1 files changed, 52 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/rtcp_receiver_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
new file mode 100644
index 0000000000..8bad9e456a
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
@@ -0,0 +1,52 @@
+/*
+ *  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 "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
+#include "modules/rtp_rtcp/source/rtcp_receiver.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kRtcpIntervalMs = 1000;
+
+// RTCP is typically sent over UDP, which has a maximum payload length
+// of 65535 bytes. We err on the side of caution and check a bit above that.
+constexpr size_t kMaxInputLenBytes = 66000;
+
+class NullModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
+ public:
+  void SetTmmbn(std::vector<rtcp::TmmbItem>) override {}
+  void OnRequestSendReport() override {}
+  void OnReceivedNack(const std::vector<uint16_t>&) override {}
+  void OnReceivedRtcpReportBlocks(const ReportBlockList&) override {}
+};
+
+}  // namespace
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  if (size > kMaxInputLenBytes) {
+    return;
+  }
+
+  NullModuleRtpRtcp rtp_rtcp_module;
+  SimulatedClock clock(1234);
+
+  RtpRtcpInterface::Configuration config;
+  config.clock = &clock;
+  config.rtcp_report_interval_ms = kRtcpIntervalMs;
+  config.local_media_ssrc = 1;
+
+  RTCPReceiver receiver(config, &rtp_rtcp_module);
+
+  receiver.IncomingPacket(data, size);
+}
+}  // namespace webrtc