Hook up speaker volume.

5 files changed, 49 insertions, 886 deletions

diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index c2631b2ea3..93e9db9c1d 100644
--- a/indra/llwebrtc/llwebrtc.cpp
+++ b/indra/llwebrtc/llwebrtc.cpp
@@ -341,6 +341,28 @@ void LLWebRTCImpl::setMute(bool mute)
         });
 }
 
+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(10.0 * volume);
+
+                }
+            }
+        });
+}
+
 //
 // PeerConnectionObserver implementation.
 //
diff --git a/indra/llwebrtc/llwebrtc.h b/indra/llwebrtc/llwebrtc.h
index acc3665e95..ca558add01 100644
--- a/indra/llwebrtc/llwebrtc.h
+++ b/indra/llwebrtc/llwebrtc.h
@@ -93,6 +93,7 @@ class LLWebRTCAudioInterface
 {
   public:
     virtual void setMute(bool mute) = 0;
+    virtual void setSpeakerVolume(float volume) = 0;  // volume between 0.0 and 1.0
 };
 
 class LLWebRTCSignalingObserver
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
index d439bd253d..5c6cfcdbc6 100644
--- a/indra/llwebrtc/llwebrtc_impl.h
+++ b/indra/llwebrtc/llwebrtc_impl.h
@@ -125,6 +125,7 @@ class LLWebRTCImpl : public LLWebRTCDeviceInterface,
     // LLWebRTCAudioInterface
     //
     void setMute(bool mute) override;
+    void setSpeakerVolume(float folume) override; // range 0.0-1.0
 
     //
     // PeerConnectionObserver implementation.
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp
index 516bab1914..23033c5fee 100644
--- a/indra/newview/llvoicewebrtc.cpp
+++ b/indra/newview/llvoicewebrtc.cpp
@@ -123,14 +123,6 @@ static int scale_mic_volume(float volume)
 	return 30 + (int)(volume * 20.0f);
 }
 
-static int scale_speaker_volume(float volume)
-{
-	// incoming volume has the range [0.0 ... 1.0], with 0.5 as the default.                                                
-	// Map it to WebRTC levels as follows: 0.0 -> 30, 0.5 -> 50, 1.0 -> 70                                                   
-	return 30 + (int)(volume * 40.0f);
-	
-}
-
 
 ///////////////////////////////////////////////////////////////////////////////////////////////
 
@@ -317,7 +309,7 @@ LLWebRTCVoiceClient::LLWebRTCVoiceClient() :
     sConnected = false;
     sPump = nullptr;
 
-	mSpeakerVolume = scale_speaker_volume(0);
+	mSpeakerVolume = 0.0;
 
 	mVoiceVersion.serverVersion = "";
 	mVoiceVersion.serverType = VOICE_SERVER_TYPE;
@@ -2220,12 +2212,11 @@ void LLWebRTCVoiceClient::tuningSetMicVolume(float volume)
 }
 
 void LLWebRTCVoiceClient::tuningSetSpeakerVolume(float volume)
-{
-	int scaled_volume = scale_speaker_volume(volume);	
+{	
 
-	if(scaled_volume != mTuningSpeakerVolume)
+	if (volume != mTuningSpeakerVolume)
 	{
-		mTuningSpeakerVolume = scaled_volume;
+        mTuningSpeakerVolume      = volume;
 		mTuningSpeakerVolumeDirty = true;
 	}
 }
@@ -2663,7 +2654,7 @@ void LLWebRTCVoiceClient::sendLocalAudioUpdates()
 
 	if (mSpeakerMuteDirty && !mTuningMode)
 	{
-		const char *muteval = ((mSpeakerVolume <= scale_speaker_volume(0)) ? "true" : "false");
+		const char *muteval = ((mSpeakerVolume <= 0.0) ? "true" : "false");
 
 		mSpeakerMuteDirty = false;
 
@@ -2899,6 +2890,11 @@ void LLWebRTCVoiceClient::OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface *
     LL_INFOS("Voice") << "On AudioEstablished." << LL_ENDL;
     mWebRTCAudioInterface = audio_interface;
     audio_interface->setMute(true);
+    {
+        LLMutexLock lock(&mVoiceStateMutex);
+
+        audio_interface->setSpeakerVolume(mSpeakerVolume);
+    }
 }
 
 void LLWebRTCVoiceClient::OnRenegotiationNeeded()
@@ -4901,18 +4897,23 @@ void LLWebRTCVoiceClient::setEarLocation(S32 loc)
 
 void LLWebRTCVoiceClient::setVoiceVolume(F32 volume)
 {
-	int scaled_volume = scale_speaker_volume(volume);	
-
-	if(scaled_volume != mSpeakerVolume)
+	if (volume != mSpeakerVolume)
 	{
-	  int min_volume = scale_speaker_volume(0);
-		if((scaled_volume == min_volume) || (mSpeakerVolume == min_volume))
-		{
-			mSpeakerMuteDirty = true;
-		}
+        {
+            LLMutexLock lock(&mVoiceStateMutex);
+            int         min_volume = 0.0;
+            if ((volume == min_volume) || (mSpeakerVolume == min_volume))
+            {
+                mSpeakerMuteDirty = true;
+            }
 
