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-rw-r--r--indra/newview/llvoicewebrtc.cpp444
1 files changed, 120 insertions, 324 deletions
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp
index 9013de67f5..da27ff7320 100644
--- a/indra/newview/llvoicewebrtc.cpp
+++ b/indra/newview/llvoicewebrtc.cpp
@@ -74,6 +74,9 @@
// for base64 decoding
#include "apr_base64.h"
+#include "json/reader.h"
+#include "json/writer.h"
+
#define USE_SESSION_GROUPS 0
#define VX_NULL_POSITION -2147483648.0 /*The Silence*/
@@ -88,7 +91,7 @@ namespace {
static const std::string VOICE_SERVER_TYPE = "WebRTC";
// Don't send positional updates more frequently than this:
- const F32 UPDATE_THROTTLE_SECONDS = 0.5f;
+ const F32 UPDATE_THROTTLE_SECONDS = 0.1f;
// Timeout for connection to WebRTC
const F32 CONNECT_ATTEMPT_TIMEOUT = 300.0f;
@@ -375,7 +378,7 @@ void LLWebRTCVoiceClient::terminate()
mRelogRequested = false;
mVoiceEnabled = false;
- llwebrtc::init();
+ llwebrtc::terminate();
sShuttingDown = true;
sPump = NULL;
@@ -684,7 +687,7 @@ void LLWebRTCVoiceClient::voiceControlStateMachine()
{
performMicTuning();
}
-
+ sessionEstablished();
setVoiceControlStateUnless(VOICE_STATE_WAIT_FOR_CHANNEL, VOICE_STATE_SESSION_RETRY);
}
break;
@@ -1116,6 +1119,9 @@ bool LLWebRTCVoiceClient::addAndJoinSession(const sessionStatePtr_t &nextSession
// Just flush it all out and start new.
mWebRTCPump.discard();
+ // add 'self' participant.
+ addParticipantByID(gAgent.getID());
+
notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_JOINED);
return true;
@@ -1922,16 +1928,6 @@ void LLWebRTCVoiceClient::sessionMediaDisconnectSendMessage(const sessionStatePt
}
-void LLWebRTCVoiceClient::OnCaptureDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList& render_devices)
-{
- clearCaptureDevices();
- for (auto &device : render_devices)
- {
- LLWebRTCVoiceClient::addCaptureDevice(LLVoiceDevice(device.display_name, device.id));
- }
- LLWebRTCVoiceClient::setDevicesListUpdated(true);
-}
-
void LLWebRTCVoiceClient::clearCaptureDevices()
{
LL_DEBUGS("Voice") << "called" << LL_ENDL;
@@ -1967,13 +1963,19 @@ void LLWebRTCVoiceClient::setDevicesListUpdated(bool state)
mDevicesListUpdated = state;
}
-void LLWebRTCVoiceClient::OnRenderDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList &render_devices)
+void LLWebRTCVoiceClient::OnDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList &render_devices,
+ const llwebrtc::LLWebRTCVoiceDeviceList &capture_devices)
{
clearRenderDevices();
for (auto &device : render_devices)
{
addRenderDevice(LLVoiceDevice(device.display_name, device.id));
}
+ clearCaptureDevices();
+ for (auto &device : capture_devices)
+ {
+ addCaptureDevice(LLVoiceDevice(device.display_name, device.id));
+ }
setDevicesListUpdated(true);
}
@@ -2105,7 +2107,7 @@ bool LLWebRTCVoiceClient::deviceSettingsAvailable()
{
bool result = true;
- if(mRenderDevices.empty())
+ if(mRenderDevices.empty() || mCaptureDevices.empty())
result = false;
return result;
@@ -2113,11 +2115,7 @@ bool LLWebRTCVoiceClient::deviceSettingsAvailable()
bool LLWebRTCVoiceClient::deviceSettingsUpdated()
{
bool updated = mDevicesListUpdated;
- if (mDevicesListUpdated)
- {
- // a hot swap event or a polling of the audio devices has been parsed since the last redraw of the input and output device panel.
