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096427e494
This allows encoded frames to be written to any encoded insertable streams writer without needing to somehow set valid RTP sequence numbers. Assumes streams are using the Dependency Descriptor header ext. A short term fix while we discuss whether we can remove the sequence number check in RtpFrameReferenceFinder::ManageFrame. Bug: chromium:1439799 Change-Id: I3c1d83793cd8b6cae2a8ad2129b3b6daab1d11c4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302301 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#39966}
78 lines
2.6 KiB
C++
78 lines
2.6 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_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
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#define MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/video/encoded_frame.h"
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namespace webrtc {
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class RtpFrameObject : public EncodedFrame {
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public:
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RtpFrameObject(uint16_t first_seq_num,
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uint16_t last_seq_num,
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bool markerBit,
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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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uint32_t rtp_timestamp,
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int64_t ntp_time_ms,
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const VideoSendTiming& timing,
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uint8_t payload_type,
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VideoCodecType codec,
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VideoRotation rotation,
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VideoContentType content_type,
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const RTPVideoHeader& video_header,
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const absl::optional<webrtc::ColorSpace>& color_space,
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RtpPacketInfos packet_infos,
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rtc::scoped_refptr<EncodedImageBuffer> image_buffer);
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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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uint8_t* mutable_data() { return image_buffer_->data(); }
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const std::vector<uint32_t>& Csrcs() const { return csrcs_; }
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void SetFirstSeqNum(uint16_t first_seq_num) {
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first_seq_num_ = first_seq_num;
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}
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void SetLastSeqNum(uint16_t last_seq_num) { last_seq_num_ = last_seq_num; }
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private:
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// Reference for mutable access.
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rtc::scoped_refptr<EncodedImageBuffer> image_buffer_;
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RTPVideoHeader rtp_video_header_;
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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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std::vector<uint32_t> csrcs_;
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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 webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_FRAME_OBJECT_H_
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