Update includes for webrtc/{base => rtc_base} rename (2/3)

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"
 
 #include <algorithm>
 #include <cmath>
 #include <sstream>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/file.h"
-#include "webrtc/base/location.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/trace_event.h"
-#include "webrtc/base/weak_ptr.h"
 #include "webrtc/call/rtp_transport_controller_send_interface.h"
 #include "webrtc/common_types.h"
 #include "webrtc/common_video/include/video_bitrate_allocator.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #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"
 
 namespace webrtc {
 
 static const int kMinSendSidePacketHistorySize = 600;
 namespace {
 
@@ skipping 1296 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