08b0351ddd
to be used with rtp::Packet class BUG=webrtc:1994 R=isheriff@chromium.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/2224063004 . Cr-Commit-Position: refs/heads/master@{#14105}
92 lines
3.2 KiB
C++
92 lines
3.2 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 "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
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namespace webrtc {
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// We decide which header extensions to register by reading one byte
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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 one byte is enough.
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static_assert(kRtpExtensionNumberOfExtensions <= 8,
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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 <= 1)
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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<8> extensionMask(data[0]);
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data++;
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size--;
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RtpPacketReceived::ExtensionManager extensions;
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for (int i = 0; i < kRtpExtensionNumberOfExtensions; i++) {
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RTPExtensionType extension_type = static_cast<RTPExtensionType>(i);
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if (extensionMask[i] && extension_type != kRtpExtensionNone) {
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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; we use i.
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extensions.Register(extension_type, i);
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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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bool voice_activity;
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uint8_t audio_level;
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packet.GetExtension<AudioLevel>(&voice_activity, &audio_level);
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break;
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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 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 kRtpExtensionPlayoutDelay:
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PlayoutDelay playout;
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packet.GetExtension<PlayoutDelayLimits>(&playout);
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break;
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}
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}
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}
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} // namespace webrtc
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