blob: b0c10c07ff20a1ef93666239a0b176d444c99486 [file] [log] [blame]
* 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.
#include <memory>
#include <set>
#include <string.h>
#include <queue>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/random.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class Clock;
class CriticalSectionWrapper;
class PacketReceiver;
class NetworkPacket {
NetworkPacket(const uint8_t* data,
size_t length,
int64_t send_time,
int64_t arrival_time)
: data_(new uint8_t[length]),
arrival_time_(arrival_time) {
memcpy(data_.get(), data, length);
uint8_t* data() const { return data_.get(); }
size_t data_length() const { return data_length_; }
int64_t send_time() const { return send_time_; }
int64_t arrival_time() const { return arrival_time_; }
void IncrementArrivalTime(int64_t extra_delay) {
arrival_time_ += extra_delay;
// The packet data.
std::unique_ptr<uint8_t[]> data_;
// Length of data_.
size_t data_length_;
// The time the packet was sent out on the network.
const int64_t send_time_;
// The time the packet should arrive at the receiver.
int64_t arrival_time_;
// 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.
class FakeNetworkPipe {
struct Config {
Config() {}
// Queue length in number of packets.
size_t queue_length_packets = 0;
// Delay in addition to capacity induced delay.
int queue_delay_ms = 0;
// Standard deviation of the extra delay.
int delay_standard_deviation_ms = 0;
// Link capacity in kbps.
int link_capacity_kbps = 0;
// Random packet loss.
int loss_percent = 0;
// If packets are allowed to be reordered.
bool allow_reordering = false;
// The average length of a burst of lost packets.
int avg_burst_loss_length = -1;
FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config);
FakeNetworkPipe(Clock* clock,
const FakeNetworkPipe::Config& config,
uint64_t seed);
// 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_; }
Clock* const clock_;
rtc::CriticalSection lock_;
PacketReceiver* packet_receiver_;
std::queue<NetworkPacket*> capacity_link_;
Random random_;
// Since we need to access both the packet with the earliest and latest
// arrival time we need to use a multiset to keep all packets sorted,
// hence, we cannot use a priority queue.
struct PacketArrivalTimeComparator {
bool operator()(const NetworkPacket* p1, const NetworkPacket* p2) {
return p1->arrival_time() < p2->arrival_time();
std::multiset<NetworkPacket*, PacketArrivalTimeComparator> delay_link_;
// Link configuration.
Config config_;
// Statistics.
size_t dropped_packets_;
size_t sent_packets_;
int64_t total_packet_delay_;
// Are we currently dropping a burst of packets?
bool bursting_;
// The probability to drop the packet if we are currently dropping a
// burst of packet
double prob_loss_bursting_;
// The probability to drop a burst of packets.
double prob_start_bursting_;
int64_t next_process_time_;
} // namespace webrtc