- mDevicesListUpdated = false; // toggle the setting
- }
+ mDevicesListUpdated = false;
return updated;
}
@@ -2285,20 +2283,12 @@ void LLWebRTCVoiceClient::setHidden(bool hidden)
void LLWebRTCVoiceClient::sendPositionAndVolumeUpdate(void)
{
- std::ostringstream stream;
-
if (mSpatialCoordsDirty && inSpatialChannel())
{
LLVector3 l, u, a, vel;
LLVector3d pos;
mSpatialCoordsDirty = false;
-
- // Always send both speaker and listener positions together.
- stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.Set3DPosition.1\">"
- << "<SessionHandle>" << getAudioSessionHandle() << "</SessionHandle>";
-
- stream << "<SpeakerPosition>";
LLMatrix3 avatarRot = mAvatarRot.getMatrix3();
@@ -2321,38 +2311,6 @@ void LLWebRTCVoiceClient::sendPositionAndVolumeUpdate(void)
pos.mdV[i] = VX_NULL_POSITION;
}
}
-
- stream
- << "<Position>"
- << "<X>" << pos.mdV[VX] << "</X>"
- << "<Y>" << pos.mdV[VY] << "</Y>"
- << "<Z>" << pos.mdV[VZ] << "</Z>"
- << "</Position>"
- << "<Velocity>"
- << "<X>" << vel.mV[VX] << "</X>"
- << "<Y>" << vel.mV[VY] << "</Y>"
- << "<Z>" << vel.mV[VZ] << "</Z>"
- << "</Velocity>"
- << "<AtOrientation>"
- << "<X>" << a.mV[VX] << "</X>"
- << "<Y>" << a.mV[VY] << "</Y>"
- << "<Z>" << a.mV[VZ] << "</Z>"
- << "</AtOrientation>"
- << "<UpOrientation>"
- << "<X>" << u.mV[VX] << "</X>"
- << "<Y>" << u.mV[VY] << "</Y>"
- << "<Z>" << u.mV[VZ] << "</Z>"
- << "</UpOrientation>"
- << "<LeftOrientation>"
- << "<X>" << l.mV [VX] << "</X>"
- << "<Y>" << l.mV [VY] << "</Y>"
- << "<Z>" << l.mV [VZ] << "</Z>"
- << "</LeftOrientation>"
- ;
-
- stream << "</SpeakerPosition>";
-
- stream << "<ListenerPosition>";
LLVector3d earPosition;
LLVector3 earVelocity;
@@ -2397,40 +2355,13 @@ void LLWebRTCVoiceClient::sendPositionAndVolumeUpdate(void)
pos.mdV[i] = VX_NULL_POSITION;
}
}
-
- stream
- << "<Position>"
- << "<X>" << pos.mdV[VX] << "</X>"
- << "<Y>" << pos.mdV[VY] << "</Y>"
- << "<Z>" << pos.mdV[VZ] << "</Z>"
- << "</Position>"
- << "<Velocity>"
- << "<X>" << vel.mV[VX] << "</X>"
- << "<Y>" << vel.mV[VY] << "</Y>"
- << "<Z>" << vel.mV[VZ] << "</Z>"
- << "</Velocity>"
- << "<AtOrientation>"
- << "<X>" << a.mV[VX] << "</X>"
- << "<Y>" << a.mV[VY] << "</Y>"
- << "<Z>" << a.mV[VZ] << "</Z>"
- << "</AtOrientation>"
- << "<UpOrientation>"
- << "<X>" << u.mV[VX] << "</X>"
- << "<Y>" << u.mV[VY] << "</Y>"
- << "<Z>" << u.mV[VZ] << "</Z>"
- << "</UpOrientation>"
- << "<LeftOrientation>"
- << "<X>" << l.mV [VX] << "</X>"
- << "<Y>" << l.mV [VY] << "</Y>"
- << "<Z>" << l.mV [VZ] << "</Z>"
- << "</LeftOrientation>"
- ;
-
- stream << "</ListenerPosition>";
+ }
- stream << "<ReqDispositionType>1</ReqDispositionType>"; //do not generate responses for update requests
- stream << "</Request>\n\n\n";
- }
+ if (mWebRTCAudioInterface)
+ {
+ mWebRTCAudioInterface->requestAudioLevel();
+ }
+
if(mAudioSession && (mAudioSession->mVolumeDirty || mAudioSession->mMuteDirty))
{
@@ -2469,41 +2400,12 @@ void LLWebRTCVoiceClient::sendPositionAndVolumeUpdate(void)
}
LL_DEBUGS("Voice") << "Setting volume/mute for avatar " << p->mAvatarID << " to " << volume << (mute?"/true":"/false") << LL_ENDL;
-
- // SLIM SDK: Send both volume and mute commands.
