133 lines
3.9 KiB
C++
133 lines
3.9 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 "modules/video_coding/frame_object.h"
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#include <string.h>
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#include <utility>
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#include "api/video/encoded_image.h"
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#include "api/video/video_timing.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace video_coding {
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RtpFrameObject::RtpFrameObject(
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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bool markerBit,
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int times_nacked,
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int64_t first_packet_received_time,
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int64_t last_packet_received_time,
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uint32_t rtp_timestamp,
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int64_t ntp_time_ms,
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const VideoSendTiming& timing,
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uint8_t payload_type,
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VideoCodecType codec,
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VideoRotation rotation,
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VideoContentType content_type,
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const RTPVideoHeader& video_header,
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const absl::optional<webrtc::ColorSpace>& color_space,
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RtpPacketInfos packet_infos,
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rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
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: image_buffer_(image_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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last_packet_received_time_(last_packet_received_time),
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times_nacked_(times_nacked) {
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rtp_video_header_ = video_header;
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// EncodedFrame members
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codec_type_ = codec;
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// TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
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// VCMEncodedFrame members
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CopyCodecSpecific(&rtp_video_header_);
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_payloadType = payload_type;
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SetTimestamp(rtp_timestamp);
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ntp_time_ms_ = ntp_time_ms;
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_frameType = rtp_video_header_.frame_type;
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// Setting frame's playout delays to the same values
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// as of the first packet's.
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SetPlayoutDelay(rtp_video_header_.playout_delay);
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SetEncodedData(image_buffer_);
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_encodedWidth = rtp_video_header_.width;
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_encodedHeight = rtp_video_header_.height;
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// EncodedFrame members
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SetPacketInfos(std::move(packet_infos));
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rotation_ = rotation;
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SetColorSpace(color_space);
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content_type_ = content_type;
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if (timing.flags != VideoSendTiming::kInvalid) {
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// ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
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// as this will be dealt with at the time of reporting.
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timing_.encode_start_ms = ntp_time_ms_ + timing.encode_start_delta_ms;
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timing_.encode_finish_ms = ntp_time_ms_ + timing.encode_finish_delta_ms;
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timing_.packetization_finish_ms =
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ntp_time_ms_ + timing.packetization_finish_delta_ms;
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timing_.pacer_exit_ms = ntp_time_ms_ + timing.pacer_exit_delta_ms;
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timing_.network_timestamp_ms =
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ntp_time_ms_ + timing.network_timestamp_delta_ms;
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timing_.network2_timestamp_ms =
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ntp_time_ms_ + timing.network2_timestamp_delta_ms;
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}
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timing_.receive_start_ms = first_packet_received_time;
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timing_.receive_finish_ms = last_packet_received_time;
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timing_.flags = timing.flags;
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is_last_spatial_layer = markerBit;
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}
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RtpFrameObject::~RtpFrameObject() {
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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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VideoFrameType RtpFrameObject::frame_type() const {
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return rtp_video_header_.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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int64_t RtpFrameObject::ReceivedTime() const {
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return last_packet_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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const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const {
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return rtp_video_header_;
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}
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} // namespace video_coding
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} // namespace webrtc
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