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Start migrating away from `hasAudioLevel`, `voiceActivity`, `audioLevel` fields in RTPHeaderExtension and switch usages to a more modern absl::optional<AudioLevel> accessor instead. The old fields are preserved for compatibility with downstream projects, but will be removed in the future. Bug: webrtc:15788 Change-Id: I76599124fd68dd4d449f850df3b9814d6a002f5d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336303 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41947}
173 lines
6.1 KiB
C++
173 lines
6.1 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <bitset>
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#include <vector>
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_video_layers_allocation_extension.h"
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namespace webrtc {
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// We decide which header extensions to register by reading four bytes
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// from the beginning of `data` and interpreting it as a bitmask over
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// the RTPExtensionType enum. This assert ensures four bytes are enough.
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static_assert(kRtpExtensionNumberOfExtensions <= 32,
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"Insufficient bits read to configure all header extensions. Add "
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"an extra byte and update the switches.");
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void FuzzOneInput(const uint8_t* data, size_t size) {
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if (size <= 4)
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return;
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// Don't use the configuration byte as part of the packet.
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std::bitset<32> extensionMask(*reinterpret_cast<const uint32_t*>(data));
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data += 4;
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size -= 4;
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RtpPacketReceived::ExtensionManager extensions(/*extmap_allow_mixed=*/true);
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// Start at local_id = 1 since 0 is an invalid extension id.
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int local_id = 1;
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// Skip i = 0 since it maps to kRtpExtensionNone.
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for (int i = 1; i < kRtpExtensionNumberOfExtensions; i++) {
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RTPExtensionType extension_type = static_cast<RTPExtensionType>(i);
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if (extensionMask[i]) {
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// Extensions are registered with an ID, which you signal to the
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// peer so they know what to expect. This code only cares about
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// parsing so the value of the ID isn't relevant.
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extensions.RegisterByType(local_id++, extension_type);
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}
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}
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RtpPacketReceived packet(&extensions);
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packet.Parse(data, size);
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// Call packet accessors because they have extra checks.
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packet.Marker();
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packet.PayloadType();
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packet.SequenceNumber();
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packet.Timestamp();
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packet.Ssrc();
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packet.Csrcs();
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// Each extension has its own getter. It is supported behaviour to
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// call GetExtension on an extension which was not registered, so we
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// don't check the bitmask here.
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for (int i = 0; i < kRtpExtensionNumberOfExtensions; i++) {
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switch (static_cast<RTPExtensionType>(i)) {
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case kRtpExtensionNone:
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case kRtpExtensionNumberOfExtensions:
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break;
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case kRtpExtensionTransmissionTimeOffset:
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int32_t offset;
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packet.GetExtension<TransmissionOffset>(&offset);
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break;
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case kRtpExtensionAudioLevel: {
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AudioLevel audio_level;
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packet.GetExtension<AudioLevelExtension>(&audio_level);
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break;
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}
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case kRtpExtensionCsrcAudioLevel: {
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std::vector<uint8_t> audio_levels;
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packet.GetExtension<CsrcAudioLevel>(&audio_levels);
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break;
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}
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case kRtpExtensionAbsoluteSendTime:
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uint32_t sendtime;
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packet.GetExtension<AbsoluteSendTime>(&sendtime);
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break;
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case kRtpExtensionAbsoluteCaptureTime: {
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AbsoluteCaptureTime extension;
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packet.GetExtension<AbsoluteCaptureTimeExtension>(&extension);
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break;
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}
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case kRtpExtensionVideoRotation:
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uint8_t rotation;
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packet.GetExtension<VideoOrientation>(&rotation);
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break;
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case kRtpExtensionTransportSequenceNumber:
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uint16_t seqnum;
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packet.GetExtension<TransportSequenceNumber>(&seqnum);
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break;
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case kRtpExtensionTransportSequenceNumber02: {
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uint16_t seqnum;
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absl::optional<FeedbackRequest> feedback_request;
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packet.GetExtension<TransportSequenceNumberV2>(&seqnum,
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&feedback_request);
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break;
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}
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case kRtpExtensionPlayoutDelay: {
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VideoPlayoutDelay playout;
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packet.GetExtension<PlayoutDelayLimits>(&playout);
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break;
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}
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case kRtpExtensionVideoContentType:
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VideoContentType content_type;
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packet.GetExtension<VideoContentTypeExtension>(&content_type);
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break;
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case kRtpExtensionVideoTiming: {
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VideoSendTiming timing;
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packet.GetExtension<VideoTimingExtension>(&timing);
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break;
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}
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case kRtpExtensionRtpStreamId: {
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std::string rsid;
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packet.GetExtension<RtpStreamId>(&rsid);
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break;
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}
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case kRtpExtensionRepairedRtpStreamId: {
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std::string rsid;
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packet.GetExtension<RepairedRtpStreamId>(&rsid);
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break;
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}
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case kRtpExtensionMid: {
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std::string mid;
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packet.GetExtension<RtpMid>(&mid);
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break;
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}
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case kRtpExtensionGenericFrameDescriptor: {
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RtpGenericFrameDescriptor descriptor;
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packet.GetExtension<RtpGenericFrameDescriptorExtension00>(&descriptor);
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break;
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}
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case kRtpExtensionColorSpace: {
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ColorSpace color_space;
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packet.GetExtension<ColorSpaceExtension>(&color_space);
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break;
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}
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case kRtpExtensionInbandComfortNoise: {
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absl::optional<uint8_t> noise_level;
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packet.GetExtension<InbandComfortNoiseExtension>(&noise_level);
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break;
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}
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case kRtpExtensionVideoLayersAllocation: {
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VideoLayersAllocation allocation;
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packet.GetExtension<RtpVideoLayersAllocationExtension>(&allocation);
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break;
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}
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case kRtpExtensionVideoFrameTrackingId: {
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uint16_t tracking_id;
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packet.GetExtension<VideoFrameTrackingIdExtension>(&tracking_id);
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break;
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}
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case kRtpExtensionDependencyDescriptor:
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// This extension requires state to read and so complicated that
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// deserves own fuzzer.
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break;
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}
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}
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// Check that zero-ing mutable extensions wouldn't cause any problems.
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packet.ZeroMutableExtensions();
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}
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} // namespace webrtc
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