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-rw-r--r--indra/newview/llvoicewebrtc.cpp2885
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diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp
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+ /**
+ * @file LLWebRTCVoiceClient.cpp
+ * @brief Implementation of LLWebRTCVoiceClient class which is the interface to the voice client process.
+ *
+ * $LicenseInfo:firstyear=2001&license=viewerlgpl$
+ * Second Life Viewer Source Code
+ * Copyright (C) 2023, Linden Research, Inc.
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation
+ * version 2.1 of the License only.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ *
+ * Linden Research, Inc., 945 Battery Street, San Francisco, CA 94111 USA
+ * $/LicenseInfo$
+ */
+#include <algorithm>
+#include "llvoicewebrtc.h"
+
+#include "llsdutil.h"
+
+// Linden library includes
+#include "llavatarnamecache.h"
+#include "llvoavatarself.h"
+#include "llbufferstream.h"
+#include "llfile.h"
+#include "llmenugl.h"
+#ifdef LL_USESYSTEMLIBS
+# include "expat.h"
+#else
+# include "expat/expat.h"
+#endif
+#include "llcallbacklist.h"
+#include "llviewernetwork.h" // for gGridChoice
+#include "llbase64.h"
+#include "llviewercontrol.h"
+#include "llappviewer.h" // for gDisconnected, gDisableVoice
+#include "llprocess.h"
+
+// Viewer includes
+#include "llmutelist.h" // to check for muted avatars
+#include "llagent.h"
+#include "llcachename.h"
+#include "llimview.h" // for LLIMMgr
+#include "llworld.h"
+#include "llparcel.h"
+#include "llviewerparcelmgr.h"
+#include "llfirstuse.h"
+#include "llspeakers.h"
+#include "lltrans.h"
+#include "llrand.h"
+#include "llviewerwindow.h"
+#include "llviewercamera.h"
+#include "llversioninfo.h"
+
+#include "llviewernetwork.h"
+#include "llnotificationsutil.h"
+
+#include "llcorehttputil.h"
+#include "lleventfilter.h"
+
+#include "stringize.h"
+
+#include "llwebrtc.h"
+
+// for base64 decoding
+#include "apr_base64.h"
+
+#include "json/reader.h"
+#include "json/writer.h"
+
+const std::string WEBRTC_VOICE_SERVER_TYPE = "webrtc";
+
+namespace {
+
+ const F32 MAX_AUDIO_DIST = 50.0f;
+ const F32 VOLUME_SCALE_WEBRTC = 0.01f;
+ const F32 LEVEL_SCALE_WEBRTC = 0.008f;
+
+ const F32 SPEAKING_AUDIO_LEVEL = 0.40;
+
+ static const std::string REPORTED_VOICE_SERVER_TYPE = "Secondlife WebRTC Gateway";
+
+ // Don't send positional updates more frequently than this:
+ const F32 UPDATE_THROTTLE_SECONDS = 0.1f;
+
+ // Cosine of a "trivially" small angle
+ const F32 FOUR_DEGREES = 4.0f * (F_PI / 180.0f);
+ const F32 MINUSCULE_ANGLE_COS = (F32) cos(0.5f * FOUR_DEGREES);
+
+} // namespace
+
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+
+void LLVoiceWebRTCStats::reset()
+{
+ mStartTime = -1.0f;
+ mConnectCycles = 0;
+ mConnectTime = -1.0f;
+ mConnectAttempts = 0;
+ mProvisionTime = -1.0f;
+ mProvisionAttempts = 0;
+ mEstablishTime = -1.0f;
+ mEstablishAttempts = 0;
+}
+
+LLVoiceWebRTCStats::LLVoiceWebRTCStats()
+{
+ reset();
+}
+
+LLVoiceWebRTCStats::~LLVoiceWebRTCStats()
+{
+}
+
+void LLVoiceWebRTCStats::connectionAttemptStart()
+{
+ if (!mConnectAttempts)
+ {
+ mStartTime = LLTimer::getTotalTime();
+ mConnectCycles++;
+ }
+ mConnectAttempts++;
+}
+
+void LLVoiceWebRTCStats::connectionAttemptEnd(bool success)
+{
+ if ( success )
+ {
+ mConnectTime = (LLTimer::getTotalTime() - mStartTime) / USEC_PER_SEC;
+ }
+}
+
+void LLVoiceWebRTCStats::provisionAttemptStart()
+{
+ if (!mProvisionAttempts)
+ {
+ mStartTime = LLTimer::getTotalTime();
+ }
+ mProvisionAttempts++;
+}
+
+void LLVoiceWebRTCStats::provisionAttemptEnd(bool success)
+{
+ if ( success )
+ {
+ mProvisionTime = (LLTimer::getTotalTime() - mStartTime) / USEC_PER_SEC;
+ }
+}
+
+void LLVoiceWebRTCStats::establishAttemptStart()
+{
+ if (!mEstablishAttempts)
+ {
+ mStartTime = LLTimer::getTotalTime();
+ }
+ mEstablishAttempts++;
+}
+
+void LLVoiceWebRTCStats::establishAttemptEnd(bool success)
+{
+ if ( success )
+ {
+ mEstablishTime = (LLTimer::getTotalTime() - mStartTime) / USEC_PER_SEC;
+ }
+}
+
+LLSD LLVoiceWebRTCStats::read()
+{
+ LLSD stats(LLSD::emptyMap());
+
+ stats["connect_cycles"] = LLSD::Integer(mConnectCycles);
+ stats["connect_attempts"] = LLSD::Integer(mConnectAttempts);
+ stats["connect_time"] = LLSD::Real(mConnectTime);
+
+ stats["provision_attempts"] = LLSD::Integer(mProvisionAttempts);
+ stats["provision_time"] = LLSD::Real(mProvisionTime);
+
+ stats["establish_attempts"] = LLSD::Integer(mEstablishAttempts);
+ stats["establish_time"] = LLSD::Real(mEstablishTime);
+
+ return stats;
+}
+
+///////////////////////////////////////////////////////////////////////////////////////////////
+
+bool LLWebRTCVoiceClient::sShuttingDown = false;
+
+LLWebRTCVoiceClient::LLWebRTCVoiceClient() :
+ mHidden(false),
+ mTuningMode(false),
+ mTuningMicGain(0.0),
+ mTuningSpeakerVolume(50), // Set to 50 so the user can hear themselves when he sets his mic volume
+ mDevicesListUpdated(false),
+
+ mSpatialCoordsDirty(false),
+
+ mMuteMic(false),
+
+ mEarLocation(0),
+ mMicGain(0.0),
+
+ mVoiceEnabled(false),
+ mProcessChannels(false),
+
+ mAvatarNameCacheConnection(),
+ mIsInTuningMode(false),
+ mIsProcessingChannels(false),
+ mIsCoroutineActive(false),
+ mWebRTCPump("WebRTCClientPump"),
+ mWebRTCDeviceInterface(nullptr)
+{
+ sShuttingDown = false;
+
+ mSpeakerVolume = 0.0;
+
+ mVoiceVersion.serverVersion = "";
+ mVoiceVersion.voiceServerType = REPORTED_VOICE_SERVER_TYPE;
+ mVoiceVersion.internalVoiceServerType = WEBRTC_VOICE_SERVER_TYPE;
+ mVoiceVersion.minorVersion = 0;
+ mVoiceVersion.majorVersion = 2;
+ mVoiceVersion.mBuildVersion = "";
+}
+
+//---------------------------------------------------
+
+LLWebRTCVoiceClient::~LLWebRTCVoiceClient()
+{
+ if (mAvatarNameCacheConnection.connected())
+ {
+ mAvatarNameCacheConnection.disconnect();
+ }
+ sShuttingDown = true;
+}
+
+//---------------------------------------------------
+
+void LLWebRTCVoiceClient::init(LLPumpIO* pump)
+{
+ // constructor will set up LLVoiceClient::getInstance()
+ llwebrtc::init();
+
+ mWebRTCDeviceInterface = llwebrtc::getDeviceInterface();
+ mWebRTCDeviceInterface->setDevicesObserver(this);
+}
+
+void LLWebRTCVoiceClient::terminate()
+{
+ if (sShuttingDown)
+ {
+ return;
+ }
+
+ mVoiceEnabled = false;
+ llwebrtc::terminate();
+
+ sShuttingDown = true;
+}
+
+//---------------------------------------------------
+
+void LLWebRTCVoiceClient::cleanUp()
+{
+ mNextSession.reset();
+ mSession.reset();
+ mNeighboringRegions.clear();
+ sessionState::for_each(boost::bind(predShutdownSession, _1));
+ LL_DEBUGS("Voice") << "Exiting" << LL_ENDL;
+}
+
+//---------------------------------------------------
+
+const LLVoiceVersionInfo& LLWebRTCVoiceClient::getVersion()
+{
+ return mVoiceVersion;
+}
+
+//---------------------------------------------------
+
+void LLWebRTCVoiceClient::updateSettings()
+{
+ setVoiceEnabled(LLVoiceClient::getInstance()->voiceEnabled());
+ setEarLocation(gSavedSettings.getS32("VoiceEarLocation"));
+
+ std::string inputDevice = gSavedSettings.getString("VoiceInputAudioDevice");
+ setCaptureDevice(inputDevice);
+ std::string outputDevice = gSavedSettings.getString("VoiceOutputAudioDevice");
+ setRenderDevice(outputDevice);
+ F32 mic_level = gSavedSettings.getF32("AudioLevelMic");
+ setMicGain(mic_level);
+}
+
+// Observers
+void LLWebRTCVoiceClient::addObserver(LLVoiceClientParticipantObserver *observer)
+{
+ mParticipantObservers.insert(observer);
+}
+
+void LLWebRTCVoiceClient::removeObserver(LLVoiceClientParticipantObserver *observer)
+{
+ mParticipantObservers.erase(observer);
+}
+
+void LLWebRTCVoiceClient::notifyParticipantObservers()
+{
+ for (observer_set_t::iterator it = mParticipantObservers.begin(); it != mParticipantObservers.end();)
+ {
+ LLVoiceClientParticipantObserver *observer = *it;
+ observer->onParticipantsChanged();
+ // In case onParticipantsChanged() deleted an entry.
+ it = mParticipantObservers.upper_bound(observer);
+ }
+}
+
+void LLWebRTCVoiceClient::addObserver(LLVoiceClientStatusObserver *observer)
+{
+ mStatusObservers.insert(observer);
+}
+
+void LLWebRTCVoiceClient::removeObserver(LLVoiceClientStatusObserver *observer)
+{
+ mStatusObservers.erase(observer);
+}
+
+void LLWebRTCVoiceClient::notifyStatusObservers(LLVoiceClientStatusObserver::EStatusType status)
+{
+ LL_DEBUGS("Voice") << "( " << LLVoiceClientStatusObserver::status2string(status) << " )"
+ << " mSession=" << mSession << LL_ENDL;
+
+ LL_DEBUGS("Voice") << " " << LLVoiceClientStatusObserver::status2string(status) << ", session channelInfo "
+ << getAudioSessionChannelInfo() << ", proximal is " << inSpatialChannel() << LL_ENDL;
+
+ mIsProcessingChannels = status == LLVoiceClientStatusObserver::STATUS_JOINED;
+
+ LLSD channelInfo = getAudioSessionChannelInfo();
+ for (status_observer_set_t::iterator it = mStatusObservers.begin(); it != mStatusObservers.end();)
+ {
+ LLVoiceClientStatusObserver *observer = *it;
+ observer->onChange(status, channelInfo, inSpatialChannel());
+ // In case onError() deleted an entry.
+ it = mStatusObservers.upper_bound(observer);
+ }
+
+ // skipped to avoid speak button blinking
+ if (status != LLVoiceClientStatusObserver::STATUS_JOINING &&
+ status != LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL &&
+ status != LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED)
+ {
+ bool voice_status = LLVoiceClient::getInstance()->voiceEnabled() && LLVoiceClient::getInstance()->isVoiceWorking();
+
+ gAgent.setVoiceConnected(voice_status);
+
+ if (voice_status)
+ {
+ LLFirstUse::speak(true);
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::addObserver(LLFriendObserver *observer)
+{
+}
+
+void LLWebRTCVoiceClient::removeObserver(LLFriendObserver *observer)
+{
+}
+
+//---------------------------------------------------
+// Primary voice loop.
