Update smoothed bitrate.

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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 314 matching lines @@
                                        const char* data,
                                        unsigned short dataLengthInBytes);
   int GetRTPStatistics(unsigned int& averageJitterMs,
                        unsigned int& maxJitterMs,
                        unsigned int& discardedPackets);
   int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
   int GetRTPStatistics(CallStatistics& stats);
   int SetCodecFECStatus(bool enable);
   bool GetCodecFECStatus();
   void SetNACKStatus(bool enable, int maxNumberOfPackets);
+  void UpdateSmoothedBitrate();
 
   // From AudioPacketizationCallback in the ACM
   int32_t SendData(FrameType frameType,
                    uint8_t payloadType,
                    uint32_t timeStamp,
                    const uint8_t* payloadData,
                    size_t payloadSize,
                    const RTPFragmentationHeader* fragmentation) override;
 
   // From ACMVADCallback in the ACM
@@ skipping 194 matching lines @@
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 
   SmoothingFilterImpl bitrate_smoother_;
+  rtc::Optional<int> last_bitrate_bps_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_