#5018 add webrtc connection statistics

maxim-productengine · maxim-productengine · commit 90f5ad365142 · 2025-11-26T18:41:30.000+02:00
diff --git a/indra/llwebrtc/llwebrtc.cpp b/indra/llwebrtc/llwebrtc.cpp
@@ -1521,6 +1521,57 @@ void LLWebRTCPeerConnectionImpl::unsetDataObserver(LLWebRTCDataObserver* observe
     }
 }
 
+class LLStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback
+{
+public:
+    typedef std::function<void(const LLWebRTCStatsMap&)> StatsCallback;
+
+    LLStatsCollectorCallback(StatsCallback callback) : callback_(callback) {}
+
+    void OnStatsDelivered(const webrtc::scoped_refptr<const webrtc::RTCStatsReport>& report) override
+    {
+        if (callback_)
+        {
+            // Transform RTCStatsReport stats to simple map
+            LLWebRTCStatsMap stats_map;
+            for (const auto& stats : *report)
+            {
+                std::map<std::string, std::string> stat_attributes;
+
+                // Convert each attribute to string format
+                for (const auto& attribute : stats.Attributes())
+                {
+                    stat_attributes[attribute.name()] = attribute.ToString();
+                }
+                stats_map[stats.id()] = stat_attributes;
+            }
+            callback_(stats_map);
+        }
+    }
+
+private:
+    StatsCallback callback_;
+};
+
+void LLWebRTCPeerConnectionImpl::gatherConnectionStats()
+{
+    if (!mPeerConnection)
+    {
+        return;
+    }
+
+    auto stats_callback = webrtc::make_ref_counted<LLStatsCollectorCallback>(
+        [this](const LLWebRTCStatsMap& generic_stats)
+        {
+            for (auto& observer : mSignalingObserverList)
+            {
+                observer->OnStatsDelivered(generic_stats);
+            }
+        });
+
+     mPeerConnection->GetStats(stats_callback.get());
+}
+
 LLWebRTCImpl * gWebRTCImpl = nullptr;
 LLWebRTCDeviceInterface * getDeviceInterface()
 {
diff --git a/indra/llwebrtc/llwebrtc.h b/indra/llwebrtc/llwebrtc.h
@@ -38,6 +38,7 @@
 #ifndef LLWEBRTC_H
 #define LLWEBRTC_H
 
+#include <map>
 #include <string>
 #include <vector>
 
@@ -55,6 +56,7 @@
 
 namespace llwebrtc
 {
+typedef std::map<std::string, std::map<std::string, std::string>> LLWebRTCStatsMap;
 
 class LLWebRTCLogCallback
 {
@@ -240,6 +242,8 @@ class LLWebRTCSignalingObserver
     // Called when the data channel has been established and data
     // transfer can begin.
     virtual void OnDataChannelReady(LLWebRTCDataInterface *data_interface) = 0;
+
+    virtual void OnStatsDelivered(const LLWebRTCStatsMap& stats_data) {}
 };
 
 // LLWebRTCPeerConnectionInterface representsd a connection to a peer,
@@ -273,6 +277,8 @@ class LLWebRTCPeerConnectionInterface
     virtual void unsetSignalingObserver(LLWebRTCSignalingObserver* observer) = 0;
 
     virtual void AnswerAvailable(const std::string &sdp) = 0;
+
+    virtual void gatherConnectionStats() = 0;
 };
 
 // The following define the dynamic linked library
diff --git a/indra/llwebrtc/llwebrtc_impl.h b/indra/llwebrtc/llwebrtc_impl.h
@@ -648,6 +648,8 @@ class LLWebRTCPeerConnectionImpl : public LLWebRTCPeerConnectionInterface,
     void enableSenderTracks(bool enable);
     void enableReceiverTracks(bool enable);
 
+    void gatherConnectionStats();
+
   protected:
 