-		mSpeakerVolume = scaled_volume;
-		mSpeakerVolumeDirty = true;
+            mSpeakerVolume      = volume;
+            mSpeakerVolumeDirty = true;
+        }
+        if (mWebRTCAudioInterface)
+        {
+            mWebRTCAudioInterface->setSpeakerVolume(volume);
+        }
 	}
 }
 
@@ -6546,755 +6547,7 @@ void LLWebRTCVoiceClient::captureBufferPlayStopSendMessage()
 	}
 }
 
-LLWebRTCProtocolParser::LLWebRTCProtocolParser()
-{
-	parser = XML_ParserCreate(NULL);
-	
-	reset();
-}
-
-void LLWebRTCProtocolParser::reset()
-{
-	responseDepth = 0;
-	ignoringTags = false;
-	accumulateText = false;
-	energy = 0.f;
-	hasText = false;
-	hasAudio = false;
-	hasVideo = false;
-	terminated = false;
-	ignoreDepth = 0;
-	isChannel = false;
-	incoming = false;
-	enabled = false;
-	isEvent = false;
-	isLocallyMuted = false;
-	isModeratorMuted = false;
-	isSpeaking = false;
-	participantType = 0;
-	returnCode = -1;
-	state = 0;
-	statusCode = 0;
-	volume = 0;
-	textBuffer.clear();
-	alias.clear();
-	numberOfAliases = 0;
-	applicationString.clear();
-}
-
-//virtual 
-LLWebRTCProtocolParser::~LLWebRTCProtocolParser()
-{
-	if (parser)
-		XML_ParserFree(parser);
-}
-
-static LLTrace::BlockTimerStatHandle FTM_WebRTC_PROCESS("WebRTC Process");
-
-// virtual
-LLIOPipe::EStatus LLWebRTCProtocolParser::process_impl(
-													  const LLChannelDescriptors& channels,
-													  buffer_ptr_t& buffer,
-													  bool& eos,
-													  LLSD& context,
-													  LLPumpIO* pump)
-{
-	LL_RECORD_BLOCK_TIME(FTM_WebRTC_PROCESS);
-	LLBufferStream istr(channels, buffer.get());
-	std::ostringstream ostr;
-	while (istr.good())
-	{
-		char buf[1024];
-		istr.read(buf, sizeof(buf));
-		mInput.append(buf, istr.gcount());
-	}
-	
-	// Look for input delimiter(s) in the input buffer.  If one is found, send the message to the xml parser.
-	int start = 0;
-	int delim;
-	while((delim = mInput.find("\n\n\n", start)) != std::string::npos)
-	{	
-		
-		// Reset internal state of the LLWebRTCProtocolParser (no effect on the expat parser)
-		reset();
-		
-		XML_ParserReset(parser, NULL);
-		XML_SetElementHandler(parser, ExpatStartTag, ExpatEndTag);
-		XML_SetCharacterDataHandler(parser, ExpatCharHandler);
-		XML_SetUserData(parser, this);	
-		XML_Parse(parser, mInput.data() + start, delim - start, false);
-		
-        LL_DEBUGS("WebRTCProtocolParser") << "parsing: " << mInput.substr(start, delim - start) << LL_ENDL;
-		start = delim + 3;
-	}
-	
-	if(start != 0)
-		mInput = mInput.substr(start);
-	
-	LL_DEBUGS("WebRTCProtocolParser") << "at end, mInput is: " << mInput << LL_ENDL;
-	
-	if(!LLWebRTCVoiceClient::sConnected)
-	{
-		// If voice has been disabled, we just want to close the socket.  This does so.
-		LL_INFOS("Voice") << "returning STATUS_STOP" << LL_ENDL;
-		return STATUS_STOP;
-	}
-	
-	return STATUS_OK;
-}
-
-void XMLCALL LLWebRTCProtocolParser::ExpatStartTag(void *data, const char *el, const char **attr)
-{
-	if (data)
-	{
-		LLWebRTCProtocolParser	*object = (LLWebRTCProtocolParser*)data;
-		object->StartTag(el, attr);
-	}
-}
-
-// --------------------------------------------------------------------------------
-
-void XMLCALL LLWebRTCProtocolParser::ExpatEndTag(void *data, const char *el)
-{
-	if (data)
-	{
-		LLWebRTCProtocolParser	*object = (LLWebRTCProtocolParser*)data;
-		object->EndTag(el);
-	}
-}
-
-// --------------------------------------------------------------------------------
-
-void XMLCALL LLWebRTCProtocolParser::ExpatCharHandler(void *data, const XML_Char *s, int len)
-{
-	if (data)
-	{
-		LLWebRTCProtocolParser	*object = (LLWebRTCProtocolParser*)data;
-		object->CharData(s, len);
-	}
-}
-
-// --------------------------------------------------------------------------------
-
-
-void LLWebRTCProtocolParser::StartTag(const char *tag, const char **attr)
-{
-	// Reset the text accumulator. We shouldn't have strings that are inturrupted by new tags
-	textBuffer.clear();
-	// only accumulate text if we're not ignoring tags.
-	accumulateText = !ignoringTags;
-	
-	if (responseDepth == 0)
-	{	
-		isEvent = !stricmp("Event", tag);
-		
-		if (!stricmp("Response", tag) || isEvent)
-		{
-			// Grab the attributes
-			while (*attr)
-			{
-				const char	*key = *attr++;
