Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies.

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  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.
  */
 
 #include "webrtc/media/engine/fakewebrtccall.h"
 
 #include <algorithm>
 #include <utility>
 
-#include "webrtc/audio_sink.h"
+#include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/media/base/rtputils.h"
 
 namespace cricket {
 FakeAudioSendStream::FakeAudioSendStream(
     const webrtc::AudioSendStream::Config& config) : config_(config) {
   RTC_DCHECK(config.voe_channel_id != -1);
 }
 
@@ skipping 390 matching lines @@
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket