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

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/call/audio_sink.h"
 #include "webrtc/api/jsepicecandidate.h"
 #include "webrtc/api/jsepsessiondescription.h"
 #include "webrtc/api/peerconnectioninterface.h"
 #include "webrtc/api/sctputils.h"
 #include "webrtc/api/webrtcsessiondescriptionfactory.h"
-#include "webrtc/audio_sink.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/bind.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/mediaconstants.h"
 #include "webrtc/media/base/videocapturer.h"
@@ skipping 2041 matching lines @@
 }
 
 void WebRtcSession::OnDtlsHandshakeError(rtc::SSLHandshakeError error) {
   if (metrics_observer_) {
     metrics_observer_->IncrementEnumCounter(
         webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
         static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
   }
 }
 }  // namespace webrtc