video/timing/simulator/rtp_packet_simulator.cc

/*
 *  Copyright (c) 2025 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "video/timing/simulator/rtp_packet_simulator.h"

#include "api/environment/environment.h"
#include "api/rtp_headers.h"
#include "logging/rtc_event_log/events/logged_rtp_rtcp.h"
#include "logging/rtc_event_log/rtc_event_log_parser.h"
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"

namespace webrtc::video_timing_simulator {

RtpPacketSimulator::RtpPacketSimulator(const Environment& env)
    : env_(env),
      rtp_header_extension_map_(
          ParsedRtcEventLog::GetDefaultHeaderExtensionMap()) {}

RtpPacketReceived RtpPacketSimulator::SimulateRtpPacketReceived(
    const LoggedRtpPacket& logged_packet) const {
  RtpPacketReceived rtp_packet(&rtp_header_extension_map_);
  rtp_packet.set_arrival_time(env_.clock().CurrentTime());

  // RTP header.
  const RTPHeader& header = logged_packet.header;
  rtp_packet.SetMarker(header.markerBit);
  rtp_packet.SetPayloadType(header.payloadType);
  rtp_packet.SetSequenceNumber(header.sequenceNumber);
  rtp_packet.SetTimestamp(header.timestamp);
  rtp_packet.SetSsrc(header.ssrc);

  // RTP header extensions.
  const RTPHeaderExtension& extension = header.extension;
  if (extension.hasTransportSequenceNumber) {
    rtp_packet.SetExtension<TransportSequenceNumber>(
        extension.transportSequenceNumber);
  }
  if (extension.hasTransmissionTimeOffset) {
    rtp_packet.SetExtension<TransmissionOffset>(
        extension.transmissionTimeOffset);
  }
  if (extension.hasAbsoluteSendTime) {
    rtp_packet.SetExtension<AbsoluteSendTime>(extension.absoluteSendTime);
  }
  rtp_packet.SetRawExtension<RtpDependencyDescriptorExtension>(
      logged_packet.dependency_descriptor_wire_format);

  // Payload and padding.
  rtp_packet.AllocatePayload(logged_packet.total_length -
                             logged_packet.header_length -
                             header.paddingLength);
  rtp_packet.SetPadding(header.paddingLength);

  return rtp_packet;
}

}  // namespace webrtc::video_timing_simulator