Fixing incorrect use of erase/remove idiom.

webrtc/pc/statscollector.cc

 /*
  *  Copyright 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 417 matching lines @@
   StatsReport* report = reports_.Find(id);
   if (!report) {
     report = reports_.InsertNew(id);
     report->AddString(StatsReport::kStatsValueNameTrackId, audio_track->id());
   }
 }
 
 void StatsCollector::RemoveLocalAudioTrack(AudioTrackInterface* audio_track,
                                            uint32_t ssrc) {
   RTC_DCHECK(audio_track != NULL);
-  local_audio_tracks_.erase(std::remove_if(local_audio_tracks_.begin(),
-      local_audio_tracks_.end(),
-      [audio_track, ssrc](const LocalAudioTrackVector::value_type& track) {
-        return track.first == audio_track && track.second == ssrc;
-      }));
+  local_audio_tracks_.erase(
+      std::remove_if(
+          local_audio_tracks_.begin(), local_audio_tracks_.end(),
+          [audio_track, ssrc](const LocalAudioTrackVector::value_type& track) {
+            return track.first == audio_track && track.second == ssrc;
+          }),
+      local_audio_tracks_.end());
 }
 
 void StatsCollector::GetStats(MediaStreamTrackInterface* track,
                               StatsReports* reports) {
   RTC_DCHECK(pc_->session()->signaling_thread()->IsCurrent());
   RTC_DCHECK(reports != NULL);
   RTC_DCHECK(reports->empty());
 
   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
 
@@ skipping 549 matching lines @@
     StatsReport* report = entry.second;
     report->set_timestamp(stats_gathering_started_);
   }
 }
 
 void StatsCollector::ClearUpdateStatsCacheForTest() {
   stats_gathering_started_ = 0;
 }
 
 }  // namespace webrtc