blob: 3e33699088886b4edd2b95baa88522f7d8769b85 [file] [log] [blame]
/*
* 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/include/rtp_header_extension_map.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 two bytes
// from the beginning of |data| and interpreting it as a bitmask over
// the RTPExtensionType enum. This assert ensures two bytes are enough.
static_assert(kRtpExtensionNumberOfExtensions <= 16,
"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 <= 2)
return;
// Don't use the configuration bytes as part of the packet.
std::bitset<16> extensionMask(*reinterpret_cast<const uint16_t*>(data));
data += 2;
size -= 2;
RtpPacketReceived::ExtensionManager extensions;
// Skip i = 0 since it maps to ExtensionNone and extension id = 0 is invalid.
for (int i = 1; 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.RegisterByType(i, 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 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;
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;
}
}
}
}
} // namespace webrtc