blob: 66e8997ca3ced908d6af0b861018fc55cc5e4c89 [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 "test/fake_network_pipe.h"
#include <assert.h>
#include <math.h>
#include <string.h>
#include <algorithm>
#include <cmath>
#include "call/call.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace {
constexpr int64_t kDefaultProcessIntervalMs = 5;
}
DemuxerImpl::DemuxerImpl(const std::map<uint8_t, MediaType>& payload_type_map)
: packet_receiver_(nullptr), payload_type_map_(payload_type_map) {}
void DemuxerImpl::SetReceiver(PacketReceiver* receiver) {
packet_receiver_ = receiver;
}
void DemuxerImpl::DeliverPacket(const NetworkPacket* packet,
const PacketTime& packet_time) {
// No packet receiver means that this demuxer will terminate the flow of
// packets.
if (!packet_receiver_)
return;
const uint8_t* const packet_data = packet->data();
const size_t packet_length = packet->data_length();
MediaType media_type = MediaType::ANY;
if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) {
RTC_CHECK_GE(packet_length, 2);
const uint8_t payload_type = packet_data[1] & 0x7f;
std::map<uint8_t, MediaType>::const_iterator it =
payload_type_map_.find(payload_type);
RTC_CHECK(it != payload_type_map_.end())
<< "payload type " << static_cast<int>(payload_type) << " unknown.";
media_type = it->second;
}
packet_receiver_->DeliverPacket(media_type, packet_data, packet_length,
packet_time);
}
FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
const FakeNetworkPipe::Config& config,
std::unique_ptr<Demuxer> demuxer)
: FakeNetworkPipe(clock, config, std::move(demuxer), 1) {}
FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
const FakeNetworkPipe::Config& config,
std::unique_ptr<Demuxer> demuxer,
uint64_t seed)
: clock_(clock),
demuxer_(std::move(demuxer)),
random_(seed),
config_(),
dropped_packets_(0),
sent_packets_(0),
total_packet_delay_(0),
bursting_(false),
next_process_time_(clock_->TimeInMilliseconds()),
last_log_time_(clock_->TimeInMilliseconds()) {
SetConfig(config);
}
FakeNetworkPipe::~FakeNetworkPipe() {
while (!capacity_link_.empty()) {
delete capacity_link_.front();
capacity_link_.pop();
}
while (!delay_link_.empty()) {
delete *delay_link_.begin();
delay_link_.erase(delay_link_.begin());
}
}
void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) {
RTC_CHECK(demuxer_);
demuxer_->SetReceiver(receiver);
}
void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) {
rtc::CritScope crit(&lock_);
config_ = config; // Shallow copy of the struct.
double prob_loss = config.loss_percent / 100.0;
if (config_.avg_burst_loss_length == -1) {
// Uniform loss
prob_loss_bursting_ = prob_loss;
prob_start_bursting_ = prob_loss;
} else {
// Lose packets according to a gilbert-elliot model.
int avg_burst_loss_length = config.avg_burst_loss_length;
int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));
RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
<< "For a total packet loss of " << config.loss_percent << "%% then"
<< " avg_burst_loss_length must be " << min_avg_burst_loss_length + 1
<< " or higher.";
prob_loss_bursting_ = (1.0 - 1.0 / avg_burst_loss_length);
prob_start_bursting_ = prob_loss / (1 - prob_loss) / avg_burst_loss_length;
}
}
void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) {
RTC_CHECK(demuxer_);
rtc::CritScope crit(&lock_);
if (config_.queue_length_packets > 0 &&
capacity_link_.size() >= config_.queue_length_packets) {
// Too many packet on the link, drop this one.
