audio_coding: rename interface -> include

webrtc/modules/audio_coding/codecs/opus/opus_fec_test.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 "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
+#include "webrtc/modules/audio_coding/codecs/opus/include/opus_interface.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 using ::std::string;
 using ::std::tr1::tuple;
 using ::std::tr1::get;
 using ::testing::TestWithParam;
 
 namespace webrtc {
 
 // Define coding parameter as <channels, bit_rate, filename, extension>.
@@ skipping 207 matching lines @@
      ::std::tr1::make_tuple(1, 32000, string("audio_coding/testfile32kHz"),
                             string("pcm")),
      ::std::tr1::make_tuple(2, 64000, string("audio_coding/teststereo32kHz"),
                             string("pcm"))};
 
 // 64 kbps, stereo
 INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest,
                         ::testing::ValuesIn(param_set));
 
 }  // namespace webrtc