-				const char	*value = *attr++;
-				
-				if (!stricmp("requestId", key))
-				{
-					requestId = value;
-				}
-				else if (!stricmp("action", key))
-				{
-					actionString = value;
-				}
-				else if (!stricmp("type", key))
-				{
-					eventTypeString = value;
-				}
-			}
-		}
-		LL_DEBUGS("WebRTCProtocolParser") << tag << " (" << responseDepth << ")"  << LL_ENDL;
-	}
-	else
-	{
-		if (ignoringTags)
-		{
-			LL_DEBUGS("WebRTCProtocolParser") << "ignoring tag " << tag << " (depth = " << responseDepth << ")" << LL_ENDL;
-		}
-		else
-		{
-			LL_DEBUGS("WebRTCProtocolParser") << tag << " (" << responseDepth << ")"  << LL_ENDL;
-			
-			// Ignore the InputXml stuff so we don't get confused
-			if (!stricmp("InputXml", tag))
-			{
-				ignoringTags = true;
-				ignoreDepth = responseDepth;
-				accumulateText = false;
-				
-				LL_DEBUGS("WebRTCProtocolParser") << "starting ignore, ignoreDepth is " << ignoreDepth << LL_ENDL;
-			}
-			else if (!stricmp("CaptureDevices", tag))
-			{
-				LLWebRTCVoiceClient::getInstance()->clearCaptureDevices();
-			}			
-			else if (!stricmp("RenderDevices", tag))
-			{
-				LLWebRTCVoiceClient::getInstance()->clearRenderDevices();
-			}
-			else if (!stricmp("CaptureDevice", tag))
-			{
-				deviceString.clear();
-			}
-			else if (!stricmp("RenderDevice", tag))
-			{
-				deviceString.clear();
-			}			
-			else if (!stricmp("SessionFont", tag))
-			{
-				id = 0;
-				nameString.clear();
-				descriptionString.clear();
-				expirationDate = LLDate();
-				hasExpired = false;
-				fontType = 0;
-				fontStatus = 0;
-			}
-			else if (!stricmp("TemplateFont", tag))
-			{
-				id = 0;
-				nameString.clear();
-				descriptionString.clear();
-				expirationDate = LLDate();
-				hasExpired = false;
-				fontType = 0;
-				fontStatus = 0;
-			}
-			else if (!stricmp("MediaCompletionType", tag))
-			{
-				mediaCompletionType.clear();
-			}
-		}
-	}
-	responseDepth++;
-}
-
-// --------------------------------------------------------------------------------
-
-void LLWebRTCProtocolParser::EndTag(const char *tag)
-{
-	const std::string& string = textBuffer;
-	
-	responseDepth--;
-	
-	if (ignoringTags)
-	{
-		if (ignoreDepth == responseDepth)
-		{
-			LL_DEBUGS("WebRTCProtocolParser") << "end of ignore" << LL_ENDL;
-			ignoringTags = false;
-		}
-		else
-		{
-			LL_DEBUGS("WebRTCProtocolParser") << "ignoring tag " << tag << " (depth = " << responseDepth << ")" << LL_ENDL;
-		}
-	}
-	
-	if (!ignoringTags)
-	{
-		LL_DEBUGS("WebRTCProtocolParser") << "processing tag " << tag << " (depth = " << responseDepth << ")" << LL_ENDL;
-		
-		// Closing a tag. Finalize the text we've accumulated and reset
-		if (!stricmp("ReturnCode", tag))
-			returnCode = strtol(string.c_str(), NULL, 10);
-		else if (!stricmp("SessionHandle", tag))
-			sessionHandle = string;
-		else if (!stricmp("SessionGroupHandle", tag))
-			sessionGroupHandle = string;
-		else if (!stricmp("StatusCode", tag))
-			statusCode = strtol(string.c_str(), NULL, 10);
-		else if (!stricmp("StatusString", tag))
-			statusString = string;
-		else if (!stricmp("ParticipantURI", tag))
-			uriString = string;
-		else if (!stricmp("Volume", tag))
-			volume = strtol(string.c_str(), NULL, 10);
-		else if (!stricmp("Energy", tag))
-			energy = (F32)strtod(string.c_str(), NULL);
-		else if (!stricmp("IsModeratorMuted", tag))
-			isModeratorMuted = !stricmp(string.c_str(), "true");
-		else if (!stricmp("IsSpeaking", tag))
-			isSpeaking = !stricmp(string.c_str(), "true");
-		else if (!stricmp("Alias", tag))
-			alias = string;
-		else if (!stricmp("NumberOfAliases", tag))
-			numberOfAliases = strtol(string.c_str(), NULL, 10);
-		else if (!stricmp("Application", tag))
-			applicationString = string;
-		else if (!stricmp("ConnectorHandle", tag))
-			connectorHandle = string;
-		else if (!stricmp("VersionID", tag))
-			versionID = string;
-		else if (!stricmp("Version", tag))
-			mBuildID = string;
-		else if (!stricmp("AccountHandle", tag))
-			accountHandle = string;
-		else if (!stricmp("State", tag))
-			state = strtol(string.c_str(), NULL, 10);
-		else if (!stricmp("URI", tag))
-			uriString = string;
-		else if (!stricmp("IsChannel", tag))
-			isChannel = !stricmp(string.c_str(), "true");
