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 975d56aa986538e0522a4fce38d358b455067ca5..c1e3704fda059cc7e01d72c7d0d1f522371cce6d 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1318,12 +1318,6 @@ void Channel::SetBitRate(int bitrate_bps) {
   constexpr int64_t kDefaultBitrateSmoothingTimeConstantMs = 20000;
   bitrate_smoother_.SetTimeConstantMs(kDefaultBitrateSmoothingTimeConstantMs);
   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) {
@@ -2720,6 +2714,18 @@ void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
     audio_coding_->DisableNack();
 }
 
+void Channel::AdaptCodec() {
+  rtc::Optional<float> smoothed_bitrate = bitrate_smoother_.GetAverage();
+  if (smoothed_bitrate) {
+    audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+      if (*encoder) {
+        (*encoder)->OnReceivedUplinkBandwidth(
+            static_cast<int>(*smoothed_bitrate));
+      }
+    });
+  }
+}
+
 // 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);