Modified the rtp_receiver_unittests.

webrtc/api/rtpreceiverinterface.h

Index: webrtc/api/rtpreceiverinterface.h
diff --git a/webrtc/api/rtpreceiverinterface.h b/webrtc/api/rtpreceiverinterface.h
index fd233abe317609d265ca187f54be4781a1a47b75..66958ff6bd422a6d683ccc46fc50bc40601f34f7 100644
--- a/webrtc/api/rtpreceiverinterface.h
+++ b/webrtc/api/rtpreceiverinterface.h
@@ -55,6 +55,11 @@ class RtpSource {
   // TODO(zhihuang): Implement this to return real audio level.
   rtc::Optional<int8_t> audio_level() const { return rtc::Optional<int8_t>(); }
 
+  bool operator==(const RtpSource& o) const {
+    return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
+           source_type_ == o.source_type();
+  }
+
  private:
   int64_t timestamp_ms_;
   uint32_t source_id_;