+// This voice loop is called every 100ms plus the time it
+// takes to process the various functions called in the loop
+// The loop does the following:
+// * gates whether we do channel processing depending on
+// whether we're running a WebRTC voice channel or
+// one from another voice provider.
+// * If in spatial voice, it determines whether we've changed
+// parcels, whether region/parcel voice settings have changed,
+// etc. and manages whether the voice channel needs to change.
+// * calls the state machines for the sessions to negotiate
+// connection to various voice channels.
+// * Sends updates to the voice server when this agent's
+// voice levels, or positions have changed.
+void LLWebRTCVoiceClient::voiceConnectionCoro()
+{
+ LL_DEBUGS("Voice") << "starting" << LL_ENDL;
+ mIsCoroutineActive = true;
+ LLCoros::set_consuming(true);
+ try
+ {
+ LLMuteList::getInstance()->addObserver(this);
+ while (!sShuttingDown)
+ {
+ // TODO: Doing some measurement and calculation here,
+ // we could reduce the timeout to take into account the
+ // time spent on the previous loop to have the loop
+ // cycle at exactly 100ms, instead of 100ms + loop
+ // execution time.
+ // Could help with voice updates making for smoother
+ // voice when we're busy.
+ llcoro::suspendUntilTimeout(UPDATE_THROTTLE_SECONDS);
+ bool voiceEnabled = mVoiceEnabled;
+
+ if (!isAgentAvatarValid())
+ {
+ continue;
+ }
+
+ LLViewerRegion *regionp = gAgent.getRegion();
+ if (!regionp)
+ {
+ continue;
+ }
+
+ if (!mProcessChannels)
+ {
+ // we've switched away from webrtc voice, so shut all channels down.
+ // leave channel can be called again and again without adverse effects.
+ // it merely tells channels to shut down if they're not already doing so.
+ leaveChannel(false);
+ }
+ else if (inSpatialChannel())
+ {
+ bool useEstateVoice = true;
+ // add session for region or parcel voice.
+ if (!regionp || regionp->getRegionID().isNull())
+ {
+ // no region, no voice.
+ continue;
+ }
+
+ voiceEnabled = voiceEnabled && regionp->isVoiceEnabled();
+
+ if (voiceEnabled)
+ {
+ LLParcel *parcel = LLViewerParcelMgr::getInstance()->getAgentParcel();
+ // check to see if parcel changed.
+ if (parcel && parcel->getLocalID() != INVALID_PARCEL_ID)
+ {
+ // parcel voice
+ if (!parcel->getParcelFlagAllowVoice())
+ {
+ voiceEnabled = false;
+ }
+ else if (!parcel->getParcelFlagUseEstateVoiceChannel())
+ {
+ // use the parcel-specific voice channel.
+ S32 parcel_local_id = parcel->getLocalID();
+ std::string channelID = regionp->getRegionID().asString() + "-" + std::to_string(parcel->getLocalID());
+
+ useEstateVoice = false;
+ if (!inOrJoiningChannel(channelID))
+ {
+ startParcelSession(channelID, parcel_local_id);
+ }
+ }
+ }
+ if (useEstateVoice && !inEstateChannel())
+ {
+ // estate voice
+ startEstateSession();
+ }
+ }
+ if (!voiceEnabled)
+ {
+ // voice is disabled, so leave and disable PTT
+ leaveChannel(true);
+ }
+ else
+ {
+ // we're in spatial voice, and voice is enabled, so determine positions in order
+ // to send position updates.
+ updatePosition();
+ }
+ }
+
+ sessionState::processSessionStates();
+ if (mProcessChannels && voiceEnabled && !mHidden)
+ {
+ sendPositionUpdate(true);
+ updateOwnVolume();
+ }
+ }
+ }
+ catch (const LLCoros::Stop&)
+ {
+ LL_DEBUGS("LLWebRTCVoiceClient") << "Received a shutdown exception" << LL_ENDL;
+ }
+ catch (const LLContinueError&)
+ {
+ LOG_UNHANDLED_EXCEPTION("LLWebRTCVoiceClient");
+ }
+ catch (...)
+ {
+ // Ideally for Windows need to log SEH exception instead or to set SEH
+ // handlers but bugsplat shows local variables for windows, which should
+ // be enough
+ LL_WARNS("Voice") << "voiceConnectionStateMachine crashed" << LL_ENDL;
+ throw;
+ }
+
+ cleanUp();
+}
+
+// For spatial, determine which neighboring regions to connect to
+// for cross-region voice.
+void LLWebRTCVoiceClient::updateNeighboringRegions()
+{
+ static const std::vector<LLVector3d> neighbors {LLVector3d(0.0f, 1.0f, 0.0f), LLVector3d(0.707f, 0.707f, 0.0f),
+ LLVector3d(1.0f, 0.0f, 0.0f), LLVector3d(0.707f, -0.707f, 0.0f),
+ LLVector3d(0.0f, -1.0f, 0.0f), LLVector3d(-0.707f, -0.707f, 0.0f),
+ LLVector3d(-1.0f, 0.0f, 0.0f), LLVector3d(-0.707f, 0.707f, 0.0f)};
+
+ // Estate voice requires connection to neighboring regions.
+ mNeighboringRegions.clear();
+
+ mNeighboringRegions.insert(gAgent.getRegion()->getRegionID());
+
+ // base off of speaker position as it'll move more slowly than camera position.
+ // Once we have hysteresis, we may be able to track off of speaker and camera position at 50m
+ // TODO: Add hysteresis so we don't flip-flop connections to neighbors
+ LLVector3d speaker_pos = LLWebRTCVoiceClient::getInstance()->getSpeakerPosition();
+ for (auto &neighbor_pos : neighbors)
+ {
+ // include every region within 100m (2*MAX_AUDIO_DIST) to deal witht he fact that the camera
+ // can stray 50m away from the avatar.
+ LLViewerRegion *neighbor = LLWorld::instance().getRegionFromPosGlobal(speaker_pos + 2 * MAX_AUDIO_DIST * neighbor_pos);
+ if (neighbor && !neighbor->getRegionID().isNull())
+ {
+ mNeighboringRegions.insert(neighbor->getRegionID());
+ }
+ }
+}
+
+//=========================================================================
+// shut down the current audio session to make room for the next one.
+void LLWebRTCVoiceClient::leaveAudioSession()
+{
+ if(mSession)
+ {
+ LL_DEBUGS("Voice") << "leaving session: " << mSession->mChannelID << LL_ENDL;
+ mSession->shutdownAllConnections();
+ }
+ else
+ {
+ LL_WARNS("Voice") << "called with no active session" << LL_ENDL;
+ }
+}
+
+//=========================================================================
+// Device Management
+void LLWebRTCVoiceClient::clearCaptureDevices()
+{
+ LL_DEBUGS("Voice") << "called" << LL_ENDL;
+ mCaptureDevices.clear();
+}
+
+void LLWebRTCVoiceClient::addCaptureDevice(const LLVoiceDevice& device)
+{
+ LL_DEBUGS("Voice") << "display: '" << device.display_name << "' device: '" << device.full_name << "'" << LL_ENDL;
+ mCaptureDevices.push_back(device);
+}
+
+LLVoiceDeviceList& LLWebRTCVoiceClient::getCaptureDevices()
+{
+ return mCaptureDevices;
+}
+
+void LLWebRTCVoiceClient::setCaptureDevice(const std::string& name)
+{
+ mWebRTCDeviceInterface->setCaptureDevice(name);
+}
+void LLWebRTCVoiceClient::setDevicesListUpdated(bool state)
+{
+ mDevicesListUpdated = state;
+}
+
+void LLWebRTCVoiceClient::OnDevicesChanged(const llwebrtc::LLWebRTCVoiceDeviceList &render_devices,
+ const llwebrtc::LLWebRTCVoiceDeviceList &capture_devices)
+{
+ std::string inputDevice = gSavedSettings.getString("VoiceInputAudioDevice");
+ std::string outputDevice = gSavedSettings.getString("VoiceOutputAudioDevice");
+
+ clearRenderDevices();
+ bool renderDeviceSet = false;
+ for (auto &device : render_devices)
+ {
+ addRenderDevice(LLVoiceDevice(device.mDisplayName, device.mID));
+ if (device.mCurrent && outputDevice == device.mID)
+ {
+ setRenderDevice(outputDevice);
+ renderDeviceSet = true;
+ }
+ }
+ if (!renderDeviceSet)
+ {
+ setRenderDevice("Default");
+ }
+
+ clearCaptureDevices();
+ bool captureDeviceSet = false;
+ for (auto &device : capture_devices)
+ {
+ addCaptureDevice(LLVoiceDevice(device.mDisplayName, device.mID));
+ if (device.mCurrent && inputDevice == device.mID)
+ {
+ setCaptureDevice(outputDevice);
+ captureDeviceSet = true;
+ }
+ }
+ if (!captureDeviceSet)
+ {
+ setCaptureDevice("Default");
+ }
+
+ setDevicesListUpdated(true);
+}
+
+void LLWebRTCVoiceClient::clearRenderDevices()
+{
+ LL_DEBUGS("Voice") << "called" << LL_ENDL;
+ mRenderDevices.clear();
+}
+
+void LLWebRTCVoiceClient::addRenderDevice(const LLVoiceDevice& device)
+{
+ LL_DEBUGS("Voice") << "display: '" << device.display_name << "' device: '" << device.full_name << "'" << LL_ENDL;
+ mRenderDevices.push_back(device);
+
+}
+
+LLVoiceDeviceList& LLWebRTCVoiceClient::getRenderDevices()
+{
+ return mRenderDevices;
+}
+
+void LLWebRTCVoiceClient::setRenderDevice(const std::string& name)
+{
+ mWebRTCDeviceInterface->setRenderDevice(name);
+}
+
+void LLWebRTCVoiceClient::tuningStart()
+{
+ if (!mIsInTuningMode)
+ {
+ mWebRTCDeviceInterface->setTuningMode(true);
+ mIsInTuningMode = true;
+ }
+}
+
+void LLWebRTCVoiceClient::tuningStop()
+{
+ if (mIsInTuningMode)
+ {
+ mWebRTCDeviceInterface->setTuningMode(false);
+ mIsInTuningMode = false;
+ }
+}
+
+bool LLWebRTCVoiceClient::inTuningMode()
+{
+ return mIsInTuningMode;
+}
+
+void LLWebRTCVoiceClient::tuningSetMicVolume(float volume)
+{
+ mTuningMicGain = volume;
+}
+
+void LLWebRTCVoiceClient::tuningSetSpeakerVolume(float volume)
+{
+
+ if (volume != mTuningSpeakerVolume)
+ {
+ mTuningSpeakerVolume = volume;
+ }
+}
+
+float LLWebRTCVoiceClient::getAudioLevel()
+{
+ if (mIsInTuningMode)
+ {
+ return (1.0 - mWebRTCDeviceInterface->getTuningAudioLevel() * LEVEL_SCALE_WEBRTC) * mTuningMicGain / 2.1;
+ }
+ else
+ {
+ return (1.0 - mWebRTCDeviceInterface->getPeerConnectionAudioLevel() * LEVEL_SCALE_WEBRTC) * mMicGain / 2.1;
+ }
+}
+
+float LLWebRTCVoiceClient::tuningGetEnergy(void)
+{
+ return getAudioLevel();
+}
+
+bool LLWebRTCVoiceClient::deviceSettingsAvailable()
+{
+ bool result = true;
+
+ if(mRenderDevices.empty() || mCaptureDevices.empty())
+ result = false;
+
+ return result;
+}
+bool LLWebRTCVoiceClient::deviceSettingsUpdated()
+{
+ bool updated = mDevicesListUpdated;
+ mDevicesListUpdated = false;
+ return updated;
+}
+
+void LLWebRTCVoiceClient::refreshDeviceLists(bool clearCurrentList)
+{
+ if(clearCurrentList)
+ {
+ clearCaptureDevices();
+ clearRenderDevices();
+ }
+ mWebRTCDeviceInterface->refreshDevices();
+}
+
+
+void LLWebRTCVoiceClient::setHidden(bool hidden)
+{
+ mHidden = hidden;
+
+ if (inSpatialChannel())
+ {
+ if (mHidden)
+ {
+ // get out of the channel entirely
+ // mute the microphone.
