Relanding of "Adding debug dump to audio network adaptor."

webrtc/modules/audio_coding/audio_network_adaptor/controller.h

 /*
  *  Copyright (c) 2016 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_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
 
 #include "webrtc/base/optional.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 
 namespace webrtc {
 
 class Controller {
  public:
   struct NetworkMetrics {
     NetworkMetrics();
     ~NetworkMetrics();
     rtc::Optional<int> uplink_bandwidth_bps;
     rtc::Optional<float> uplink_packet_loss_fraction;
     rtc::Optional<int> target_audio_bitrate_bps;
+    rtc::Optional<int> rtt_ms;
   };
 
   struct Constraints {
     Constraints();
     ~Constraints();
     struct FrameLengthRange {
       FrameLengthRange(int min_frame_length_ms, int max_frame_length_ms);
       ~FrameLengthRange();
       int min_frame_length_ms;
       int max_frame_length_ms;
     };
     rtc::Optional<FrameLengthRange> receiver_frame_length_range;
   };
 
   virtual ~Controller() = default;
 
   virtual void MakeDecision(
       const NetworkMetrics& metrics,
       AudioNetworkAdaptor::EncoderRuntimeConfig* config) = 0;
 
   virtual void SetConstraints(const Constraints& constraints);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_