1 files changed, 63 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/receive_side_congestion_controller_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/receive_side_congestion_controller_fuzzer.cc
new file mode 100644
index 0000000000..8f548c2b90
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/receive_side_congestion_controller_fuzzer.cc
@@ -0,0 +1,63 @@
+/*
+ *  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 <algorithm>
+#include <cstddef>
+#include <cstdint>
+
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  Timestamp arrival_time = Timestamp::Micros(123'456'789);
+  SimulatedClock clock(arrival_time);
+  ReceiveSideCongestionController cc(
+      &clock,
+      /*feedback_sender=*/[](auto...) {},
+      /*remb_sender=*/[](auto...) {},
+      /*network_state_estimator=*/nullptr);
+  RtpHeaderExtensionMap extensions;
+  extensions.Register<TransmissionOffset>(1);
+  extensions.Register<AbsoluteSendTime>(2);
+  extensions.Register<TransportSequenceNumber>(3);
+  extensions.Register<TransportSequenceNumberV2>(4);
+  RtpPacketReceived rtp_packet(&extensions);
+
+  constexpr int kMinPacketSize = sizeof(uint16_t) + sizeof(uint8_t) + 12;
+  const uint8_t* const end_data = data + size;
+  while (end_data - data >= kMinPacketSize) {
+    size_t packet_size = ByteReader<uint16_t>::ReadBigEndian(data) % 1500;
+    data += sizeof(uint16_t);
+    arrival_time += TimeDelta::Millis(ByteReader<uint8_t>::ReadBigEndian(data));
+    data += sizeof(uint8_t);
+    packet_size = std::min<size_t>(end_data - data, packet_size);
+    auto raw_packet = rtc::MakeArrayView(data, packet_size);
+    data += packet_size;
+
+    if (!rtp_packet.Parse(raw_packet)) {
+      continue;
+    }
+    rtp_packet.set_arrival_time(arrival_time);
+
+    cc.OnReceivedPacket(rtp_packet, MediaType::VIDEO);
+    clock.AdvanceTimeMilliseconds(5);
+    cc.MaybeProcess();
+  }
+}
+}  // namespace webrtc