chromium/third_party/webrtc/modules/rtp_rtcp/source/frame_object.cc

/*
 *  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 <string.h>

#include <utility>

#include "api/video/encoded_image.h"
#include "api/video/video_timing.h"
#include "rtc_base/checks.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 absl::optional<webrtc::ColorSpace>& color_space,
    RtpPacketInfos packet_infos,
    rtc::scoped_refptr<EncodedImageBuffer> image_buffer)
    : … { … }

RtpFrameObject::~RtpFrameObject() { … }

uint16_t RtpFrameObject::first_seq_num() const { … }

uint16_t RtpFrameObject::last_seq_num() const { … }

int RtpFrameObject::times_nacked() const { … }

VideoFrameType RtpFrameObject::frame_type() const { … }

VideoCodecType RtpFrameObject::codec_type() const { … }

int64_t RtpFrameObject::ReceivedTime() const { … }

int64_t RtpFrameObject::RenderTime() const { … }

bool RtpFrameObject::delayed_by_retransmission() const { … }

const RTPVideoHeader& RtpFrameObject::GetRtpVideoHeader() const { … }

void RtpFrameObject::SetHeaderFromMetadata(const VideoFrameMetadata& metadata) { … }
}  // namespace webrtc