Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)"

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.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/modules/rtp_rtcp/source/rtp_receiver_impl.h"
 
 #include <assert.h>
 #include <math.h>
 #include <stdlib.h>
 #include <string.h>
 
 #include <set>
 #include <vector>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
-#include "webrtc/rtc_base/logging.h"
 
 namespace webrtc {
 
 using RtpUtility::Payload;
 
 // Only return the sources in the last 10 seconds.
 const int64_t kGetSourcesTimeoutMs = 10000;
 
 RtpReceiver* RtpReceiver::CreateVideoReceiver(
     Clock* clock,
@@ skipping 508 matching lines @@
   for (vec_it = ssrc_sources_.begin(); vec_it != ssrc_sources_.end();
        ++vec_it) {
     if ((now_ms - vec_it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
       break;
     }
   }
   ssrc_sources_.erase(ssrc_sources_.begin(), vec_it);
 }
 
 }  // namespace webrtc