Removing the dependency on //third_party/webrtc:webrtc and replacing with rtc_base.

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"
@@ skipping 117 matching lines @@
   int32_t GetLoudspeakerStatus(bool* enabled) const override;
   bool BuiltInAECIsAvailable() const override;
   bool BuiltInAGCIsAvailable() const override;
   bool BuiltInNSIsAvailable() const override;
   int32_t EnableBuiltInAEC(bool enable) override;
   int32_t EnableBuiltInAGC(bool enable) override;
   int32_t EnableBuiltInNS(bool enable) override;
 
 // Only supported on iOS.
 #if defined(WEBRTC_IOS)
-  int GetPlayoutAudioParameters(AudioParameters* params) const override;
-  int GetRecordAudioParameters(AudioParameters* params) const override;
+  int GetPlayoutAudioParameters(webrtc::AudioParameters* params) const override;
+  int GetRecordAudioParameters(webrtc::AudioParameters* params) const override;
 #endif  // WEBRTC_IOS
 
  private:
   void StartPlayoutOnAudioThread();
   void StopPlayoutOnAudioThread();
 
   void PollFromSource();
 
   scoped_refptr<base::SingleThreadTaskRunner> audio_task_runner_;
 
   // |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_|.
   base::RepeatingTimer poll_timer_;
 };
 
 }  // namespace protocol
 }  // namespace remoting
 
 #endif  // REMOTING_PROTOCOL_WEBRTC_AUDIO_MODULE_H_