Move RtpExtension to api/ directory and config.h/.cc to call/.

webrtc/modules/audio_processing/audio_processing_impl_unittest.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 "webrtc/modules/audio_processing/audio_processing_impl.h"
 
-#include "webrtc/config.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/gmock.h"
 #include "webrtc/test/gtest.h"
 
 using ::testing::Invoke;
 using ::testing::Return;
 
 namespace webrtc {
 namespace {
@@ skipping 47 matching lines @@
   frame.num_channels_ = 2;
   EXPECT_NOERR(mock.ProcessReverseStream(&frame));
 
   // A new sample rate passed to ProcessReverseStream should cause an init.
   SetFrameSampleRate(&frame, 16000);
   EXPECT_CALL(mock, InitializeLocked()).Times(1);
   EXPECT_NOERR(mock.ProcessReverseStream(&frame));
 }
 
 }  // namespace webrtc