     LLWebRTCImpl * mWebRTCImpl;
diff --git a/indra/newview/app_settings/settings.xml b/indra/newview/app_settings/settings.xml
@@ -6060,6 +6060,39 @@
       <key>Value</key>
       <integer>0</integer>
     </map>
+    <key>OpenDebugStatVoice</key>
+    <map>
+      <key>Comment</key>
+      <string>Expand Voice (WebRTC) stats display</string>
+      <key>Persist</key>
+      <integer>1</integer>
+      <key>Type</key>
+      <string>Boolean</string>
+      <key>Value</key>
+      <integer>1</integer>
+    </map>
+    <key>OpenDebugStatVoiceOutgoing</key>
+    <map>
+      <key>Comment</key>
+      <string>Expand Outgoing audio (Voice) stats display</string>
+      <key>Persist</key>
+      <integer>1</integer>
+      <key>Type</key>
+      <string>Boolean</string>
+      <key>Value</key>
+      <integer>1</integer>
+    </map>
+    <key>OpenDebugStatVoiceIncoming</key>
+    <map>
+      <key>Comment</key>
+      <string>Expand Incoming audio (Voice) stats display</string>
+      <key>Persist</key>
+      <integer>1</integer>
+      <key>Type</key>
+      <string>Boolean</string>
+      <key>Value</key>
+      <integer>1</integer>
+    </map>
     <key>OutBandwidth</key>
     <map>
       <key>Comment</key>
diff --git a/indra/newview/llviewerstats.cpp b/indra/newview/llviewerstats.cpp
@@ -263,6 +263,20 @@ LLTrace::SampleStatHandle<LLUnit<F32, LLUnits::Percent> >  HUDS_FRAME_PCT("huds_
 LLTrace::SampleStatHandle<LLUnit<F32, LLUnits::Percent> >  UI_FRAME_PCT("ui_frame_pct");
 LLTrace::SampleStatHandle<LLUnit<F32, LLUnits::Percent> >  SWAP_FRAME_PCT("swap_frame_pct");
 LLTrace::SampleStatHandle<LLUnit<F32, LLUnits::Percent> >  IDLE_FRAME_PCT("idle_frame_pct");
+
+
+
+LLTrace::SampleStatHandle<U32> WEBRTC_PACKETS_IN_LOST("webrtc_packets_in_lost", "Lost incoming packets"),
+    WEBRTC_PACKETS_IN_RECEIVED("webrtc_packets_in_recv", "Incoming packets received"),
+    WEBRTC_PACKETS_OUT_SENT("webrtc_packets_out_sent", "Outgoing packets sent"),
+    WEBRTC_PACKETS_OUT_LOST("webrtc_packets_out_lost", "Lost outgoing packets");
+
+LLTrace::SampleStatHandle<F32> WEBRTC_JITTER_OUT("webrtc_jitter_out", "Timing variation of outgoing audio"),
+    WEBRTC_JITTER_IN("webrtc_jitter_in", "Timing variation of incoming audio"),
+    WEBRTC_LATENCY("webrtc_latency", "Round-trip audio delay"),
+    WEBRTC_UPLOAD_BANDWIDTH("webrtc_upload_bandwidth", "Estimated upload bandwidth"),
+    WEBRTC_JITTER_BUFFER("webrtc_jitter_buffer", "Average delay added to smooth incoming audio");
+
 }
 
 LLViewerStats::LLViewerStats()
diff --git a/indra/newview/llviewerstats.h b/indra/newview/llviewerstats.h
@@ -229,6 +229,8 @@ extern LLTrace::EventStatHandle<F64Seconds >    AVATAR_EDIT_TIME,
 
 extern LLTrace::EventStatHandle<LLUnit<F32, LLUnits::Percent> > OBJECT_CACHE_HIT_RATE;
 
