modules: more interface -> include renames

webrtc/modules/audio_processing/audio_buffer.h

 /*
  *  Copyright (c) 2011 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_AUDIO_PROCESSING_AUDIO_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
 
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/channel_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/splitting_filter.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/scoped_vector.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class PushSincResampler;
 class IFChannelBuffer;
 
 enum Band {
   kBand0To8kHz = 0,
@@ skipping 125 matching lines @@
   rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
   rtc::scoped_ptr<IFChannelBuffer> output_buffer_;
   rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
   ScopedVector<PushSincResampler> input_resamplers_;
   ScopedVector<PushSincResampler> output_resamplers_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_