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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc')
-rw-r--r-- | third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc | 64 |
1 files changed, 64 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc new file mode 100644 index 0000000000..ae8b8728fb --- /dev/null +++ b/third_party/libwebrtc/test/fuzzers/rtp_video_layers_allocation_fuzzer.cc @@ -0,0 +1,64 @@ +/* + * Copyright (c) 2021 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 <cstddef> +#include <cstdint> +#include <limits> + +#include "api/array_view.h" +#include "api/video/video_layers_allocation.h" +#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h" +#include "rtc_base/checks.h" + +namespace webrtc { + +void FuzzOneInput(const uint8_t* data, size_t size) { + // Video layers allocation is an rtp header extension. + // Per https://datatracker.ietf.org/doc/html/rfc8285#section-4.3 + // rtp header extension uses up to one byte to store the size, i.e. + // maximum size of any rtp header extension is 255 bytes. + constexpr int kMaxSize = std::numeric_limits<uint8_t>::max(); + if (size > kMaxSize) { + return; + } + auto raw = rtc::MakeArrayView(data, size); + + VideoLayersAllocation allocation1; + if (!RtpVideoLayersAllocationExtension::Parse(raw, &allocation1)) { + // Ignore invalid buffer and move on. + return; + } + + // Write parsed allocation back into raw buffer. + size_t value_size = RtpVideoLayersAllocationExtension::ValueSize(allocation1); + // Check `writer` use minimal number of bytes to pack the extension by + // checking it doesn't use more than reader consumed. + RTC_CHECK_LE(value_size, raw.size()); + uint8_t some_memory[kMaxSize]; + RTC_CHECK_LE(value_size, kMaxSize); + rtc::ArrayView<uint8_t> write_buffer(some_memory, value_size); + RTC_CHECK( + RtpVideoLayersAllocationExtension::Write(write_buffer, allocation1)); + + // Parse what Write assembled. + // Unlike random input that should always succeed. + VideoLayersAllocation allocation2; + RTC_CHECK( + RtpVideoLayersAllocationExtension::Parse(write_buffer, &allocation2)); + + RTC_CHECK_EQ(allocation1.rtp_stream_index, allocation2.rtp_stream_index); + RTC_CHECK_EQ(allocation1.resolution_and_frame_rate_is_valid, + allocation2.resolution_and_frame_rate_is_valid); + RTC_CHECK_EQ(allocation1.active_spatial_layers.size(), + allocation2.active_spatial_layers.size()); + RTC_CHECK(allocation1 == allocation2); +} + +} // namespace webrtc |