|
||||
////////////////////////////////////////
|
||||
// voice sessions.
|
||||
typedef std::set<sessionStatePtr_t> sessionSet;
|
||||
|
||||
typedef sessionSet::iterator sessionIterator;
|
||||
sessionIterator sessionsBegin(void);
|
||||
sessionIterator sessionsEnd(void);
|
||||
#endif
|
||||
void sessionEstablished();
|
||||
sessionStatePtr_t findSession(const std::string &handle);
|
||||
sessionStatePtr_t findSession(const LLUUID &participant_id);
|
||||
void sessionEstablished(const LLUUID& region_id);
|
||||
sessionStatePtr_t findP2PSession(const LLUUID &agent_id);
|
||||
|
||||
sessionStatePtr_t addSession(const std::string &uri, const std::string &handle = std::string());
|
||||
void clearSessionHandle(const sessionStatePtr_t &session);
|
||||
void setSessionHandle(const sessionStatePtr_t &session, const std::string &handle);
|
||||
void setSessionURI(const sessionStatePtr_t &session, const std::string &uri);
|
||||
sessionStatePtr_t addSession(const std::string& channel_id, S32 parcel_local_id);
|
||||
void deleteSession(const sessionStatePtr_t &session);
|
||||
void deleteAllSessions(void);
|
||||
|
||||
void verifySessionState(void);
|
||||
|
||||
|
|
@ -518,11 +468,7 @@ protected:
|
|||
|
||||
// This is called in several places where the session _may_ need to be deleted.
|
||||
// It contains logic for whether to delete the session or keep it around.
|
||||
void reapSession(const sessionStatePtr_t &session);
|
||||
|
||||
// Returns true if the session seems to indicate we've moved to a region on a different voice server
|
||||
bool sessionNeedsRelog(const sessionStatePtr_t &session);
|
||||
|
||||
void reapSession(const sessionStatePtr_t &session);
|
||||
|
||||
//////////////////////////////////////
|
||||
// buddy list stuff, needed for SLIM later
|
||||
|
|
@ -583,41 +529,9 @@ private:
|
|||
|
||||
// Coroutine support methods
|
||||
//---
|
||||
void voiceControlCoro();
|
||||
void voiceControlStateMachine();
|
||||
void voiceConnectionCoro();
|
||||
|
||||
int mVoiceControlState;
|
||||
LLMutex mVoiceStateMutex;
|
||||
void setVoiceControlStateUnless(int new_voice_control_state, int unless=-1)
|
||||
{
|
||||
LLMutexLock lock(&mVoiceStateMutex);
|
||||
if (mVoiceControlState != unless)
|
||||
{
|
||||
mVoiceControlState = new_voice_control_state;
|
||||
}
|
||||
}
|
||||
int getVoiceControlState()
|
||||
{
|
||||
LLMutexLock lock(&mVoiceStateMutex);
|
||||
return mVoiceControlState;
|
||||
}
|
||||
|
||||
bool callbackEndDaemon(const LLSD& data);
|
||||
bool provisionVoiceAccount();
|
||||
void OnVoiceAccountProvisioned(const LLSD& body);
|
||||
void OnVoiceAccountProvisionFailure(std::string url, int retries, LLSD body, const LLSD& result);
|
||||
bool establishVoiceConnection();
|
||||
bool breakVoiceConnection(bool wait);
|
||||
bool loginToWebRTC();
|
||||
void logoutOfWebRTC(bool wait);
|
||||
|
||||
bool requestParcelVoiceInfo();
|
||||
|
||||
bool addAndJoinSession(const sessionStatePtr_t &nextSession);
|
||||
bool terminateAudioSession(bool wait);
|
||||
|
||||
bool waitForChannel();
|
||||
bool runSession(const sessionStatePtr_t &session);
|
||||
void voiceConnectionStateMachine();
|
||||
|
||||
bool performMicTuning();
|
||||
//---
|
||||
|
|
@ -632,72 +546,38 @@ private:
|
|||
int mSpatialJoiningNum;
|
||||
|
||||
static void idle(void *user_data);
|
||||
|
||||
LLHost mDaemonHost;
|
||||
LLSocket::ptr_t mSocket;
|
||||
|
||||
// We should kill the voice daemon in case of connection alert
|
||||
bool mTerminateDaemon;
|
||||
|
||||
std::string mAccountName;
|
||||
std::string mAccountPassword;
|
||||
std::string mChannelSDP;
|
||||
std::string mRemoteChannelSDP;
|
||||
std::string mAccountDisplayName;
|
||||
|
||||
|
||||
bool mTuningMode;
|
||||
float mTuningEnergy;
|
||||
std::string mTuningAudioFile;
|
||||
int mTuningMicVolume;
|
||||
bool mTuningMicVolumeDirty;
|
||||
int mTuningSpeakerVolume;
|
||||
bool mTuningSpeakerVolumeDirty;
|
||||
bool mDevicesListUpdated; // set to true when the device list has been updated
|
||||
// and false when the panelvoicedevicesettings has queried for an update status.
