156 lines
4.6 KiB
C++
156 lines
4.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
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#define WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
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#include <memory>
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#include <set>
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#include <string.h>
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#include <queue>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/random.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class Clock;
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class CriticalSectionWrapper;
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class PacketReceiver;
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class NetworkPacket {
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public:
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NetworkPacket(const uint8_t* data,
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size_t length,
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int64_t send_time,
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int64_t arrival_time)
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: data_(new uint8_t[length]),
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data_length_(length),
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send_time_(send_time),
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arrival_time_(arrival_time) {
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memcpy(data_.get(), data, length);
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}
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uint8_t* data() const { return data_.get(); }
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size_t data_length() const { return data_length_; }
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int64_t send_time() const { return send_time_; }
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int64_t arrival_time() const { return arrival_time_; }
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void IncrementArrivalTime(int64_t extra_delay) {
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arrival_time_ += extra_delay;
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}
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private:
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// The packet data.
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std::unique_ptr<uint8_t[]> data_;
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// Length of data_.
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size_t data_length_;
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// The time the packet was sent out on the network.
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const int64_t send_time_;
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// The time the packet should arrive at the receiver.
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int64_t arrival_time_;
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};
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// Class faking a network link. This is a simple and naive solution just faking
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// capacity and adding an extra transport delay in addition to the capacity
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// introduced delay.
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// TODO(mflodman) Add random and bursty packet loss.
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class FakeNetworkPipe {
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public:
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struct Config {
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Config() {}
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// Queue length in number of packets.
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size_t queue_length_packets = 0;
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// Delay in addition to capacity induced delay.
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int queue_delay_ms = 0;
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// Standard deviation of the extra delay.
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int delay_standard_deviation_ms = 0;
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// Link capacity in kbps.
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int link_capacity_kbps = 0;
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// Random packet loss.
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int loss_percent = 0;
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// If packets are allowed to be reordered.
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bool allow_reordering = false;
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// The average length of a burst of lost packets.
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int avg_burst_loss_length = -1;
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};
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FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config);
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FakeNetworkPipe(Clock* clock,
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const FakeNetworkPipe::Config& config,
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uint64_t seed);
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~FakeNetworkPipe();
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// Must not be called in parallel with SendPacket or Process.
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void SetReceiver(PacketReceiver* receiver);
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// Sets a new configuration. This won't affect packets already in the pipe.
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void SetConfig(const FakeNetworkPipe::Config& config);
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// Sends a new packet to the link.
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void SendPacket(const uint8_t* packet, size_t packet_length);
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// Processes the network queues and trigger PacketReceiver::IncomingPacket for
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// packets ready to be delivered.
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void Process();
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int64_t TimeUntilNextProcess() const;
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// Get statistics.
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float PercentageLoss();
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int AverageDelay();
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size_t dropped_packets() { return dropped_packets_; }
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size_t sent_packets() { return sent_packets_; }
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private:
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Clock* const clock_;
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rtc::CriticalSection lock_;
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PacketReceiver* packet_receiver_;
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std::queue<NetworkPacket*> capacity_link_;
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Random random_;
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// Since we need to access both the packet with the earliest and latest
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// arrival time we need to use a multiset to keep all packets sorted,
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// hence, we cannot use a priority queue.
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struct PacketArrivalTimeComparator {
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bool operator()(const NetworkPacket* p1, const NetworkPacket* p2) {
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return p1->arrival_time() < p2->arrival_time();
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}
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};
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std::multiset<NetworkPacket*, PacketArrivalTimeComparator> delay_link_;
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// Link configuration.
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Config config_;
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// Statistics.
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size_t dropped_packets_;
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size_t sent_packets_;
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int64_t total_packet_delay_;
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// Are we currently dropping a burst of packets?
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bool bursting_;
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// The probability to drop the packet if we are currently dropping a
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// burst of packet
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double prob_loss_bursting_;
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// The probability to drop a burst of packets.
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double prob_start_bursting_;
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int64_t next_process_time_;
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RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
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};
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} // namespace webrtc
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#endif // WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
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