-
- // Send a "volume for me" command for the user.
- stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.SetParticipantVolumeForMe.1\">"
- << "<SessionHandle>" << getAudioSessionHandle() << "</SessionHandle>"
- << "<ParticipantURI>" << p->mURI << "</ParticipantURI>"
- << "<Volume>" << volume << "</Volume>"
- << "</Request>\n\n\n";
-
- if(!mAudioSession->mIsP2P)
- {
- // Send a "mute for me" command for the user
- // Doesn't work in P2P sessions
- stream << "<Request requestId=\"" << mCommandCookie++ << "\" action=\"Session.SetParticipantMuteForMe.1\">"
- << "<SessionHandle>" << getAudioSessionHandle() << "</SessionHandle>"
- << "<ParticipantURI>" << p->mURI << "</ParticipantURI>"
- << "<Mute>" << (mute?"1":"0") << "</Mute>"
- << "<Scope>Audio</Scope>"
- << "</Request>\n\n\n";
- }
}
p->mVolumeDirty = false;
}
}
}
-
- std::string update(stream.str());
- if(!update.empty())
- {
- LL_DEBUGS("VoiceUpdate") << "sending update " << update << LL_ENDL;
- writeString(update);
- }
-
}
void LLWebRTCVoiceClient::sendLocalAudioUpdates()
@@ -2700,6 +2602,7 @@ void LLWebRTCVoiceClient::OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface *
{
LL_INFOS("Voice") << "On AudioEstablished." << LL_ENDL;
mWebRTCAudioInterface = audio_interface;
+ mWebRTCAudioInterface->setAudioObserver(this);
float speaker_volume = 0;
audio_interface->setMute(true);
{
@@ -2710,9 +2613,68 @@ void LLWebRTCVoiceClient::OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface *
setVoiceControlStateUnless(VOICE_STATE_SESSION_ESTABLISHED, VOICE_STATE_SESSION_RETRY);
}
+void LLWebRTCVoiceClient::OnAudioLevel(float level)
+{
+ if (mWebRTCDataInterface)
+ {
+ Json::FastWriter writer;
+ Json::Value root;
+ root["p"] = (UINT32) (level * 256);
+ std::string json_data = writer.write(root);
+
+ mWebRTCDataInterface->sendData(json_data, false);
+ }
+}
+
void LLWebRTCVoiceClient::OnDataReceived(const std::string& data, bool binary)
{
-
+ // incoming data will be a json structure (if it's not binary.) We may pack
+ // binary for size reasons. Most of the keys in the json objects are
+ // single or double characters for size reasons.
+ // The primary element is:
+ // An object where each key is an agent id. (in the future, we may allow
+ // integer indices into an agentid list, populated on join commands. For size.
+ // Each key will point to a json object with keys identifying what's updated.
+ // 'p' - audio source power (level/volume) (int8 as int)
+ // 'j' - join - object of join data (TBD) (true for now)
+ // 'l' - boolean, always true if exists.