|
||||
|
||||
std::string mSpatialSessionURI;
|
||||
std::string mSpatialSessionCredentials;
|
||||
|
||||
std::string mMainSessionGroupHandle; // handle of the "main" session group.
|
||||
|
||||
std::string mChannelName; // Name of the channel to be looked up
|
||||
bool mAreaVoiceDisabled;
|
||||
sessionStatePtr_t mAudioSession; // Session state for the current audio session
|
||||
sessionStatePtr_t mAudioSession; // Session state for the current audio session
|
||||
bool mAudioSessionChanged; // set to true when the above pointer gets changed, so observers can be notified.
|
||||
|
||||
sessionStatePtr_t mNextAudioSession; // Session state for the audio session we're trying to join
|
||||
|
||||
S32 mCurrentParcelLocalID; // Used to detect parcel boundary crossings
|
||||
std::string mCurrentRegionName; // Used to detect parcel boundary crossings
|
||||
|
||||
bool mConnectorEstablished; // set by "Create Connector" response
|
||||
bool mAccountLoggedIn; // set by login message
|
||||
int mNumberOfAliases;
|
||||
U32 mCommandCookie;
|
||||
|
||||
int mLoginRetryCount;
|
||||
|
||||
sessionMap mSessionsByHandle; // Active sessions, indexed by session handle. Sessions which are being initiated may not be in this map.
|
||||
#if 0
|
||||
sessionSet mSessions; // All sessions, not indexed. This is the canonical session list.
|
||||
#endif
|
||||
|
||||
|
||||
bool mBuddyListMapPopulated;
|
||||
bool mBlockRulesListReceived;
|
||||
bool mAutoAcceptRulesListReceived;
|
||||
buddyListMap mBuddyListMap;
|
||||
|
||||
llwebrtc::LLWebRTCDeviceInterface *mWebRTCDeviceInterface;
|
||||
llwebrtc::LLWebRTCPeerConnection *mWebRTCPeerConnection;
|
||||
llwebrtc::LLWebRTCAudioInterface *mWebRTCAudioInterface;
|
||||
llwebrtc::LLWebRTCDataInterface *mWebRTCDataInterface;
|
||||
|
||||
LLVoiceDeviceList mCaptureDevices;
|
||||
LLVoiceDeviceList mRenderDevices;
|
||||
std::vector<llwebrtc::LLWebRTCIceCandidate> mIceCandidates;
|
||||
bool mIceCompleted;
|
||||
bool mTrickling;
|
||||
|
||||
uint32_t mAudioLevel;
|
||||
|
||||
|
|
@ -705,7 +585,12 @@ private:
|
|||
bool mShutdownComplete;
|
||||
|
||||
bool checkParcelChanged(bool update = false);
|
||||
bool switchChannel(std::string uri = std::string(), bool spatial = true, bool no_reconnect = false, bool is_p2p = false, std::string hash = "");
|
||||
bool switchChannel(const std::string channelID,
|
||||
bool spatial = true,
|
||||
bool no_reconnect = false,
|
||||
bool is_p2p = false,
|
||||
std::string hash = "",
|
||||
S32 parcel_local_id = INVALID_PARCEL_ID);
|
||||
void joinSession(const sessionStatePtr_t &session);
|
||||
|
||||
std::string nameFromAvatar(LLVOAvatar *avatar);
|
||||
|
|
@ -716,11 +601,8 @@ private:
|
|||
|
||||
bool inSpatialChannel(void);
|
||||
std::string getAudioSessionURI();
|
||||
std::string getAudioSessionHandle();
|
||||
|
||||
void setHidden(bool hidden) override; //virtual
|
||||
