Merge pull request #5843 from secondlife/geenz/26.2-to-26.3

26.2 to 26.3

1 files changed, 14 insertions, 0 deletions

diff --git a/indra/newview/llviewerstats.cpp b/indra/newview/llviewerstats.cpp
index 4dcd1cc03a..6490fe76a2 100644
--- a/indra/newview/llviewerstats.cpp
+++ b/indra/newview/llviewerstats.cpp
@@ -262,6 +262,20 @@ 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()