
Add RTC_DEPRACATed anonymous unions to not break downstream projects.
Orignal issue's description:
> commit 0ad21111fcc57a7e978edba3c4263f0062d7f9ff
> Author: danilchap <danilchap@webrtc.org>
> Date: Mon Dec 19 09:36:33 2016 -0800
>
> Revert of Rename RTPVideoHeader.isFirstPacket to
> .is_first_packet_in_frame. (patchset #1 id:1 of
> https://codereview.webrtc.org/2574943003/ )
>
> Reason for revert:
> breaks downstream project.
>
> Can you make this change in a compatible way using anonymous
> union:
> union {
> bool is_first_packet_in_frame;
> RTC_DEPRECATED bool isFirstPacket;
> };
> (unfortunetly this this treak breaks braced initialization in
> rtp_rtcp_impl_unittest.cc,
> so that should be rewritting in a more classic way)
>
> Original issue's description:
> > Rename RTPVideoHeader.isFirstPacket to
> > .is_first_packet_in_frame.
> >
> > Name should represent the actual meaning.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2574943003
> > Cr-Commit-Position: refs/heads/master@{#15684}
> > Committed:
> > efde908380
>
> TBR=stefan@webrtc.org,sprang@webrtc.org,johan@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days
> ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2589783003
> Cr-Commit-Position: refs/heads/master@{#15686}
>
BUG=None
Review-Url: https://codereview.webrtc.org/2614503002
Cr-Commit-Position: refs/heads/master@{#15987}
143 lines
4.1 KiB
C++
143 lines
4.1 KiB
C++
/*
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* Copyright (c) 2011 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/video_coding/packet.h"
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#include <assert.h>
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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VCMPacket::VCMPacket()
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: payloadType(0),
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timestamp(0),
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ntp_time_ms_(0),
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seqNum(0),
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dataPtr(NULL),
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sizeBytes(0),
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markerBit(false),
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timesNacked(-1),
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frameType(kEmptyFrame),
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codec(kVideoCodecUnknown),
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is_first_packet_in_frame(false),
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completeNALU(kNaluUnset),
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insertStartCode(false),
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width(0),
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height(0),
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video_header() {
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video_header.playout_delay = {-1, -1};
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}
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VCMPacket::VCMPacket(const uint8_t* ptr,
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const size_t size,
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const WebRtcRTPHeader& rtpHeader)
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: payloadType(rtpHeader.header.payloadType),
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timestamp(rtpHeader.header.timestamp),
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ntp_time_ms_(rtpHeader.ntp_time_ms),
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seqNum(rtpHeader.header.sequenceNumber),
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dataPtr(ptr),
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sizeBytes(size),
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markerBit(rtpHeader.header.markerBit),
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timesNacked(-1),
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frameType(rtpHeader.frameType),
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codec(kVideoCodecUnknown),
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is_first_packet_in_frame(rtpHeader.type.Video.is_first_packet_in_frame),
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completeNALU(kNaluComplete),
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insertStartCode(false),
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width(rtpHeader.type.Video.width),
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height(rtpHeader.type.Video.height),
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video_header(rtpHeader.type.Video) {
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CopyCodecSpecifics(rtpHeader.type.Video);
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if (markerBit) {
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video_header.rotation = rtpHeader.type.Video.rotation;
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}
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// Playout decisions are made entirely based on first packet in a frame.
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if (is_first_packet_in_frame) {
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video_header.playout_delay = rtpHeader.type.Video.playout_delay;
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} else {
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video_header.playout_delay = {-1, -1};
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}
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}
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void VCMPacket::Reset() {
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payloadType = 0;
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timestamp = 0;
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ntp_time_ms_ = 0;
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seqNum = 0;
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dataPtr = NULL;
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sizeBytes = 0;
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markerBit = false;
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timesNacked = -1;
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frameType = kEmptyFrame;
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codec = kVideoCodecUnknown;
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is_first_packet_in_frame = false;
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completeNALU = kNaluUnset;
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insertStartCode = false;
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width = 0;
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height = 0;
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memset(&video_header, 0, sizeof(RTPVideoHeader));
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}
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void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {
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switch (videoHeader.codec) {
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case kRtpVideoVp8:
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// Handle all packets within a frame as depending on the previous packet
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// TODO(holmer): This should be changed to make fragments independent
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// when the VP8 RTP receiver supports fragments.
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if (is_first_packet_in_frame && markerBit)
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completeNALU = kNaluComplete;
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else if (is_first_packet_in_frame)
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completeNALU = kNaluStart;
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else if (markerBit)
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completeNALU = kNaluEnd;
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else
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completeNALU = kNaluIncomplete;
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codec = kVideoCodecVP8;
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return;
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case kRtpVideoVp9:
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if (is_first_packet_in_frame && markerBit)
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completeNALU = kNaluComplete;
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else if (is_first_packet_in_frame)
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completeNALU = kNaluStart;
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else if (markerBit)
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completeNALU = kNaluEnd;
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else
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completeNALU = kNaluIncomplete;
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codec = kVideoCodecVP9;
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return;
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case kRtpVideoH264:
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is_first_packet_in_frame = videoHeader.is_first_packet_in_frame;
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if (is_first_packet_in_frame)
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insertStartCode = true;
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if (is_first_packet_in_frame && markerBit) {
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completeNALU = kNaluComplete;
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} else if (is_first_packet_in_frame) {
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completeNALU = kNaluStart;
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} else if (markerBit) {
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completeNALU = kNaluEnd;
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} else {
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completeNALU = kNaluIncomplete;
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}
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codec = kVideoCodecH264;
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return;
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case kRtpVideoGeneric:
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case kRtpVideoNone:
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codec = kVideoCodecUnknown;
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return;
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}
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}
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} // namespace webrtc
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