+ sessionState::for_each(boost::bind(predSetMuteMic, _1, true));
+ }
+ else
+ {
+ // and put it back
+ sessionState::for_each(boost::bind(predSetMuteMic, _1, mMuteMic));
+ updatePosition();
+ sendPositionUpdate(true);
+ }
+ }
+}
+
+/////////////////////////////
+// session control messages.
+//
+// these are called by the sessions to report
+// status for a given channel. By filtering
+// on channel and region, these functions
+// can send various notifications to
+// other parts of the viewer, as well as
+// managing housekeeping
+
+// A connection to a channel was successfully established,
+// so shut down the current session and move on to the next
+// if one is available.
+// if the current session is the one that was established,
+// notify the observers.
+void LLWebRTCVoiceClient::OnConnectionEstablished(const std::string &channelID, const LLUUID &regionID)
+{
+ if (gAgent.getRegion()->getRegionID() == regionID)
+ {
+ if (mNextSession && mNextSession->mChannelID == channelID)
+ {
+ if (mSession)
+ {
+ mSession->shutdownAllConnections();
+ }
+ mSession = mNextSession;
+ mNextSession.reset();
+
+ // Add ourselves as a participant.
+ mSession->addParticipant(gAgentID);
+ }
+
+ // The current session was established.
+ if (mSession && mSession->mChannelID == channelID)
+ {
+ LLWebRTCVoiceClient::getInstance()->notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LOGGED_IN);
+
+ // only set status to joined if asked to. This will happen in the case where we're not
+ // doing an ad-hoc based p2p session. Those sessions expect a STATUS_JOINED when the peer
+ // has, in fact, joined, which we detect elsewhere.
+ if (!mSession->mNotifyOnFirstJoin)
+ {
+ LLWebRTCVoiceClient::getInstance()->notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_JOINED);
+ }
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::OnConnectionShutDown(const std::string &channelID, const LLUUID &regionID)
+{
+ if (gAgent.getRegion()->getRegionID() == regionID)
+ {
+ if (mSession && mSession->mChannelID == channelID)
+ {
+ LL_DEBUGS("Voice") << "Main WebRTC Connection Shut Down." << LL_ENDL;
+ }
+ }
+}
+void LLWebRTCVoiceClient::OnConnectionFailure(const std::string &channelID, const LLUUID &regionID)
+{
+ LL_DEBUGS("Voice") << "A connection failed. channel:" << channelID << LL_ENDL;
+ if (gAgent.getRegion()->getRegionID() == regionID)
+ {
+ if (mNextSession && mNextSession->mChannelID == channelID)
+ {
+ LLWebRTCVoiceClient::getInstance()->notifyStatusObservers(LLVoiceClientStatusObserver::ERROR_UNKNOWN);
+ }
+ else if (mSession && mSession->mChannelID == channelID)
+ {
+ LLWebRTCVoiceClient::getInstance()->notifyStatusObservers(LLVoiceClientStatusObserver::ERROR_UNKNOWN);
+ }
+ }
+}
+
+// -----------------------------------------------------------
+// positional functionality.
+void LLWebRTCVoiceClient::setEarLocation(S32 loc)
+{
+ if (mEarLocation != loc)
+ {
+ LL_DEBUGS("Voice") << "Setting mEarLocation to " << loc << LL_ENDL;
+
+ mEarLocation = loc;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+void LLWebRTCVoiceClient::updatePosition(void)
+{
+ LLViewerRegion *region = gAgent.getRegion();
+ if (region && isAgentAvatarValid())
+ {
+ // get the avatar position.
+ LLVector3d avatar_pos = gAgentAvatarp->getPositionGlobal();
+ LLQuaternion avatar_qrot = gAgentAvatarp->getRootJoint()->getWorldRotation();
+
+ avatar_pos += LLVector3d(0.f, 0.f, 1.f); // bump it up to head height
+
+ LLVector3d earPosition;
+ LLQuaternion earRot;
+ switch (mEarLocation)
+ {
+ case earLocCamera:
+ default:
+ earPosition = region->getPosGlobalFromRegion(LLViewerCamera::getInstance()->getOrigin());
+ earRot = LLViewerCamera::getInstance()->getQuaternion();
+ break;
+
+ case earLocAvatar:
+ earPosition = mAvatarPosition;
+ earRot = mAvatarRot;
+ break;
+
+ case earLocMixed:
+ earPosition = mAvatarPosition;
+ earRot = LLViewerCamera::getInstance()->getQuaternion();
+ break;
+ }
+ setListenerPosition(earPosition, // position
+ LLVector3::zero, // velocity
+ earRot); // rotation matrix
+
+ setAvatarPosition(avatar_pos, // position
+ LLVector3::zero, // velocity
+ avatar_qrot); // rotation matrix
+
+ enforceTether();
+
+ updateNeighboringRegions();
+ }
+}
+
+void LLWebRTCVoiceClient::setListenerPosition(const LLVector3d &position, const LLVector3 &velocity, const LLQuaternion &rot)
+{
+ mListenerRequestedPosition = position;
+
+ if (mListenerVelocity != velocity)
+ {
+ mListenerVelocity = velocity;
+ mSpatialCoordsDirty = true;
+ }
+
+ if (mListenerRot != rot)
+ {
+ mListenerRot = rot;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+void LLWebRTCVoiceClient::setAvatarPosition(const LLVector3d &position, const LLVector3 &velocity, const LLQuaternion &rot)
+{
+ if (dist_vec_squared(mAvatarPosition, position) > 0.01)
+ {
+ mAvatarPosition = position;
+ mSpatialCoordsDirty = true;
+ }
+
+ if (mAvatarVelocity != velocity)
+ {
+ mAvatarVelocity = velocity;
+ mSpatialCoordsDirty = true;
+ }
+
+ // If the two rotations are not exactly equal test their dot product
+ // to get the cos of the angle between them.
+ // If it is too small, don't update.
+ F32 rot_cos_diff = llabs(dot(mAvatarRot, rot));
+ if ((mAvatarRot != rot) && (rot_cos_diff < MINUSCULE_ANGLE_COS))
+ {
+ mAvatarRot = rot;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+// The listener (camera) must be within 50m of the
+// avatar. Enforce it on the client.
+// This will also be enforced on the voice server
+// based on position sent from the simulator to the
+// voice server.
+void LLWebRTCVoiceClient::enforceTether()
+{
+ LLVector3d tethered = mListenerRequestedPosition;
+
+ // constrain 'tethered' to within 50m of mAvatarPosition.
+ {
+ LLVector3d camera_offset = mListenerRequestedPosition - mAvatarPosition;
+ F32 camera_distance = (F32) camera_offset.magVec();
+ if (camera_distance > MAX_AUDIO_DIST)
+ {
+ tethered = mAvatarPosition + (MAX_AUDIO_DIST / camera_distance) * camera_offset;
+ }
+ }
+
+ if (dist_vec_squared(mListenerPosition, tethered) > 0.01)
+ {
+ mListenerPosition = tethered;
+ mSpatialCoordsDirty = true;
+ }
+}
+
+// We send our position via a WebRTC data channel to the WebRTC
+// server for fine-grained, low latency updates. On the server,
+// these updates will be 'tethered' to the actual position of the avatar.
+// Those updates are higher latency, however.
+// This mechanism gives low latency spatial updates and server-enforced
+// prevention of 'evesdropping' by sending camera updates beyond the
+// standard 50m
+void LLWebRTCVoiceClient::sendPositionUpdate(bool force)
+{
+ Json::FastWriter writer;
+ std::string spatial_data;
+
+ if (mSpatialCoordsDirty || force)
+ {
+ Json::Value spatial = Json::objectValue;
+
+ spatial["sp"] = Json::objectValue;
+ spatial["sp"]["x"] = (int) (mAvatarPosition[0] * 100);
+ spatial["sp"]["y"] = (int) (mAvatarPosition[1] * 100);
+ spatial["sp"]["z"] = (int) (mAvatarPosition[2] * 100);
+ spatial["sh"] = Json::objectValue;
+ spatial["sh"]["x"] = (int) (mAvatarRot[0] * 100);
+ spatial["sh"]["y"] = (int) (mAvatarRot[1] * 100);
+ spatial["sh"]["z"] = (int) (mAvatarRot[2] * 100);
+ spatial["sh"]["w"] = (int) (mAvatarRot[3] * 100);
+
+ spatial["lp"] = Json::objectValue;
+ spatial["lp"]["x"] = (int) (mListenerPosition[0] * 100);
+ spatial["lp"]["y"] = (int) (mListenerPosition[1] * 100);
+ spatial["lp"]["z"] = (int) (mListenerPosition[2] * 100);
+ spatial["lh"] = Json::objectValue;
+ spatial["lh"]["x"] = (int) (mListenerRot[0] * 100);
+ spatial["lh"]["y"] = (int) (mListenerRot[1] * 100);
+ spatial["lh"]["z"] = (int) (mListenerRot[2] * 100);
+ spatial["lh"]["w"] = (int) (mListenerRot[3] * 100);
+
+ mSpatialCoordsDirty = false;
+ spatial_data = writer.write(spatial);
+
+ sessionState::for_each(boost::bind(predSendData, _1, spatial_data));
+ }
+}
+
+// Update our own volume on our participant, so it'll show up
+// in the UI. This is done on all sessions, so switching
+// sessions retains consistent volume levels.
+void LLWebRTCVoiceClient::updateOwnVolume() {
+ F32 audio_level = 0.0;
+ if (!mMuteMic && !mTuningMode)
+ {
+ audio_level = getAudioLevel();
+ }
+
+ sessionState::for_each(boost::bind(predUpdateOwnVolume, _1, audio_level));
+}
+
+////////////////////////////////////
+// Managing list of participants
+
+// Provider-level participant management
+
+BOOL LLWebRTCVoiceClient::isParticipantAvatar(const LLUUID &id)
+{
+ // WebRTC participants are always SL avatars.
+ return TRUE;
+}
+
+void LLWebRTCVoiceClient::getParticipantList(std::set<LLUUID> &participants)
+{
+ if (mProcessChannels && mSession)
+ {
+ for (participantUUIDMap::iterator iter = mSession->mParticipantsByUUID.begin();
+ iter != mSession->mParticipantsByUUID.end();
+ iter++)
+ {
+ participants.insert(iter->first);
+ }
+ }
+}
+
+bool LLWebRTCVoiceClient::isParticipant(const LLUUID &speaker_id)
+{
+ if (mProcessChannels && mSession)
+ {
+ return (mSession->mParticipantsByUUID.find(speaker_id) != mSession->mParticipantsByUUID.end());
+ }
+ return false;
+}
+
+// protected provider-level participant management.
