6b4b5f3770
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}
79 lines
2.1 KiB
C++
79 lines
2.1 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/modules/video_coding/frame_object.h"
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#include "webrtc/base/criticalsection.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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: 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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times_nacked_(times_nacked) {
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size = frame_size;
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VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num);
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if (packet) {
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frame_type_ = packet->frameType;
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codec_type_ = packet->codec;
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}
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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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RTPVideoTypeHeader* RtpFrameObject::GetCodecHeader() const {
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VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
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if (!packet)
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return nullptr;
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return &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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