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

webrtc/api/webrtcsession.cc

 /*
  *  Copyright 2012 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/api/webrtcsession.h"
 
 #include <limits.h>
 
 #include <algorithm>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/api/jsepicecandidate.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/mediaconstraintsinterface.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/sctputils.h"
 #include "webrtc/api/webrtcsessiondescriptionfactory.h"
-#include "webrtc/audio/audio_sink.h"
+#include "webrtc/audio_sink.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/call.h"
 #include "webrtc/media/base/constants.h"
 #include "webrtc/media/base/videocapturer.h"
 #include "webrtc/p2p/base/portallocator.h"
@@ skipping 2082 matching lines @@
   }
 }
 
 void WebRtcSession::OnSentPacket_w(cricket::TransportChannel* channel,
                                    const rtc::SentPacket& sent_packet) {
   RTC_DCHECK(worker_thread()->IsCurrent());
   media_controller_->call_w()->OnSentPacket(sent_packet);
 }
 
 }  // namespace webrtc