Allow packets to be reordered in the fake network pipe.

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 18 matching lines @@
   double uniform2 = (rand() + 1.0) / (RAND_MAX + 1.0);  // NOLINT
   return static_cast<int>(mean_delay_ms + standard_deviation_ms *
                           sqrt(-2 * log(uniform1)) * cos(2 * kPi * uniform2));
 }
 
 static bool UniformLoss(int loss_percent) {
   int outcome = rand() % 100;
   return outcome < loss_percent;
 }
 
-class NetworkPacket {
- public:
-  NetworkPacket(const uint8_t* data, size_t length, int64_t send_time,
-      int64_t arrival_time)
-      : data_(NULL),
-        data_length_(length),
-        send_time_(send_time),
-        arrival_time_(arrival_time) {
-    data_ = new uint8_t[length];
-    memcpy(data_, data, length);
-  }
-  ~NetworkPacket() {
-    delete [] data_;
-  }
-
-  uint8_t* data() const { return data_; }
-  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;
-  }
-
- private:
-  // The packet data.
-  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 reciver.
-  int64_t arrival_time_;
-};
-
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
                                  const FakeNetworkPipe::Config& config)
     : clock_(clock),
       packet_receiver_(NULL),
       config_(config),
       dropped_packets_(0),
       sent_packets_(0),
       total_packet_delay_(0),
       next_process_time_(clock_->TimeInMilliseconds()) {}
 
 FakeNetworkPipe::~FakeNetworkPipe() {
   while (!capacity_link_.empty()) {
     delete capacity_link_.front();
     capacity_link_.pop();
   }
   while (!delay_link_.empty()) {
-    delete delay_link_.front();
-    delay_link_.pop();
+    delete *delay_link_.begin();
+    delay_link_.erase(delay_link_.begin());
   }
 }
 
 void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) {
   packet_receiver_ = receiver;
 }
 
 void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) {
   rtc::CritScope crit(&lock_);
   config_ = config;  // Shallow copy of the struct.
@@ skipping 60 matching lines @@
       NetworkPacket* packet = capacity_link_.front();
       capacity_link_.pop();
 
       // Packets are randomly dropped after being affected by the bottleneck.
       if (UniformLoss(config_.loss_percent)) {
         delete packet;
         continue;
       }
 
       // Add extra delay and jitter, but make sure the arrival time is not
       // earlier than the last packet in the queue.

[stefan-webrtc, 2016/01/20 10:57:33]

Half of this comment should be moved down to line 144 or something.

[philipel, 2016/01/20 14:22:57]

Done.

       int extra_delay = GaussianRandom(config_.queue_delay_ms,

[stefan-webrtc, 2016/01/20 10:57:33]

I think it would be good to rename this random_jitter_ms or maybe
extra_jitter_ms. What do you think?

[philipel, 2016/01/20 14:22:57]

Renamed to arrival_time_jitter.

                                        config_.delay_standard_deviation_ms);
-      if (delay_link_.size() > 0 &&
+      if (!config_.allow_reordering &&
+          delay_link_.size() > 0 &&

[stefan-webrtc, 2016/01/20 10:57:33]

!delay_link_.empty()

[philipel, 2016/01/20 14:22:56]

Done.

           packet->arrival_time() + extra_delay <
-          delay_link_.back()->arrival_time()) {
-        extra_delay = delay_link_.back()->arrival_time() -
+          (*delay_link_.rbegin())->arrival_time()) {
+        extra_delay = (*delay_link_.rbegin())->arrival_time() -
             packet->arrival_time();
       }
       packet->IncrementArrivalTime(extra_delay);
       if (packet->arrival_time() < next_process_time_)
         next_process_time_ = packet->arrival_time();
-      delay_link_.push(packet);
+      delay_link_.insert(packet);
     }
 
     // Check the extra delay queue.
     while (delay_link_.size() > 0 &&
-           time_now >= delay_link_.front()->arrival_time()) {
+           time_now >= (*delay_link_.begin())->arrival_time()) {
       // Deliver this packet.
-      NetworkPacket* packet = delay_link_.front();
+      NetworkPacket* packet = *delay_link_.begin();
       packets_to_deliver.push(packet);
-      delay_link_.pop();
+      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();
     packet_receiver_->DeliverPacket(MediaType::ANY, packet->data(),
                                     packet->data_length(), PacketTime());
     delete packet;
   }
 }
 
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
   rtc::CritScope crit(&lock_);
   const int64_t kDefaultProcessIntervalMs = 30;
   if (capacity_link_.size() == 0 || delay_link_.size() == 0)
     return kDefaultProcessIntervalMs;
   return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
                            0);
 }
 
 }  // namespace webrtc