-		else if (!stricmp("Incoming", tag))
-			incoming = !stricmp(string.c_str(), "true");
-		else if (!stricmp("Enabled", tag))
-			enabled = !stricmp(string.c_str(), "true");
-		else if (!stricmp("Name", tag))
-			nameString = string;
-		else if (!stricmp("AudioMedia", tag))
-			audioMediaString = string;
-		else if (!stricmp("ChannelName", tag))
-			nameString = string;
-		else if (!stricmp("DisplayName", tag))
-			displayNameString = string;
-		else if (!stricmp("Device", tag))
-			deviceString = string;		
-		else if (!stricmp("AccountName", tag))
-			nameString = string;
-		else if (!stricmp("ParticipantType", tag))
-			participantType = strtol(string.c_str(), NULL, 10);
-		else if (!stricmp("IsLocallyMuted", tag))
-			isLocallyMuted = !stricmp(string.c_str(), "true");
-		else if (!stricmp("MicEnergy", tag))
-			energy = (F32)strtod(string.c_str(), NULL);
-		else if (!stricmp("ChannelName", tag))
-			nameString = string;
-		else if (!stricmp("ChannelURI", tag))
-			uriString = string;
-		else if (!stricmp("BuddyURI", tag))
-			uriString = string;
-		else if (!stricmp("Presence", tag))
-			statusString = string;
-		else if (!stricmp("CaptureDevices", tag))
-		{
-			LLWebRTCVoiceClient::getInstance()->setDevicesListUpdated(true);
-		}
-		else if (!stricmp("RenderDevices", tag))
-		{
-			LLWebRTCVoiceClient::getInstance()->setDevicesListUpdated(true);
-		}
-		else if (!stricmp("CaptureDevice", tag))
-		{
-			LLWebRTCVoiceClient::getInstance()->addCaptureDevice(LLVoiceDevice(displayNameString, deviceString));
-		}
-		else if (!stricmp("RenderDevice", tag))
-		{
-			LLWebRTCVoiceClient::getInstance()->addRenderDevice(LLVoiceDevice(displayNameString, deviceString));
-		}
-		else if (!stricmp("BlockMask", tag))
-			blockMask = string;
-		else if (!stricmp("PresenceOnly", tag))
-			presenceOnly = string;
-		else if (!stricmp("AutoAcceptMask", tag))
-			autoAcceptMask = string;
-		else if (!stricmp("AutoAddAsBuddy", tag))
-			autoAddAsBuddy = string;
-		else if (!stricmp("MessageHeader", tag))
-			messageHeader = string;
-		else if (!stricmp("MessageBody", tag))
-			messageBody = string;
-		else if (!stricmp("NotificationType", tag))
-			notificationType = string;
-		else if (!stricmp("HasText", tag))
-			hasText = !stricmp(string.c_str(), "true");
-		else if (!stricmp("HasAudio", tag))
-			hasAudio = !stricmp(string.c_str(), "true");
-		else if (!stricmp("HasVideo", tag))
-			hasVideo = !stricmp(string.c_str(), "true");
-		else if (!stricmp("Terminated", tag))
-			terminated = !stricmp(string.c_str(), "true");
-		else if (!stricmp("SubscriptionHandle", tag))
-			subscriptionHandle = string;
-		else if (!stricmp("SubscriptionType", tag))
-			subscriptionType = string;
-		else if (!stricmp("SessionFont", tag))
-		{
-			LLWebRTCVoiceClient::getInstance()->addVoiceFont(id, nameString, descriptionString, expirationDate, hasExpired, fontType, fontStatus, false);
-		}
-		else if (!stricmp("TemplateFont", tag))
-		{
-			LLWebRTCVoiceClient::getInstance()->addVoiceFont(id, nameString, descriptionString, expirationDate, hasExpired, fontType, fontStatus, true);
-		}
-		else if (!stricmp("ID", tag))
-		{
-			id = strtol(string.c_str(), NULL, 10);
-		}
-		else if (!stricmp("Description", tag))
-		{
-			descriptionString = string;
-		}
-		else if (!stricmp("ExpirationDate", tag))
-		{
-			expirationDate = expiryTimeStampToLLDate(string);
-		}
-		else if (!stricmp("Expired", tag))
-		{
-			hasExpired = !stricmp(string.c_str(), "1");
-		}
-		else if (!stricmp("Type", tag))
-		{
-			fontType = strtol(string.c_str(), NULL, 10);
-		}
-		else if (!stricmp("Status", tag))
-		{
-			fontStatus = strtol(string.c_str(), NULL, 10);
-		}
-		else if (!stricmp("MediaCompletionType", tag))
-		{
-			mediaCompletionType = string;;
-		}
-
-		textBuffer.clear();
-		accumulateText= false;
-		
-		if (responseDepth == 0)
-		{
-			// We finished all of the XML, process the data
-			processResponse(tag);
-		}
-	}
-}
-
-// --------------------------------------------------------------------------------
-
-void LLWebRTCProtocolParser::CharData(const char *buffer, int length)
-{
-	/*
-	 This method is called for anything that isn't a tag, which can be text you
-	 want that lies between tags, and a lot of stuff you don't want like file formatting
