modules: more interface -> include renames

webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

 /*
  *  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.
  */
 
 #include <vector>
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
 
 namespace webrtc {
 namespace {
 const size_t kMaxPayloadSize = 1200;
 const size_t kLengthFieldLength = 2;
 
 enum Nalu {
   kSlice = 1,
@@ skipping 536 matching lines @@
   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
 }
 
 TEST_F(RtpDepacketizerH264Test, TestShortSpsPacket) {
   const uint8_t kPayload[] = {0x27, 0x80, 0x00};
   RtpDepacketizer::ParsedPayload payload;
   EXPECT_TRUE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
 }
 
 }  // namespace webrtc