webrtc_m130/test/fuzzers/rtp_packet_fuzzer.cc

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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>
Reland "Extend TransportSequenceNumber RTP header extension" This reverts commit 109b5fb5f5b2f46e1798c91c4a024ce26f57f0b0. Reason for revert: The failing libfuzzer was fixed in commit d6c6f16063b81fc60206618ba06198e34ee0eacb Original change's description: > Revert "Extend TransportSequenceNumber RTP header extension" > > This reverts commit 28c7362bc485d22bdc8c744bc725022780187a96. > > Reason for revert: It breaks Linux64 Release (libfuzzer): > https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket.appspot.com/8921003137877469920/+/steps/compile/0/stdout > > Original change's description: > > Extend TransportSequenceNumber RTP header extension > > > > Extend TransportSequenceNumber RTP header extension to support > > feedback on sender request. > > > > Bug: webrtc:10262 > > Change-Id: Ibc1cf18162d15cd102e951c9dc697d8ea536ebb6 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123233 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26766} > > TBR=danilchap@webrtc.org,aleloi@webrtc.org,kron@webrtc.org > > Change-Id: Ie8a73f5fdffd99919ceaa1ae8911a1645f2077e9 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10262 > Reviewed-on: https://webrtc-review.googlesource.com/c/123522 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26767} TBR=danilchap@webrtc.org,mbonadei@webrtc.org,aleloi@webrtc.org,kron@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10262 Change-Id: I0f854299a46c042cfbdf8b8cc8cd965a228142c8 Reviewed-on: https://webrtc-review.googlesource.com/c/123764 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26798}
2019-02-21 14:09:20 +00:00
#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"
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 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;
Reland "Extend TransportSequenceNumber RTP header extension" This reverts commit 109b5fb5f5b2f46e1798c91c4a024ce26f57f0b0. Reason for revert: The failing libfuzzer was fixed in commit d6c6f16063b81fc60206618ba06198e34ee0eacb Original change's description: > Revert "Extend TransportSequenceNumber RTP header extension" > > This reverts commit 28c7362bc485d22bdc8c744bc725022780187a96. > > Reason for revert: It breaks Linux64 Release (libfuzzer): > https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket.appspot.com/8921003137877469920/+/steps/compile/0/stdout > > Original change's description: > > Extend TransportSequenceNumber RTP header extension > > > > Extend TransportSequenceNumber RTP header extension to support > > feedback on sender request. > > > > Bug: webrtc:10262 > > Change-Id: Ibc1cf18162d15cd102e951c9dc697d8ea536ebb6 > > Reviewed-on: https://webrtc-review.googlesource.com/c/123233 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Alex Loiko <aleloi@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26766} > > TBR=danilchap@webrtc.org,aleloi@webrtc.org,kron@webrtc.org > > Change-Id: Ie8a73f5fdffd99919ceaa1ae8911a1645f2077e9 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10262 > Reviewed-on: https://webrtc-review.googlesource.com/c/123522 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26767} TBR=danilchap@webrtc.org,mbonadei@webrtc.org,aleloi@webrtc.org,kron@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10262 Change-Id: I0f854299a46c042cfbdf8b8cc8cd965a228142c8 Reviewed-on: https://webrtc-review.googlesource.com/c/123764 Reviewed-by: Johannes Kron <kron@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26798}
2019-02-21 14:09:20 +00:00
case kRtpExtensionTransportSequenceNumber02: {
uint16_t seqnum;
absl::optional<FeedbackRequest> feedback_request;
packet.GetExtension<TransportSequenceNumberV2>(&seqnum,
&feedback_request);
break;
}
Revert "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery" This reverts commit 4f68f5398d7fa3d47c449e99893c9bea07bf7ca2. Reason for revert: Breaks downstream project Original change's description: > Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery > > The PlayoutDelayOracle was responsible for making sure the PlayoutDelay > header extension was successfully propagated to the receiving side. Once > it was determined that the receiver had received a frame with the new > delay tag, it's no longer necessary to propagate. > > The issue with this implementation is that it is based on max > extended sequence number reported via RTCP, which makes it often slow > to react, could theoretically fail to produce desired outcome (max > received > X does not guarantee X was fully received and decoded), and > added a lot of code complexity. > > The guarantee of delivery can in fact be accomplished more reliably and > with less code by making sure to tag each frame until an undiscardable > frame is sent. > > This allows containing the logic fully within RTPSenderVideo. > > Bug: webrtc:11340 > Change-Id: I2d1d2b6b67f4f07b8b33336f8fcfcde724243eef > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168221 > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Erik Språng <sprang@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#30473} TBR=sprang@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Ide922e680ae36bb69b95e58002482cf5ed57e254 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11340 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168304 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30475}
2020-02-06 16:04:48 +00:00
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;
}
case kRtpExtensionInbandComfortNoise: {
absl::optional<uint8_t> noise_level;
packet.GetExtension<InbandComfortNoiseExtension>(&noise_level);
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