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

webrtc/modules/audio_processing/audio_processing_performance_unittest.cc

 /*
  *  Copyright (c) 2015 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 <math.h>
 
 #include <algorithm>
 #include <memory>
 #include <vector>
 
-#include "webrtc/config.h"
 #include "webrtc/modules/audio_processing/test/test_utils.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/rtc_base/array_view.h"
 #include "webrtc/rtc_base/atomicops.h"
 #include "webrtc/rtc_base/platform_thread.h"
 #include "webrtc/rtc_base/random.h"
 #include "webrtc/rtc_base/safe_conversions.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/gtest.h"
@@ skipping 689 matching lines @@
   // Run test and verify that it did not time out.
   EXPECT_EQ(kEventSignaled, Run());
 }
 
 INSTANTIATE_TEST_CASE_P(
     AudioProcessingPerformanceTest,
     CallSimulator,
     ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
 
 }  // namespace webrtc