-	 (tabs, spaces, CR/LF, etc).
-	 
-	 Only copy text if we are in accumulate mode...
-	 */
-	if (accumulateText)
-		textBuffer.append(buffer, length);
-}
-
-// --------------------------------------------------------------------------------
-
-LLDate LLWebRTCProtocolParser::expiryTimeStampToLLDate(const std::string& WebRTC_ts)
-{
-	// *HACK: WebRTC reports the time incorrectly. LLDate also only parses a
-	// subset of valid ISO 8601 dates (only handles Z, not offsets).
-	// So just use the date portion and fix the time here.
-	std::string time_stamp = WebRTC_ts.substr(0, 10);
-	time_stamp += VOICE_FONT_EXPIRY_TIME;
-
-	LL_DEBUGS("WebRTCProtocolParser") << "WebRTC timestamp " << WebRTC_ts << " modified to: " << time_stamp << LL_ENDL;
-
-	return LLDate(time_stamp);
-}
-
-// --------------------------------------------------------------------------------
-
-void LLWebRTCProtocolParser::processResponse(std::string tag)
-{
-	LL_DEBUGS("WebRTCProtocolParser") << tag << LL_ENDL;
-	
-	// SLIM SDK: the SDK now returns a statusCode of "200" (OK) for success.  This is a change vs. previous SDKs.
-	// According to Mike S., "The actual API convention is that responses with return codes of 0 are successful, regardless of the status code returned",
-	// so I believe this will give correct behavior.
-	
-	if(returnCode == 0)
-		statusCode = 0;
-	
-	if (isEvent)
-	{
-		const char *eventTypeCstr = eventTypeString.c_str();
-        LL_DEBUGS("LowVoice") << eventTypeCstr << LL_ENDL;
-
-		if (!stricmp(eventTypeCstr, "ParticipantUpdatedEvent"))
-		{
-			// These happen so often that logging them is pretty useless.
-            LL_DEBUGS("LowVoice") << "Updated Params: " << sessionHandle << ", " << sessionGroupHandle << ", " << uriString << ", " << alias << ", " << isModeratorMuted << ", " << isSpeaking << ", " << volume << ", " << energy << LL_ENDL;
-            LLWebRTCVoiceClient::getInstance()->participantUpdatedEvent(sessionHandle, sessionGroupHandle, uriString, alias, isModeratorMuted, isSpeaking, volume, energy);
-		}
-		else if (!stricmp(eventTypeCstr, "AccountLoginStateChangeEvent"))
-		{
-			LLWebRTCVoiceClient::getInstance()->accountLoginStateChangeEvent(accountHandle, statusCode, statusString, state);
-		}
-		else if (!stricmp(eventTypeCstr, "SessionAddedEvent"))
-		{
-			/*
-			 <Event type="SessionAddedEvent">
-			 <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg0</SessionGroupHandle>
-			 <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==0</SessionHandle>
-			 <Uri>sip:confctl-1408789@bhr.WebRTC.com</Uri>
-			 <IsChannel>true</IsChannel>
-			 <Incoming>false</Incoming>
-			 <ChannelName />
-			 </Event>
-			 */
-			LLWebRTCVoiceClient::getInstance()->sessionAddedEvent(uriString, alias, sessionHandle, sessionGroupHandle, isChannel, incoming, nameString, applicationString);
-		}
-		else if (!stricmp(eventTypeCstr, "SessionRemovedEvent"))
-		{
-			LLWebRTCVoiceClient::getInstance()->sessionRemovedEvent(sessionHandle, sessionGroupHandle);
-		}
-		else if (!stricmp(eventTypeCstr, "SessionGroupUpdatedEvent"))
-		{
-			//nothng useful to process for this event, but we should not WARN that we have received it.
-		}
-		else if (!stricmp(eventTypeCstr, "SessionGroupAddedEvent"))
-		{
-			LLWebRTCVoiceClient::getInstance()->sessionGroupAddedEvent(sessionGroupHandle);
-		}
-		else if (!stricmp(eventTypeCstr, "MediaStreamUpdatedEvent"))
-		{
-			/*
-			 <Event type="MediaStreamUpdatedEvent">
-			 <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg0</SessionGroupHandle>
-			 <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==0</SessionHandle>
-			 <StatusCode>200</StatusCode>
-			 <StatusString>OK</StatusString>
-			 <State>2</State>
-			 <Incoming>false</Incoming>
-			 </Event>
-			 */
-			LLWebRTCVoiceClient::getInstance()->mediaStreamUpdatedEvent(sessionHandle, sessionGroupHandle, statusCode, statusString, state, incoming);
-		}
-		else if (!stricmp(eventTypeCstr, "MediaCompletionEvent"))
-		{
-			/*
-			<Event type="MediaCompletionEvent">
-			<SessionGroupHandle />
-			<MediaCompletionType>AuxBufferAudioCapture</MediaCompletionType>
-			</Event>
-			*/
-			LLWebRTCVoiceClient::getInstance()->mediaCompletionEvent(sessionGroupHandle, mediaCompletionType);