+
+ if (binary)
+ {
+ LL_WARNS("Voice") << "Binary data received from data channel." << LL_ENDL;
+ return;
+ }
+
+ Json::Reader reader;
+ Json::Value voice_data;
+ if (reader.parse(data, voice_data, false)) // don't collect comments
+ {
+ if (!voice_data.isObject())
+ {
+ LL_WARNS("Voice") << "Expected object from data channel:" << data << LL_ENDL;
+ return;
+ }
+ for (auto &participant_id : voice_data.getMemberNames())
+ {
+ std::string foo = participant_id;
+ LL_WARNS("Voice") << "Participant ID (" << participant_id << "):" << data << LL_ENDL;
+
+ LLUUID agent_id(participant_id);
+ if (agent_id.isNull())
+ {
+ LL_WARNS("Voice") << "Bad participant ID from data channel (" << participant_id << "):" << data << LL_ENDL;
+ continue;
+ }
+ participantStatePtr_t participant = findParticipantByID(agent_id);
+ if (participant)
+ {
+ participant->mPower = (F32) (voice_data[participant_id].get("p", Json::Value(participant->mPower)).asInt()) / 256;
+ /* WebRTC appears to have deprecated VAD, but it's still in the Audio Processing Module so maybe we
+ can use it at some point when we actually process frames. */
+ participant->mIsSpeaking = participant->mPower > 0.05;
+ }
+ }
+ }
}
@@ -2994,7 +2956,7 @@ void LLWebRTCVoiceClient::sessionAddedEvent(
session->mAlternateSIPURI = session->mSIPURI;
// and generate a proper URI from the ID.
- setSessionURI(session, sipURIFromID(session->mCallerID));
+ setSessionURI(session, session->mCallerID.asString());
}
else
{
@@ -3065,7 +3027,7 @@ void LLWebRTCVoiceClient::joinedAudioSession(const sessionStatePtr_t &session)
if(!session->mIsChannel)
{
// this is a p2p session. Make sure the other end is added as a participant.
- participantStatePtr_t participant(session->addParticipant(session->mSIPURI));
+ participantStatePtr_t participant(session->addParticipant(LLUUID(session->mSIPURI)));
if(participant)
{
if(participant->mAvatarIDValid)
@@ -3365,7 +3327,7 @@ void LLWebRTCVoiceClient::participantAddedEvent(
sessionStatePtr_t session(findSession(sessionHandle));
if(session)
{
- participantStatePtr_t participant(session->addParticipant(uriString));
+ participantStatePtr_t participant(session->addParticipant(LLUUID(uriString)));
if(participant)
{
participant->mAccountName = nameString;
@@ -3424,92 +3386,6 @@ void LLWebRTCVoiceClient::participantRemovedEvent(
}
}
-
-void LLWebRTCVoiceClient::participantUpdatedEvent(
- std::string &sessionHandle,
- std::string &sessionGroupHandle,
- std::string &uriString,
- std::string &alias,
- bool isModeratorMuted,
- bool isSpeaking,
- int volume,
- F32 energy)
-{
- sessionStatePtr_t session(findSession(sessionHandle));
- if(session)
- {
- participantStatePtr_t participant(session->findParticipant(uriString));
-
- if(participant)
- {
- //LL_INFOS("Voice") << "Participant Update for " << participant->mDisplayName << LL_ENDL;
-
- participant->mIsSpeaking = isSpeaking;
- participant->mIsModeratorMuted = isModeratorMuted;
-
- // SLIM SDK: convert range: ensure that energy is set to zero if is_speaking is false
- if (isSpeaking)
- {
- participant->mSpeakingTimeout.reset();
- participant->mPower = energy;
- }
- else
- {
- participant->mPower = 0.0f;
- }
-
- // Ignore incoming volume level if it has been explicitly set, or there
- // is a volume or mute change pending.
- if ( !participant->mVolumeSet && !participant->mVolumeDirty)
- {
- participant->mVolume = (F32)volume * VOLUME_SCALE_WEBRTC;
- }
-
- // *HACK: mantipov: added while working on EXT-3544
- /*
- Sometimes LLVoiceClient::participantUpdatedEvent callback is called BEFORE
- LLViewerChatterBoxSessionAgentListUpdates::post() sometimes AFTER.
-
- participantUpdatedEvent updates voice participant state in particular participantState::mIsModeratorMuted
- Originally we wanted to update session Speaker Manager to fire LLSpeakerVoiceModerationEvent to fix the EXT-3544 bug.
- Calling of the LLSpeakerMgr::update() method was added into LLIMMgr::processAgentListUpdates.
-
- But in case participantUpdatedEvent() is called after LLViewerChatterBoxSessionAgentListUpdates::post()
- voice participant mIsModeratorMuted is changed after speakers are updated in Speaker Manager
- and event is not fired.
