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9ecc76e15b
Bug: webrtc:13757 Change-Id: I7f8fc1a9c4cbf464b3969c4754ce5aa9c5b5f076 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303500 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39960}
77 lines
2.8 KiB
C++
77 lines
2.8 KiB
C++
/*
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* Copyright (c) 2021 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_PACKET_SEQUENCER_H_
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#define MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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// Helper class used to assign RTP sequence numbers and populate some fields for
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// padding packets based on the last sequenced packets.
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// This class is not thread safe, the caller must provide that.
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class PacketSequencer {
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public:
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// If `require_marker_before_media_padding_` is true, padding packets on the
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// media ssrc is not allowed unless the last sequenced media packet had the
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// marker bit set (i.e. don't insert padding packets between the first and
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// last packets of a video frame).
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// Packets with unknown SSRCs will be ignored.
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PacketSequencer(uint32_t media_ssrc,
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absl::optional<uint32_t> rtx_ssrc,
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bool require_marker_before_media_padding,
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Clock* clock);
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// Assigns sequence number, and in the case of non-RTX padding also timestamps
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// and payload type.
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void Sequence(RtpPacketToSend& packet);
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void set_media_sequence_number(uint16_t sequence_number) {
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media_sequence_number_ = sequence_number;
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}
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void set_rtx_sequence_number(uint16_t sequence_number) {
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rtx_sequence_number_ = sequence_number;
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}
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void SetRtpState(const RtpState& state);
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void PopulateRtpState(RtpState& state) const;
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uint16_t media_sequence_number() const { return media_sequence_number_; }
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uint16_t rtx_sequence_number() const { return rtx_sequence_number_; }
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// Checks whether it is allowed to send padding on the media SSRC at this
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// time, e.g. that we don't send padding in the middle of a video frame.
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bool CanSendPaddingOnMediaSsrc() const;
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private:
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void UpdateLastPacketState(const RtpPacketToSend& packet);
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void PopulatePaddingFields(RtpPacketToSend& packet);
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const uint32_t media_ssrc_;
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const absl::optional<uint32_t> rtx_ssrc_;
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const bool require_marker_before_media_padding_;
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Clock* const clock_;
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uint16_t media_sequence_number_;
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uint16_t rtx_sequence_number_;
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int8_t last_payload_type_;
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uint32_t last_rtp_timestamp_;
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Timestamp last_capture_time_ = Timestamp::MinusInfinity();
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Timestamp last_timestamp_time_ = Timestamp::MinusInfinity();
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bool last_packet_marker_bit_;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_PACKET_SEQUENCER_H_
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