Update smoothed bitrate.

webrtc/call/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.
  */
 
@@ skipping 82 matching lines @@
     // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
     // disable audio bitrate adaptation.
     // Note: This is still an experimental feature and not ready for real usage.
     int min_bitrate_bps = -1;
     int max_bitrate_bps = -1;
 
     // Defines whether to turn on audio network adaptor, and defines its config
     // string.
     rtc::Optional<std::string> audio_network_adaptor_config;
 
+    // Interval in which adapt codec is called. Default is an interval of 200ms.
+    uint32_t adapt_codec_interval_ms = 200;
+
     struct SendCodecSpec {
       SendCodecSpec();
       std::string ToString() const;
 
       bool operator==(const SendCodecSpec& rhs) const;
       bool operator!=(const SendCodecSpec& rhs) const {
         return !(*this == rhs);
       }
 
       bool nack_enabled = false;
@@ skipping 23 matching lines @@
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_