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/*
* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/rtp_rtcp/source/frame_object.h"
#include <cstdint>
#include <optional>
#include <utility>
#include <variant>
#include "api/rtp_packet_infos.h"
#include "api/scoped_refptr.h"
#include "api/video/color_space.h"
#include "api/video/encoded_image.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_metadata.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "common_video/frame_instrumentation_data.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
namespace webrtc {
RtpFrameObject::RtpFrameObject(
uint16_t first_seq_num,
uint16_t last_seq_num,
bool markerBit,
int times_nacked,
int64_t first_packet_received_time,
int64_t last_packet_received_time,
uint32_t rtp_timestamp,
int64_t ntp_time_ms,
const VideoSendTiming& timing,
uint8_t payload_type,
VideoCodecType codec,
VideoRotation rotation,
VideoContentType content_type,
const RTPVideoHeader& video_header,
const std::optional<webrtc::ColorSpace>& color_space,
const std::optional<
std::variant<FrameInstrumentationSyncData, FrameInstrumentationData>>&
frame_instrumentation_data,
RtpPacketInfos packet_infos,
scoped_refptr<EncodedImageBuffer> image_buffer)
: image_buffer_(image_buffer),
first_seq_num_(first_seq_num),
last_seq_num_(last_seq_num),
last_packet_received_time_(last_packet_received_time),
times_nacked_(times_nacked) {
rtp_video_header_ = video_header;
// EncodedFrame members
codec_type_ = codec;
// TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
// VCMEncodedFrame members
_codecSpecificInfo.frame_instrumentation_data = frame_instrumentation_data;
CopyCodecSpecific(&rtp_video_header_);
_payloadType = payload_type;
SetRtpTimestamp(rtp_timestamp);
ntp_time_ms_ = ntp_time_ms;
_frameType = rtp_video_header_.frame_type;
// Setting frame's playout delays to the same values
// as of the first packet's.
SetPlayoutDelay(rtp_video_header_.playout_delay);
SetEncodedData(image_buffer_);
_encodedWidth = rtp_video_header_.width;
_encodedHeight = rtp_video_header_.height;
if (packet_infos.begin() != packet_infos.end()) {
csrcs_ = packet_infos.begin()->csrcs();
}
// EncodedFrame members
SetPacketInfos(std::move(packet_infos));
rotation_ = rotation;
SetColorSpace(color_space);
SetVideoFrameTrackingId(rtp_video_header_.video_frame_tracking_id);
content_type_ = content_type;
if (timing.flags != VideoSendTiming::kInvalid) {
// ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
// as this will be dealt with at the time of reporting.
timing_.encode_start_ms = ntp_time_ms_ + timing.encode_start_delta_ms;
timing_.encode_finish_ms = ntp_time_ms_ + timing.encode_finish_delta_ms;
timing_.packetization_finish_ms =
ntp_time_ms_ + timing.packetization_finish_delta_ms;
timing_.pacer_exit_ms = ntp_time_ms_ + timing.pacer_exit_delta_ms;
timing_.network_timestamp_ms =
ntp_time_ms_ + timing.network_timestamp_delta_ms;
timing_.network2_timestamp_ms =
ntp_time_ms_ + timing.network2_timestamp_delta_ms;
}
timing_.receive_start_ms = first_packet_received_time;
timing_.receive_finish_ms = last_packet_received_time;
timing_.flags = timing.flags;
is_last_spatial_layer = markerBit;
}
RtpFrameObject::~RtpFrameObject() {}
uint16_t RtpFrameObject::first_seq_num() const {
return first_seq_num_;
}
uint16_t RtpFrameObject::last_seq_num() const {
return last_seq_num_;
}
int RtpFrameObject::times_nacked() const {
return times_nacked_;
}
VideoFrameType RtpFrameObject::frame_type() const {
return rtp_video_header_.frame_type;
}
VideoCodecType RtpFrameObject::codec_type() const {
return codec_type_;
}
int64_t RtpFrameObject::ReceivedTime() const {
return last_packet_received_time_;
}
int64_t RtpFrameObject::RenderTime() const {
return _renderTimeMs;
}
bool RtpFrameObject::delayed_by_retransmission() const {
return times_nacked() > 0;
}
const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const {
return rtp_video_header_;
}
void RtpFrameObject::SetHeaderFromMetadata(const VideoFrameMetadata& metadata) {
rtp_video_header_.SetFromMetadata(metadata);
}
} // namespace webrtc
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