Repo created

TMessagesProj/jni/voip/webrtc/call/simulated_packet_receiver.h

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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_