-
- So, we have to call LLSpeakerMgr::update() here.
- */
- LLVoiceChannel* voice_cnl = LLVoiceChannel::getCurrentVoiceChannel();
-
- // ignore session ID of local chat
- if (voice_cnl && voice_cnl->getSessionID().notNull())
- {
- LLSpeakerMgr* speaker_manager = LLIMModel::getInstance()->getSpeakerManager(voice_cnl->getSessionID());
- if (speaker_manager)
- {
- speaker_manager->update(true);
-
- // also initialize voice moderate_mode depend on Agent's participant. See EXT-6937.
- // *TODO: remove once a way to request the current voice channel moderation mode is implemented.
- if (gAgent.getID() == participant->mAvatarID)
- {
- speaker_manager->initVoiceModerateMode();
- }
- }
- }
- }
- else
- {
- LL_WARNS("Voice") << "unknown participant: " << uriString << LL_ENDL;
- }
- }
- else
- {
- LL_DEBUGS("Voice") << "unknown session " << sessionHandle << LL_ENDL;
- }
-}
-
void LLWebRTCVoiceClient::messageEvent(
std::string &sessionHandle,
std::string &uriString,
@@ -3743,8 +3619,9 @@ void LLWebRTCVoiceClient::muteListChanged()
/////////////////////////////
// Managing list of participants
-LLWebRTCVoiceClient::participantState::participantState(const std::string &uri) :
- mURI(uri),
+LLWebRTCVoiceClient::participantState::participantState(const LLUUID& agent_id) :
+ mURI(agent_id.asString()),
+ mAvatarID(agent_id),
mPTT(false),
mIsSpeaking(false),
mIsModeratorMuted(false),
@@ -3760,55 +3637,27 @@ LLWebRTCVoiceClient::participantState::participantState(const std::string &uri)
{
}
-LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::sessionState::addParticipant(const std::string &uri)
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::sessionState::addParticipant(const LLUUID& agent_id)
{
participantStatePtr_t result;
- bool useAlternateURI = false;
- // Note: this is mostly the body of LLWebRTCVoiceClient::sessionState::findParticipant(), but since we need to know if it
- // matched the alternate SIP URI (so we can add it properly), we need to reproduce it here.
- {
- participantMap::iterator iter = mParticipantsByURI.find(uri);
+ participantUUIDMap::iterator iter = mParticipantsByUUID.find(agent_id);
- if(iter == mParticipantsByURI.end())
- {
- if(!mAlternateSIPURI.empty() && (uri == mAlternateSIPURI))
- {
- // This is a p2p session (probably with the SLIM client) with an alternate URI for the other participant.
- // Use mSIPURI instead, since it will be properly encoded.
- iter = mParticipantsByURI.find(mSIPURI);
- useAlternateURI = true;
- }
- }
- if(iter != mParticipantsByURI.end())
- {
- result = iter->second;
- }
+ if (iter != mParticipantsByUUID.end())
+ {
+ result = iter->second;
}
if(!result)
{
// participant isn't already in one list or the other.
- result.reset(new participantState(useAlternateURI?mSIPURI:uri));
- mParticipantsByURI.insert(participantMap::value_type(result->mURI, result));
+ result.reset(new participantState(agent_id));
+ mParticipantsByURI.insert(participantMap::value_type(agent_id.asString(), result));
mParticipantsChanged = true;
-
- // Try to do a reverse transform on the URI to get the GUID back.
- {
- LLUUID id;
- if(LLWebRTCVoiceClient::getInstance()->IDFromName(result->mURI, id))
- {
- result->mAvatarIDValid = true;
- result->mAvatarID = id;
- }
- else
- {
- // Create a UUID by hashing the URI, but do NOT set mAvatarIDValid.
- // This indicates that the ID will not be in the name cache.
