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/*
* Copyright (c) 2021 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.
*/
#ifndef CALL_RTP_TRANSPORT_CONFIG_H_
#define CALL_RTP_TRANSPORT_CONFIG_H_
#include <memory>
#include "absl/types/optional.h"
#include "api/environment/environment.h"
#include "api/network_state_predictor.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/network_control.h"
#include "api/units/time_delta.h"
namespace webrtc {
struct RtpTransportConfig {
Environment env;
// Bitrate config used until valid bitrate estimates are calculated. Also
// used to cap total bitrate used. This comes from the remote connection.
BitrateConstraints bitrate_config;
// NetworkStatePredictor to use for this call.
NetworkStatePredictorFactoryInterface* network_state_predictor_factory =
nullptr;
// Network controller factory to use for this call.
NetworkControllerFactoryInterface* network_controller_factory = nullptr;
// The burst interval of the pacer, see TaskQueuePacedSender constructor.
absl::optional<TimeDelta> pacer_burst_interval;
};
} // namespace webrtc
#endif // CALL_RTP_TRANSPORT_CONFIG_H_
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