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Diffstat (limited to 'third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc')
| -rw-r--r-- | third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc | 182 |
1 files changed, 182 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc b/third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc new file mode 100644 index 0000000000..5d117529bb --- /dev/null +++ b/third_party/libwebrtc/test/fuzzers/rtp_packet_fuzzer.cc @@ -0,0 +1,182 @@ +/* + * Copyright (c) 2016 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 <bitset> +#include <vector> + +#include "absl/types/optional.h" +#include "modules/rtp_rtcp/include/rtp_header_extension_map.h" +#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h" +#include "modules/rtp_rtcp/source/rtp_header_extensions.h" +#include "modules/rtp_rtcp/source/rtp_packet_received.h" +#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h" + +namespace webrtc { +// We decide which header extensions to register by reading four bytes +// from the beginning of `data` and interpreting it as a bitmask over +// the RTPExtensionType enum. This assert ensures four bytes are enough. +static_assert(kRtpExtensionNumberOfExtensions <= 32, + "Insufficient bits read to configure all header extensions. Add " + "an extra byte and update the switches."); + +void FuzzOneInput(const uint8_t* data, size_t size) { + if (size <= 4) + return; + + // Don't use the configuration byte as part of the packet. + std::bitset<32> extensionMask(*reinterpret_cast<const uint32_t*>(data)); + data += 4; + size -= 4; + + RtpPacketReceived::ExtensionManager extensions(/*extmap_allow_mixed=*/true); + // Start at local_id = 1 since 0 is an invalid extension id. + int local_id = 1; + // Skip i = 0 since it maps to kRtpExtensionNone. + for (int i = 1; i < kRtpExtensionNumberOfExtensions; i++) { + RTPExtensionType extension_type = static_cast<RTPExtensionType>(i); + if (extensionMask[i]) { + // Extensions are registered with an ID, which you signal to the + // peer so they know what to expect. This code only cares about + // parsing so the value of the ID isn't relevant. + extensions.RegisterByType(local_id++, extension_type); + } + } + + RtpPacketReceived packet(&extensions); + packet.Parse(data, size); + + // Call packet accessors because they have extra checks. + packet.Marker(); + packet.PayloadType(); + packet.SequenceNumber(); + packet.Timestamp(); + packet.Ssrc(); + packet.Csrcs(); + + // Each extension has its own getter. It is supported behaviour to + // call GetExtension on an extension which was not registered, so we + // don't check the bitmask here. + for (int i = 0; i < kRtpExtensionNumberOfExtensions; i++) { + switch (static_cast<RTPExtensionType>(i)) { + case kRtpExtensionNone: + case kRtpExtensionNumberOfExtensions: + break; + case kRtpExtensionTransmissionTimeOffset: + int32_t offset; + packet.GetExtension<TransmissionOffset>(&offset); + break; + case kRtpExtensionAudioLevel: + bool voice_activity; + uint8_t audio_level; + packet.GetExtension<AudioLevel>(&voice_activity, &audio_level); + break; +#if !defined(WEBRTC_MOZILLA_BUILD) + case kRtpExtensionCsrcAudioLevel: { + std::vector<uint8_t> audio_levels; + packet.GetExtension<CsrcAudioLevel>(&audio_levels); + break; + } +#endif + case kRtpExtensionAbsoluteSendTime: + uint32_t sendtime; + packet.GetExtension<AbsoluteSendTime>(&sendtime); + break; + case kRtpExtensionAbsoluteCaptureTime: { + AbsoluteCaptureTime extension; + packet.GetExtension<AbsoluteCaptureTimeExtension>(&extension); + break; + } + case kRtpExtensionVideoRotation: + uint8_t rotation; + packet.GetExtension<VideoOrientation>(&rotation); + break; + case kRtpExtensionTransportSequenceNumber: + uint16_t seqnum; + packet.GetExtension<TransportSequenceNumber>(&seqnum); + break; + case kRtpExtensionTransportSequenceNumber02: { + uint16_t seqnum; + absl::optional<FeedbackRequest> feedback_request; + packet.GetExtension<TransportSequenceNumberV2>(&seqnum, + &feedback_request); + break; + } + case kRtpExtensionPlayoutDelay: { + VideoPlayoutDelay playout; + packet.GetExtension<PlayoutDelayLimits>(&playout); + break; + } + case kRtpExtensionVideoContentType: + VideoContentType content_type; + packet.GetExtension<VideoContentTypeExtension>(&content_type); + break; + case kRtpExtensionVideoTiming: { + VideoSendTiming timing; + packet.GetExtension<VideoTimingExtension>(&timing); + break; + } + case kRtpExtensionRtpStreamId: { + std::string rsid; + packet.GetExtension<RtpStreamId>(&rsid); + break; + } + case kRtpExtensionRepairedRtpStreamId: { + std::string rsid; + packet.GetExtension<RepairedRtpStreamId>(&rsid); + break; + } + case kRtpExtensionMid: { + std::string mid; + packet.GetExtension<RtpMid>(&mid); + break; + } + case kRtpExtensionGenericFrameDescriptor: { + RtpGenericFrameDescriptor descriptor; + packet.GetExtension<RtpGenericFrameDescriptorExtension00>(&descriptor); + break; + } + case kRtpExtensionColorSpace: { + ColorSpace color_space; + packet.GetExtension<ColorSpaceExtension>(&color_space); + break; + } + case kRtpExtensionInbandComfortNoise: { + absl::optional<uint8_t> noise_level; + packet.GetExtension<InbandComfortNoiseExtension>(&noise_level); + break; + } + case kRtpExtensionVideoLayersAllocation: { + VideoLayersAllocation allocation; + packet.GetExtension<RtpVideoLayersAllocationExtension>(&allocation); + break; + } + case kRtpExtensionVideoFrameTrackingId: { + uint16_t tracking_id; + packet.GetExtension<VideoFrameTrackingIdExtension>(&tracking_id); + break; + } + case kRtpExtensionDependencyDescriptor: + // This extension requires state to read and so complicated that + // deserves own fuzzer. + break; +#if defined(WEBRTC_MOZILLA_BUILD) + case kRtpExtensionCsrcAudioLevel: { + CsrcAudioLevelList levels; + packet.GetExtension<CsrcAudioLevel>(&levels); + break; + } +#endif + } + } + + // Check that zero-ing mutable extensions wouldn't cause any problems. + packet.ZeroMutableExtensions(); +} +} // namespace webrtc |