Don't copy loop variables when we don't have to

webrtc/api/rtcstatscollector.cc

 /*
  *  Copyright 2016 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 232 matching lines @@
     return;
   for (size_t i = 0; i < streams->count(); ++i) {
     MediaStreamInterface* stream = streams->at(i);
 
     std::unique_ptr<RTCMediaStreamStats> stream_stats(
         new RTCMediaStreamStats("RTCMediaStream_" + stream->label(),
                                 timestamp_us));
     stream_stats->stream_identifier = stream->label();
     stream_stats->track_ids = std::vector<std::string>();
     // Audio Tracks
-    for (rtc::scoped_refptr<AudioTrackInterface> audio_track :
+    for (const rtc::scoped_refptr<AudioTrackInterface>& audio_track :
          stream->GetAudioTracks()) {
       std::string id = RTCMediaStreamTrackStatsIDFromMediaStreamTrackInterface(
         *audio_track.get());
       if (report->Get(id)) {
         // Skip track, stats already exist for it.
         continue;
       }
       std::unique_ptr<RTCMediaStreamTrackStats> audio_track_stats(
           new RTCMediaStreamTrackStats(id, timestamp_us));
       stream_stats->track_ids->push_back(audio_track_stats->id());
@@ skipping 13 matching lines @@
         AudioProcessorInterface::AudioProcessorStats audio_processor_stats;
         audio_track->GetAudioProcessor()->GetStats(&audio_processor_stats);
         audio_track_stats->echo_return_loss = static_cast<double>(
             audio_processor_stats.echo_return_loss);
         audio_track_stats->echo_return_loss_enhancement = static_cast<double>(
             audio_processor_stats.echo_return_loss_enhancement);
       }
       report->AddStats(std::move(audio_track_stats));
     }
     // Video Tracks
-    for (rtc::scoped_refptr<VideoTrackInterface> video_track :
+    for (const rtc::scoped_refptr<VideoTrackInterface>& video_track :
          stream->GetVideoTracks()) {
       std::string id = RTCMediaStreamTrackStatsIDFromMediaStreamTrackInterface(
           *video_track.get());
       if (report->Get(id)) {
         // Skip track, stats already exist for it.
         continue;
       }
       std::unique_ptr<RTCMediaStreamTrackStats> video_track_stats(
           new RTCMediaStreamTrackStats(id, timestamp_us));
       stream_stats->track_ids->push_back(video_track_stats->id());
@@ skipping 492 matching lines @@
     const std::string& type) {
   return CandidateTypeToRTCIceCandidateType(type);
 }
 
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state) {
   return DataStateToRTCDataChannelState(state);
 }
 
 }  // namespace webrtc