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

webrtc/video/rtp_streams_synchronizer.cc

 /*
  *  Copyright (c) 2012 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/rtp_streams_synchronizer.h"
 
-#include "webrtc/base/checks.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/trace_event.h"
 #include "webrtc/call/syncable.h"
 #include "webrtc/modules/video_coding/video_coding_impl.h"
+#include "webrtc/rtc_base/checks.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/timeutils.h"
+#include "webrtc/rtc_base/trace_event.h"
 
 namespace webrtc {
 namespace {
 bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
                         const Syncable::Info& info) {
   RTC_DCHECK(stream);
   stream->latest_timestamp = info.latest_received_capture_timestamp;
   stream->latest_receive_time_ms = info.latest_receive_time_ms;
   bool new_rtcp_sr = false;
   if (!stream->rtp_to_ntp.UpdateMeasurements(info.capture_time_ntp_secs,
@@ skipping 116 matching lines @@
   int64_t time_to_render_ms = render_time_ms - rtc::TimeMillis();
   if (time_to_render_ms > 0)
     latest_video_ntp += time_to_render_ms;
 
   *stream_offset_ms = latest_audio_ntp - latest_video_ntp;
   *estimated_freq_khz = video_measurement_.rtp_to_ntp.params().frequency_khz;
   return true;
 }
 
 }  // namespace webrtc