modules: more interface -> include renames

webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.cc

 /*
  *  Copyright (c) 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 <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
 #include "webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
-#include "webrtc/modules/utility/interface/process_thread.h"
+#include "webrtc/modules/utility/include/process_thread.h"
 
 namespace webrtc {
 
 const int64_t kNoTimestamp = -1;
 const int64_t kSendTimeHistoryWindowMs = 10000;
 const int64_t kBaseTimestampScaleFactor =
     rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
 
 TransportFeedbackAdapter::TransportFeedbackAdapter(
@@ skipping 97 matching lines @@
   rtcp_bandwidth_observer_->OnReceivedEstimatedBitrate(bitrate);
 }
 
 void TransportFeedbackAdapter::OnRttUpdate(int64_t avg_rtt_ms,
                                            int64_t max_rtt_ms) {
   RTC_DCHECK(bitrate_estimator_.get() != nullptr);
   bitrate_estimator_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
 }
 
 }  // namespace webrtc