diff options
Diffstat (limited to 'indra/newview/llvoicewebrtc.cpp')
| -rw-r--r-- | indra/newview/llvoicewebrtc.cpp | 211 |
1 files changed, 195 insertions, 16 deletions
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp index 06ccd23670..c6967a8743 100644 --- a/indra/newview/llvoicewebrtc.cpp +++ b/indra/newview/llvoicewebrtc.cpp @@ -61,10 +61,12 @@ #include "llrand.h" #include "llviewerwindow.h" #include "llviewercamera.h" +#include "llviewerstats.h" #include "llversioninfo.h" #include "llviewernetwork.h" #include "llnotificationsutil.h" +#include "llnearbyvoicemoderation.h" #include "llcorehttputil.h" #include "lleventfilter.h" @@ -80,6 +82,8 @@ const std::string WEBRTC_VOICE_SERVER_TYPE = "webrtc"; +const F32 STATS_TIMER_DELAY = 2.0; + namespace { const F32 MAX_AUDIO_DIST = 50.0f; @@ -2496,17 +2500,18 @@ void LLVoiceWebRTCConnection::processIceUpdatesCoro(connectionPtr_t connection) // callback from llwebrtc void LLVoiceWebRTCConnection::OnOfferAvailable(const std::string &sdp) { + connectionPtr_t connection = shared_from_this(); LL::WorkQueue::postMaybe(mMainQueue, - [=, this] { - if (mShutDown) + [=] { + if (connection->mShutDown) { return; } LL_DEBUGS("Voice") << "On Offer Available." << LL_ENDL; - mChannelSDP = sdp; - if (mVoiceConnectionState == VOICE_STATE_WAIT_FOR_SESSION_START) + connection->mChannelSDP = sdp; + if (connection->mVoiceConnectionState == VOICE_STATE_WAIT_FOR_SESSION_START) { - mVoiceConnectionState = VOICE_STATE_REQUEST_CONNECTION; + connection->mVoiceConnectionState = VOICE_STATE_REQUEST_CONNECTION; } }); } @@ -2523,16 +2528,17 @@ void LLVoiceWebRTCConnection::OnOfferAvailable(const std::string &sdp) // callback from llwebrtc void LLVoiceWebRTCConnection::OnAudioEstablished(llwebrtc::LLWebRTCAudioInterface* audio_interface) { + connectionPtr_t connection = shared_from_this(); LL::WorkQueue::postMaybe(mMainQueue, - [=, this] { - if (mShutDown) + [=] { + if (connection->mShutDown) { return; } LL_DEBUGS("Voice") << "On AudioEstablished." << LL_ENDL; - mWebRTCAudioInterface = audio_interface; - mWebRTCAudioInterface->setMute(true); // mute will be set appropriately later when we finish setting up. - setVoiceConnectionState(VOICE_STATE_SESSION_ESTABLISHED); + connection->mWebRTCAudioInterface = audio_interface; + connection->mWebRTCAudioInterface->setMute(true); // mute will be set appropriately later when we finish setting up. + connection->setVoiceConnectionState(VOICE_STATE_SESSION_ESTABLISHED); }); } @@ -2902,6 +2908,7 @@ bool LLVoiceWebRTCConnection::connectionStateMachine() } mWebRTCAudioInterface->setReceiveVolume(mSpeakerVolume); LLWebRTCVoiceClient::getInstance()->OnConnectionEstablished(mChannelID, mRegionID); + resetConnectionStats(); setVoiceConnectionState(VOICE_STATE_WAIT_FOR_DATA_CHANNEL); break; } @@ -2955,6 +2962,13 @@ bool LLVoiceWebRTCConnection::connectionStateMachine() sendJoin(); } } + + static LLTimer stats_timer; + if (stats_timer.getElapsedTimeF32() > STATS_TIMER_DELAY) + { + mWebRTCPeerConnectionInterface->gatherConnectionStats(); + stats_timer.reset(); + } } break; } @@ -3086,6 +3100,13 @@ void LLVoiceWebRTCConnection::OnDataReceivedImpl(const std::string &data, bool b LL_WARNS("Voice") << "Expected object from data channel:" << data << LL_ENDL; return; } + + bool is_primary_region = mPrimary; + if (!mPrimary && isSpatial() && gAgent.getRegion()) + { + is_primary_region = (mRegionID == gAgent.getRegion()->getRegionID()); + LL_WARNS() << "mPrimary is false, expected: " << is_primary_region << " connection state: " << getVoiceConnectionState() << LL_ENDL; + } boost::json::object voice_data = voice_data_parsed.as_object(); boost::json::object mute; boost::json::object user_gain; @@ -3171,12 +3192,63 @@ void LLVoiceWebRTCConnection::OnDataReceivedImpl(const std::string &data, bool b if (participant_obj.contains("m") && participant_obj["m"].is_bool()) { - participant->mIsModeratorMuted = participant_obj["m"].as_bool(); + bool is_moderator_muted = participant_obj["m"].as_bool(); + if (isSpatial()) + { + // ignore muted flags from non-primary server + if (is_primary_region || primary) + { + participant->mIsModeratorMuted = is_moderator_muted; + if (gAgentID == agent_id) + { + LLNearbyVoiceModeration::getInstance()->setMutedInfo(mChannelID, is_moderator_muted); + } + } + } + else + { + participant->mIsModeratorMuted = is_moderator_muted; + } } } } - } + else + { + if (isSpatial() && (is_primary_region || primary)) + { + // mute info message can be received before join message, so try to mute again later + if (participant_obj.contains("m") && participant_obj["m"].is_bool()) + { + LL_WARNS() << "Mute info msg received: " << participant_obj["m"].as_bool() + << " but participant " << agent_id + << " was not found in channel " << mChannelID << LL_ENDL; + bool is_moderator_muted = participant_obj["m"].as_bool(); + std::string channel_id = mChannelID; + F32 delay { 1.5f }; + doAfterInterval( + [channel_id, agent_id, is_moderator_muted]() + { + LLWebRTCVoiceClient::participantStatePtr_t participant = + LLWebRTCVoiceClient::getInstance()->findParticipantByID(channel_id, agent_id); + if (participant) + { + participant->mIsModeratorMuted = is_moderator_muted; + LL_WARNS() << "Participant " << agent_id << " is found after delay, is_muted: " << is_moderator_muted << LL_ENDL; + if (gAgentID == agent_id) + { + LLNearbyVoiceModeration::getInstance()->setMutedInfo(channel_id, is_moderator_muted); + } + } + else + { + LL_WARNS() << "Participant " << agent_id << " is still not found in channel " << channel_id << LL_ENDL; + } + }, delay); + } + } + } + } // tell the simulator to set the mute and volume data for this // participant, if there are any updates. boost::json::object root; @@ -3207,17 +3279,18 @@ void LLVoiceWebRTCConnection::OnDataReceivedImpl(const std::string &data, bool b // llwebrtc callback void LLVoiceWebRTCConnection::OnDataChannelReady(llwebrtc::LLWebRTCDataInterface *data_interface) { + connectionPtr_t connection = shared_from_this(); LL::WorkQueue::postMaybe(mMainQueue, - [=, this] { - if (mShutDown) + [=] { + if (connection->mShutDown) { return; } if (data_interface) { - mWebRTCDataInterface = data_interface; - mWebRTCDataInterface->setDataObserver(this); + connection->mWebRTCDataInterface = data_interface; + connection->mWebRTCDataInterface->setDataObserver(connection.get()); } }); } @@ -3247,6 +3320,112 @@ void LLVoiceWebRTCConnection::sendJoin() mWebRTCDataInterface->sendData(json_data, false); } +void LLVoiceWebRTCConnection::OnStatsDelivered(const llwebrtc::LLWebRTCStatsMap& stats_data) +{ + LL::WorkQueue::postMaybe(mMainQueue, [=, this] + { + if (mShutDown) + { + return; + } + for (const auto& [stats_id, attributes] : stats_data) + { + if (attributes.contains("currentRoundTripTime")) + { + F32 