| /* |
| * 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 "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
| |
| namespace webrtc { |
| |
| // We decide which header extensions to register by reading one byte |
| // from the beginning of |data| and interpreting it as a bitmask over |
| // the RTPExtensionType enum. This assert ensures one byte is enough. |
| static_assert(kRtpExtensionNumberOfExtensions <= 8, |
| "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 <= 1) |
| return; |
| |
| // Don't use the configuration byte as part of the packet. |
| std::bitset<8> extensionMask(data[0]); |
| data++; |
| size--; |
| |
| RtpPacketReceived::ExtensionManager extensions; |
| for (int i = 0; i < kRtpExtensionNumberOfExtensions; i++) { |
| RTPExtensionType extension_type = static_cast<RTPExtensionType>(i); |
| if (extensionMask[i] && extension_type != kRtpExtensionNone) { |
| // 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; we use i. |
| extensions.Register(extension_type, i); |
| } |
| } |
| |
| 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; |
| } |
| } |
| } |
| } // namespace webrtc |