/*
 *  Copyright (c) 2018 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 "call/call_config.h"

#include "api/environment/environment.h"
#include "api/task_queue/task_queue_base.h"
#include "call/rtp_transport_config.h"

namespace webrtc {

CallConfig::CallConfig(const Environment& env,
                       TaskQueueBase* network_task_queue)
    : env(env), network_task_queue_(network_task_queue) {}

RtpTransportConfig CallConfig::ExtractTransportConfig() const {
  RtpTransportConfig transport_config = {.env = env};
  transport_config.bitrate_config = bitrate_config;
  transport_config.network_controller_factory =
      per_call_network_controller_factory
          ? per_call_network_controller_factory.get()
          : network_controller_factory;
  transport_config.network_state_predictor_factory =
      network_state_predictor_factory;
  if (pacer_burst_interval.has_value()) {
    transport_config.default_pacing_time_window = *pacer_burst_interval;
  }
  return transport_config;
}

CallConfig::~CallConfig() = default;

}  // namespace webrtc
