Fixes a bug where a video stream can get stuck in the suspended state.

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 15 matching lines @@
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
                                  const FakeNetworkPipe::Config& config)
     : FakeNetworkPipe(clock, config, 1) {}
 
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
                                  const FakeNetworkPipe::Config& config,
                                  uint64_t seed)
     : clock_(clock),
       packet_receiver_(NULL),
       random_(seed),
-      config_(config),
+      config_(),
       dropped_packets_(0),
       sent_packets_(0),
       total_packet_delay_(0),
       bursting_(false),
       next_process_time_(clock_->TimeInMilliseconds()),
       last_log_time_(clock_->TimeInMilliseconds()) {
-  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;
-  }
+  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) {
   packet_receiver_ = 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) {
   // A NULL packet_receiver_ means that this pipe will terminate the flow of
   // packets.
   if (packet_receiver_ == NULL)
     return;
   rtc::CritScope crit(&lock_);
   if (config_.queue_length_packets > 0 &&
       capacity_link_.size() >= config_.queue_length_packets) {
@@ skipping 114 matching lines @@
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
   rtc::CritScope crit(&lock_);
   const int64_t kDefaultProcessIntervalMs = 5;
   if (capacity_link_.empty() || delay_link_.empty())
     return kDefaultProcessIntervalMs;
   return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
                            0);
 }
 
 }  // namespace webrtc