modules: more interface -> include renames

webrtc/modules/audio_conference_mixer/source/audio_frame_manipulator.cc

 /*
  *  Copyright (c) 2012 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/audio_conference_mixer/source/audio_frame_manipulator.h"
-#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace {
 // Linear ramping over 80 samples.
 // TODO(hellner): ramp using fix point?
 const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f,
                            0.0506f, 0.0633f, 0.0759f, 0.0886f,
                            0.1013f, 0.1139f, 0.1266f, 0.1392f,
                            0.1519f, 0.1646f, 0.1772f, 0.1899f,
                            0.2025f, 0.2152f, 0.2278f, 0.2405f,
@@ skipping 45 matching lines @@
     {
         const size_t rampPos = rampSize - 1 - i;
         audioFrame.data_[i] = static_cast<int16_t>(rampArray[rampPos] *
                                                    audioFrame.data_[i]);
     }
     memset(&audioFrame.data_[rampSize], 0,
            (audioFrame.samples_per_channel_ - rampSize) *
            sizeof(audioFrame.data_[0]));
 }
 }  // namespace webrtc