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ff7a557f2e
The purpose is to be able to create a RtpPacketSendInfo from Pacing and RtpPacketSendInfo only. This allow further refactoring where we directly in PacketRouter can notify BWE and early loss detection that a packet will be sent. RtpPacketSendInfo::From is mostly added to be able to test conversion. Bug: webrtc:15368 Change-Id: I5ebe2dc91d2eedf2c86e62c3f9738437082a49e4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343766 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41961}
63 lines
2.3 KiB
C++
63 lines
2.3 KiB
C++
/*
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* Copyright (c) 2024 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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#include <cstdint>
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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namespace webrtc {
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RtpPacketSendInfo RtpPacketSendInfo::From(const RtpPacketToSend& packet,
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const PacedPacketInfo& pacing_info) {
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RtpPacketSendInfo packet_info;
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if (packet.transport_sequence_number()) {
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packet_info.transport_sequence_number =
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*packet.transport_sequence_number() & 0xFFFF;
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} else {
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absl::optional<uint16_t> packet_id =
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packet.GetExtension<TransportSequenceNumber>();
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if (packet_id) {
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packet_info.transport_sequence_number = *packet_id;
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}
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}
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packet_info.rtp_timestamp = packet.Timestamp();
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packet_info.length = packet.size();
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packet_info.pacing_info = pacing_info;
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packet_info.packet_type = packet.packet_type();
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switch (*packet_info.packet_type) {
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case RtpPacketMediaType::kAudio:
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case RtpPacketMediaType::kVideo:
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packet_info.media_ssrc = packet.Ssrc();
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packet_info.rtp_sequence_number = packet.SequenceNumber();
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break;
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case RtpPacketMediaType::kRetransmission:
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RTC_DCHECK(packet.original_ssrc() &&
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packet.retransmitted_sequence_number());
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// For retransmissions, we're want to remove the original media packet
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// if the retransmit arrives - so populate that in the packet info.
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packet_info.media_ssrc = packet.original_ssrc().value_or(0);
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packet_info.rtp_sequence_number =
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packet.retransmitted_sequence_number().value_or(0);
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break;
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case RtpPacketMediaType::kPadding:
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case RtpPacketMediaType::kForwardErrorCorrection:
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// We're not interested in feedback about these packets being received
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// or lost.
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break;
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}
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return packet_info;
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}
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} // namespace webrtc
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