64e739aeae
Use it to separate UMA e2e delay metric between screenshare from video. Content type extension is set based on encoder settings and processed and decoders. Also, Fix full-stack-tests to calculate RTT correctly, so new metric could be tested. BUG=webrtc:7420 Review-Url: https://codereview.webrtc.org/2772033002 Cr-Commit-Position: refs/heads/master@{#17640}
139 lines
4.4 KiB
C++
139 lines
4.4 KiB
C++
/*
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* Copyright (c) 2016 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/base/checks.h"
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#include "webrtc/modules/video_coding/frame_object.h"
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#include "webrtc/modules/video_coding/packet_buffer.h"
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namespace webrtc {
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namespace video_coding {
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FrameObject::FrameObject()
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: picture_id(0),
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spatial_layer(0),
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timestamp(0),
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num_references(0),
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inter_layer_predicted(false) {}
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RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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size_t frame_size,
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int times_nacked,
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int64_t received_time)
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: packet_buffer_(packet_buffer),
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first_seq_num_(first_seq_num),
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last_seq_num_(last_seq_num),
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received_time_(received_time),
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times_nacked_(times_nacked) {
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VCMPacket* first_packet = packet_buffer_->GetPacket(first_seq_num);
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RTC_DCHECK(first_packet);
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// RtpFrameObject members
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frame_type_ = first_packet->frameType;
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codec_type_ = first_packet->codec;
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// TODO(philipel): Remove when encoded image is replaced by FrameObject.
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// VCMEncodedFrame members
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CopyCodecSpecific(&first_packet->video_header);
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_completeFrame = true;
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_payloadType = first_packet->payloadType;
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_timeStamp = first_packet->timestamp;
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ntp_time_ms_ = first_packet->ntp_time_ms_;
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// Since FFmpeg use an optimized bitstream reader that reads in chunks of
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// 32/64 bits we have to add at least that much padding to the buffer
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// to make sure the decoder doesn't read out of bounds.
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// NOTE! EncodedImage::_size is the size of the buffer (think capacity of
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// an std::vector) and EncodedImage::_length is the actual size of
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// the bitstream (think size of an std::vector).
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if (codec_type_ == kVideoCodecH264)
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_size = frame_size + EncodedImage::kBufferPaddingBytesH264;
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else
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_size = frame_size;
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_buffer = new uint8_t[_size];
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_length = frame_size;
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_frameType = first_packet->frameType;
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GetBitstream(_buffer);
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_encodedWidth = first_packet->width;
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_encodedHeight = first_packet->height;
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// FrameObject members
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timestamp = first_packet->timestamp;
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VCMPacket* last_packet = packet_buffer_->GetPacket(last_seq_num);
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RTC_DCHECK(last_packet && last_packet->markerBit);
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// http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
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// ts_126114v120700p.pdf Section 7.4.5.
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// The MTSI client shall add the payload bytes as defined in this clause
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// onto the last RTP packet in each group of packets which make up a key
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// frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
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// (HEVC)).
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rotation_ = last_packet->video_header.rotation;
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_rotation_set = true;
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content_type_ = last_packet->video_header.content_type;
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}
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RtpFrameObject::~RtpFrameObject() {
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packet_buffer_->ReturnFrame(this);
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}
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uint16_t RtpFrameObject::first_seq_num() const {
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return first_seq_num_;
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}
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uint16_t RtpFrameObject::last_seq_num() const {
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return last_seq_num_;
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}
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int RtpFrameObject::times_nacked() const {
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return times_nacked_;
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}
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FrameType RtpFrameObject::frame_type() const {
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return frame_type_;
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}
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VideoCodecType RtpFrameObject::codec_type() const {
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return codec_type_;
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}
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bool RtpFrameObject::GetBitstream(uint8_t* destination) const {
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return packet_buffer_->GetBitstream(*this, destination);
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}
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uint32_t RtpFrameObject::Timestamp() const {
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return timestamp_;
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}
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int64_t RtpFrameObject::ReceivedTime() const {
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return received_time_;
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}
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int64_t RtpFrameObject::RenderTime() const {
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return _renderTimeMs;
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}
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bool RtpFrameObject::delayed_by_retransmission() const {
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return times_nacked() > 0;
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}
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rtc::Optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
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rtc::CritScope lock(&packet_buffer_->crit_);
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VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
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if (!packet)
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return rtc::Optional<RTPVideoTypeHeader>();
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return rtc::Optional<RTPVideoTypeHeader>(packet->video_header.codecHeader);
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}
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} // namespace video_coding
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} // namespace webrtc
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