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https://github.com/mollyim/webrtc.git
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ef615ea7a3
Today we use |is_first_packet_in_frame| to know when a frame begins and the |markerBit| to know when it ends, but the markerbit does not actually mark the end of a frame, it marks the end of a picture. Bug: webrtc:9361 Change-Id: Icc70e6075590cdc31e875a4eb9d489868adbb67c Reviewed-on: https://webrtc-review.googlesource.com/100160 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24722}
84 lines
2.6 KiB
C++
84 lines
2.6 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(kVideoCodecGeneric),
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is_first_packet_in_frame(false),
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is_last_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(rtpHeader.video_header().codec),
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is_first_packet_in_frame(
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rtpHeader.video_header().is_first_packet_in_frame),
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is_last_packet_in_frame(rtpHeader.video_header().is_last_packet_in_frame),
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completeNALU(kNaluIncomplete),
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insertStartCode(rtpHeader.video_header().codec == kVideoCodecH264 &&
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rtpHeader.video_header().is_first_packet_in_frame),
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width(rtpHeader.video_header().width),
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height(rtpHeader.video_header().height),
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video_header(rtpHeader.video_header()) {
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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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if (markerBit) {
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video_header.rotation = rtpHeader.video_header().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.video_header().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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} // namespace webrtc
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