audio_coding: rename interface -> include

webrtc/modules/audio_coding/main/test/insert_packet_with_timing.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 <stdio.h>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.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/main/test/Channel.h"
 #include "webrtc/modules/audio_coding/main/test/PCMFile.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 // Codec.
 DEFINE_string(codec, "opus", "Codec Name");
 DEFINE_int32(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
 DEFINE_int32(codec_channels, 1, "Number of channels of the codec.");
@@ skipping 274 matching lines @@
       if (delay_log != NULL) {
         fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
       }
     }
   }
   std::cout << std::endl;
   test.TearDown();
   if (delay_log != NULL)
     fclose(delay_log);
 }