-		}
-		else if (!stricmp(eventTypeCstr, "ParticipantAddedEvent"))
-		{
-			/* 
-			 <Event type="ParticipantAddedEvent">
-			 <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg4</SessionGroupHandle>
-			 <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==4</SessionHandle>
-			 <ParticipantUri>sip:xI5auBZ60SJWIk606-1JGRQ==@bhr.WebRTC.com</ParticipantUri>
-			 <AccountName>xI5auBZ60SJWIk606-1JGRQ==</AccountName>
-			 <DisplayName />
-			 <ParticipantType>0</ParticipantType>
-			 </Event>
-			 */
-            LL_DEBUGS("LowVoice") << "Added Params: " << sessionHandle << ", " << sessionGroupHandle << ", " << uriString << ", " << alias << ", " << nameString << ", " << displayNameString << ", " << participantType << LL_ENDL;
-			LLWebRTCVoiceClient::getInstance()->participantAddedEvent(sessionHandle, sessionGroupHandle, uriString, alias, nameString, displayNameString, participantType);
-		}
-		else if (!stricmp(eventTypeCstr, "ParticipantRemovedEvent"))
-		{
-			/*
-			 <Event type="ParticipantRemovedEvent">
-			 <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg4</SessionGroupHandle>
-			 <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==4</SessionHandle>
-			 <ParticipantUri>sip:xtx7YNV-3SGiG7rA1fo5Ndw==@bhr.WebRTC.com</ParticipantUri>
-			 <AccountName>xtx7YNV-3SGiG7rA1fo5Ndw==</AccountName>
-			 </Event>
-			 */
-            LL_DEBUGS("LowVoice") << "Removed params:" << sessionHandle << ", " << sessionGroupHandle << ", " << uriString << ", " << alias << ", " << nameString << LL_ENDL;
-
-			LLWebRTCVoiceClient::getInstance()->participantRemovedEvent(sessionHandle, sessionGroupHandle, uriString, alias, nameString);
-		}
-		else if (!stricmp(eventTypeCstr, "AuxAudioPropertiesEvent"))
-		{
-			// These are really spammy in tuning mode
-			LLWebRTCVoiceClient::getInstance()->auxAudioPropertiesEvent(energy);
-		}
-		else if (!stricmp(eventTypeCstr, "MessageEvent"))  
-		{
-			//TODO:  This probably is not received any more, it was used to support SLim clients
-			LLWebRTCVoiceClient::getInstance()->messageEvent(sessionHandle, uriString, alias, messageHeader, messageBody, applicationString);
-		}
-		else if (!stricmp(eventTypeCstr, "SessionNotificationEvent"))  
-		{
-			//TODO:  This probably is not received any more, it was used to support SLim clients
-			LLWebRTCVoiceClient::getInstance()->sessionNotificationEvent(sessionHandle, uriString, notificationType);
-		}
-		else if (!stricmp(eventTypeCstr, "SessionUpdatedEvent"))
-		{
-			/*
-			 <Event type="SessionUpdatedEvent">
-			 <SessionGroupHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==_sg0</SessionGroupHandle>
-			 <SessionHandle>c1_m1000xFnPP04IpREWNkuw1cOXlhw==0</SessionHandle>
-			 <Uri>sip:confctl-9@bhd.WebRTC.com</Uri>
-			 <IsMuted>0</IsMuted>
-			 <Volume>50</Volume>
-			 <TransmitEnabled>1</TransmitEnabled>
-			 <IsFocused>0</IsFocused>
-			 <SpeakerPosition><Position><X>0</X><Y>0</Y><Z>0</Z></Position></SpeakerPosition>
-			 <SessionFontID>0</SessionFontID>
-			 </Event>
-			 */
-			// We don't need to process this, but we also shouldn't warn on it, since that confuses people.
-		}
-		else if (!stricmp(eventTypeCstr, "SessionGroupRemovedEvent"))
-		{
-			// We don't need to process this, but we also shouldn't warn on it, since that confuses people.
-		}
-		else if (!stricmp(eventTypeCstr, "VoiceServiceConnectionStateChangedEvent"))
-		{
-			LLWebRTCVoiceClient::getInstance()->voiceServiceConnectionStateChangedEvent(statusCode, statusString, mBuildID);
-		}
-		else if (!stricmp(eventTypeCstr, "AudioDeviceHotSwapEvent"))
-		{
-			/*
-			<Event type = "AudioDeviceHotSwapEvent">
-			<EventType>RenderDeviceChanged< / EventType>
-			<RelevantDevice>
-			<Device>Speakers(Turtle Beach P11 Headset)< / Device>
-			<DisplayName>Speakers(Turtle Beach P11 Headset)< / DisplayName>
-			<Type>SpecificDevice< / Type>
-			< / RelevantDevice>
-			< / Event>
-			*/
-			// an audio device was removed or added, fetch and update the local list of audio devices.
-		}
-		else
-		{
-			LL_WARNS("WebRTCProtocolParser") << "Unknown event type " << eventTypeString << LL_ENDL;
-		}
-	}
-	else
-	{
-		const char *actionCstr = actionString.c_str();
-        LL_DEBUGS("LowVoice") << actionCstr << LL_ENDL;
 
