Update smoothed bitrate.

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

 /*
  *  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 47 matching lines @@
                          rtc::Buffer* encoded) override;
   void Reset() override;
   bool SetFec(bool enable) override;
   bool SetDtx(bool enable) override;
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
   rtc::ArrayView<std::unique_ptr<AudioEncoder>> ReclaimContainedEncoders()
       override;
   void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) override;
-  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedTargetAudioBitrate(
+      int target_audio_bitrate_bps,
+      rtc::Optional<int64_t> probing_interval_ms) override;
 
  private:
   EncodedInfo EncodePassive(size_t frames_to_encode,
                             rtc::Buffer* encoded);
   EncodedInfo EncodeActive(size_t frames_to_encode,
                            rtc::Buffer* encoded);
   size_t SamplesPer10msFrame() const;
 
   std::unique_ptr<AudioEncoder> speech_encoder_;
   const int cng_payload_type_;
   const int num_cng_coefficients_;
   const int sid_frame_interval_ms_;
   std::vector<int16_t> speech_buffer_;
   std::vector<uint32_t> rtp_timestamps_;
   bool last_frame_active_;
   std::unique_ptr<Vad> vad_;
   std::unique_ptr<ComfortNoiseEncoder> cng_encoder_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_