test/fake_network_pipe.cc - src/webrtc - Git at Google

 /*
  *  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 "webrtc/test/fake_network_pipe.h"

 #include <assert.h>
 #include <math.h>
 #include <string.h>

 #include <algorithm>
 #include <cmath>

 #include "webrtc/call/call.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/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
	/*
	* 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 "webrtc/test/fake_network_pipe.h"

	#include <assert.h>
	#include <math.h>
	#include <string.h>

	#include <algorithm>
	#include <cmath>

	#include "webrtc/call/call.h"
	#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
	#include "webrtc/rtc_base/logging.h"
	#include "webrtc/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