Removing dependencies on stub headers within WebRTC.

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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/video/video_send_stream.h"
@@ skipping 18 matching lines @@
 #include "webrtc/modules/video_coding/utility/ivf_file_writer.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/file.h"
 #include "webrtc/rtc_base/location.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/trace_event.h"
 #include "webrtc/rtc_base/weak_ptr.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/payload_router.h"
-#include "webrtc/video_send_stream.h"
+#include "webrtc/call/video_send_stream.h"
 
 namespace webrtc {
 
 static const int kMinSendSidePacketHistorySize = 600;
 namespace {
 
 // We don't do MTU discovery, so assume that we have the standard ethernet MTU.
 const size_t kPathMTU = 1500;
 
 std::vector<RtpRtcp*> CreateRtpRtcpModules(
@@ skipping 1198 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc