Allow FakeNetworkPipe to drop packets in bursts.

webrtc/test/fake_network_pipe.cc

 /*
  *  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.
  */
 
@@ skipping 16 matching lines @@
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
                                  const FakeNetworkPipe::Config& config,
                                  uint64_t seed)
     : clock_(clock),
       packet_receiver_(NULL),
       random_(seed),
       config_(config),
       dropped_packets_(0),
       sent_packets_(0),
       total_packet_delay_(0),
-      next_process_time_(clock_->TimeInMilliseconds()) {}
+      bursting_(false),
+      next_process_time_(clock_->TimeInMilliseconds()) {
+  double prob_loss = config.loss_percent / 100.0;
+  if (config_.avg_burst_loss_length == -1) {
+    // Uniform loss
+    prob_loss_bursting_ = 0;

[terelius, 2016/05/30 15:40:23]

This won't really be uniform. After each loss, the probability is zero to lose
another packet.

[philipel, 2016/05/30 16:05:39]

Fixed.

+    prob_start_bursting_ = prob_loss;
+  } else {
+    // Lose packets according to geometric distribution.
+    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;
+  }
+}
 
 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());
   }
@@ skipping 63 matching lines @@
   std::queue<NetworkPacket*> packets_to_deliver;
   {
     rtc::CritScope crit(&lock_);
     // 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();
 
-      // Packets are randomly dropped after being affected by the bottleneck.
-      if (random_.Rand(100) < static_cast<uint32_t>(config_.loss_percent)) {
+      // 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()) {
@@ skipping 32 matching lines @@
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
   rtc::CritScope crit(&lock_);
   const int64_t kDefaultProcessIntervalMs = 30;
   if (capacity_link_.empty() || delay_link_.empty())
     return kDefaultProcessIntervalMs;
   return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
                            0);
 }
 
 }  // namespace webrtc