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::findParticipantByID(const std::string &channelID, const LLUUID &id)
+{
+ participantStatePtr_t result;
+ LLWebRTCVoiceClient::sessionState::ptr_t session = sessionState::matchSessionByChannelID(channelID);
+
+ if (session)
+ {
+ result = session->findParticipantByID(id);
+ }
+
+ return result;
+}
+
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::addParticipantByID(const std::string &channelID, const LLUUID &id)
+{
+ participantStatePtr_t result;
+ LLWebRTCVoiceClient::sessionState::ptr_t session = sessionState::matchSessionByChannelID(channelID);
+ if (session)
+ {
+ result = session->addParticipant(id);
+ if (session->mNotifyOnFirstJoin && (id != gAgentID))
+ {
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_JOINED);
+ }
+ }
+ return result;
+}
+
+void LLWebRTCVoiceClient::removeParticipantByID(const std::string &channelID, const LLUUID &id)
+{
+ participantStatePtr_t result;
+ LLWebRTCVoiceClient::sessionState::ptr_t session = sessionState::matchSessionByChannelID(channelID);
+ if (session)
+ {
+ participantStatePtr_t participant = session->findParticipantByID(id);
+ if (participant)
+ {
+ session->removeParticipant(participant);
+ if (session->mHangupOnLastLeave && (id != gAgentID) && (session->mParticipantsByUUID.size() <= 1))
+ {
+ notifyStatusObservers(LLVoiceClientStatusObserver::STATUS_LEFT_CHANNEL);
+ }
+ }
+ }
+}
+
+
+// participantState level participant management
+LLWebRTCVoiceClient::participantState::participantState(const LLUUID& agent_id) :
+ mURI(agent_id.asString()),
+ mAvatarID(agent_id),
+ mIsSpeaking(false),
+ mIsModeratorMuted(false),
+ mLevel(0.f),
+ mVolume(LLVoiceClient::VOLUME_DEFAULT)
+{
+}
+
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::sessionState::addParticipant(const LLUUID& agent_id)
+{
+ participantStatePtr_t result;
+
+ participantUUIDMap::iterator iter = mParticipantsByUUID.find(agent_id);
+
+ if (iter != mParticipantsByUUID.end())
+ {
+ result = iter->second;
+ }
+
+ if(!result)
+ {
+ // participant isn't already in one list or the other.
+ result.reset(new participantState(agent_id));
+ mParticipantsByUUID.insert(participantUUIDMap::value_type(agent_id, result));
+ result->mAvatarID = agent_id;
+
+ LLWebRTCVoiceClient::getInstance()->lookupName(agent_id);
+
+ LLSpeakerVolumeStorage::getInstance()->getSpeakerVolume(result->mAvatarID, result->mVolume);
+ if (!LLWebRTCVoiceClient::sShuttingDown)
+ {
+ LLWebRTCVoiceClient::getInstance()->notifyParticipantObservers();
+ }
+
+ LL_DEBUGS("Voice") << "Participant \"" << result->mURI << "\" added." << LL_ENDL;
+ }
+
+ return result;
+}
+
+
+// session-level participant management
+
+LLWebRTCVoiceClient::participantStatePtr_t LLWebRTCVoiceClient::sessionState::findParticipantByID(const LLUUID& id)
+{
+ participantStatePtr_t result;
+ participantUUIDMap::iterator iter = mParticipantsByUUID.find(id);
+
+ if(iter != mParticipantsByUUID.end())
+ {
+ result = iter->second;
+ }
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::sessionState::removeParticipant(const LLWebRTCVoiceClient::participantStatePtr_t &participant)
+{
+ if (participant)
+ {
+ participantUUIDMap::iterator iter = mParticipantsByUUID.find(participant->mAvatarID);
+
+ LL_DEBUGS("Voice") << "participant \"" << participant->mURI << "\" (" << participant->mAvatarID << ") removed." << LL_ENDL;
+
+ if (iter == mParticipantsByUUID.end())
+ {
+ LL_WARNS("Voice") << "Internal error: participant ID " << participant->mAvatarID << " not in UUID map" << LL_ENDL;
+ }
+ else
+ {
+ mParticipantsByUUID.erase(iter);
+ if (!LLWebRTCVoiceClient::sShuttingDown)
+ {
+ LLWebRTCVoiceClient::getInstance()->notifyParticipantObservers();
+ }
+ }
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::removeAllParticipants()
+{
+ LL_DEBUGS("Voice") << "called" << LL_ENDL;
+
+ while (!mParticipantsByUUID.empty())
+ {
+ removeParticipant(mParticipantsByUUID.begin()->second);
+ }
+}
+
+// Initiated the various types of sessions.
+bool LLWebRTCVoiceClient::startEstateSession()
+{
+ leaveChannel(false);
+ mNextSession = addSession("Estate", sessionState::ptr_t(new estateSessionState()));
+ return true;
+}
+
+bool LLWebRTCVoiceClient::startParcelSession(const std::string &channelID, S32 parcelID)
+{
+ leaveChannel(false);
+ mNextSession = addSession(channelID, sessionState::ptr_t(new parcelSessionState(channelID, parcelID)));
+ return true;
+}
+
+bool LLWebRTCVoiceClient::startAdHocSession(const LLSD& channelInfo, bool notify_on_first_join, bool hangup_on_last_leave)
+{
+ leaveChannel(false);
+ LL_WARNS("Voice") << "Start AdHoc Session " << channelInfo << LL_ENDL;
+ std::string channelID = channelInfo["channel_uri"];
+ std::string credentials = channelInfo["channel_credentials"];
+ mNextSession = addSession(channelID,
+ sessionState::ptr_t(new adhocSessionState(channelID,
+ credentials,
+ notify_on_first_join,
+ hangup_on_last_leave)));
+ return true;
+}
+
+bool LLWebRTCVoiceClient::isVoiceWorking() const
+{
+ return mIsProcessingChannels;
+}
+
+// Returns true if calling back the session URI after the session has closed is possible.
+// Currently this will be false only for PSTN P2P calls.
+BOOL LLWebRTCVoiceClient::isSessionCallBackPossible(const LLUUID &session_id)
+{
+ sessionStatePtr_t session(findP2PSession(session_id));
+ return session && session->isCallbackPossible() ? TRUE : FALSE;
+}
+
+// Channel Management
+void LLWebRTCVoiceClient::leaveNonSpatialChannel()
+{
+ LL_DEBUGS("Voice") << "Request to leave non-spatial channel." << LL_ENDL;
+
+ // make sure we're not simply rejoining the current session
+ deleteSession(mNextSession);
+
+ leaveChannel(true);
+}
+
+// determine whether we're processing channels, or whether
+// another voice provider is.
+void LLWebRTCVoiceClient::processChannels(bool process)
+{
+ mProcessChannels = process;
+}
+
+bool LLWebRTCVoiceClient::inProximalChannel()
+{
+ return inSpatialChannel();
+}
+
+bool LLWebRTCVoiceClient::inOrJoiningChannel(const std::string& channelID)
+{
+ return (mSession && mSession->mChannelID == channelID) || (mNextSession && mNextSession->mChannelID == channelID);
+}
+
+bool LLWebRTCVoiceClient::inEstateChannel()
+{
+ return (mSession && mSession->isEstate()) || (mNextSession && mNextSession->isEstate());
+}
+
+bool LLWebRTCVoiceClient::inSpatialChannel()
+{
+ bool result = true;
+
+ if (mNextSession)
+ {
+ result = mNextSession->isSpatial();
+ }
+ else if(mSession)
+ {
+ result = mSession->isSpatial();
+ }
+
+ return result;
+}
+
+// retrieves information used to negotiate p2p, adhoc, and group
+// channels
+LLSD LLWebRTCVoiceClient::getAudioSessionChannelInfo()
+{
+ LLSD result;
+
+ if (mSession)
+ {
+ result["voice_server_type"] = WEBRTC_VOICE_SERVER_TYPE;
+ result["channel_uri"] = mSession->mChannelID;
+ }
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::leaveChannel(bool stopTalking)
+{
+ if (mSession)
+ {
+ deleteSession(mSession);
+ }
+
+ if (mNextSession)
+ {
+ deleteSession(mNextSession);
+ }
+
+ // If voice was on, turn it off
+ if (stopTalking && LLVoiceClient::getInstance()->getUserPTTState())
+ {
+ LLVoiceClient::getInstance()->setUserPTTState(false);
+ }
+}
+
+bool LLWebRTCVoiceClient::isCurrentChannel(const LLSD &channelInfo)
+{
+ if (!mProcessChannels || (channelInfo["voice_server_type"].asString() != WEBRTC_VOICE_SERVER_TYPE))
+ {
+ return false;
+ }
+
+ if (mSession)
+ {
+ if (!channelInfo["sessionHandle"].asString().empty())
+ {
+ return mSession->mHandle == channelInfo["session_handle"].asString();
+ }
+ return channelInfo["channel_uri"].asString() == mSession->mChannelID;
+ }
+ return false;
+}
+
+bool LLWebRTCVoiceClient::compareChannels(const LLSD &channelInfo1, const LLSD &channelInfo2)
+{
+ return (channelInfo1["voice_server_type"] == WEBRTC_VOICE_SERVER_TYPE) &&
+ (channelInfo1["voice_server_type"] == channelInfo2["voice_server_type"]) &&
+ (channelInfo1["sip_uri"] == channelInfo2["sip_uri"]);
+}
+
+
+//----------------------------------------------
+// Audio muting, volume, gain, etc.
+
+// we're muting the mic, so tell each session such
+void LLWebRTCVoiceClient::setMuteMic(bool muted)
+{
+ mMuteMic = muted;
+ // when you're hidden, your mic is always muted.
+ if (!mHidden)
+ {
+ sessionState::for_each(boost::bind(predSetMuteMic, _1, muted));
+ }
+}
+
+void LLWebRTCVoiceClient::predSetMuteMic(const LLWebRTCVoiceClient::sessionStatePtr_t &session, bool muted)
+{
+ participantStatePtr_t participant = session->findParticipantByID(gAgentID);
+ if (participant)
+ {
+ participant->mLevel = 0.0;
+ }
+ session->setMuteMic(muted);
+}
+
+void LLWebRTCVoiceClient::setVoiceVolume(F32 volume)
+{
+ if (volume != mSpeakerVolume)
+ {
+ {
+ mSpeakerVolume = volume;
+ }
+ sessionState::for_each(boost::bind(predSetSpeakerVolume, _1, volume));
+ }
+}
+
+void LLWebRTCVoiceClient::predSetSpeakerVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 volume)
+{
+ session->setSpeakerVolume(volume);
+}
+
+void LLWebRTCVoiceClient::setMicGain(F32 gain)
+{
+ if (gain != mMicGain)
+ {
+ mMicGain = gain;
+ sessionState::for_each(boost::bind(predSetMicGain, _1, gain));
+ }
+}
+
+void LLWebRTCVoiceClient::predSetMicGain(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 gain)
+{
+ session->setMicGain(gain);
+}
+
+void LLWebRTCVoiceClient::setVoiceEnabled(bool enabled)
+{
+ LL_DEBUGS("Voice")
+ << "( " << (enabled ? "enabled" : "disabled") << " )"
+ << " was "<< (mVoiceEnabled ? "enabled" : "disabled")
+ << " coro "<< (mIsCoroutineActive ? "active" : "inactive")
+ << LL_ENDL;
+
+ if (enabled != mVoiceEnabled)
+ {
+ // TODO: Refactor this so we don't call into LLVoiceChannel, but simply
+ // use the status observer
+ mVoiceEnabled = enabled;
+ LLVoiceClientStatusObserver::EStatusType status;
+
+ if (enabled)
+ {
+ LL_DEBUGS("Voice") << "enabling" << LL_ENDL;
+ LLVoiceChannel::getCurrentVoiceChannel()->activate();
+ status = LLVoiceClientStatusObserver::STATUS_VOICE_ENABLED;
+ mSpatialCoordsDirty = true;
+ updatePosition();
+ if (!mIsCoroutineActive)
+ {
+ LLCoros::instance().launch("LLWebRTCVoiceClient::voiceConnectionCoro",
+ boost::bind(&LLWebRTCVoiceClient::voiceConnectionCoro, LLWebRTCVoiceClient::getInstance()));
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << "coro should be active.. not launching" << LL_ENDL;
+ }
+ }
+ else
+ {
+ // Turning voice off looses your current channel -- this makes sure the UI isn't out of sync when you re-enable it.
