Update smoothed bitrate.

webrtc/audio/audio_send_stream.h

Index: webrtc/audio/audio_send_stream.h
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 23a748501438178c1bfce5c1b9330be3a56f7314..a732f48f0f092b95a165422bee154dcdb7a2c30c 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -16,7 +16,9 @@
 #include "webrtc/api/call/audio_send_stream.h"
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/onetimeevent.h"
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/base/weak_ptr.h"
 #include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
@@ -65,9 +67,12 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   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_;
@@ -76,6 +81,10 @@ class AudioSendStream final : public webrtc::AudioSendStream,
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
+  OneTimeEvent first_update_bitrate_;
+
+  rtc::WeakPtr<AudioSendStream> weak_ptr_;
+  std::unique_ptr<rtc::WeakPtrFactory<AudioSendStream>> weak_ptr_factory_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };