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

webrtc/pc/channel.cc

 /*
  *  Copyright 2004 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 <utility>
 
 #include "webrtc/pc/channel.h"
 
-#include "webrtc/audio_sink.h"
+#include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/media/base/mediaconstants.h"
@@ skipping 2384 matching lines @@
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
   rtc::TypedMessageData<uint32_t>* message =
       new rtc::TypedMessageData<uint32_t>(sid);
   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_STREAMCLOSEDREMOTELY,
                            message);
 }
 
 }  // namespace cricket