/* * 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.
*/
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.");
// 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);
}
}
// 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: {
AudioLevel audio_level;
packet.GetExtension<AudioLevelExtension>(&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;
std::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: {
std::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; case kRtpExtensionCorruptionDetection: {
CorruptionDetectionMessage message;
packet.GetExtension<CorruptionDetectionExtension>(&message); break; #ifdefined(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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