Introduce RtpTransportInternal and SrtpTransport.

webrtc/pc/srtptransport_unittest.cc

Index: webrtc/pc/srtptransport_unittest.cc
diff --git a/webrtc/pc/srtptransport_unittest.cc b/webrtc/pc/srtptransport_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e54dac3ea29dab437e636365ce71f69246366578
--- /dev/null
+++ b/webrtc/pc/srtptransport_unittest.cc
@@ -0,0 +1,76 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/pc/srtptransport.h"
+
+#include "webrtc/pc/rtptransport.h"
+#include "webrtc/pc/rtptransporttestutil.h"
+#include "webrtc/rtc_base/asyncpacketsocket.h"
+#include "webrtc/rtc_base/gunit.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+
+using testing::_;
+using testing::Return;
+
+class MockRtpTransport : public RtpTransport {
+ public:
+  MockRtpTransport() : RtpTransport(true) {}
+
+  MOCK_METHOD4(SendPacket,
+               bool(bool rtcp,
+                    rtc::CopyOnWriteBuffer* packet,
+                    const rtc::PacketOptions& options,
+                    int flags));
+
+  void PretendReceivedPacket() {
+    bool rtcp = false;
+    rtc::CopyOnWriteBuffer buffer;
+    rtc::PacketTime time;
+    SignalPacketReceived(rtcp, &buffer, time);
+  }
+};
+
+TEST(SrtpTransportTest, SendPacket) {
+  auto rtp_transport = rtc::MakeUnique<MockRtpTransport>();
+  EXPECT_CALL(*rtp_transport, SendPacket(_, _, _, _)).WillOnce(Return(true));
+
+  SrtpTransport srtp_transport(std::move(rtp_transport), "a");
+
+  const bool rtcp = false;
+  rtc::CopyOnWriteBuffer packet;
+  rtc::PacketOptions options;
+  int flags = 0;
+  EXPECT_TRUE(srtp_transport.SendPacket(rtcp, &packet, options, flags));
+
+  // TODO(zstein): Also verify that the packet received by RtpTransport has been
+  // protected once SrtpTransport handles that.
+}
+
+// Test that SrtpTransport fires SignalPacketReceived when the underlying
+// RtpTransport fires SignalPacketReceived.
+TEST(SrtpTransportTest, SignalPacketReceived) {
+  auto rtp_transport = rtc::MakeUnique<MockRtpTransport>();
+  MockRtpTransport* rtp_transport_raw = rtp_transport.get();
+  SrtpTransport srtp_transport(std::move(rtp_transport), "a");
+
+  SignalPacketReceivedCounter counter(&srtp_transport);
+
+  rtp_transport_raw->PretendReceivedPacket();
+
+  EXPECT_EQ(1, counter.rtp_count());
+
+  // TODO(zstein): Also verify that the packet is unprotected once SrtpTransport
+  // handles that.
+}
+
+}  // namespace webrtc