+ LLVoiceChannel::getCurrentVoiceChannel()->deactivate();
+ gAgent.setVoiceConnected(false);
+ status = LLVoiceClientStatusObserver::STATUS_VOICE_DISABLED;
+ cleanUp();
+ }
+
+ notifyStatusObservers(status);
+ }
+ else
+ {
+ LL_DEBUGS("Voice") << " no-op" << LL_ENDL;
+ }
+}
+
+
+/////////////////////////////
+// Accessors for data related to nearby speakers
+
+std::string LLWebRTCVoiceClient::getDisplayName(const LLUUID& id)
+{
+ std::string result;
+ if (mProcessChannels && mSession)
+ {
+ participantStatePtr_t participant(mSession->findParticipantByID(id));
+ if (participant)
+ {
+ result = participant->mDisplayName;
+ }
+ }
+ return result;
+}
+
+BOOL LLWebRTCVoiceClient::getIsSpeaking(const LLUUID& id)
+{
+ BOOL result = FALSE;
+ if (mProcessChannels && mSession)
+ {
+ participantStatePtr_t participant(mSession->findParticipantByID(id));
+ if (participant)
+ {
+ result = participant->mIsSpeaking;
+ }
+ }
+ return result;
+}
+
+// TODO: Need to pull muted status from the webrtc server
+BOOL LLWebRTCVoiceClient::getIsModeratorMuted(const LLUUID& id)
+{
+ BOOL result = FALSE;
+ if (mProcessChannels && mSession)
+ {
+ participantStatePtr_t participant(mSession->findParticipantByID(id));
+ if (participant)
+ {
+ result = participant->mIsModeratorMuted;
+ }
+ }
+ return result;
+}
+
+F32 LLWebRTCVoiceClient::getCurrentPower(const LLUUID &id)
+{
+ F32 result = 0.0;
+ if (!mProcessChannels || !mSession)
+ {
+ return result;
+ }
+ participantStatePtr_t participant(mSession->findParticipantByID(id));
+ if (participant)
+ {
+ if (participant->mIsSpeaking)
+ {
+ result = participant->mLevel;
+ }
+ }
+ return result;
+}
+
+// External accessors.
+F32 LLWebRTCVoiceClient::getUserVolume(const LLUUID& id)
+{
+ // Minimum volume will be returned for users with voice disabled
+ F32 result = LLVoiceClient::VOLUME_MIN;
+
+ participantStatePtr_t participant(mSession->findParticipantByID(id));
+ if(participant)
+ {
+ result = participant->mVolume;
+ }
+
+ return result;
+}
+
+void LLWebRTCVoiceClient::setUserVolume(const LLUUID& id, F32 volume)
+{
+ F32 clamped_volume = llclamp(volume, LLVoiceClient::VOLUME_MIN, LLVoiceClient::VOLUME_MAX);
+ if(mSession)
+ {
+ participantStatePtr_t participant(mSession->findParticipantByID(id));
+ if (participant && (participant->mAvatarID != gAgentID))
+ {
+ if (!is_approx_equal(volume, LLVoiceClient::VOLUME_DEFAULT))
+ {
+ // Store this volume setting for future sessions if it has been
+ // changed from the default
+ LLSpeakerVolumeStorage::getInstance()->storeSpeakerVolume(id, volume);
+ }
+ else
+ {
+ // Remove stored volume setting if it is returned to the default
+ LLSpeakerVolumeStorage::getInstance()->removeSpeakerVolume(id);
+ }
+
+ participant->mVolume = clamped_volume;
+ }
+ }
+ sessionState::for_each(boost::bind(predSetUserVolume, _1, id, clamped_volume));
+}
+
+// set volume level (gain level) for another user.
+void LLWebRTCVoiceClient::predSetUserVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const LLUUID &id, F32 volume)
+{
+ session->setUserVolume(id, volume);
+}
+
+////////////////////////
+///LLMuteListObserver
+///
+
+void LLWebRTCVoiceClient::onChange()
+{
+}
+
+void LLWebRTCVoiceClient::onChangeDetailed(const LLMute& mute)
+{
+ if (mute.mType == LLMute::AGENT)
+ {
+ bool muted = ((mute.mFlags & LLMute::flagVoiceChat) == 0);
+ sessionState::for_each(boost::bind(predSetUserMute, _1, mute.mID, muted));
+ }
+}
+
+void LLWebRTCVoiceClient::predSetUserMute(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const LLUUID &id, bool mute)
+{
+ session->setUserMute(id, mute);
+}
+
+//------------------------------------------------------------------------
+// Sessions
+
+std::map<std::string, LLWebRTCVoiceClient::sessionState::ptr_t> LLWebRTCVoiceClient::sessionState::mSessions;
+
+
+LLWebRTCVoiceClient::sessionState::sessionState() :
+ mHangupOnLastLeave(false),
+ mNotifyOnFirstJoin(false),
+ mMicGain(1.0),
+ mMuted(false),
+ mSpeakerVolume(1.0),
+ mShuttingDown(false)
+{
+}
+// ------------------------------------------------------------------
+// Predicates, for calls to all sessions
+
+void LLWebRTCVoiceClient::predUpdateOwnVolume(const LLWebRTCVoiceClient::sessionStatePtr_t &session, F32 audio_level)
+{
+ participantStatePtr_t participant = session->findParticipantByID(gAgentID);
+ if (participant)
+ {
+ participant->mLevel = audio_level;
+ // TODO: Add VAD for our own voice.
+ participant->mIsSpeaking = audio_level > SPEAKING_AUDIO_LEVEL;
+ }
+}
+
+void LLWebRTCVoiceClient::predSendData(const LLWebRTCVoiceClient::sessionStatePtr_t &session, const std::string &spatial_data)
+{
+ if (session->isSpatial() && !spatial_data.empty())
+ {
+ session->sendData(spatial_data);
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::sendData(const std::string &data)
+{
+ for (auto &connection : mWebRTCConnections)
+ {
+ connection->sendData(data);
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::setMuteMic(bool muted)
+{
+ mMuted = muted;
+ for (auto &connection : mWebRTCConnections)
+ {
+ connection->setMuteMic(muted);
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::setMicGain(F32 gain)
+{
+ mMicGain = gain;
+ for (auto &connection : mWebRTCConnections)
+ {
+ connection->setMicGain(gain);
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::setSpeakerVolume(F32 volume)
+{
+ mSpeakerVolume = volume;
+ for (auto &connection : mWebRTCConnections)
+ {
+ connection->setSpeakerVolume(volume);
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::setUserVolume(const LLUUID &id, F32 volume)
+{
+ if (mParticipantsByUUID.find(id) == mParticipantsByUUID.end())
+ {
+ return;
+ }
+ for (auto &connection : mWebRTCConnections)
+ {
+ connection->setUserVolume(id, volume);
+ }
+}
+
+void LLWebRTCVoiceClient::sessionState::setUserMute(const LLUUID &id, bool mute)
+{
+ if (mParticipantsByUUID.find(id) == mParticipantsByUUID.end())
+ {
+ return;
+ }
+ for (auto &connection : mWebRTCConnections)
+ {
+ connection->setUserMute(id, mute);
+ }
+}
+/*static*/
+void LLWebRTCVoiceClient::sessionState::addSession(
+ const std::string & channelID,
+ LLWebRTCVoiceClient::sessionState::ptr_t& session)
+{
+ mSessions[channelID] = session;
+}
+
+LLWebRTCVoiceClient::sessionState::~sessionState()
+{
+ LL_DEBUGS("Voice") << "Destroying session CHANNEL=" << mChannelID << LL_ENDL;
+
+ removeAllParticipants();
+}
+
+/*static*/
+LLWebRTCVoiceClient::sessionState::ptr_t LLWebRTCVoiceClient::sessionState::matchSessionByChannelID(const std::string& channel_id)
+{
+ sessionStatePtr_t result;
+
+ // *TODO: My kingdom for a lambda!
+ std::map<std::string, ptr_t>::iterator it = mSessions.find(channel_id);
+ if (it != mSessions.end())
+ {
+ result = (*it).second;
+ }
+ return result;
+}
+
+void LLWebRTCVoiceClient::sessionState::for_each(sessionFunc_t func)
+{
+ std::for_each(mSessions.begin(), mSessions.end(), boost::bind(for_eachPredicate, _1, func));
+}
+
+void LLWebRTCVoiceClient::sessionState::reapEmptySessions()
+{
+ std::map<std::string, ptr_t>::iterator iter;
+ for (iter = mSessions.begin(); iter != mSessions.end();)
+ {
+ if (iter->second->isEmpty())
+ {
+ iter = mSessions.erase(iter);
+ }
+ else
+ {
+ ++iter;
+ }
+ }
+}
+
+/*static*/
+void LLWebRTCVoiceClient::sessionState::for_eachPredicate(const std::pair<std::string, LLWebRTCVoiceClient::sessionState::wptr_t> &a, sessionFunc_t func)
+{
+ ptr_t aLock(a.second.lock());
+
+ if (aLock)
+ func(aLock);
+ else
+ {
+ LL_WARNS("Voice") << "Stale handle in session map!" << LL_ENDL;
+ }
+}
+
+LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::addSession(const std::string &channel_id, sessionState::ptr_t session)
+{
+ sessionStatePtr_t existingSession = sessionState::matchSessionByChannelID(channel_id);
+ if (!existingSession)
+ {
+ // No existing session found.
+
+ LL_DEBUGS("Voice") << "adding new session with channel: " << channel_id << LL_ENDL;
+ session->setMuteMic(mMuteMic);
+ session->setMicGain(mMicGain);
+ session->setSpeakerVolume(mSpeakerVolume);
+
+ sessionState::addSession(channel_id, session);
+ return session;
+ }
+ else
+ {
+ // Found an existing session
+ LL_DEBUGS("Voice") << "Attempting to add already-existing session " << channel_id << LL_ENDL;
+ existingSession->revive();
+
+ return existingSession;
+ }
+}
+
+LLWebRTCVoiceClient::sessionStatePtr_t LLWebRTCVoiceClient::findP2PSession(const LLUUID &agent_id)
+{
+ sessionStatePtr_t result = sessionState::matchSessionByChannelID(agent_id.asString());
+ if (result && !result->isSpatial())
+ {
+ return result;
+ }
+
+ result.reset();
+ return result;
+}
+
+void LLWebRTCVoiceClient::sessionState::shutdownAllConnections()
+{
+ mShuttingDown = true;
+ for (auto &&connection : mWebRTCConnections)
+ {
+ connection->shutDown();
+ }
+}
+
+// in case we drop into a session (spatial, etc.) right after
+// telling the session to shut down, revive it so it reconnects.
+void LLWebRTCVoiceClient::sessionState::revive()
+{
+ mShuttingDown = false;
+}
+
+//=========================================================================
+// the following are methods to support the coroutine implementation of the
+// voice connection and processing. They should only be called in the context
+// of a coroutine.
+//
+//
+
+void LLWebRTCVoiceClient::sessionState::processSessionStates()
+{
+ auto iter = mSessions.begin();
+ while (iter != mSessions.end())
+ {
+ if (!iter->second->processConnectionStates() && iter->second->mShuttingDown)
+ {
+ // if the connections associated with a session are gone,
+ // and this session is shutting down, remove it.
+ iter = mSessions.erase(iter);
+ }
+ else
+ {
+ iter++;
+ }
+ }
+}
+
+// process the states on each connection associated with a session.
+bool LLWebRTCVoiceClient::sessionState::processConnectionStates()
+{
+ std::list<connectionPtr_t>::iterator iter = mWebRTCConnections.begin();
+ while (iter != mWebRTCConnections.end())
+ {
+ if (!iter->get()->connectionStateMachine())
+ {
+ // if the state machine returns false, the connection is shut down
+ // so delete it.
+ iter = mWebRTCConnections.erase(iter);
+ }
+ else
+ {
+ ++iter;
+ }
+ }
+ return !mWebRTCConnections.empty();
+}
+
+// processing of spatial voice connection states requires special handling.
+// as neighboring regions need to be started up or shut down depending
+// on our location.
+bool LLWebRTCVoiceClient::estateSessionState::processConnectionStates()
+{
+ if (!mShuttingDown)
+ {
+ // Estate voice requires connection to neighboring regions.
+ std::set<LLUUID> neighbor_ids = LLWebRTCVoiceClient::getInstance()->getNeighboringRegions();
+
+ for (auto &connection : mWebRTCConnections)
+ {
+ boost::shared_ptr<LLVoiceWebRTCSpatialConnection> spatialConnection =
+ boost::static_pointer_cast<LLVoiceWebRTCSpatialConnection>(connection);
+
+ LLUUID regionID = spatialConnection.get()->getRegionID();
+
+ if (neighbor_ids.find(regionID) == neighbor_ids.end())
+ {
+ // shut down connections to neighbors that are too far away.
