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With this turned on, packets will be sequence number after the pacing stage rather that during packetization. This avoids a race where packets may be sent out of order, and paves the way for the ability to cull packets from the pacer queue without causing sequence number gaps. For now, the feature is off by default. Follow-ups will enable it for video and audio separately. Bug: webrtc:11340, webrtc:12470 Change-Id: I6d411d8c85b9047e3e9b05ff4c2c3ed97c579aa1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208584 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34661}
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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int64_t last_capture_time_ms_;
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int64_t last_timestamp_time_ms_;
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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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