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

webrtc/test/call_test.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/test/call_test.h"
 
 #include <algorithm>
 
 #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
 #include "webrtc/call/rtp_transport_controller_send.h"
-#include "webrtc/config.h"
+#include "webrtc/call/video_config.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/event.h"
 #include "webrtc/rtc_base/ptr_util.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
 namespace webrtc {
 namespace test {
 
@@ skipping 575 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc