Set / verify stats report timestamps.

talk/app/webrtc/test/mockpeerconnectionobservers.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 109 matching lines @@
   MockStatsObserver() : called_(false), stats_() {}
   virtual ~MockStatsObserver() {}
 
   virtual void OnComplete(const StatsReports& reports) {
     ASSERT(!called_);
     called_ = true;
     stats_.Clear();
     stats_.number_of_reports = reports.size();
     for (const auto* r : reports) {
       if (r->type() == StatsReport::kStatsReportTypeSsrc) {
+        stats_.timestamp = r->timestamp();
         GetIntValue(r, StatsReport::kStatsValueNameAudioOutputLevel,
             &stats_.audio_output_level);
         GetIntValue(r, StatsReport::kStatsValueNameAudioInputLevel,
             &stats_.audio_input_level);
         GetIntValue(r, StatsReport::kStatsValueNameBytesReceived,
             &stats_.bytes_received);
         GetIntValue(r, StatsReport::kStatsValueNameBytesSent,
             &stats_.bytes_sent);
       } else if (r->type() == StatsReport::kStatsReportTypeBwe) {
+        stats_.timestamp = r->timestamp();
         GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth,
             &stats_.available_receive_bandwidth);
       } else if (r->type() == StatsReport::kStatsReportTypeComponent) {
+        stats_.timestamp = r->timestamp();
         GetStringValue(r, StatsReport::kStatsValueNameDtlsCipher,
             &stats_.dtls_cipher);
         GetStringValue(r, StatsReport::kStatsValueNameSrtpCipher,
             &stats_.srtp_cipher);
       }
     }
   }
 
   bool called() const { return called_; }
   size_t number_of_reports() const { return stats_.number_of_reports; }
+  double timestamp() const { return stats_.timestamp; }
 
   int AudioOutputLevel() const {
     ASSERT(called_);
     return stats_.audio_output_level;
   }
 
   int AudioInputLevel() const {
     ASSERT(called_);
     return stats_.audio_input_level;
   }
@@ skipping 41 matching lines @@
     const StatsReport::Value* v = report->FindValue(name);
     if (v)
       *value = v->ToString();
     return v != nullptr;
   }
 
   bool called_;
   struct {
     void Clear() {
       number_of_reports = 0;
+      timestamp = 0;
       audio_output_level = 0;
       audio_input_level = 0;
       bytes_received = 0;
       bytes_sent = 0;
       available_receive_bandwidth = 0;
       dtls_cipher.clear();
       srtp_cipher.clear();
     }
 
     size_t number_of_reports;
+    double timestamp;
     int audio_output_level;
     int audio_input_level;
     int bytes_received;
     int bytes_sent;
     int available_receive_bandwidth;
     std::string dtls_cipher;
     std::string srtp_cipher;
   } stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_