rtt_seconds = 0.0f; + LLStringUtil::convertToF32(attributes.at("currentRoundTripTime"), rtt_seconds); + sample(LLStatViewer::WEBRTC_LATENCY, rtt_seconds * 1000.0f); + } + if (attributes.contains("availableOutgoingBitrate")) + { + F32 bitrate_bps = 0.0f; + LLStringUtil::convertToF32(attributes.at("availableOutgoingBitrate"), bitrate_bps); + sample(LLStatViewer::WEBRTC_UPLOAD_BANDWIDTH, bitrate_bps / 1000.0f); + } + + // Stat type detection below is heuristic-based. + // It's relied on specific fields to distinguish outbound-rtp, remote-inbound-rtp, and inbound-rtp. + // This approach works with current WebRTC stats but may need updating later. + + // Outbound RTP + if (attributes.contains("mediaSourceId")) + { + U32 out_packets_sent = 0; + LLStringUtil::convertToU32(attributes.at("packetsSent"), out_packets_sent); + sample(LLStatViewer::WEBRTC_PACKETS_OUT_SENT, out_packets_sent); + } + // Remote-Inbound RTP + else if (attributes.contains("localId")) + { + if (attributes.contains("packetsLost")) + { + U32 out_packets_lost = 0; + LLStringUtil::convertToU32(attributes.at("packetsLost"), out_packets_lost); + sample(LLStatViewer::WEBRTC_PACKETS_OUT_LOST, out_packets_lost); + } + if (attributes.contains("jitter")) + { + F32 jitter_seconds = 0.0f; + LLStringUtil::convertToF32(attributes.at("jitter"), jitter_seconds); + sample(LLStatViewer::WEBRTC_JITTER_OUT, jitter_seconds * 1000.0f); + } + } + // Inbound RTP + else if (attributes.contains("jitterBufferDelay")) + { + if (attributes.contains("packetsLost")) + { + U32 in_packets_lost = 0; + LLStringUtil::convertToU32(attributes.at("packetsLost"), in_packets_lost); + sample(LLStatViewer::WEBRTC_PACKETS_IN_LOST, in_packets_lost); + } + if (attributes.contains("packetsReceived")) + { + U32 in_packets_recv = 0; + LLStringUtil::convertToU32(attributes.at("packetsReceived"), in_packets_recv); + sample(LLStatViewer::WEBRTC_PACKETS_IN_RECEIVED, in_packets_recv); + } + if (attributes.contains("jitter")) + { + F32 jitter_seconds = 0.0f; + LLStringUtil::convertToF32(attributes.at("jitter"), jitter_seconds); + sample(LLStatViewer::WEBRTC_JITTER_IN, jitter_seconds * 1000.0f); + } + if (attributes.contains("jitterBufferDelay") && attributes.contains("jitterBufferEmittedCount")) + { + F32 total_delay_seconds = 0.0f; + F32 emitted_count_f = 0.0f; + + // total delay in seconds + LLStringUtil::convertToF32(attributes.at("jitterBufferDelay"), total_delay_seconds); + + // number of packets played out + LLStringUtil::convertToF32(attributes.at("jitterBufferEmittedCount"), emitted_count_f); + if (emitted_count_f > 0.0f) + { + F32 avg_delay_seconds = total_delay_seconds / emitted_count_f; + F32 avg_delay_ms = avg_delay_seconds * 1000.0f; + sample(LLStatViewer::WEBRTC_JITTER_BUFFER, avg_delay_seconds * 1000.0f); + } + } + } + } + }); +} + +void LLVoiceWebRTCConnection::resetConnectionStats() +{ + sample(LLStatViewer::WEBRTC_JITTER_BUFFER, 0); + sample(LLStatViewer::WEBRTC_JITTER_IN, 0); + sample(LLStatViewer::WEBRTC_JITTER_OUT, 0); + sample(LLStatViewer::WEBRTC_LATENCY, 0); + sample(LLStatViewer::WEBRTC_PACKETS_IN_LOST, 0); + sample(LLStatViewer::WEBRTC_PACKETS_IN_RECEIVED, 0); + sample(LLStatViewer::WEBRTC_PACKETS_OUT_SENT, 0); + sample(LLStatViewer::WEBRTC_PACKETS_OUT_LOST, 0); + sample(LLStatViewer::WEBRTC_UPLOAD_BANDWIDTH, 0); +} + ///////////////////////////// // WebRTC Spatial Connection |