audio_coding: rename interface -> include

webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc

 /*
  *  Copyright (c) 2013 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 <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"
+#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 struct WebRtcISACStruct;
 
 namespace webrtc {
 
 // Number of samples in a 60 ms, sampled at 32 kHz.
 const int kIsacNumberOfSamples = 320 * 6;
 // Maximum number of bytes in output bitstream.
 const size_t kMaxBytes = 1000;
@@ skipping 78 matching lines @@
   // Call to update bandwidth estimator with real data.
   EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_,
                                            static_cast<size_t>(encoded_bytes),
                                            1, 12345, 56789));
 
   // Free memory.
   EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
 }
 
 }  // namespace webrtc