04f4d126f8
Timing information is gathered in EncodedImage, starting at encoders. Then it's sent using RTP header extension. In the end, it's gathered at the GenericDecoder. Actual reporting and tests will be in the next CLs. BUG=webrtc:7594 Review-Url: https://codereview.webrtc.org/2911193002 Cr-Commit-Position: refs/heads/master@{#18659}
111 lines
4.0 KiB
C++
111 lines
4.0 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/include/rtp_header_extension_map.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 two 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 two bytes are enough.
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static_assert(kRtpExtensionNumberOfExtensions <= 16,
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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 <= 2)
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return;
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// Don't use the configuration bytes as part of the packet.
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std::bitset<16> extensionMask(*reinterpret_cast<const uint16_t*>(data));
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data += 2;
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size -= 2;
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RtpPacketReceived::ExtensionManager extensions;
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// Skip i = 0 since it maps to ExtensionNone and extension id = 0 is invalid.
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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] && 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.RegisterByType(i, 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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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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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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VideoTiming timing;
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packet.GetExtension<VideoTimingExtension>(&timing);
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break;
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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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}
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}
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}
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} // namespace webrtc
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