Remove webrtc/test/channel_transport/include

webrtc/tools/e2e_quality/audio/audio_e2e_harness.cc

 /*
  *  Copyright (c) 2012 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.
  */
 
 // Sets up a simple VoiceEngine loopback call with the default audio devices
 // and runs forever. Some parameters can be configured through command-line
 // flags.
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/test/channel_transport/include/channel_transport.h"
+#include "webrtc/test/channel_transport/channel_transport.h"
 #include "webrtc/voice_engine/include/voe_audio_processing.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_hardware.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 
 DEFINE_string(render, "render", "render device name");
 DEFINE_string(codec, "ISAC", "codec name");
 DEFINE_int32(rate, 16000, "codec sample rate in Hz");
 
@@ skipping 69 matching lines @@
   }
 }
 
 }  // namespace test
 }  // namespace webrtc
 
 int main(int argc, char** argv) {
   google::ParseCommandLineFlags(&argc, &argv, true);
   webrtc::test::RunHarness();
 }