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/*
* 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.
*/
#ifndef CALL_SIMULATED_PACKET_RECEIVER_H_
#define CALL_SIMULATED_PACKET_RECEIVER_H_
#include "api/test/simulated_network.h"
#include "call/packet_receiver.h"
namespace webrtc {
// Private API that is fixing surface between DirectTransport and underlying
// network conditions simulation implementation.
class SimulatedPacketReceiverInterface : public PacketReceiver {
public:
// Must not be called in parallel with DeliverPacket or Process.
// Destination receiver will be injected with this method
virtual void SetReceiver(PacketReceiver* receiver) = 0;
// Reports average packet delay.
virtual int AverageDelay() = 0;
// Process any pending tasks such as timeouts.
// Called on a worker thread.
virtual void Process() = 0;
// Returns the time until next process or nullopt to indicate that the next
// process time is unknown. If the next process time is unknown, this should
// be checked again any time a packet is enqueued.
virtual absl::optional<int64_t> TimeUntilNextProcess() = 0;
};
} // namespace webrtc
#endif // CALL_SIMULATED_PACKET_RECEIVER_H_
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