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be9b576188
Step 1 of combining the sender and receiver types Also moved the RtpFrameObject to rtp_rtcp/source, as it's heavily used by the transformable receiver frame, I couldn't work out a better way of managing the dependencies, and everything else seemed to work fine. Bug: chromium:1412687 Change-Id: I55e816a0d7aa2962560ff9ebaf30ad63ab0b9810 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291710 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Tony Herre <herre@google.com> Cr-Commit-Position: refs/heads/main@{#39255}
73 lines
2.4 KiB
C++
73 lines
2.4 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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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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