-		if (!stricmp(actionCstr, "Session.Set3DPosition.1"))
-		{
-			// We don't need to process these
-		}
-		else if (!stricmp(actionCstr, "Connector.Create.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->connectorCreateResponse(statusCode, statusString, connectorHandle, versionID);
-		}
-		else if (!stricmp(actionCstr, "Account.Login.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->loginResponse(statusCode, statusString, accountHandle, numberOfAliases);
-		}
-		else if (!stricmp(actionCstr, "Session.Create.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->sessionCreateResponse(requestId, statusCode, statusString, sessionHandle);			
-		}
-		else if (!stricmp(actionCstr, "SessionGroup.AddSession.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->sessionGroupAddSessionResponse(requestId, statusCode, statusString, sessionHandle);			
-		}
-		else if (!stricmp(actionCstr, "Session.Connect.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->sessionConnectResponse(requestId, statusCode, statusString);			
-		}
-		else if (!stricmp(actionCstr, "Account.Logout.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->logoutResponse(statusCode, statusString);			
-		}
-		else if (!stricmp(actionCstr, "Connector.InitiateShutdown.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->connectorShutdownResponse(statusCode, statusString);			
-		}
-		else if (!stricmp(actionCstr, "Account.GetSessionFonts.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->accountGetSessionFontsResponse(statusCode, statusString);
-		}
-		else if (!stricmp(actionCstr, "Account.GetTemplateFonts.1"))
-		{
-			LLWebRTCVoiceClient::getInstance()->accountGetTemplateFontsResponse(statusCode, statusString);
-		}
-		else if (!stricmp(actionCstr, "Aux.SetVadProperties.1"))
-		{
-			// both values of statusCode (old and more recent) indicate valid requests
-			if (statusCode != 0 && statusCode != 200)
-			{
-				LL_WARNS("Voice") << "Aux.SetVadProperties.1 request failed: "
-					<< "statusCode: " << statusCode
-					<< " and "
-					<< "statusString: " << statusString
-					<< LL_ENDL;
-			}
-		}
-		/*
-		 else if (!stricmp(actionCstr, "Account.ChannelGetList.1"))
-		 {
-		 LLVoiceClient::getInstance()->channelGetListResponse(statusCode, statusString);
-		 }
-		 else if (!stricmp(actionCstr, "Connector.AccountCreate.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Connector.MuteLocalMic.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Connector.MuteLocalSpeaker.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Connector.SetLocalMicVolume.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Connector.SetLocalSpeakerVolume.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Session.ListenerSetPosition.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Session.SpeakerSetPosition.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Session.AudioSourceSetPosition.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Session.GetChannelParticipants.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelCreate.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelUpdate.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelDelete.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelCreateAndInvite.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelFolderCreate.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelFolderUpdate.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelFolderDelete.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelAddModerator.1"))
-		 {
-		 
-		 }
-		 else if (!stricmp(actionCstr, "Account.ChannelDeleteModerator.1"))
-		 {
-		 
-		 }
-		 */
-	}
-}
 
