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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-06-12 05:35:37 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-06-12 05:35:37 +0000 |
commit | a90a5cba08fdf6c0ceb95101c275108a152a3aed (patch) | |
tree | 532507288f3defd7f4dcf1af49698bcb76034855 /third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc | |
parent | Adding debian version 126.0.1-1. (diff) | |
download | firefox-a90a5cba08fdf6c0ceb95101c275108a152a3aed.tar.xz firefox-a90a5cba08fdf6c0ceb95101c275108a152a3aed.zip |
Merging upstream version 127.0.
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc')
-rw-r--r-- | third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc | 146 |
1 files changed, 73 insertions, 73 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc index c3c055de0f..93706e9253 100644 --- a/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc +++ b/third_party/libwebrtc/test/fuzzers/rtp_format_vp8_fuzzer.cc @@ -1,73 +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_vp8.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);
-
- RTPVideoHeaderVP8 hdr_info;
- hdr_info.InitRTPVideoHeaderVP8();
- uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0);
- hdr_info.pictureId =
- picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff;
-
- // Main function under test: RtpPacketizerVp8's constructor.
- RtpPacketizerVp8 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
+/* + * 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_vp8.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); + + RTPVideoHeaderVP8 hdr_info; + hdr_info.InitRTPVideoHeaderVP8(); + uint16_t picture_id = fuzz_input.ReadOrDefaultValue<uint16_t>(0); + hdr_info.pictureId = + picture_id >= 0x8000 ? kNoPictureId : picture_id & 0x7fff; + + // Main function under test: RtpPacketizerVp8's constructor. + RtpPacketizerVp8 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 |