
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}
144 lines
4.1 KiB
C++
144 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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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 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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