Update smoothed bitrate.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index dc93dc7e06c30483a10ee658d78da4d9ca2e3c89..2b8a5e8bac3a68d33fbadbcde1467d0cdcc8ef30 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1345,14 +1345,9 @@ void Channel::SetBitRate(int bitrate_bps, int64_t probing_interval_ms) {
   // the next probing, we would choose a time constant that fulfills
   // 1 - e^(-probing_interval_ms / time_constant) < 0.25
   // Then 4 * probing_interval_ms is a good choice.
+  rtc::CritScope lock(&bitrate_smoother_lock_);
   bitrate_smoother_.SetTimeConstantMs(probing_interval_ms * 4);
   bitrate_smoother_.AddSample(bitrate_bps);
-  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
-    if (*encoder) {
-      (*encoder)->OnReceivedUplinkBandwidth(
-          static_cast<int>(*bitrate_smoother_.GetAverage()));
-    }
-  });
 }
 
 void Channel::OnIncomingFractionLoss(int fraction_lost) {
@@ -2749,6 +2744,16 @@ void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
     audio_coding_->DisableNack();
 }
 
+void Channel::AdaptCodec() {
+  audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+    rtc::CritScope lock(&bitrate_smoother_lock_);
+    if (*encoder) {
+      (*encoder)->OnReceivedUplinkBandwidth(
+          static_cast<int>(*bitrate_smoother_.GetAverage()));
+    }
+  });
+}
+
 // Called when we are missing one or more packets.
 int Channel::ResendPackets(const uint16_t* sequence_numbers, int length) {
   return _rtpRtcpModule->SendNACK(sequence_numbers, length);