Json::Value getPositionAndVolumeUpdateJson(bool force);
|
||||
void sendPositionAndVolumeUpdate();
|
||||
|
||||
void enforceTether(void);
|
||||
|
||||
|
|
@ -759,9 +641,6 @@ private:
|
|||
bool mMicVolumeDirty;
|
||||
|
||||
bool mVoiceEnabled;
|
||||
bool mWriteInProgress;
|
||||
std::string mWriteString;
|
||||
size_t mWriteOffset;
|
||||
|
||||
BOOL mLipSyncEnabled;
|
||||
|
||||
|
|
@ -781,15 +660,13 @@ private:
|
|||
|
||||
S32 mPlayRequestCount;
|
||||
bool mIsInTuningMode;
|
||||
bool mIsInChannel;
|
||||
bool mIsJoiningSession;
|
||||
bool mIsWaitingForFonts;
|
||||
bool mIsLoggingIn;
|
||||
bool mIsLoggedIn;
|
||||
bool mIsProcessingChannels;
|
||||
bool mIsCoroutineActive;
|
||||
|
||||
// This variables can last longer than WebRTC in coroutines so we need them as static
|
||||
// These variables can last longer than WebRTC in coroutines so we need them as static
|
||||
static bool sShuttingDown;
|
||||
static bool sConnected;
|
||||
static LLPumpIO* sPump;
|
||||
|
|
@ -797,19 +674,6 @@ private:
|
|||
LLEventMailDrop mWebRTCPump;
|
||||
};
|
||||
|
||||
class LLWebRTCSecurity : public LLSingleton<LLWebRTCSecurity>
|
||||
{
|
||||
LLSINGLETON(LLWebRTCSecurity);
|
||||
virtual ~LLWebRTCSecurity();
|
||||
|
||||
public:
|
||||
std::string connectorHandle() { return mConnectorHandle; };
|
||||
std::string accountHandle() { return mAccountHandle; };
|
||||
|
||||
private:
|
||||
std::string mConnectorHandle;
|
||||
std::string mAccountHandle;
|
||||
};
|
||||
|
||||
class LLVoiceWebRTCStats : public LLSingleton<LLVoiceWebRTCStats>
|
||||
{
|
||||
|
|
@ -843,5 +707,119 @@ class LLVoiceWebRTCStats : public LLSingleton<LLVoiceWebRTCStats>
|
|||
LLSD read();
|
||||
};
|
||||
|
||||
class LLVoiceWebRTCConnection :
|
||||
public llwebrtc::LLWebRTCSignalingObserver,
|
||||
public llwebrtc::LLWebRTCDataObserver
|
||||
{
|
||||
public:
|
||||
LLVoiceWebRTCConnection(const LLUUID& regionID, S32 parcelLocalID, const std::string& channelID);
|
||||
|
||||
virtual ~LLVoiceWebRTCConnection();
|
||||
|
||||
//////////////////////////////
|
||||
/// @name Signaling notification
|
||||
// LLWebRTCSignalingObserver
|
||||
//@{
|
||||
void OnIceGatheringState(IceGatheringState state) override;
|
||||
void OnIceCandidate(const llwebrtc::LLWebRTCIceCandidate &candidate) override;
|
||||
void OnOfferAvailable(const std::string &sdp) override;
|
||||
void OnRenegotiationNeeded() override;
|
||||
void OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface *audio_interface) override;
|
||||
//@}
|
||||
|
||||
/////////////////////////
|
||||
/// @name Data Notification
|
||||
/// LLWebRTCDataObserver
|
||||
//@{
|
||||
void OnDataReceived(const std::string &data, bool binary) override;
|
||||
void OnDataChannelReady(llwebrtc::LLWebRTCDataInterface *data_interface) override;
|
||||
//@}
|
||||
|
||||
void processIceUpdates();
|
||||
void onIceUpdateComplete(bool ice_completed, const LLSD &result);
|
||||
