Introduce RtpTransportInternal and SrtpTransport.

webrtc/pc/rtptransporttestutil.h

Index: webrtc/pc/rtptransporttestutil.h
diff --git a/webrtc/pc/rtptransporttestutil.h b/webrtc/pc/rtptransporttestutil.h
new file mode 100644
index 0000000000000000000000000000000000000000..92cfaec703a23a21628cbc063b93b3e67a5b5ad0
--- /dev/null
+++ b/webrtc/pc/rtptransporttestutil.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2017 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_PC_RTPTRANSPORTTESTUTIL_H_
+#define WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_
+
+#include "webrtc/pc/rtptransportinternal.h"
+#include "webrtc/rtc_base/sigslot.h"
+
+namespace webrtc {
+
+class SignalPacketReceivedCounter : public sigslot::has_slots<> {
+ public:
+  explicit SignalPacketReceivedCounter(RtpTransportInternal* transport) {
+    transport->SignalPacketReceived.connect(
+        this, &SignalPacketReceivedCounter::OnPacketReceived);
+  }
+  int rtcp_count() const { return rtcp_count_; }
+  int rtp_count() const { return rtp_count_; }
+
+ private:
+  void OnPacketReceived(bool rtcp,
+                        rtc::CopyOnWriteBuffer*,
+                        const rtc::PacketTime&) {
+    if (rtcp) {
+      ++rtcp_count_;
+    } else {
+      ++rtp_count_;
+    }
+  }
+  int rtcp_count_ = 0;
+  int rtp_count_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_PC_RTPTRANSPORTTESTUTIL_H_