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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.
*/
#ifndef MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
#define MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
#include <cstdint>
#include <optional>
#include <variant>
#include <vector>
#include "api/rtp_packet_infos.h"
#include "api/scoped_refptr.h"
#include "api/video/color_space.h"
#include "api/video/encoded_frame.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 {
class RtpFrameObject : public EncodedFrame {
public:
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);
~RtpFrameObject() override;
uint16_t first_seq_num() const;
uint16_t last_seq_num() const;
int times_nacked() const;
VideoFrameType frame_type() const;
VideoCodecType codec_type() const;
int64_t ReceivedTime() const override;
int64_t RenderTime() const override;
bool delayed_by_retransmission() const override;
const RTPVideoHeader& GetRtpVideoHeader() const;
uint8_t* mutable_data() { return image_buffer_->data(); }
const std::vector<uint32_t>& Csrcs() const { return csrcs_; }
void SetFirstSeqNum(uint16_t first_seq_num) {
first_seq_num_ = first_seq_num;
}
void SetLastSeqNum(uint16_t last_seq_num) { last_seq_num_ = last_seq_num; }
void SetHeaderFromMetadata(const VideoFrameMetadata& metadata);
private:
// Reference for mutable access.
scoped_refptr<EncodedImageBuffer> image_buffer_;
RTPVideoHeader rtp_video_header_;
VideoCodecType codec_type_;
uint16_t first_seq_num_;
uint16_t last_seq_num_;
int64_t last_packet_received_time_;
std::vector<uint32_t> csrcs_;
// Equal to times nacked of the packet with the highet times nacked
// belonging to this frame.
int times_nacked_;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
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