Update smoothed bitrate.

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/audio_state.h"
+#include "webrtc/base/weak_ptr.h"
 #include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
 class CongestionController;
 class VoiceEngine;
 class RtcEventLog;
 class RtcpRttStats;
 class PacketRouter;
 
 namespace voe {
@@ skipping 28 matching lines @@
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
                             int64_t rtt,
                             int64_t probing_interval_ms) override;
 
   const webrtc::AudioSendStream::Config& config() const;
   void SetTransportOverhead(int transport_overhead_per_packet);
 
  private:
+  class AdaptCodecTask;
+
   VoiceEngine* voice_engine() const;
 
   bool SetupSendCodec();
+  void AdaptCodec();
 
   rtc::ThreadChecker thread_checker_;
   rtc::TaskQueue* worker_queue_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
 
+  std::unique_ptr<rtc::WeakPtrFactory<AudioSendStream>> weak_ptr_factory_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_