1 files changed, 73 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc
new file mode 100644
index 0000000000..3b5e67f697
--- /dev/null
+++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc
@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2024 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 <stddef.h>
+#include <stdint.h>
+
+#include "api/video/video_frame_type.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "test/fuzzers/fuzz_data_helper.h"
+
+namespace webrtc {
+void FuzzOneInput(const uint8_t* data, size_t size) {
+  test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 1200;
+  // Read uint8_t to be sure reduction_lens are much smaller than
+  // max_payload_len and thus limits structure is valid.
+  limits.first_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.last_packet_reduction_len = fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+  limits.single_packet_reduction_len =
+      fuzz_input.ReadOrDefaultValue<uint8_t>(0);
+
+  RTPVideoHeaderVP9 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP9();
+  uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
+  hdr_info.picture_id =
+      picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
+
+  // Main function under test: RtpPacketizerVp9's constructor.
+  RtpPacketizerVp9 packetizer(fuzz_input.ReadByteArray(fuzz_input.BytesLeft()),
+                              limits, hdr_info);
+
+  size_t num_packets = packetizer.NumPackets();
+  if (num_packets == 0) {
+    return;
+  }
+  // When packetization was successful, validate NextPacket function too.
+  // While at it, check that packets respect the payload size limits.
+  RtpPacketToSend rtp_packet(nullptr);
+  // Single packet.
+  if (num_packets == 1) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+    RTC_CHECK_LE(rtp_packet.payload_size(),
+                 limits.max_payload_len - limits.single_packet_reduction_len);
+    return;
+  }
+  // First packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.first_packet_reduction_len);
+  // Middle packets.
+  for (size_t i = 1; i < num_packets - 1; ++i) {
+    RTC_CHECK(packetizer.NextPacket(&rtp_packet))
+        << "Failed to get packet#" << i;
+    RTC_CHECK_LE(rtp_packet.payload_size(), limits.max_payload_len)
+        << "Packet #" << i << " exceeds it's limit";
+  }
+  // Last packet.
+  RTC_CHECK(packetizer.NextPacket(&rtp_packet));
+  RTC_CHECK_LE(rtp_packet.payload_size(),
+               limits.max_payload_len - limits.last_packet_reduction_len);
+}
+}  // namespace webrtc