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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_VIDEO_CODING_FRAME_OBJECT_H_
#define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
#include "absl/types/optional.h"
#include "api/video/encoded_frame.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
namespace webrtc {
namespace video_coding {
class PacketBuffer;
class RtpFrameObject : public EncodedFrame {
public:
RtpFrameObject(PacketBuffer* packet_buffer,
uint16_t first_seq_num,
uint16_t last_seq_num,
size_t frame_size,
int times_nacked,
int64_t first_packet_received_time,
int64_t last_packet_received_time);
~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;
absl::optional<RTPVideoHeader> GetRtpVideoHeader() const;
absl::optional<RtpGenericFrameDescriptor> GetGenericFrameDescriptor() const;
absl::optional<FrameMarking> GetFrameMarking() const;
private:
void AllocateBitstreamBuffer(size_t frame_size);
rtc::scoped_refptr<PacketBuffer> packet_buffer_;
VideoFrameType frame_type_;
VideoCodecType codec_type_;
uint16_t first_seq_num_;
uint16_t last_seq_num_;
int64_t last_packet_received_time_;
// Equal to times nacked of the packet with the highet times nacked
// belonging to this frame.
int times_nacked_;
};
} // namespace video_coding
} // namespace webrtc
#endif // MODULES_VIDEO_CODING_FRAME_OBJECT_H_