Reland of move ignored return code from modules.

webrtc/video/call_stats.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.
  */
 
@@ skipping 91 matching lines @@
 
 CallStats::~CallStats() {
   RTC_DCHECK(observers_.empty());
   UpdateHistograms();
 }
 
 int64_t CallStats::TimeUntilNextProcess() {
   return last_process_time_ + kUpdateIntervalMs - clock_->TimeInMilliseconds();
 }
 
-int32_t CallStats::Process() {
+void CallStats::Process() {
   rtc::CritScope cs(&crit_);
   int64_t now = clock_->TimeInMilliseconds();
   if (now < last_process_time_ + kUpdateIntervalMs)
-    return 0;
+    return;
 
   last_process_time_ = now;
 
   RemoveOldReports(now, &reports_);
   max_rtt_ms_ = GetMaxRttMs(&reports_);
   UpdateAvgRttMs(&reports_, &avg_rtt_ms_);
 
   // If there is a valid rtt, update all observers with the max rtt.
   if (max_rtt_ms_ >= 0) {
     RTC_DCHECK_GE(avg_rtt_ms_, 0);
     for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
          it != observers_.end(); ++it) {
       (*it)->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_);
     }
     // Sum for Histogram of average RTT reported over the entire call.
     sum_avg_rtt_ms_ += avg_rtt_ms_;
     ++num_avg_rtt_;
   }
-  return 0;
 }
 
 int64_t CallStats::avg_rtt_ms() const {
   rtc::CritScope cs(&crit_);
   return avg_rtt_ms_;
 }
 
 RtcpRttStats* CallStats::rtcp_rtt_stats() const {
   return rtcp_rtt_stats_.get();
 }
@@ skipping 35 matching lines @@
   int64_t elapsed_sec =
       (clock_->TimeInMilliseconds() - time_of_first_rtt_ms_) / 1000;
   if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
     int64_t avg_rtt_ms = (sum_avg_rtt_ms_ + num_avg_rtt_ / 2) / num_avg_rtt_;
     RTC_HISTOGRAM_COUNTS_10000(
         "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms);
   }
 }
 
 }  // namespace webrtc