b4d31085b4
Added various timestamps to the FrameObject class which are needed to calculate the jitter delay. BUG=webrtc:5514 Review-Url: https://codereview.webrtc.org/2124943002 Cr-Commit-Position: refs/heads/master@{#13434}
110 lines
3.0 KiB
C++
110 lines
3.0 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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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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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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// TODO(philipel): Remove when encoded image is replaced by FrameObject.
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// VCMEncodedFrame members
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CopyCodecSpecific(&packet->video_header);
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_completeFrame = true;
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_payloadType = packet->payloadType;
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_timeStamp = packet->timestamp;
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ntp_time_ms_ = packet->ntp_time_ms_;
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_buffer = new uint8_t[frame_size]();
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_size = frame_size;
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_length = frame_size;
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_frameType = packet->frameType;
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GetBitstream(_buffer);
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// RtpFrameObject members
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frame_type_ = packet->frameType;
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codec_type_ = packet->codec;
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// FrameObject members
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timestamp = packet->timestamp;
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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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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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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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