This CL adds support for an extension on RTP frames to allow the sender to specify the minimum and maximum playout delay limits. The receiver makes a best-effort attempt to keep the capture-to-render delay within this range. This allows different types of application to specify different end-to-end delay goals. For example gaming can support rendering of frames as soon as received on receiver to minimize delay. A movie playback application can specify a minimum playout delay to allow fixed buffering in presence of network jitter. There are no tests at this time and most of testing is done with chromium webrtc prototype. On chromoting performance tests, this extension helps bring down end-to-end delay by about 150 ms on small frames. BUG=webrtc:5895 Review-Url: https://codereview.webrtc.org/2007743003 Cr-Commit-Position: refs/heads/master@{#13059}
165 lines
4.5 KiB
C++
165 lines
4.5 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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isFirstPacket(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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isFirstPacket(rtpHeader.type.Video.isFirstPacket),
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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 (isFirstPacket) {
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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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VCMPacket::VCMPacket(const uint8_t* ptr,
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size_t size,
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uint16_t seq,
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uint32_t ts,
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bool mBit)
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: payloadType(0),
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timestamp(ts),
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ntp_time_ms_(0),
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seqNum(seq),
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dataPtr(ptr),
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sizeBytes(size),
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markerBit(mBit),
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timesNacked(-1),
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frameType(kVideoFrameDelta),
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codec(kVideoCodecUnknown),
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isFirstPacket(false),
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completeNALU(kNaluComplete),
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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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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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isFirstPacket = 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 (isFirstPacket && markerBit)
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completeNALU = kNaluComplete;
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else if (isFirstPacket)
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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 (isFirstPacket && markerBit)
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completeNALU = kNaluComplete;
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else if (isFirstPacket)
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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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isFirstPacket = videoHeader.isFirstPacket;
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if (isFirstPacket)
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insertStartCode = true;
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if (isFirstPacket && markerBit) {
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completeNALU = kNaluComplete;
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} else if (isFirstPacket) {
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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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