Adding integration test for unsignaled inbound RTP stream stats.

webrtc/pc/test/mockpeerconnectionobservers.h

 /*
  *  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 207 matching lines @@
     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_;
 };
 
+// Helper class that just stores the report from the callback.
+class MockRTCStatsCollectorCallback : public webrtc::RTCStatsCollectorCallback {
+ public:
+  rtc::scoped_refptr<const RTCStatsReport> report() { return report_; }
+
+  bool called() const { return called_; }
+
+ protected:
+  void OnStatsDelivered(
+      const rtc::scoped_refptr<const RTCStatsReport>& report) override {
+    report_ = report;
+    called_ = true;
+  }
+
+ private:
+  bool called_ = false;
+  rtc::scoped_refptr<const RTCStatsReport> report_;
+};
+
 }  // namespace webrtc
 
 #endif  // WEBRTC_PC_TEST_MOCKPEERCONNECTIONOBSERVERS_H_