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90612a681b
This is a reland of 20f2133d5d
Original change's description:
> Add stereo codec header and pass it through RTP
>
> - Defines CodecSpecificInfoStereo that carries stereo specific header info from
> encoded image.
> - Defines RTPVideoHeaderStereo that carries the above info to packetizer,
> see module_common_types.h.
> - Adds an RTPPacketizer and RTPDepacketizer that supports passing specific stereo
> header.
> - Uses new data containers in StereoAdapter classes.
>
> This CL is the step 3 for adding alpha channel support over the wire in webrtc.
> See https://webrtc-review.googlesource.com/c/src/+/7800 for the experimental
> CL that gives an idea about how it will come together.
> Design Doc: https://goo.gl/sFeSUT
>
> Bug: webrtc:7671
> Change-Id: Ia932568fdd7065ba104afd2bc0ecf25a765748ab
> Reviewed-on: https://webrtc-review.googlesource.com/22900
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20920}
TBR=danilchap@webrtc.org, sprang@webrtc.org, niklas.enbom@webrtc.org
Bug: webrtc:7671
Change-Id: If8f0c7e6e3a2a704f19161f0e8bf1880906e7fe0
Reviewed-on: https://webrtc-review.googlesource.com/27160
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20946}
148 lines
4.2 KiB
C++
148 lines
4.2 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 "modules/video_coding/packet.h"
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#include <assert.h>
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#include "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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receive_time_ms(0) {
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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 kRtpVideoStereo:
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codec = kVideoCodecStereo;
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return;
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case kRtpVideoGeneric:
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codec = kVideoCodecGeneric;
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return;
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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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