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 |