+ spatialConnection.get()->shutDown();
+ }
+ neighbor_ids.erase(regionID);
+ }
+
+ // add new connections for new neighbors
+ for (auto &neighbor : neighbor_ids)
+ {
+ connectionPtr_t connection(new LLVoiceWebRTCSpatialConnection(neighbor, INVALID_PARCEL_ID, mChannelID));
+
+ mWebRTCConnections.push_back(connection);
+ connection->setMicGain(mMicGain);
+ connection->setMuteMic(mMuted);
+ connection->setSpeakerVolume(mSpeakerVolume);
+ }
+ }
+ return LLWebRTCVoiceClient::sessionState::processConnectionStates();
+}
+
+// Various session state constructors.
+
+LLWebRTCVoiceClient::estateSessionState::estateSessionState()
+{
+ mHangupOnLastLeave = false;
+ mNotifyOnFirstJoin = false;
+ mChannelID = "Estate";
+ LLUUID region_id = gAgent.getRegion()->getRegionID();
+
+ mWebRTCConnections.emplace_back(new LLVoiceWebRTCSpatialConnection(region_id, INVALID_PARCEL_ID, "Estate"));
+}
+
+LLWebRTCVoiceClient::parcelSessionState::parcelSessionState(const std::string &channelID, S32 parcel_local_id)
+{
+ mHangupOnLastLeave = false;
+ mNotifyOnFirstJoin = false;
+ LLUUID region_id = gAgent.getRegion()->getRegionID();
+ mChannelID = channelID;
+ mWebRTCConnections.emplace_back(new LLVoiceWebRTCSpatialConnection(region_id, parcel_local_id, channelID));
+}
+
+LLWebRTCVoiceClient::adhocSessionState::adhocSessionState(const std::string &channelID,
+ const std::string &credentials,
+ bool notify_on_first_join,
+ bool hangup_on_last_leave) :
+ mCredentials(credentials)
+{
+ mHangupOnLastLeave = hangup_on_last_leave;
+ mNotifyOnFirstJoin = notify_on_first_join;
+ LLUUID region_id = gAgent.getRegion()->getRegionID();
+ mChannelID = channelID;
+ mWebRTCConnections.emplace_back(new LLVoiceWebRTCAdHocConnection(region_id, channelID, credentials));
+}
+
+void LLWebRTCVoiceClient::predShutdownSession(const LLWebRTCVoiceClient::sessionStatePtr_t& session)
+{
+ session->shutdownAllConnections();
+}
+
+void LLWebRTCVoiceClient::deleteSession(const sessionStatePtr_t &session)
+{
+ if (!session)
+ {
+ return;
+ }
+
+ // At this point, the session should be unhooked from all lists and all state should be consistent.
+ session->shutdownAllConnections();
+ // If this is the current audio session, clean up the pointer which will soon be dangling.
+ bool deleteAudioSession = mSession == session;
+ bool deleteNextAudioSession = mNextSession == session;
+ if (deleteAudioSession)
+ {
+ mSession.reset();
+ }
+
+ // ditto for the next audio session
+ if (deleteNextAudioSession)
+ {
+ mNextSession.reset();
+ }
+}
+
+
+// Name resolution
+void LLWebRTCVoiceClient::lookupName(const LLUUID &id)
+{
+ if (mAvatarNameCacheConnection.connected())
+ {
+ mAvatarNameCacheConnection.disconnect();
+ }
+ mAvatarNameCacheConnection = LLAvatarNameCache::get(id, boost::bind(&LLWebRTCVoiceClient::onAvatarNameCache, this, _1, _2));
+}
+
+void LLWebRTCVoiceClient::onAvatarNameCache(const LLUUID& agent_id,
+ const LLAvatarName& av_name)
+{
+ mAvatarNameCacheConnection.disconnect();
+ std::string display_name = av_name.getDisplayName();
+ avatarNameResolved(agent_id, display_name);
+}
+
+void LLWebRTCVoiceClient::predAvatarNameResolution(const LLWebRTCVoiceClient::sessionStatePtr_t &session, LLUUID id, std::string name)
+{
+ participantStatePtr_t participant(session->findParticipantByID(id));
+ if (participant)
+ {
+ // Found -- fill in the name
+ participant->mDisplayName = name;
+ // and post a "participants updated" message to listeners later.
+ LLWebRTCVoiceClient::getInstance()->notifyParticipantObservers();
+ }
+}
+
+void LLWebRTCVoiceClient::avatarNameResolved(const LLUUID &id, const std::string &name)
+{
+ sessionState::for_each(boost::bind(predAvatarNameResolution, _1, id, name));
+}
+
+// Leftover from vivox PTSN
+std::string LLWebRTCVoiceClient::sipURIFromID(const LLUUID& id)
+{
+ return id.asString();
+}
+
+
+/////////////////////////////
+// LLVoiceWebRTCConnection
+// These connections manage state transitions, negotiating webrtc connections,
+// and other such things for a single connection to a Secondlife WebRTC server.
+// Multiple of these connections may be active at once, in the case of
+// cross-region voice, or when a new connection is being created before the old
+// has a chance to shut down.
+LLVoiceWebRTCConnection::LLVoiceWebRTCConnection(const LLUUID &regionID, const std::string &channelID) :
+ mWebRTCAudioInterface(nullptr),
+ mWebRTCDataInterface(nullptr),
+ mVoiceConnectionState(VOICE_STATE_START_SESSION),
+ mMuted(true),
+ mShutDown(false),
+ mTrickling(false),
+ mIceCompleted(false),
+ mSpeakerVolume(0.0),
+ mMicGain(0.0),
+ mOutstandingRequests(0),
+ mChannelID(channelID),
+ mRegionID(regionID)
+{
+ mWebRTCPeerConnectionInterface = llwebrtc::newPeerConnection();
+ mWebRTCPeerConnectionInterface->setSignalingObserver(this);
+}
+
+LLVoiceWebRTCConnection::~LLVoiceWebRTCConnection()
+{
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ // peer connection and observers will be cleaned up
+ // by llwebrtc::terminate() on shutdown.
+ return;
+ }
+ if (mWebRTCPeerConnectionInterface)
+ {
+ llwebrtc::freePeerConnection(mWebRTCPeerConnectionInterface);
+ mWebRTCPeerConnectionInterface = nullptr;
+ }
+}
+
+
+// ICE (Interactive Connectivity Establishment)
+// When WebRTC tries to negotiate a connection to the Secondlife WebRTC Server,
+// the negotiation will result in a few updates about the best path
+// to which to connect.
+// The Secondlife servers are configured for ICE trickling, where, after a session is partially
+// negotiated, updates about the best connectivity paths may trickle in. These need to be
+// sent to the Secondlife WebRTC server via the simulator so that both sides have a clear
+// view of the network environment.
+void LLVoiceWebRTCConnection::OnIceGatheringState(llwebrtc::LLWebRTCSignalingObserver::EIceGatheringState state)
+{
+ LL_DEBUGS("Voice") << "Ice Gathering voice account. " << state << LL_ENDL;
+
+ switch (state)
+ {
+ case llwebrtc::LLWebRTCSignalingObserver::EIceGatheringState::ICE_GATHERING_COMPLETE:
+ {
+ LLMutexLock lock(&mVoiceStateMutex);
+ mIceCompleted = true;
+ break;
+ }
+ case llwebrtc::LLWebRTCSignalingObserver::EIceGatheringState::ICE_GATHERING_NEW:
+ {
+ LLMutexLock lock(&mVoiceStateMutex);
+ mIceCompleted = false;
+ }
+ default:
+ break;
+ }
+}
+
+void LLVoiceWebRTCConnection::OnIceCandidate(const llwebrtc::LLWebRTCIceCandidate &candidate)
+{
+ LLMutexLock lock(&mVoiceStateMutex);
+ mIceCandidates.push_back(candidate);
+}
+
+void LLVoiceWebRTCConnection::onIceUpdateComplete(bool ice_completed, const LLSD &result)
+{
+ mOutstandingRequests--;
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ return;
+ }
+ mTrickling = false;
+}
+
+void LLVoiceWebRTCConnection::onIceUpdateError(int retries, std::string url, LLSD body, bool ice_completed, const LLSD &result)
+{
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ mOutstandingRequests--;
+ return;
+ }
+ LLCore::HttpRequest::policy_t httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
+ LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t httpAdapter(new LLCoreHttpUtil::HttpCoroutineAdapter("voiceAccountProvision", httpPolicy));
+
+ if (retries >= 0)
+ {
+ LL_WARNS("Voice") << "Unable to complete ice trickling voice account, retrying. " << result << LL_ENDL;
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLVoiceWebRTCConnection::onIceUpdateComplete, this, ice_completed, _1),
+ boost::bind(&LLVoiceWebRTCConnection::onIceUpdateError, this, retries - 1, url, body, ice_completed, _1));
+ return;
+ }
+
+ LL_WARNS("Voice") << "Unable to complete ice trickling voice account, restarting connection. " << result << LL_ENDL;
+ if (!mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_SESSION_RETRY);
+ }
+ mTrickling = false;
+
+ mOutstandingRequests--;
+}
+
+
+// Ice candidates may be streamed in before or after the SDP offer is available (see below)
+// This function determines whether candidates are available to send to the Secondlife WebRTC
+// server via the simulator. If so, and there are no more candidates, this code
+// will make the cap call to the server sending up the ICE candidates.
+void LLVoiceWebRTCConnection::processIceUpdates()
+{
+ if (mShutDown || LLWebRTCVoiceClient::isShuttingDown())
+ {
+ return;
+ }
+
+ bool iceCompleted = false;
+ LLSD body;
+ {
+ if (!mTrickling)
+ {
+ if (!mIceCandidates.empty() || mIceCompleted)
+ {
+ LLViewerRegion *regionp = LLWorld::instance().getRegionFromID(mRegionID);
+ if (!regionp || !regionp->capabilitiesReceived())
+ {
+ LL_DEBUGS("Voice") << "no capabilities for ice gathering; waiting " << LL_ENDL;
+ return;
+ }
+
+ std::string url = regionp->getCapability("VoiceSignalingRequest");
+ if (url.empty())
+ {
+ return;
+ }
+
+ LL_DEBUGS("Voice") << "region ready to complete voice signaling; url=" << url << LL_ENDL;
+ if (!mIceCandidates.empty())
+ {
+ LLSD candidates = LLSD::emptyArray();
+ for (auto &ice_candidate : mIceCandidates)
+ {
+ LLSD body_candidate;
+ body_candidate["sdpMid"] = ice_candidate.mSdpMid;
+ body_candidate["sdpMLineIndex"] = ice_candidate.mMLineIndex;
+ body_candidate["candidate"] = ice_candidate.mCandidate;
+ candidates.append(body_candidate);
+ }
+ body["candidates"] = candidates;
+ mIceCandidates.clear();
+ }
+ else if (mIceCompleted)
+ {
+ LLSD body_candidate;
+ body_candidate["completed"] = true;
+ body["candidate"] = body_candidate;
+ iceCompleted = mIceCompleted;
+ mIceCompleted = false;
+ }
+
+ body["viewer_session"] = mViewerSession;
+ body["voice_server_type"] = WEBRTC_VOICE_SERVER_TYPE;
+
+ LLCore::HttpRequest::policy_t httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
+ LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t httpAdapter(
+ new LLCoreHttpUtil::HttpCoroutineAdapter("voiceAccountProvision", httpPolicy));
+ LLCore::HttpRequest::ptr_t httpRequest(new LLCore::HttpRequest);
+ LLCore::HttpOptions::ptr_t httpOpts = LLCore::HttpOptions::ptr_t(new LLCore::HttpOptions);
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLVoiceWebRTCSpatialConnection::onIceUpdateComplete, this, iceCompleted, _1),
+ boost::bind(&LLVoiceWebRTCSpatialConnection::onIceUpdateError, this, 3, url, body, iceCompleted, _1));
+ mOutstandingRequests++;
+ mTrickling = true;
+ }
+ }
+ }
+}
+
+
+// An 'Offer' comes in the form of a SDP (Session Description Protocol)
+// which contains all sorts of info about the session, from network paths
+// to the type of session (audio, video) to characteristics (the encoder type.)
