| /* |
| * 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; |
| case kRtpExtensionCsrcAudioLevel: { |
| std::vector<uint8_t> audio_levels; |
| packet.GetExtension<CsrcAudioLevel>(&audio_levels); |
| break; |
| } |
| 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 kRtpExtensionGenericFrameDescriptor00: { |
| 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 kRtpExtensionGenericFrameDescriptor02: |
| // This extension requires state to read and so complicated that |
| // deserves own fuzzer. |
| break; |
| } |
| } |
| |
| // Check that zero-ing mutable extensions wouldn't cause any problems. |
| packet.ZeroMutableExtensions(); |
| } |
| } // namespace webrtc |