Add RtcpRttStats to AudioStream

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/api/call/audio_send_stream.h"
 #include "webrtc/api/call/audio_state.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/call/bitrate_allocator.h"
 
 namespace webrtc {
 class CongestionController;
 class VoiceEngine;
 class RtcEventLog;
+class RtcpRttStats;
 
 namespace voe {
 class ChannelProxy;
 }  // namespace voe
 
 namespace internal {
 class AudioSendStream final : public webrtc::AudioSendStream,
                               public webrtc::BitrateAllocatorObserver {
  public:
   AudioSendStream(const webrtc::AudioSendStream::Config& config,
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
                   CongestionController* congestion_controller,
                   BitrateAllocator* bitrate_allocator,
-                  RtcEventLog* event_log);
+                  RtcEventLog* event_log,
+                  RtcpRttStats* rtcp_rtt_stats);
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
   void Start() override;
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
@@ skipping 17 matching lines @@
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_