+// This SDP also serves as the 'ticket' to the server, security-wise.
+// The Offer is retrieved from the WebRTC library on the client,
+// and is passed to the simulator via a CAP, which then passes
+// it on to the Secondlife WebRTC server.
+
+void LLVoiceWebRTCConnection::OnOfferAvailable(const std::string &sdp)
+{
+ LL_DEBUGS("Voice") << "On Offer Available." << LL_ENDL;
+ LLMutexLock lock(&mVoiceStateMutex);
+ mChannelSDP = sdp;
+ if (mVoiceConnectionState == VOICE_STATE_WAIT_FOR_SESSION_START)
+ {
+ mVoiceConnectionState = VOICE_STATE_REQUEST_CONNECTION;
+ }
+}
+
+// Notifications from the webrtc library.
+void LLVoiceWebRTCConnection::OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface *audio_interface)
+{
+ LL_DEBUGS("Voice") << "On AudioEstablished." << LL_ENDL;
+ mWebRTCAudioInterface = audio_interface;
+ setVoiceConnectionState(VOICE_STATE_SESSION_ESTABLISHED);
+}
+
+void LLVoiceWebRTCConnection::OnRenegotiationNeeded()
+{
+ LL_DEBUGS("Voice") << "Voice channel requires renegotiation." << LL_ENDL;
+ if (!mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_SESSION_RETRY);
+ }
+}
+
+void LLVoiceWebRTCConnection::OnPeerConnectionShutdown()
+{
+ setVoiceConnectionState(VOICE_STATE_SESSION_EXIT);
+ mOutstandingRequests--; // shut down is an async call which is handled on a webrtc thread.
+}
+
+void LLVoiceWebRTCConnection::setMuteMic(bool muted)
+{
+ mMuted = muted;
+ if (mWebRTCAudioInterface)
+ {
+ mWebRTCAudioInterface->setMute(muted);
+ }
+}
+
+void LLVoiceWebRTCConnection::setMicGain(F32 gain)
+{
+ mMicGain = gain;
+ if (mWebRTCAudioInterface)
+ {
+ mWebRTCAudioInterface->setSendVolume(gain);
+ }
+}
+
+void LLVoiceWebRTCConnection::setSpeakerVolume(F32 volume)
+{
+ mSpeakerVolume = volume;
+ if (mWebRTCAudioInterface)
+ {
+ mWebRTCAudioInterface->setReceiveVolume(volume);
+ }
+}
+
+void LLVoiceWebRTCConnection::setUserVolume(const LLUUID& id, F32 volume)
+{
+ Json::Value root = Json::objectValue;
+ Json::Value user_gain = Json::objectValue;
+ user_gain[id.asString()] = (uint32_t)(volume*200); // give it two decimal places with a range from 0-200, where 100 is normal
+ root["ug"] = user_gain;
+ Json::FastWriter writer;
+ std::string json_data = writer.write(root);
+ mWebRTCDataInterface->sendData(json_data, false);
+}
+
+void LLVoiceWebRTCConnection::setUserMute(const LLUUID& id, bool mute)
+{
+ Json::Value root = Json::objectValue;
+ Json::Value muted = Json::objectValue;
+ muted[id.asString()] = mute;
+ root["m"] = muted;
+ Json::FastWriter writer;
+ std::string json_data = writer.write(root);
+ mWebRTCDataInterface->sendData(json_data, false);
+}
+
+
+// Send data to the Secondlife WebRTC server via the webrtc
+// data channel.
+void LLVoiceWebRTCConnection::sendData(const std::string &data)
+{
+ if (getVoiceConnectionState() == VOICE_STATE_SESSION_UP && mWebRTCDataInterface)
+ {
+ mWebRTCDataInterface->sendData(data, false);
+ }
+}
+
+// Tell the simulator that we're shutting down a voice connection.
+// The simulator will pass this on to the Secondlife WebRTC server.
+bool LLVoiceWebRTCConnection::breakVoiceConnection(bool corowait)
+{
+ LL_DEBUGS("Voice") << "Disconnecting voice." << LL_ENDL;
+ if (mWebRTCDataInterface)
+ {
+ mWebRTCDataInterface->unsetDataObserver(this);
+ mWebRTCDataInterface = nullptr;
+ }
+ mWebRTCAudioInterface = nullptr;
+ LLViewerRegion *regionp = LLWorld::instance().getRegionFromID(mRegionID);
+ if (!regionp || !regionp->capabilitiesReceived())
+ {
+ LL_DEBUGS("Voice") << "no capabilities for voice provisioning; waiting " << LL_ENDL;
+ setVoiceConnectionState(VOICE_STATE_SESSION_EXIT);
+ return false;
+ }
+
+ std::string url = regionp->getCapability("ProvisionVoiceAccountRequest");
+ if (url.empty())
+ {
+ return false;
+ }
+
+ LL_DEBUGS("Voice") << "region ready for voice break; url=" << url << LL_ENDL;
+
+ LLCore::HttpRequest::policy_t httpPolicy(LLCore::HttpRequest::DEFAULT_POLICY_ID);
+ LLCoreHttpUtil::HttpCoroutineAdapter::ptr_t httpAdapter(new LLCoreHttpUtil::HttpCoroutineAdapter("parcelVoiceInfoRequest", httpPolicy));
+ LLCore::HttpRequest::ptr_t httpRequest(new LLCore::HttpRequest);
+
+ LLVoiceWebRTCStats::getInstance()->provisionAttemptStart();
+ LLSD body;
+ body["logout"] = TRUE;
+ body["viewer_session"] = mViewerSession;
+ body["voice_server_type"] = REPORTED_VOICE_SERVER_TYPE;
+
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLVoiceWebRTCSpatialConnection::OnVoiceDisconnectionRequestSuccess, this, _1),
+ boost::bind(&LLVoiceWebRTCSpatialConnection::OnVoiceDisconnectionRequestFailure, this, url, 3, body, _1));
+ setVoiceConnectionState(VOICE_STATE_WAIT_FOR_EXIT);
+ mOutstandingRequests++;
+ return true;
+}
+
+void LLVoiceWebRTCConnection::OnVoiceDisconnectionRequestSuccess(const LLSD &result)
+{
+ mOutstandingRequests--;
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ return;
+ }
+
+ if (mWebRTCPeerConnectionInterface)
+ {
+ if (mWebRTCPeerConnectionInterface->shutdownConnection())
+ {
+ mOutstandingRequests++;
+ }
+ }
+ else
+ {
+ setVoiceConnectionState(VOICE_STATE_SESSION_EXIT);
+ }
+}
+
+void LLVoiceWebRTCConnection::OnVoiceDisconnectionRequestFailure(std::string url, int retries, LLSD body, const LLSD &result)
+{
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ mOutstandingRequests--;
+ return;
+ }
+ if (retries >= 0)
+ {
+ // retry a few times.
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLVoiceWebRTCSpatialConnection::OnVoiceDisconnectionRequestSuccess, this, _1),
+ boost::bind(&LLVoiceWebRTCSpatialConnection::OnVoiceDisconnectionRequestFailure, this, url, retries - 1, body, _1));
+ return;
+ }
+ if (mWebRTCPeerConnectionInterface)
+ {
+ mOutstandingRequests++;
+ mWebRTCPeerConnectionInterface->shutdownConnection();
+ }
+ else
+ {
+ setVoiceConnectionState(VOICE_STATE_SESSION_EXIT);
+ }
+ mOutstandingRequests--;
+}
+
+
+// Tell the simulator to tell the Secondlife WebRTC server that we want a voice
+// connection. The SDP is sent up as part of this, and the simulator will respond
+// with an 'answer' which is in the form of another SDP. The webrtc library
+// will use the offer and answer to negotiate the session.
+bool LLVoiceWebRTCSpatialConnection::requestVoiceConnection()
+{
+ LLViewerRegion *regionp = LLWorld::instance().getRegionFromID(mRegionID);
+
+ LL_DEBUGS("Voice") << "Requesting voice connection." << LL_ENDL;
+ if (!regionp || !regionp->capabilitiesReceived())
+ {
+ LL_DEBUGS("Voice") << "no capabilities for voice provisioning; waiting " << LL_ENDL;
+ return false;
+ }
+
+ std::string url = regionp->getCapability("ProvisionVoiceAccountRequest");
+ if (url.empty())
+ {
+ return false;
+ }
+
+ LL_DEBUGS("Voice") << "region ready for voice provisioning; url=" << url << LL_ENDL;
+
+ LLVoiceWebRTCStats::getInstance()->provisionAttemptStart();
+ LLSD body;
+ LLSD jsep;
+ jsep["type"] = "offer";
+ {
+ LLMutexLock lock(&mVoiceStateMutex);
+ jsep["sdp"] = mChannelSDP;
+ }
+ body["jsep"] = jsep;
+ if (mParcelLocalID != INVALID_PARCEL_ID)
+ {
+ body["parcel_local_id"] = mParcelLocalID;
+ }
+
+ body["voice_server_type"] = WEBRTC_VOICE_SERVER_TYPE;
+
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLVoiceWebRTCSpatialConnection::OnVoiceConnectionRequestSuccess, this, _1),
+ boost::bind(&LLVoiceWebRTCSpatialConnection::OnVoiceConnectionRequestFailure, this, url, 3, body, _1));
+ mOutstandingRequests++;
+ return true;
+}
+
+void LLVoiceWebRTCConnection::OnVoiceConnectionRequestSuccess(const LLSD &result)
+{
+ mOutstandingRequests--;
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ return;
+ }
+ LLVoiceWebRTCStats::getInstance()->provisionAttemptEnd(true);
+
+ if (result.has("viewer_session") && result.has("jsep") && result["jsep"].has("type") && result["jsep"]["type"] == "answer" &&
+ result["jsep"].has("sdp"))
+ {
+ mRemoteChannelSDP = result["jsep"]["sdp"].asString();
+ mViewerSession = result["viewer_session"];
+ }
+ else
+ {
+ LL_WARNS("Voice") << "Invalid voice provision request result:" << result << LL_ENDL;
+ setVoiceConnectionState(VOICE_STATE_SESSION_RETRY);
+ return;
+ }
+
+ LL_DEBUGS("Voice") << "ProvisionVoiceAccountRequest response"
+ << " channel sdp " << mRemoteChannelSDP << LL_ENDL;
+
+ mWebRTCPeerConnectionInterface->AnswerAvailable(mRemoteChannelSDP);
+}
+
+
+void LLVoiceWebRTCConnection::OnVoiceConnectionRequestFailure(std::string url, int retries, LLSD body, const LLSD &result)
+{
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ mOutstandingRequests--;
+ return;
+ }
+ if (retries >= 0)
+ {
+ LL_WARNS("Voice") << "Failure connecting to voice, retrying." << body << " RESULT: " << result << LL_ENDL;
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLVoiceWebRTCConnection::OnVoiceConnectionRequestSuccess, this, _1),
+ boost::bind(&LLVoiceWebRTCConnection::OnVoiceConnectionRequestFailure, this, url, retries - 1, body, _1));
+ return;
+ }
+ LL_WARNS("Voice") << "Unable to connect voice." << body << " RESULT: " << result << LL_ENDL;
+ setVoiceConnectionState(VOICE_STATE_SESSION_RETRY);
+ mOutstandingRequests--;
+}
+
+
+// Primary state machine for negotiating a single voice connection to the
+// Secondlife WebRTC server.
+bool LLVoiceWebRTCConnection::connectionStateMachine()
+{
+ processIceUpdates();
+
+ switch (getVoiceConnectionState())
+ {
+ case VOICE_STATE_START_SESSION:
+ {
+ if (mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ break;
+ }
+ mTrickling = false;
+ mIceCompleted = false;
+ setVoiceConnectionState(VOICE_STATE_WAIT_FOR_SESSION_START);
+ // tell the webrtc library that we want a connection. The library will
+ // respond with an offer on a separate thread, which will cause
+ // the session state to change.
