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This CL is a followup to https://webrtc-review.googlesource.com/c/src/+/38481. With the new approach we can just use the generic RTP packetizer to pass frames over the wire as the specific info is contained within the bitstream. This makes the new codec more modular and reduces its footprint. Bug: webrtc:7671 Change-Id: Ib07f72a9d338e3cbfdbf39cba9891a959b5f7552 Reviewed-on: https://webrtc-review.googlesource.com/43220 Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21753}
145 lines
4.1 KiB
C++
145 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 "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 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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