audio_coding: rename interface -> include

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_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 "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.h"
+#include "webrtc/modules/audio_coding/codecs/opus/include/audio_encoder_opus.h"
 
 namespace webrtc {
 
 namespace {
 const CodecInst kOpusSettings = {105, "opus", 48000, 960, 1, 32000};
 }  // namespace
 
 class AudioEncoderOpusTest : public ::testing::Test {
  protected:
   void CreateCodec(int num_channels) {
@@ skipping 117 matching lines @@
 
   TestSetPacketLossRate(encoder_.get(), I(1.00      , 0.18 + eps), 0.20);
   TestSetPacketLossRate(encoder_.get(), I(0.18 - eps, 0.09 + eps), 0.10);
   TestSetPacketLossRate(encoder_.get(), I(0.09 - eps, 0.04 + eps), 0.05);
   TestSetPacketLossRate(encoder_.get(), I(0.04 - eps, 0.01 + eps), 0.01);
   TestSetPacketLossRate(encoder_.get(), I(0.01 - eps, 0.00      ), 0.00);
   // clang-format on
 }
 
 }  // namespace webrtc