 LLWebRTCSecurity::LLWebRTCSecurity()
 {
diff --git a/indra/newview/llvoicewebrtc.h b/indra/newview/llvoicewebrtc.h
index f4e2ac0b7f..e040552ab2 100644
--- a/indra/newview/llvoicewebrtc.h
+++ b/indra/newview/llvoicewebrtc.h
@@ -712,8 +712,6 @@ private:
 	// We should kill the voice daemon in case of connection alert 
 	bool mTerminateDaemon;
 	
-	friend class LLWebRTCProtocolParser;
-	
 	std::string mAccountName;
 	std::string mAccountPassword;
     std::string mChannelSDP;
@@ -956,118 +954,6 @@ private:
     LLEventMailDrop mWebRTCPump;
 };
 
-
-/** 
- * @class LLWebRTCProtocolParser
- * @brief This class helps construct new LLIOPipe specializations
- * @see LLIOPipe
- *
- * THOROUGH_DESCRIPTION
- */
-class LLWebRTCProtocolParser : public LLIOPipe
-{
-	LOG_CLASS(LLWebRTCProtocolParser);
-public:
-	LLWebRTCProtocolParser();
-	virtual ~LLWebRTCProtocolParser();
-	
-protected:
-	/* @name LLIOPipe virtual implementations
-	 */
-	//@{
-	/** 
-	 * @brief Process the data in buffer
-	 */
-	virtual EStatus process_impl(
-								 const LLChannelDescriptors& channels,
-								 buffer_ptr_t& buffer,
-								 bool& eos,
-								 LLSD& context,
-								 LLPumpIO* pump);
-	//@}
-	
-	std::string 	mInput;
-	
-	// Expat control members
-	XML_Parser		parser;
-	int				responseDepth;
-	bool			ignoringTags;
-	bool			isEvent;
-	int				ignoreDepth;
-	
-	// Members for processing responses. The values are transient and only valid within a call to processResponse().
-	int				returnCode;
-	int				statusCode;
-	std::string		statusString;
-	std::string		requestId;
-	std::string		actionString;
-	std::string		connectorHandle;
-	std::string		versionID;
-	std::string		mBuildID;
-	std::string		accountHandle;
-	std::string		sessionHandle;
-	std::string		sessionGroupHandle;
-	std::string		alias;
-	std::string		applicationString;
-	
-	// Members for processing events. The values are transient and only valid within a call to processResponse().
-	std::string		eventTypeString;
-	int				state;
-	std::string		uriString;
-	bool			isChannel;
-	bool			incoming;
-	bool			enabled;
-	std::string		nameString;
-	std::string		audioMediaString;
-	std::string     deviceString;
-	std::string		displayNameString;
-	int				participantType;
-	bool			isLocallyMuted;
-	bool			isModeratorMuted;
-	bool			isSpeaking;
-	int				volume;
-	F32				energy;
-	std::string		messageHeader;
-	std::string		messageBody;
-	std::string		notificationType;
-	bool			hasText;
-	bool			hasAudio;
-	bool			hasVideo;
-	bool			terminated;
-	std::string		blockMask;
-	std::string		presenceOnly;
-	std::string		autoAcceptMask;
-	std::string		autoAddAsBuddy;
-	int				numberOfAliases;
-	std::string		subscriptionHandle;
-	std::string		subscriptionType;
-	S32				id;
-	std::string		descriptionString;
-	LLDate			expirationDate;
-	bool			hasExpired;
-	S32				fontType;
-	S32				fontStatus;
-	std::string		mediaCompletionType;
-	
-	// Members for processing text between tags
-	std::string		textBuffer;
-	bool			accumulateText;
-	
-	void			reset();
-	
-	void			processResponse(std::string tag);
-	
-	static void XMLCALL ExpatStartTag(void *data, const char *el, const char **attr);
-	static void XMLCALL ExpatEndTag(void *data, const char *el);
-	static void XMLCALL ExpatCharHandler(void *data, const XML_Char *s, int len);
-	
-	void			StartTag(const char *tag, const char **attr);
-	void			EndTag(const char *tag);
-	void			CharData(const char *buffer, int length);
-	LLDate			expiryTimeStampToLLDate(const std::string& WebRTC_ts);
-
-};
-
 class LLWebRTCSecurity :	public LLSingleton<LLWebRTCSecurity>
 {
     LLSINGLETON(LLWebRTCSecurity);