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a370556270
The packets belonging to a frame were kept in PacketBuffer until the frame was decoded. This CL clears the dependencies of an existing RtpFrameObject to PacketBuffer so that we can free up PacketBuffer as soon as the RtpFrameObject is created. Bug: none Change-Id: Ic939be91815519ae1d1c67ada82006417b2d26a3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149818 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28977}
65 lines
2.1 KiB
C++
65 lines
2.1 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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#ifndef MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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#define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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#include "absl/types/optional.h"
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#include "api/video/encoded_frame.h"
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
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namespace webrtc {
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namespace video_coding {
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class PacketBuffer;
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class RtpFrameObject : public EncodedFrame {
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public:
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RtpFrameObject(PacketBuffer* packet_buffer,
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uint16_t first_seq_num,
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uint16_t last_seq_num,
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size_t frame_size,
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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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RtpPacketInfos packet_infos);
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~RtpFrameObject() override;
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uint16_t first_seq_num() const;
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uint16_t last_seq_num() const;
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int times_nacked() const;
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VideoFrameType frame_type() const;
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VideoCodecType codec_type() const;
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int64_t ReceivedTime() const override;
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int64_t RenderTime() const override;
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bool delayed_by_retransmission() const override;
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const RTPVideoHeader& GetRtpVideoHeader() const;
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const absl::optional<RtpGenericFrameDescriptor>& GetGenericFrameDescriptor()
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const;
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const FrameMarking& GetFrameMarking() const;
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private:
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RTPVideoHeader rtp_video_header_;
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absl::optional<RtpGenericFrameDescriptor> rtp_generic_frame_descriptor_;
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VideoCodecType codec_type_;
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uint16_t first_seq_num_;
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uint16_t last_seq_num_;
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int64_t last_packet_received_time_;
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// Equal to times nacked of the packet with the highet times nacked
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// belonging to this frame.
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int times_nacked_;
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};
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} // namespace video_coding
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} // namespace webrtc
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#endif // MODULES_VIDEO_CODING_FRAME_OBJECT_H_
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