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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:35:49 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-05-15 03:35:49 +0000 |
commit | d8bbc7858622b6d9c278469aab701ca0b609cddf (patch) | |
tree | eff41dc61d9f714852212739e6b3738b82a2af87 /third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc | |
parent | Releasing progress-linux version 125.0.3-1~progress7.99u1. (diff) | |
download | firefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.tar.xz firefox-d8bbc7858622b6d9c278469aab701ca0b609cddf.zip |
Merging upstream version 126.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/test/fuzzers/rtp_format_vp9_fuzzer.cc | 73 |
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
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