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/*
* Copyright (c) 2012 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 WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
#define WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
#include <queue>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class CriticalSectionWrapper;
class NetworkPacket;
class PacketReceiver;
// Class faking a network link. This is a simple and naive solution just faking
// capacity and adding an extra transport delay in addition to the capacity
// introduced delay.
// TODO(mflodman) Add random and bursty packet loss.
class FakeNetworkPipe {
public:
struct Config {
Config()
: queue_length_packets(0),
queue_delay_ms(0),
delay_standard_deviation_ms(0),
link_capacity_kbps(0),
loss_percent(0) {
}
// Queue length in number of packets.
size_t queue_length_packets;
// Delay in addition to capacity induced delay.
int queue_delay_ms;
// Standard deviation of the extra delay.
int delay_standard_deviation_ms;
// Link capacity in kbps.
int link_capacity_kbps;
// Random packet loss.
int loss_percent;
};
explicit FakeNetworkPipe(const FakeNetworkPipe::Config& config);
~FakeNetworkPipe();
// Must not be called in parallel with SendPacket or Process.
void SetReceiver(PacketReceiver* receiver);
// Sets a new configuration. This won't affect packets already in the pipe.
void SetConfig(const FakeNetworkPipe::Config& config);
// Sends a new packet to the link.
void SendPacket(const uint8_t* packet, size_t packet_length);
// Processes the network queues and trigger PacketReceiver::IncomingPacket for
// packets ready to be delivered.
void Process();
int64_t TimeUntilNextProcess() const;
// Get statistics.
float PercentageLoss();
int AverageDelay();
size_t dropped_packets() { return dropped_packets_; }
size_t sent_packets() { return sent_packets_; }
private:
mutable rtc::CriticalSection lock_;
PacketReceiver* packet_receiver_;
std::queue<NetworkPacket*> capacity_link_;
std::queue<NetworkPacket*> delay_link_;
// Link configuration.
Config config_;
// Statistics.
size_t dropped_packets_;
size_t sent_packets_;
int total_packet_delay_;
int64_t next_process_time_;
RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
};
} // namespace webrtc
#endif // WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
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