- result->mAvatarID.generate(uri);
- }
- }
+
+ result->mAvatarIDValid = true;
+ result->mAvatarID = agent_id;
if(result->updateMuteState())
{
@@ -3983,6 +3832,16 @@ LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::findParticipantB
return result;
}
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::addParticipantByID(const LLUUID &id)
+{
+ participantStatePtr_t result;
+ if (mAudioSession)
+ {
+ result = mAudioSession->addParticipant(id);
+ }
+ return result;
+}
+
// Check for parcel boundary crossing
@@ -4156,68 +4015,14 @@ bool LLWebRTCVoiceClient::setSpatialChannel(
void LLWebRTCVoiceClient::callUser(const LLUUID &uuid)
{
- std::string userURI = sipURIFromID(uuid);
-
- switchChannel(userURI, false, true, true);
+ switchChannel(uuid.asString(), false, true, true);
}
-#if 0
-// WebRTC text IMs are not in use.
-LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::startUserIMSession(const LLUUID &uuid)
-{
- // Figure out if a session with the user already exists
- sessionStatePtr_t session(findSession(uuid));
- if(!session)
- {
- // No session with user, need to start one.
- std::string uri = sipURIFromID(uuid);
- session = addSession(uri);
-
- llassert(session);
- if (!session)
- return session;
- session->mIsSpatial = false;
- session->mReconnect = false;
- session->mIsP2P = true;
- session->mCallerID = uuid;
- }
-
- if(session->mHandle.empty())
- {
- // Session isn't active -- start it up.
- sessionCreateSendMessage(session, false, false);
- }
- else
- {
- // Session is already active -- start up text.
- sessionTextConnectSendMessage(session);
- }
-
- return session;
-}
-#endif
void LLWebRTCVoiceClient::endUserIMSession(const LLUUID &uuid)
{
-#if 0
- // WebRTC text IMs are not in use.
-
- // Figure out if a session with the user exists
- sessionStatePtr_t session(findSession(uuid));
- if(session)
- {
- // found the session
- if(!session->mHandle.empty())
- {
- // sessionTextDisconnectSendMessage(session); // a SLim leftover, not used any more.
- }
- }
- else
- {
- LL_DEBUGS("Voice") << "Session not found for participant ID " << uuid << LL_ENDL;
- }
-#endif
+
}
bool LLWebRTCVoiceClient::isValidChannel(std::string &sessionHandle)
{
@@ -4361,16 +4166,6 @@ bool LLWebRTCVoiceClient::inProximalChannel()
return result;
}
-std::string LLWebRTCVoiceClient::sipURIFromID(const LLUUID &id)
-{
- std::string result;
- result = "sip:";
- result += nameFromID(id);
- result += "@";
-
- return result;
-}
-
std::string LLWebRTCVoiceClient::nameFromAvatar(LLVOAvatar *avatar)
{
std::string result;
@@ -4785,10 +4580,6 @@ BOOL LLWebRTCVoiceClient::getIsSpeaking(const LLUUID& id)
participantStatePtr_t participant(findParticipantByID(id));
if(participant)
{
- if (participant->mSpeakingTimeout.getElapsedTimeF32() > SPEAKING_TIMEOUT)
- {
- participant->mIsSpeaking = FALSE;
- }
result = participant->mIsSpeaking;
}
@@ -4814,7 +4605,8 @@ F32 LLWebRTCVoiceClient::getCurrentPower(const LLUUID& id)
participantStatePtr_t participant(findParticipantByID(id));
if(participant)
{
- result = participant->mPower;
+ LL_WARNS("Voice") << "Power:" << participant->mPower << LL_ENDL;
+ result = participant->mPower*4;
}
return result;
@@ -5178,7 +4970,10 @@ void LLWebRTCVoiceClient::sessionState::for_eachPredicate(const LLWebRTCVoiceCli
}
}
-
+void LLWebRTCVoiceClient::sessionEstablished()
+{
+ addSession(gAgent.getRegion()->getRegionID().asString());
+}
LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::findSession(const std::string &handle)
{
@@ -6367,6 +6162,7 @@ void LLWebRTCVoiceClient::captureBufferPlayStopSendMessage()
}
}
+std::string LLWebRTCVoiceClient::sipURIFromID(const LLUUID& id) { return id.asString(); }
LLWebRTCSecurity::LLWebRTCSecurity()