Moved call.h and most of api/call/* into call/

webrtc/audio/audio_state.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_STATE_H_
 #define WEBRTC_AUDIO_AUDIO_STATE_H_
 
-#include "webrtc/api/call/audio_state.h"
 #include "webrtc/audio/audio_transport_proxy.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_checker.h"
+#include "webrtc/call/audio_state.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
 namespace webrtc {
 namespace internal {
 
 class AudioState final : public webrtc::AudioState,
                          public webrtc::VoiceEngineObserver {
  public:
   explicit AudioState(const AudioState::Config& config);
   ~AudioState() override;
@@ skipping 29 matching lines @@
   // Transports mixed audio from the mixer to the audio device and
   // recorded audio to the VoE AudioTransport.
   AudioTransportProxy audio_transport_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_STATE_H_