RTCStatsCollector collecting stats on multiple threads.

webrtc/stats/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.
  */
 
 #ifndef WEBRTC_STATS_RTCSTATSCOLLECTOR_H_
 #define WEBRTC_STATS_RTCSTATSCOLLECTOR_H_
 
 #include <memory>
+#include <vector>
 
 #include "webrtc/api/rtcstats_objects.h"
 #include "webrtc/api/rtcstatsreport.h"
+#include "webrtc/base/asyncinvoker.h"
+#include "webrtc/base/criticalsection.h"
+#include "webrtc/base/refcount.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/base/timing.h"
 
 namespace webrtc {
 
 class PeerConnection;
 
-// All calls to the collector and gathering of stats is performed on the
-// signaling thread. A stats report is cached for |cache_lifetime_| ms.
-class RTCStatsCollector {
+class RTCStatsCollectorCallback : public virtual rtc::RefCountInterface {
  public:
-  explicit RTCStatsCollector(
+  virtual ~RTCStatsCollectorCallback() {}
+
+  virtual void OnStatsDelivered(
+      const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;
+};
+
+// All public methods of the collector are to be called on the signaling thread.
+// Stats are gathered on the signaling, worker and network threads
+// asynchronously. The callback is invoked on the signaling thread. Resulting
+// reports are cached for |cache_lifetime_| ms.
+class RTCStatsCollector : public virtual rtc::RefCountInterface {
+ public:
+  static rtc::scoped_refptr<RTCStatsCollector> Create(
       PeerConnection* pc,
       double cache_lifetime = 0.05,
       std::unique_ptr<rtc::Timing> timing = std::unique_ptr<rtc::Timing>(
           new rtc::Timing()));
 
   // 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.
-  rtc::scoped_refptr<const RTCStatsReport> GetStatsReport();
+  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();
 
+ protected:
+  // Takes ownership of |timing| (not using |std::unique_ptr| due to
+  // incompatibility with |rtc::RefCountedObject| constructors).
+  RTCStatsCollector(
+      PeerConnection* pc,
+      double cache_lifetime,
+      rtc::Timing* timing);
+
+  // Stats gathering on a particular thread. Calls |AddPartialResults| before
+  // returning. Virtual for the sake of testing.
+  virtual void ProducePartialResultsOnSignalingThread_s(double timestamp);
+  virtual void ProducePartialResultsOnWorkerThread_w(double timestamp);
+  virtual void ProducePartialResultsOnNetworkThread_n(double timestamp);

[hta-webrtc, 2016/08/31 09:57:36]

Is the _s/_n/_w suffix a WebRTC convention that we have to adhere to?
It seems to be breaking naming rules & repeating the same info twice.

[hbos, 2016/09/01 14:16:16]

Probably not, I'll skip it for these three.

+
+  // Can be called on any thread.
+  void AddPartialResults(
+      const rtc::scoped_refptr<RTCStatsReport>& partial_report,
+      double timestamp);
+
  private:
-  bool IsOnSignalingThread() const;
+  void AddPartialResults_s(rtc::scoped_refptr<RTCStatsReport> partial_report,
+                           double timestamp);
 
-  std::unique_ptr<RTCPeerConnectionStats> ProducePeerConnectionStats() const;
+  std::unique_ptr<RTCPeerConnectionStats> ProducePeerConnectionStats_s(
+      double timestamp) const;
 
   PeerConnection* const pc_;
-  mutable std::unique_ptr<rtc::Timing> timing_;
+  rtc::Thread* const signaling_thread_;
+  rtc::Thread* const worker_thread_;
+  rtc::Thread* const network_thread_;
+  rtc::AsyncInvoker invoker_;
+
+  rtc::CriticalSection lock_;

[hta-webrtc, 2016/08/31 09:57:36]

Please add GUARDED_BY statements on the stuff that's guarded by this lock. It's
good documentation, and there are tools that do checking for you.

[hbos, 2016/09/01 14:16:16]

Done.

+  int num_pending_partial_reports_;
+  rtc::scoped_refptr<RTCStatsReport> partial_report_;
+  std::vector<rtc::scoped_refptr<RTCStatsCollectorCallback>> callbacks_;
+
+  const std::unique_ptr<rtc::Timing> timing_;
   // Time relative to the UNIX epoch (Jan 1, 1970, UTC), in seconds.
   double cache_timestamp_;
   double cache_lifetime_;  // In seconds.
   rtc::scoped_refptr<const RTCStatsReport> cached_report_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_STATS_RTCSTATSCOLLECTOR_H_