+extern LLTrace::SampleStatHandle<U32> WEBRTC_PACKETS_IN_LOST, WEBRTC_PACKETS_IN_RECEIVED, WEBRTC_PACKETS_OUT_SENT, WEBRTC_PACKETS_OUT_LOST;
+extern LLTrace::SampleStatHandle<F32> WEBRTC_JITTER_OUT, WEBRTC_JITTER_IN, WEBRTC_LATENCY, WEBRTC_UPLOAD_BANDWIDTH, WEBRTC_JITTER_BUFFER;
 }
 
 class LLViewerStats : public LLSingleton<LLViewerStats>
diff --git a/indra/newview/llvoicewebrtc.cpp b/indra/newview/llvoicewebrtc.cpp
@@ -61,6 +61,7 @@
 #include "llrand.h"
 #include "llviewerwindow.h"
 #include "llviewercamera.h"
+#include "llviewerstats.h"
 #include "llversioninfo.h"
 
 #include "llviewernetwork.h"
@@ -81,6 +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;
@@ -2905,6 +2908,7 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
             }
             mWebRTCAudioInterface->setReceiveVolume(mSpeakerVolume);
             LLWebRTCVoiceClient::getInstance()->OnConnectionEstablished(mChannelID, mRegionID);
+            resetConnectionStats();
             setVoiceConnectionState(VOICE_STATE_WAIT_FOR_DATA_CHANNEL);
             break;
         }
@@ -2954,6 +2958,13 @@ bool LLVoiceWebRTCConnection::connectionStateMachine()
                         sendJoin();
                     }
                 }
+
+                static LLTimer stats_timer;
+                if (stats_timer.getElapsedTimeF32() > STATS_TIMER_DELAY)
+                {
+                    mWebRTCPeerConnectionInterface->gatherConnectionStats();
+                    stats_timer.reset();
+                }
             }
             break;
         }
@@ -3288,6 +3299,112 @@ 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
 
diff --git a/indra/newview/llvoicewebrtc.h b/indra/newview/llvoicewebrtc.h
@@ -540,6 +540,8 @@ class LLWebRTCVoiceClient : public LLSingleton<LLWebRTCVoiceClient>,
     static bool sShuttingDown;
 
     LLEventMailDrop mWebRTCPump;
+
+    LLSD mLastWebRTCStats;
 };
 
 
@@ -603,6 +605,8 @@ class LLVoiceWebRTCConnection :
     //@{
     void OnDataReceived(const std::string &data, bool binary) override;
     void OnDataChannelReady(llwebrtc::LLWebRTCDataInterface *data_interface) override;
+
+    void OnStatsDelivered(const llwebrtc::LLWebRTCStatsMap& stats_data) override;
     //@}
 
     void OnDataReceivedImpl(const std::string &data, bool binary);
@@ -638,6 +642,8 @@ class LLVoiceWebRTCConnection :
 
     void OnVoiceConnectionRequestSuccess(const LLSD &body);
 
+    void resetConnectionStats();
+
   protected:
     typedef enum e_voice_connection_state
     {
diff --git a/indra/newview/skins/default/xui/en/floater_stats.xml b/indra/newview/skins/default/xui/en/floater_stats.xml

Original file line number	Diff line number	Diff line change
`@@ -229,6 +229,8 @@ extern LLTrace::EventStatHandle<F64Seconds > AVATAR_EDIT_TIME,`
`229`	`229`
`230`	`230`	`extern LLTrace::EventStatHandle<LLUnit<F32, LLUnits::Percent> > OBJECT_CACHE_HIT_RATE;`
`231`	`231`
	`232`	`+extern LLTrace::SampleStatHandle<U32> WEBRTC_PACKETS_IN_LOST, WEBRTC_PACKETS_IN_RECEIVED, WEBRTC_PACKETS_OUT_SENT, WEBRTC_PACKETS_OUT_LOST;`
	`233`	`+extern LLTrace::SampleStatHandle<F32> WEBRTC_JITTER_OUT, WEBRTC_JITTER_IN, WEBRTC_LATENCY, WEBRTC_UPLOAD_BANDWIDTH, WEBRTC_JITTER_BUFFER;`
`232`	`234`	`}`
`233`	`235`
`234`	`236`	`class LLViewerStats : public LLSingleton<LLViewerStats>`