Revert 25680c62320767f590d037d301edfe15e9c55650

remoting/protocol/webrtc_audio_module.h

 // Copyright 2016 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_
 #define REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_
 
 #include "base/memory/ref_counted.h"
 #include "base/synchronization/lock.h"
+#include "base/timer/timer.h"
 #include "third_party/webrtc/modules/audio_device/include/audio_device.h"
 
 namespace base {
-class RepeatingTimer;
 class SingleThreadTaskRunner;
 }  // namespace base
 
 namespace remoting {
 namespace protocol {
 
 // Audio module passed to WebRTC. It doesn't access actual audio devices, but it
 // provides all functionality we need to ensure that audio streaming works
 // properly in WebRTC. Particularly it's responsible for calling AudioTransport
 // on regular intervals when playback is active. This ensures that all incoming
@@ skipping 124 matching lines @@
   // |lock_| must be locked when accessing |initialized_|, |playing_| and
   // |audio_transport_|.
   mutable base::Lock lock_;
 
   bool initialized_ = false;
   bool playing_ = false;
   webrtc::AudioTransport* audio_transport_ = nullptr;
 
   // Timer running on the |audio_task_runner_| that polls audio from
   // |audio_transport_|.
-  std::unique_ptr<base::RepeatingTimer> poll_timer_;
+  base::RepeatingTimer poll_timer_;
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_