Update smoothed bitrate.

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 173 matching lines @@
   return rtc::ArrayView<std::unique_ptr<AudioEncoder>>(&speech_encoder_, 1);
 }
 
 void AudioEncoderCng::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
   speech_encoder_->OnReceivedUplinkPacketLossFraction(
       uplink_packet_loss_fraction);
 }
 
 void AudioEncoderCng::OnReceivedTargetAudioBitrate(
-    int target_audio_bitrate_bps) {
-  speech_encoder_->OnReceivedTargetAudioBitrate(target_audio_bitrate_bps);
+    int target_audio_bitrate_bps,
+    rtc::Optional<int64_t> probing_interval_ms) {
+  speech_encoder_->OnReceivedTargetAudioBitrate(target_audio_bitrate_bps,
+                                                probing_interval_ms);
 }
 
 AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
     size_t frames_to_encode,
     rtc::Buffer* encoded) {
   bool force_sid = last_frame_active_;
   bool output_produced = false;
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   AudioEncoder::EncodedInfo info;
 
@@ skipping 44 matching lines @@
     }
   }
   return info;
 }
 
 size_t AudioEncoderCng::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
 }
 
 }  // namespace webrtc