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..97f3caf932a48b790c333884dfc0caf6656943dd 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1317,13 +1317,7 @@ void Channel::SetBitRate(int bitrate_bps) {
   // we pass the probing interval to this function.
   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()));
-    }
-  });
+  last_bitrate_bps_ = rtc::Optional<int>(bitrate_bps);
 }
 
 void Channel::OnIncomingFractionLoss(int fraction_lost) {
@@ -2720,6 +2714,18 @@ void Channel::SetNACKStatus(bool enable, int maxNumberOfPackets) {
     audio_coding_->DisableNack();
 }
 
+void Channel::UpdateSmoothedBitrate() {
+  if (last_bitrate_bps_) {
+    bitrate_smoother_.AddSample(*last_bitrate_bps_);
+    audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
+      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);