Delete empty API files and cleaned up includes.

webrtc/api/rtpsender.cc

 /*
  *  Copyright 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/api/rtpsender.h"
 
 #include "webrtc/api/localaudiosource.h"
-#include "webrtc/api/videosourceinterface.h"
+#include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/base/helpers.h"
 
 namespace webrtc {
 
 LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
 
 LocalAudioSinkAdapter::~LocalAudioSinkAdapter() {
   rtc::CritScope lock(&lock_);
   if (sink_)
     sink_->OnClose();
@@ skipping 315 matching lines @@
 
 RtpParameters VideoRtpSender::GetParameters() const {
   return provider_->GetVideoRtpParameters(ssrc_);
 }
 
 bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
   return provider_->SetVideoRtpParameters(ssrc_, parameters);
 }
 
 }  // namespace webrtc