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

webrtc/call/audio_send_stream.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_
 #define WEBRTC_CALL_AUDIO_SEND_STREAM_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_encoder_factory.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/api/call/transport.h"
-#include "webrtc/config.h"
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/call/rtp_config.h"
 #include "webrtc/rtc_base/optional.h"
+#include "webrtc/rtc_base/scoped_ref_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioSendStream {
@@ skipping 114 matching lines @@
   virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
                                   int event, int duration_ms) = 0;
 
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_SEND_STREAM_H_