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git ls-files | grep -e "\(\.h\|\.cc\)$" | grep -vE "^(rtc_base|sdk|modules|api|call|common_audio|examples|media|net|p2p|pc)/" | xargs clang-format -i ; git cl format after landing: add to .git-blame-ignore-revs Bug: webrtc:15082 Change-Id: I9c7fc4e6fbb023809fb22a89a78be713de6990d3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302063 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39978}
38 lines
1.4 KiB
C++
38 lines
1.4 KiB
C++
/*
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* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stddef.h>
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#include <stdint.h>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/rtp_rtcp/source/video_rtp_depacketizer_av1.h"
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#include "test/fuzzers/fuzz_data_helper.h"
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namespace webrtc {
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void FuzzOneInput(const uint8_t* data, size_t size) {
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std::vector<rtc::ArrayView<const uint8_t>> rtp_payloads;
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// Convert plain array of bytes into array of array bytes.
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test::FuzzDataHelper fuzz_input(rtc::MakeArrayView(data, size));
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while (fuzz_input.CanReadBytes(sizeof(uint16_t))) {
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// In practice one rtp payload can be up to ~1200 - 1500 bytes. Majority
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// of the payload is just copied. To make fuzzing more efficient limit the
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// size of rtp payload to realistic value.
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uint16_t next_size = fuzz_input.Read<uint16_t>() % 1200;
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if (next_size > fuzz_input.BytesLeft()) {
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next_size = fuzz_input.BytesLeft();
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}
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rtp_payloads.push_back(fuzz_input.ReadByteArray(next_size));
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}
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// Run code under test.
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VideoRtpDepacketizerAv1().AssembleFrame(rtp_payloads);
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}
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} // namespace webrtc
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