audio_coding: rename interface -> include

webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.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/audio_coding/main/acm2/acm_receive_test_oldapi.h"
 
 #include <assert.h>
 #include <stdio.h>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
+#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 // Returns true if the codec should be registered, otherwise false. Changes
 // the number of channels for the Opus codec to always be 1.
@@ skipping 184 matching lines @@
   if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
     output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
                           ? output_freq_hz_2_
                           : output_freq_hz_1_;
     last_toggle_time_ms_ = clock_.TimeInMilliseconds();
   }
 }
 
 }  // namespace test
 }  // namespace webrtc