Split webrtc/video into webrtc/{audio,call,video}.

webrtc/video/rampup_tests.h

 /*
  *  Copyright (c) 2014 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.
  */
 
 #ifndef WEBRTC_VIDEO_RAMPUP_TESTS_H_
 #define WEBRTC_VIDEO_RAMPUP_TESTS_H_
 
 #include <map>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
+#include "webrtc/call/transport_adapter.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/system_wrappers/interface/event_wrapper.h"
 #include "webrtc/test/call_test.h"
-#include "webrtc/video/transport_adapter.h"
 
 namespace webrtc {
 
 static const int kTransmissionTimeOffsetExtensionId = 6;
 static const int kAbsSendTimeExtensionId = 7;
 static const int kTransportSequenceNumberExtensionId = 8;
 static const unsigned int kSingleStreamTargetBps = 1000000;
 
 class Clock;
 class PacketRouter;
@@ skipping 125 matching lines @@
   void RunRampUpTest(size_t num_streams,
                      unsigned int start_bitrate_bps,
                      const std::string& extension_type,
                      bool rtx,
                      bool red);
 
   void RunRampUpDownUpTest(size_t number_of_streams, bool rtx, bool red);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_RAMPUP_TESTS_H_