+ if (!mWebRTCPeerConnectionInterface->initializeConnection())
+ {
+ setVoiceConnectionState(VOICE_STATE_SESSION_RETRY);
+ }
+ break;
+ }
+
+ case VOICE_STATE_WAIT_FOR_SESSION_START:
+ {
+ if (mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ }
+ break;
+ }
+
+ case VOICE_STATE_REQUEST_CONNECTION:
+ if (mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ break;
+ }
+ // Ask the sim to ask the Secondlife WebRTC server for a connection to
+ // a given voice channel. On completion, we'll move on to the
+ // VOICE_STATE_SESSION_ESTABLISHED via a callback on a webrtc thread.
+ if (!requestVoiceConnection())
+ {
+ setVoiceConnectionState(VOICE_STATE_SESSION_RETRY);
+ }
+ else
+ {
+ setVoiceConnectionState(VOICE_STATE_CONNECTION_WAIT);
+ }
+ break;
+
+ case VOICE_STATE_CONNECTION_WAIT:
+ if (mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ }
+ break;
+
+ case VOICE_STATE_SESSION_ESTABLISHED:
+ {
+ if (mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ break;
+ }
+ // update the peer connection with the various characteristics of
+ // this connection.
+ mWebRTCAudioInterface->setMute(mMuted);
+ mWebRTCAudioInterface->setReceiveVolume(mSpeakerVolume);
+ mWebRTCAudioInterface->setSendVolume(mMicGain);
+ LLWebRTCVoiceClient::getInstance()->OnConnectionEstablished(mChannelID, mRegionID);
+ setVoiceConnectionState(VOICE_STATE_WAIT_FOR_DATA_CHANNEL);
+ break;
+ }
+
+ case VOICE_STATE_WAIT_FOR_DATA_CHANNEL:
+ {
+ if (mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ break;
+ }
+ if (mWebRTCDataInterface) // the interface will be set when the session is negotiated.
+ {
+ sendJoin(); // tell the Secondlife WebRTC server that we're here via the data channel.
+ setVoiceConnectionState(VOICE_STATE_SESSION_UP);
+ }
+ break;
+ }
+
+ case VOICE_STATE_SESSION_UP:
+ {
+ // we'll stay here as long as the session remains up.
+ if (mShutDown)
+ {
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ }
+ break;
+ }
+
+ case VOICE_STATE_SESSION_RETRY:
+ // something went wrong, so notify that the connection has failed.
+ LLWebRTCVoiceClient::getInstance()->OnConnectionFailure(mChannelID, mRegionID);
+ setVoiceConnectionState(VOICE_STATE_DISCONNECT);
+ break;
+
+ case VOICE_STATE_DISCONNECT:
+ breakVoiceConnection(true);
+ break;
+
+ case VOICE_STATE_WAIT_FOR_EXIT:
+ break;
+
+ case VOICE_STATE_SESSION_EXIT:
+ {
+ {
+ LLMutexLock lock(&mVoiceStateMutex);
+ if (!mShutDown)
+ {
+ mVoiceConnectionState = VOICE_STATE_START_SESSION;
+ }
+ else
+ {
+ // if we still have outstanding http or webrtc calls, wait for them to
+ // complete so we don't delete objects while they still may be used.
+ if (mOutstandingRequests <= 0)
+ {
+ LLWebRTCVoiceClient::getInstance()->OnConnectionShutDown(mChannelID, mRegionID);
+ return false;
+ }
+ }
+ }
+ break;
+ }
+
+ default:
+ {
+ LL_WARNS("Voice") << "Unknown voice control state " << getVoiceConnectionState() << LL_ENDL;
+ return false;
+ }
+ }
+ return true;
+}
+
+// Data has been received on the webrtc data channel
+void LLVoiceWebRTCConnection::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' - object of join data (currently only a boolean 'p' marking a primary participant)
+ // 'l' - boolean, always true if exists.
+ // 'v' - boolean - voice activity has been detected.
+
+ 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;
+ }
+ bool new_participant = false;
+ Json::Value mute = Json::objectValue;
+ Json::Value user_gain = Json::objectValue;
+ for (auto &participant_id : voice_data.getMemberNames())
+ {
+ LLUUID agent_id(participant_id);
+ if (agent_id.isNull())
+ {
+ LL_WARNS("Voice") << "Bad participant ID from data channel (" << participant_id << "):" << data << LL_ENDL;
+ continue;
+ }
+
+ LLWebRTCVoiceClient::participantStatePtr_t participant =
+ LLWebRTCVoiceClient::getInstance()->findParticipantByID(mChannelID, agent_id);
+ bool joined = false;
+ bool primary = false; // we ignore any 'joins' reported about participants
+ // that come from voice servers that aren't their primary
+ // voice server. This will happen with cross-region voice
+ // where a participant on a neighboring region may be
+ // connected to multiple servers. We don't want to
+ // add new identical participants from all of those servers.
+ if (voice_data[participant_id].isMember("j"))
+ {
+ // a new participant has announced that they're joining.
+ joined = true;
+ primary = voice_data[participant_id]["j"].get("p", Json::Value(false)).asBool();
+
+ // track incoming participants that are muted so we can mute their connections (or set their volume)
+ bool isMuted = LLMuteList::getInstance()->isMuted(agent_id, LLMute::flagVoiceChat);
+ if (isMuted)
+ {
+ mute[participant_id] = true;
+ }
+ F32 volume;
+ if(LLSpeakerVolumeStorage::getInstance()->getSpeakerVolume(agent_id, volume))
+ {
+ user_gain[participant_id] = (uint32_t)(volume * 200);
+ }
+ }
+
+ new_participant |= joined;
+ if (!participant && joined && (primary || !isSpatial()))
+ {
+ participant = LLWebRTCVoiceClient::getInstance()->addParticipantByID(mChannelID, agent_id);
+ }
+
+ if (participant)
+ {
+ if (voice_data[participant_id].get("l", Json::Value(false)).asBool())
+ {
+ // an existing participant is leaving.
+ if (agent_id != gAgentID)
+ {
+ LLWebRTCVoiceClient::getInstance()->removeParticipantByID(mChannelID, agent_id);
+ }
+ }
+ else
+ {
+ // we got a 'power' update.
+ F32 level = (F32) (voice_data[participant_id].get("p", Json::Value(participant->mLevel)).asInt()) / 128;
+ // convert to decibles
+ participant->mLevel = level;
+
+ if (voice_data[participant_id].isMember("v"))
+ {
+ participant->mIsSpeaking = voice_data[participant_id].get("v", Json::Value(false)).asBool();
+ }
+ }
+ }
+ }
+
+ // tell the simulator to set the mute and volume data for this
+ // participant, if there are any updates.
+ Json::FastWriter writer;
+ Json::Value root = Json::objectValue;
+ if (mute.size() > 0)
+ {
+ root["m"] = mute;
+ }
+ if (user_gain.size() > 0)
+ {
+ root["ug"] = user_gain;
+ }
+ if (root.size() > 0)
+ {
+ std::string json_data = writer.write(root);
+ mWebRTCDataInterface->sendData(json_data, false);
+ }
+ }
+}
+
+void LLVoiceWebRTCConnection::OnDataChannelReady(llwebrtc::LLWebRTCDataInterface *data_interface)
+{
+ if (data_interface)
+ {
+ mWebRTCDataInterface = data_interface;
+ mWebRTCDataInterface->setDataObserver(this);
+ }
+}
+
+// tell the Secondlife WebRTC server that
+// we're joining and whether we're
+// joining a server associated with the
+// the region we currently occupy or not (primary)
+// The WebRTC voice server will pass this info
+// to peers.
+void LLVoiceWebRTCConnection::sendJoin()
+{
+ Json::FastWriter writer;
+ Json::Value root = Json::objectValue;
+ Json::Value join_obj = Json::objectValue;
+ LLUUID regionID = gAgent.getRegion()->getRegionID();
+ if ((regionID == mRegionID) || !isSpatial())
+ {
+ join_obj["p"] = true;
+ }
+ root["j"] = join_obj;
+ std::string json_data = writer.write(root);
+ mWebRTCDataInterface->sendData(json_data, false);
+}
+
+/////////////////////////////
+// WebRTC Spatial Connection
+
+LLVoiceWebRTCSpatialConnection::LLVoiceWebRTCSpatialConnection(const LLUUID &regionID,
+ S32 parcelLocalID,
+ const std::string &channelID) :
+ LLVoiceWebRTCConnection(regionID, channelID),
+ mParcelLocalID(parcelLocalID)
+{
+}
+
+LLVoiceWebRTCSpatialConnection::~LLVoiceWebRTCSpatialConnection()
+{
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ // peer connection and observers will be cleaned up
+ // by llwebrtc::terminate() on shutdown.
+ return;
+ }
+ assert(mOutstandingRequests == 0);
+ mWebRTCPeerConnectionInterface->unsetSignalingObserver(this);
+}
+
+void LLVoiceWebRTCSpatialConnection::setMuteMic(bool muted)
+{
+ if (mMuted != muted)
+ {
+ mMuted = muted;
+ if (mWebRTCAudioInterface)
+ {
+ LLViewerRegion *regionp = gAgent.getRegion();
+ if (regionp && mRegionID == regionp->getRegionID())
+ {
+ mWebRTCAudioInterface->setMute(muted);
+ }
+ else
+ {
+ // Always mute this agent with respect to neighboring regions.
+ // Peers don't want to hear this agent from multiple regions
+ // as that'll echo.
+ mWebRTCAudioInterface->setMute(true);
+ }
+ }
+ }
+}
+
+/////////////////////////////
+// WebRTC Spatial Connection
+
+LLVoiceWebRTCAdHocConnection::LLVoiceWebRTCAdHocConnection(const LLUUID &regionID,
+ const std::string& channelID,
+ const std::string& credentials) :
+ LLVoiceWebRTCConnection(regionID, channelID),
+ mCredentials(credentials)
+{
+}
+
+LLVoiceWebRTCAdHocConnection::~LLVoiceWebRTCAdHocConnection()
+{
+ if (LLWebRTCVoiceClient::isShuttingDown())
+ {
+ // peer connection and observers will be cleaned up
+ // by llwebrtc::terminate() on shutdown.
+ return;
+ }
+ assert(mOutstandingRequests == 0);
+ mWebRTCPeerConnectionInterface->unsetSignalingObserver(this);
+}
+
+// Add-hoc connections require a different channel type
+// as they go to a different set of Secondlife WebRTC servers.
+// They also require credentials for the given channels.
+// So, we have a separate requestVoiceConnection call.
+bool LLVoiceWebRTCAdHocConnection::requestVoiceConnection()
+{
+ LLViewerRegion *regionp = LLWorld::instance().getRegionFromID(mRegionID);
+
+ LL_DEBUGS("Voice") << "Requesting voice connection." << LL_ENDL;
+ if (!regionp || !regionp->capabilitiesReceived())
+ {
+ LL_DEBUGS("Voice") << "no capabilities for voice provisioning; waiting " << LL_ENDL;
+ return false;
+ }
+
+ std::string url = regionp->getCapability("ProvisionVoiceAccountRequest");
+ if (url.empty())
+ {
+ return false;
+ }
+
+ LLVoiceWebRTCStats::getInstance()->provisionAttemptStart();
+ LLSD body;
+ LLSD jsep;
+ jsep["type"] = "offer";
+ {
+ LLMutexLock lock(&mVoiceStateMutex);
+ jsep["sdp"] = mChannelSDP;
+ }
+ body["jsep"] = jsep;
+ body["credentials"] = mCredentials;
+ body["channel"] = mChannelID;
+ body["channel_type"] = "multiagent";
+ body["voice_server_type"] = WEBRTC_VOICE_SERVER_TYPE;
+
+ LLCoreHttpUtil::HttpCoroutineAdapter::callbackHttpPost(
+ url,
+ LLCore::HttpRequest::DEFAULT_POLICY_ID,
+ body,
+ boost::bind(&LLVoiceWebRTCAdHocConnection::OnVoiceConnectionRequestSuccess, this, _1),
+ boost::bind(&LLVoiceWebRTCAdHocConnection::OnVoiceConnectionRequestFailure, this, url, 3, body, _1));
+ mOutstandingRequests++;
+ return true;
+}