1 files changed, 91 insertions, 0 deletions
diff --git a/webrtc/call/packet_injection_tests.cc b/webrtc/call/packet_injection_tests.cc
new file mode 100644
index 0000000000..18ca0581d1
--- /dev/null
+++ b/webrtc/call/packet_injection_tests.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2015 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 "testing/gtest/include/gtest/gtest.h"
+
+#include "webrtc/test/call_test.h"
+#include "webrtc/test/null_transport.h"
+
+namespace webrtc {
+
+class PacketInjectionTest : public test::CallTest {
+ protected:
+  enum class CodecType {
+    kVp8,
+    kH264,
+  };
+
+  PacketInjectionTest() : rtp_header_parser_(RtpHeaderParser::Create()) {}
+
+  void InjectIncorrectPacket(CodecType codec_type,
+                             uint8_t packet_type,
+                             const uint8_t* packet,
+                             size_t length);
+
+  rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
+};
+
+void PacketInjectionTest::InjectIncorrectPacket(CodecType codec_type,
+                                                uint8_t payload_type,
+                                                const uint8_t* packet,
+                                                size_t length) {
+  CreateSenderCall(Call::Config());
+  CreateReceiverCall(Call::Config());
+
+  test::NullTransport null_transport;
+  CreateSendConfig(1, &null_transport);
+  CreateMatchingReceiveConfigs(&null_transport);
+  receive_configs_[0].decoders[0].payload_type = payload_type;
+  switch (codec_type) {
+    case CodecType::kVp8:
+      receive_configs_[0].decoders[0].payload_name = "VP8";
+      break;
+    case CodecType::kH264:
+      receive_configs_[0].decoders[0].payload_name = "H264";
+      break;
+  }
+  CreateStreams();
+
+  RTPHeader header;
+  EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
+  EXPECT_EQ(kSendSsrcs[0], header.ssrc)
+      << "Packet should have configured SSRC to not be dropped early.";
+  EXPECT_EQ(payload_type, header.payloadType);
+  Start();
+  EXPECT_EQ(PacketReceiver::DELIVERY_PACKET_ERROR,
+            receiver_call_->Receiver()->DeliverPacket(MediaType::VIDEO, packet,
+                                                      length, PacketTime()));
+  Stop();
+
+  DestroyStreams();
+}
+
+TEST_F(PacketInjectionTest, StapAPacketWithTruncatedNalUnits) {
+  const uint8_t kPacket[] = {0x80,
+                             0xE5,
+                             0xE6,
+                             0x0,
+                             0x0,
+                             0xED,
+                             0x23,
+                             0x4,
+                             0x00,
+                             0xC0,
+                             0xFF,
+                             0xED,
+                             0x58,
+                             0xCB,
+                             0xED,
+                             0xDF};
+
+  InjectIncorrectPacket(CodecType::kH264, 101, kPacket, sizeof(kPacket));
+}
+
+}  // namespace webrtc