diff --git a/webrtc/modules/video_coding/main/source/session_info.cc b/webrtc/modules/video_coding/main/source/session_info.cc
index c981829..361c0a1 100644
--- a/webrtc/modules/video_coding/main/source/session_info.cc
+++ b/webrtc/modules/video_coding/main/source/session_info.cc
@@ -14,18 +14,13 @@
 #include "webrtc/system_wrappers/interface/logging.h"
 
 namespace webrtc {
-namespace {
-// Used in determining whether a frame is decodable.
-enum {kRttThreshold = 100};  // Not decodable if Rtt is lower than this.
 
-// Do not decode frames if the number of packets is between these two
-// thresholds.
-static const float kLowPacketPercentageThreshold = 0.2f;
-static const float kHighPacketPercentageThreshold = 0.8f;
+namespace {
 
 uint16_t BufferToUWord16(const uint8_t* dataBuffer) {
   return (dataBuffer[0] << 8) | dataBuffer[1];
 }
+
 }  // namespace
 
 VCMSessionInfo::VCMSessionInfo()
@@ -233,6 +228,12 @@
     return;
   // TODO(agalusza): Account for bursty loss.
   // TODO(agalusza): Refine these values to better approximate optimal ones.
+  // Do not decode frames if the RTT is lower than this.
+  const int64_t kRttThreshold = 100;
+  // Do not decode frames if the number of packets is between these two
+  // thresholds.
+  const float kLowPacketPercentageThreshold = 0.2f;
+  const float kHighPacketPercentageThreshold = 0.8f;
   if (frame_data.rtt_ms < kRttThreshold
       || frame_type_ == kVideoFrameKey
       || !HaveFirstPacket()