|  | /* | 
|  | *  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 "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" | 
|  |  | 
|  | 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 kRtpExtensionAbsoluteSendTime: | 
|  | uint32_t sendtime; | 
|  | packet.GetExtension<AbsoluteSendTime>(&sendtime); | 
|  | break; | 
|  | case kRtpExtensionVideoRotation: | 
|  | uint8_t rotation; | 
|  | packet.GetExtension<VideoOrientation>(&rotation); | 
|  | break; | 
|  | case kRtpExtensionTransportSequenceNumber: | 
|  | uint16_t seqnum; | 
|  | packet.GetExtension<TransportSequenceNumber>(&seqnum); | 
|  | break; | 
|  | case kRtpExtensionPlayoutDelay: | 
|  | PlayoutDelay 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 kRtpExtensionFrameMarking: | 
|  | FrameMarking frame_marking; | 
|  | packet.GetExtension<FrameMarkingExtension>(&frame_marking); | 
|  | 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 kRtpExtensionGenericFrameDescriptor01: { | 
|  | RtpGenericFrameDescriptor descriptor; | 
|  | packet.GetExtension<RtpGenericFrameDescriptorExtension01>(&descriptor); | 
|  | break; | 
|  | } | 
|  | case kRtpExtensionColorSpace: { | 
|  | ColorSpace color_space; | 
|  | packet.GetExtension<ColorSpaceExtension>(&color_space); | 
|  | break; | 
|  | } | 
|  | } | 
|  | } | 
|  | } | 
|  | }  // namespace webrtc |