Move webrtc/audio_*.h to webrtc/api/call

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/api/call/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 452 matching lines @@
 }
 
 bool FakeCall::StartEventLog(rtc::PlatformFile log_file,
                              int64_t max_size_bytes) {
   return false;
 }
 
 void FakeCall::StopEventLog() {}
 
 }  // namespace cricket