Fix segfault when PeerConnection is destroyed during stats collection.

webrtc/api/rtcstatscollector.h

 /*
  *  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 52 matching lines @@
 
   // Gets a recent stats report. If there is a report cached that is still fresh
   // it is returned, otherwise new stats are gathered and returned. A report is
   // considered fresh for |cache_lifetime_| ms. const RTCStatsReports are safe
   // to use across multiple threads and may be destructed on any thread.
   void GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
   // Clears the cache's reference to the most recent stats report. Subsequently
   // calling |GetStatsReport| guarantees fresh stats.
   void ClearCachedStatsReport();
 
+  // If there are any |GetStatsReport| requests in-flight, waits until it has

[hbos, 2016/12/16 13:49:35]

nit: Will update comment to speak in singular, this makes it sound like there
could be multiple concurrent requests, but there can only be one at a time.
(Multiple GetStatsReport calls are piggybacked)

+  // been completed. Must be called on the signaling thread.
+  void WaitForPendingRequest();
+
  protected:
   RTCStatsCollector(PeerConnection* pc, int64_t cache_lifetime_us);
 
   // Stats gathering on a particular thread. Calls |AddPartialResults| before
   // returning. Virtual for the sake of testing.
   virtual void ProducePartialResultsOnSignalingThread(int64_t timestamp_us);
   virtual void ProducePartialResultsOnWorkerThread(int64_t timestamp_us);
   virtual void ProducePartialResultsOnNetworkThread(int64_t timestamp_us);
 
   // Can be called on any thread.
@@ skipping 96 matching lines @@
 };
 
 const char* CandidateTypeToRTCIceCandidateTypeForTesting(
     const std::string& type);
 const char* DataStateToRTCDataChannelStateForTesting(
     DataChannelInterface::DataState state);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_RTCSTATSCOLLECTOR_H_