++dropped_packets_;
return;
}
int64_t time_now = clock_->TimeInMilliseconds();
// Delay introduced by the link capacity.
int64_t capacity_delay_ms = 0;
if (config_.link_capacity_kbps > 0) {
const int bytes_per_millisecond = config_.link_capacity_kbps / 8;
// To round to the closest millisecond we add half a milliseconds worth of
// bytes to the delay calculation.
capacity_delay_ms = (data_length + capacity_delay_error_bytes_ +
bytes_per_millisecond / 2) /
bytes_per_millisecond;
capacity_delay_error_bytes_ +=
data_length - capacity_delay_ms * bytes_per_millisecond;
}
int64_t network_start_time = time_now;
// Check if there already are packets on the link and change network start
// time forward if there is.
if (!capacity_link_.empty() &&
network_start_time < capacity_link_.back()->arrival_time())
network_start_time = capacity_link_.back()->arrival_time();
int64_t arrival_time = network_start_time + capacity_delay_ms;
NetworkPacket* packet = new NetworkPacket(data, data_length, time_now,
arrival_time);
capacity_link_.push(packet);
}
float FakeNetworkPipe::PercentageLoss() {
rtc::CritScope crit(&lock_);
if (sent_packets_ == 0)
return 0;
return static_cast<float>(dropped_packets_) /
(sent_packets_ + dropped_packets_);
}
int FakeNetworkPipe::AverageDelay() {
rtc::CritScope crit(&lock_);
if (sent_packets_ == 0)
return 0;
return static_cast<int>(total_packet_delay_ /
static_cast<int64_t>(sent_packets_));
}
void FakeNetworkPipe::Process() {
int64_t time_now = clock_->TimeInMilliseconds();
std::queue<NetworkPacket*> packets_to_deliver;
{
rtc::CritScope crit(&lock_);
if (time_now - last_log_time_ > 5000) {
int64_t queueing_delay_ms = 0;
if (!capacity_link_.empty()) {
queueing_delay_ms = time_now - capacity_link_.front()->send_time();
}
LOG(LS_INFO) << "Network queue: " << queueing_delay_ms << " ms.";
last_log_time_ = time_now;
}
// Check the capacity link first.
while (!capacity_link_.empty() &&
time_now >= capacity_link_.front()->arrival_time()) {
// Time to get this packet.
NetworkPacket* packet = capacity_link_.front();
capacity_link_.pop();
// Drop packets at an average rate of |config_.loss_percent| with
// and average loss burst length of |config_.avg_burst_loss_length|.
if ((bursting_ && random_.Rand<double>() < prob_loss_bursting_) ||
(!bursting_ && random_.Rand<double>() < prob_start_bursting_)) {
bursting_ = true;
delete packet;
continue;
} else {
bursting_ = false;
}
int arrival_time_jitter = random_.Gaussian(
config_.queue_delay_ms, config_.delay_standard_deviation_ms);
// If reordering is not allowed then adjust arrival_time_jitter
// to make sure all packets are sent in order.
if (!config_.allow_reordering && !delay_link_.empty() &&
packet->arrival_time() + arrival_time_jitter <
(*delay_link_.rbegin())->arrival_time()) {
arrival_time_jitter =
(*delay_link_.rbegin())->arrival_time() - packet->arrival_time();
}
packet->IncrementArrivalTime(arrival_time_jitter);
delay_link_.insert(packet);
}
// Check the extra delay queue.
while (!delay_link_.empty() &&
time_now >= (*delay_link_.begin())->arrival_time()) {
// Deliver this packet.
NetworkPacket* packet = *delay_link_.begin();
packets_to_deliver.push(packet);
delay_link_.erase(delay_link_.begin());
// |time_now| might be later than when the packet should have arrived, due
// to NetworkProcess being called too late. For stats, use the time it
// should have been on the link.
total_packet_delay_ += packet->arrival_time() - packet->send_time();
}
sent_packets_ += packets_to_deliver.size();
}
while (!packets_to_deliver.empty()) {
NetworkPacket* packet = packets_to_deliver.front();
packets_to_deliver.pop();
demuxer_->DeliverPacket(packet, PacketTime());
delete packet;
}
next_process_time_ = !delay_link_.empty()
? (*delay_link_.begin())->arrival_time()
: time_now + kDefaultProcessIntervalMs;
}
int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
rtc::CritScope crit(&lock_);
return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
0);
}
} // namespace webrtc