void onIceUpdateError(int retries, std::string url, LLSD body, bool ice_completed, const LLSD &result);
|
||||
|
||||
bool requestVoiceConnection();
|
||||
void OnVoiceConnectionRequestSuccess(const LLSD &body);
|
||||
void OnVoiceConnectionRequestFailure(std::string url, int retries, LLSD body, const LLSD &result);
|
||||
|
||||
bool connectionStateMachine();
|
||||
|
||||
void sendData(const std::string &data);
|
||||
|
||||
void shutDown()
|
||||
{
|
||||
LLMutexLock lock(&mVoiceStateMutex);
|
||||
mShutDown = true;
|
||||
}
|
||||
|
||||
protected:
|
||||
typedef enum e_voice_connection_state
|
||||
{
|
||||
VOICE_STATE_ERROR = 0x0,
|
||||
VOICE_STATE_START_SESSION = 0x1,
|
||||
VOICE_STATE_WAIT_FOR_SESSION_START = 0x2,
|
||||
VOICE_STATE_REQUEST_CONNECTION = 0x4,
|
||||
VOICE_STATE_CONNECTION_WAIT = 0x8,
|
||||
VOICE_STATE_SESSION_ESTABLISHED = 0x10,
|
||||
VOICE_STATE_SESSION_UP = 0x20,
|
||||
VOICE_STATE_SESSION_RETRY = 0x40,
|
||||
VOICE_STATE_DISCONNECT = 0x80,
|
||||
VOICE_STATE_WAIT_FOR_EXIT = 0x100,
|
||||
VOICE_STATE_SESSION_EXIT = 0x200,
|
||||
VOICE_STATE_SESSION_STOPPING = 0x3c0,
|
||||
VOICE_STATE_SESSION_WAITING = 0x10a
|
||||
} EVoiceConnectionState;
|
||||
|
||||
EVoiceConnectionState mVoiceConnectionState;
|
||||
LLMutex mVoiceStateMutex;
|
||||
void setVoiceConnectionState(EVoiceConnectionState new_voice_connection_state)
|
||||
{
|
||||
LLMutexLock lock(&mVoiceStateMutex);
|
||||
|
||||
if (new_voice_connection_state & VOICE_STATE_SESSION_STOPPING)
|
||||
{
|
||||
// the new state is shutdown or restart.
|
||||
mVoiceConnectionState = new_voice_connection_state;
|
||||
return;
|
||||
}
|
||||
if (mVoiceConnectionState & VOICE_STATE_SESSION_STOPPING)
|
||||
{
|
||||
// we're currently shutting down or restarting, so ignore any
|
||||
// state changes.
|
||||
return;
|
||||
}
|
||||
|
||||
mVoiceConnectionState = new_voice_connection_state;
|
||||
}
|
||||
EVoiceConnectionState getVoiceConnectionState()
|
||||
{
|
||||
LLMutexLock lock(&mVoiceStateMutex);
|
||||
return mVoiceConnectionState;
|
||||
}
|
||||
|
||||
bool breakVoiceConnection(bool wait);
|
||||
void OnVoiceDisconnectionRequestSuccess(const LLSD &body);
|
||||
void OnVoiceDisconnectionRequestFailure(std::string url, int retries, LLSD body, const LLSD &result);
|
||||
|
||||
std::string mChannelSDP;
|
||||
std::string mRemoteChannelSDP;
|
||||
|
||||
std::string mChannelID;
|
||||
LLUUID mRegionID;
|
||||
S32 mParcelLocalID;
|
||||
|
||||
bool mShutDown;
|
||||
|
||||
std::vector<llwebrtc::LLWebRTCIceCandidate> mIceCandidates;
|
||||
bool mIceCompleted;
|
||||
bool mTrickling;
|
||||
|
||||
llwebrtc::LLWebRTCPeerConnection *mWebRTCPeerConnection;
|
||||
llwebrtc::LLWebRTCAudioInterface *mWebRTCAudioInterface;
|
||||
llwebrtc::LLWebRTCDataInterface *mWebRTCDataInterface;
|
||||
};
|
||||
|
||||
|
||||
#endif //LL_WebRTC_VOICE_CLIENT_H
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue