From 5bed20f7c67c54940ca53bbcfc2c6be3dd3e6149 Mon Sep 17 00:00:00 2001
From: Per <perkj@chromium.org>
Date: Tue, 23 Aug 2016 22:00:07 +0200
Subject: [PATCH] Do not update stats for
 WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video.

This hopefully fixes a UMA stats  regression introduced in https://chromium.googlesource.com/external/webrtc/+/71ee44cc6d3049763da69e8e42a08d4b796b97db

BUG=webrtc:6244
TBR=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2274713002 .

Cr-Commit-Position: refs/heads/master@{#13868}
---
 webrtc/call/call.cc | 6 ++++++
 1 file changed, 6 insertions(+)

diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index f6354ade28..ceec963ee2 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -735,6 +735,12 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
   // Ignore updates where the bitrate is zero because the aggregate network
   // state is down.
   if (target_bitrate_bps > 0) {
+    {
+      ReadLockScoped read_lock(*send_crit_);
+      // Do not update the stats if we are not sending video.
+      if (video_send_streams_.empty())
+        return;
+    }
     rtc::CritScope lock(&bitrate_crit_);
     // We only update these stats if we have send streams, and assume that
     // OnNetworkChanged is called roughly with a fixed frequency.