audio_coding: rename interface -> include

webrtc/modules/audio_coding/codecs/isac/unittest.cc

 /*
  *  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.
  */
 
 #include <algorithm>
 #include <numeric>
 #include <sstream>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/buffer.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 namespace {
 
 const int kIsacNumberOfSamples = 32 * 60;  // 60 ms at 32 kHz
 
 std::vector<int16_t> LoadSpeechData() {
@@ skipping 215 matching lines @@
             for (int frame_size : frame_sizes)
               if (!(sample_rate == 32000 && frame_size == 60))
                 cases.push_back(
                     {type, adaptive, channel_rate, sample_rate, frame_size});
   return cases;
 }
 
 INSTANTIATE_TEST_CASE_P(, IsacCommonTest, testing::ValuesIn(TestCases()));
 
 }  // namespace webrtc