Move RtpExtension to api/ directory and config.h/.cc to call/.

webrtc/video/payload_router_unittest.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 <memory>
 
+#include "webrtc/call/video_config.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/video_coding/include/video_codec_interface.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/video/payload_router.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 using ::testing::NiceMock;
@@ skipping 190 matching lines @@
   bitrate.SetBitrate(0, 1, 20000);
   bitrate.SetBitrate(1, 0, 40000);
   bitrate.SetBitrate(1, 1, 80000);
 
   EXPECT_CALL(rtp_1, SetVideoBitrateAllocation(bitrate)).Times(1);
 
   payload_router.OnBitrateAllocationUpdated(bitrate);
 }
 
 }  // namespace webrtc