From 8ba90d077381bf16b4ba03a0f530f76e770e69c1 Mon Sep 17 00:00:00 2001
From: Roxie Linden <roxie@lindenlab.com>
Date: Wed, 13 Sep 2023 14:49:42 -0700
Subject: [PATCH] Hook up speaker volume.
---
indra/llwebrtc/llwebrtc.cpp | 22 +
indra/llwebrtc/llwebrtc.h | 1 +
indra/llwebrtc/llwebrtc_impl.h | 1 +
indra/newview/llvoicewebrtc.cpp | 797 +-------------------------------
indra/newview/llvoicewebrtc.h | 114 -----
5 files changed, 49 insertions(+), 886 deletions(-)
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
index c2631b2ea36..93e9db9c1d6 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 acc3665e95e..ca558add014 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 d439bd253d0..5c6cfcdbc69 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 516bab19149..23033c5fee4 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 f4e2ac0b7f8..e040552ab24 100644
--- a/indra/newview/llvoicewebrtc.h
+++ b/indra/newview/llvoicewebrtc.h
@@ -712,8 +712,6 @@ class LLWebRTCVoiceClient : public LLSingleton<LLWebRTCVoiceClient>,
// 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 @@ class LLWebRTCVoiceClient : public LLSingleton<LLWebRTCVoiceClient>,
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);
--
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