Fixed the crash when adding a chromium sink to remote audio track or

content/browser/renderer_host/media/media_stream_track_metrics_host.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/browser/renderer_host/media/media_stream_track_metrics_host.h"
 
 #include "base/metrics/histogram.h"
 #include "content/common/media/media_stream_track_metrics_host_messages.h"
 
 // We use a histogram with a maximum bucket of 16 hours to infinity
@@ skipping 31 matching lines @@
     IPC_MESSAGE_HANDLER(MediaStreamTrackMetricsHost_RemoveTrack, OnRemoveTrack)
     IPC_MESSAGE_UNHANDLED(handled = false)
   IPC_END_MESSAGE_MAP()
 
   return handled;
 }
 
 void MediaStreamTrackMetricsHost::OnAddTrack(uint64 id,
                                              bool is_audio,
                                              bool is_remote) {
-  DCHECK(tracks_.find(id) == tracks_.end());
+  if (tracks_.find(id) != tracks_.end())
+    return;
+
   TrackInfo info = {is_audio, is_remote, base::TimeTicks::Now()};
   tracks_[id] = info;
 }
 
 void MediaStreamTrackMetricsHost::OnRemoveTrack(uint64 id) {
-  DCHECK(tracks_.find(id) != tracks_.end());
+  if (tracks_.find(id) == tracks_.end())
+    return;
 
   TrackInfo& info = tracks_[id];
   ReportDuration(info);
   tracks_.erase(id);
 }
 
 void MediaStreamTrackMetricsHost::ReportDuration(const TrackInfo& info) {
   base::TimeDelta duration = base::TimeTicks::Now() - info.timestamp;
   if (info.is_remote) {
     if (info.is_audio) {
       DVLOG(3) << "WebRTC.ReceivedAudioTrackDuration: " << duration.InSeconds();
       UMA_HISTOGRAM_TIMES_16H("WebRTC.ReceivedAudioTrackDuration", duration);
     } else {
       DVLOG(3) << "WebRTC.ReceivedVideoTrackDuration: " << duration.InSeconds();
       UMA_HISTOGRAM_TIMES_16H("WebRTC.ReceivedVideoTrackDuration", duration);
     }
   } else {
     if (info.is_audio) {
       DVLOG(3) << "WebRTC.SentAudioTrackDuration: " << duration.InSeconds();
       UMA_HISTOGRAM_TIMES_16H("WebRTC.SentAudioTrackDuration", duration);
     } else {
       DVLOG(3) << "WebRTC.SentVideoTrackDuration: " << duration.InSeconds();
       UMA_HISTOGRAM_TIMES_16H("WebRTC.SentVideoTrackDuration", duration);
     }
   }
 }
 
 }  // namespace content