This changes the corpus semantics, but libfuzzer should be smart enough to figure it out, and if not then we can add a seed_corpus to help. BUG=webrtc:4771 NOTRY=true Review-Url: https://codereview.webrtc.org/2072473002 Cr-Commit-Position: refs/heads/master@{#13384}
50 lines
1.9 KiB
C++
50 lines
1.9 KiB
C++
/*
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* Copyright (c) 2016 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 <bitset>
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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namespace webrtc {
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// We decide which header extensions to register by reading one byte
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// from the beginning of |data| and interpreting it as a bitmask over
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// the RTPExtensionType enum. This assert ensures one byte is enough.
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static_assert(kRtpExtensionNumberOfExtensions <= 8,
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"Insufficient bits read to configure all header extensions. Add "
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"an extra byte and update the switches.");
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void FuzzOneInput(const uint8_t* data, size_t size) {
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if (size <= 1)
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return;
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// Don't use the configuration byte as part of the packet.
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std::bitset<8> extensionMask(data[0]);
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data++;
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size--;
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RtpPacketReceived::ExtensionManager extensions;
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for (int i = 0; i < kRtpExtensionNumberOfExtensions; i++) {
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RTPExtensionType extension_type = static_cast<RTPExtensionType>(i);
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if (extensionMask[i] && extension_type != kRtpExtensionNone) {
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// Extensions are registered with an ID, which you signal to the
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// peer so they know what to expect. This code only cares about
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// parsing so the value of the ID isn't relevant; we use i.
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extensions.Register(extension_type, i);
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}
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}
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RTPHeader rtp_header;
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RtpUtility::RtpHeaderParser rtp_parser(data, size);
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rtp_parser.Parse(&rtp_header, &extensions);
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}
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} // namespace webrtc
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