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/*
* 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
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