modules: more interface -> include renames

webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.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 "webrtc/modules/interface/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/atomic32.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/test/auto_test/fixtures/before_streaming_fixture.h"
 
 using ::testing::_;
 using ::testing::AtLeast;
 using ::testing::Eq;
 using ::testing::Field;
 
 class ExtensionVerifyTransport : public webrtc::Transport {
@@ skipping 123 matching lines @@
   EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAbsoluteSenderTimeStatus(channel_, true,
                                                               3));
   EXPECT_EQ(0, voe_rtp_rtcp_->SetSendAudioLevelIndicationStatus(channel_, true,
                                                                 9));
   verifying_transport_.SetAbsoluteSenderTimeId(3);
   // Don't register audio level with header parser - unknown extensions should
   // be ignored when parsing.
   ResumePlaying();
   EXPECT_TRUE(verifying_transport_.Wait());
 }