modules: more interface -> include renames

webrtc/modules/video_coding/codecs/h264/h264_video_toolbox_nalu.h

 /*
  *  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.
  *
  */
 
 #ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H
 #define WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H
 
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 
 #if defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
 
 #include <CoreMedia/CoreMedia.h>
 
 #include "webrtc/base/buffer.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 
 namespace webrtc {
 
 // Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
 // suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
 // needs to be in Annex B format. Data is written directly to |annexb_buffer|
 // and a new RTPFragmentationHeader is returned in |out_header|.
 bool H264CMSampleBufferToAnnexBBuffer(
     CMSampleBufferRef avcc_sample_buffer,
     bool is_keyframe,
@@ skipping 58 matching lines @@
  private:
   uint8_t* const start_;
   size_t offset_;
   const size_t length_;
 };
 
 }  // namespace webrtc
 
 #endif  // defined(WEBRTC_VIDEO_TOOLBOX_SUPPORTED)
 #endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_H264_H264_VIDEO_TOOLBOX_NALU_H