「Mutilated Webrtc」Fec Encode

从这里计算丢包率

void RtpTransportControllerSend::OnReceivedRtcpReceiverReportBlocks(
    const ReportBlockList& report_blocks,
    int64_t now_ms) {
  if (report_blocks.empty())
    return;

  int total_packets_lost_delta = 0;
  int total_packets_delta = 0;

  // Compute the packet loss from all report blocks.
  for (const RTCPReportBlock& report_block : report_blocks) {
    auto it = last_report_blocks_.find(report_block.source_ssrc);
    if (it != last_report_blocks_.end()) {
      auto number_of_packets = report_block.extended_highest_sequence_number -
                               it->second.extended_highest_sequence_number;
      total_packets_delta += number_of_packets;
      auto lost_delta = report_block.packets_lost - it->second.packets_lost;
      total_packets_lost_delta += lost_delta;
    }
    last_report_blocks_[report_block.source_ssrc] = report_block;
  }
  // Can only compute delta if there has been previous blocks to compare to. If
  // not, total_packets_delta will be unchanged and there's nothing more to do.
  if (!total_packets_delta)
    return;
  int packets_received_delta = total_packets_delta - total_packets_lost_delta;
  // To detect lost packets, at least one packet has to be received. This check
  // is needed to avoid bandwith detection update in
  // VideoSendStreamTest.SuspendBelowMinBitrate

  if (packets_received_delta < 1)
    return;
  Timestamp now = Timestamp::Millis(now_ms);
  TransportLossReport msg;
  msg.packets_lost_delta = total_packets_lost_delta;
  msg.packets_received_delta = packets_received_delta;
  msg.receive_time = now;
  msg.start_time = last_report_block_time_;
  msg.end_time = now;
  if (controller_) {
    RTC_LOG(LS_ERROR) << "kiki-fec: RtpTransportControllerSend::OnReceivedRtcpReceiverReportBlocks";
    PostUpdates(controller_->OnTransportLossReport(msg));
  }
  last_report_block_time_ = now;
}

123

void RtpTransportControllerSend::PostUpdates(NetworkControlUpdate update) {
  if (update.congestion_window) {
    pacer()->SetCongestionWindow(*update.congestion_window);
  }
  if (update.pacer_config) {
    pacer()->SetPacingRates(update.pacer_config->data_rate(),
                            update.pacer_config->pad_rate());
  }
  for (const auto& probe : update.probe_cluster_configs) {
    pacer()->CreateProbeCluster(probe.target_data_rate, probe.id);
  }
  if (update.target_rate) {
    control_handler_->SetTargetRate(*update.target_rate);
    UpdateControlState();
  }
}

在这里被设置

void CongestionControlHandler::SetTargetRate(
    TargetTransferRate new_target_rate) {
  RTC_DCHECK_RUN_ON(&sequenced_checker_);
  RTC_CHECK(new_target_rate.at_time.IsFinite());
  last_incoming_ = new_target_rate;
}

这里定期获得最后的记录，传递下去。值得注意的是，这里有个标志位disable_pacer_emergency_stop_是0。

void RtpTransportControllerSend::UpdateControlState() {
  absl::optional<TargetTransferRate> update = control_handler_->GetUpdate();
  if (!update)
    return;
  retransmission_rate_limiter_.SetMaxRate(update->target_rate.bps());
  // We won't create control_handler_ until we have an observers.
  RTC_DCHECK(observer_ != nullptr);
  observer_->OnTargetTransferRate(*update);
}

absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
  RTC_DCHECK_RUN_ON(&sequenced_checker_);
  if (!last_incoming_.has_value())
    return absl::nullopt;
  TargetTransferRate new_outgoing = *last_incoming_;
  DataRate log_target_rate = new_outgoing.target_rate;
  bool pause_encoding = false;
  if (!network_available_) {
    pause_encoding = true;
  } else if (!disable_pacer_emergency_stop_ &&
             pacer_expected_queue_ms_ > PacedSender::kMaxQueueLengthMs) {
    pause_encoding = true;
  }
  if (pause_encoding)
    new_outgoing.target_rate = DataRate::Zero();
  if (!last_reported_ ||
      last_reported_->target_rate != new_outgoing.target_rate ||
      (!new_outgoing.target_rate.IsZero() &&
       (last_reported_->network_estimate.loss_rate_ratio !=
            new_outgoing.network_estimate.loss_rate_ratio ||
        last_reported_->network_estimate.round_trip_time !=
            new_outgoing.network_estimate.round_trip_time))) {
    if (encoder_paused_in_last_report_ != pause_encoding)
      RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
                       << ToString(log_target_rate) << ".";
    encoder_paused_in_last_report_ = pause_encoding;
    last_reported_ = new_outgoing;
    return new_outgoing;
  }
  return absl::nullopt;
}

 

Call::OnTargetTransferRate

void BitrateAllocator::OnNetworkEstimateChanged(TargetTransferRate msg) {

src/video/video_send_stream_impl.cc

uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) {
    ...

    rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_->GetSendFrameRate());

    ...
}

src/call/rtp_video_sender.cc

void RtpVideoSender::OnBitrateUpdated(BitrateAllocationUpdate update,
                                      int framerate) {
    ...

    encoder_target_rate_bps_ = fec_controller_->UpdateFecRates(
        payload_bitrate_bps, framerate,
        rtc::saturated_cast<uint8_t>(update.packet_loss_ratio * 256),
        loss_mask_vector_, update.round_trip_time.ms());

    ...
}