More Simulation Framework features

webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc

 /*
  *  Copyright (c) 2015 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 26 matching lines @@
       ++it;
     }
   }
   return fb_packets;
 }
 
 VideoSender::VideoSender(PacketProcessorListener* listener,
                          VideoSource* source,
                          BandwidthEstimatorType estimator_type)
     : PacketSender(listener, source->flow_id()),
+      running_(true),
       source_(source),
       bwe_(CreateBweSender(estimator_type,
                            source_->bits_per_second() / 1000,
                            this,
                            &clock_)) {
   modules_.push_back(bwe_.get());
 }
 
 VideoSender::~VideoSender() {
 }
@@ skipping 12 matching lines @@
     // Make sure to at least run Process() below every 100 ms.
     int64_t time_to_run_ms = std::min<int64_t>(time_ms, 100);
     if (!feedbacks->empty()) {
       int64_t time_until_feedback_ms =
           feedbacks->front()->send_time_us() / 1000 -
           clock_.TimeInMilliseconds();
       time_to_run_ms =
           std::max<int64_t>(std::min(time_ms, time_until_feedback_ms), 0);
     }
     Packets generated;
+
     source_->RunFor(time_to_run_ms, &generated);
     bwe_->OnPacketsSent(generated);
+
     packets->merge(generated, DereferencingComparator<Packet>);
+
     clock_.AdvanceTimeMilliseconds(time_to_run_ms);
+
     if (!feedbacks->empty()) {
       bwe_->GiveFeedback(*feedbacks->front());
       delete feedbacks->front();
       feedbacks->pop_front();
     }
+
     bwe_->Process();
+
     time_ms -= time_to_run_ms;
   } while (time_ms > 0);
   assert(feedbacks->empty());
 }
 
 int VideoSender::GetFeedbackIntervalMs() const {
   return bwe_->GetFeedbackIntervalMs();
 }
 
 void VideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
                                    uint8_t fraction_lost,
                                    int64_t rtt) {
   source_->SetBitrateBps(target_bitrate_bps);
 }
 
+void VideoSender::Pause() {
+  running_ = false;
+  source_->Pause();
+  bwe_->Pause();
+}
+
+void VideoSender::Resume() {
+  running_ = true;
+  source_->Resume();
+  bwe_->Resume();
+}
+
+uint32_t VideoSender::TargetBitrateKbps() {
+  return (source_->bits_per_second() + 500) / 1000;
+}
+
 PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener,
                                    VideoSource* source,
                                    BandwidthEstimatorType estimator)
     : VideoSender(listener, source, estimator),
       pacer_(&clock_,
              this,
              source->bits_per_second() / 1000,
              PacedSender::kDefaultPaceMultiplier * source->bits_per_second() /
                  1000,
              0) {
@@ skipping 141 matching lines @@
 
 void PacedVideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
                                         uint8_t fraction_lost,
                                         int64_t rtt) {
   VideoSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
   pacer_.UpdateBitrate(
       target_bitrate_bps / 1000,
       PacedSender::kDefaultPaceMultiplier * target_bitrate_bps / 1000, 0);
 }
 
+const int kNoLimit = std::numeric_limits<int>::max();
+const int kPacketSizeBytes = 1200;
+
+TcpSender::TcpSender(PacketProcessorListener* listener,
+                     int flow_id,
+                     int64_t offset_ms)
+    : PacketSender(listener, flow_id),
+      cwnd_(10),
+      ssthresh_(kNoLimit),
+      ack_received_(false),
+      last_acked_seq_num_(0),
+      next_sequence_number_(0),
+      offset_ms_(offset_ms),
+      last_reduction_time_ms_(-1),
+      last_rtt_ms_(0),
+      total_sent_bytes_(0),
+      send_limit_bytes_(kNoLimit),
+      running_(true),
+      last_generated_packets_ms_(0),
+      num_recent_sent_packets_(0),
+      bitrate_kbps_(0) {
+}
+
+TcpSender::TcpSender(PacketProcessorListener* listener,
+                     int flow_id,
+                     int64_t offset_ms,
+                     int send_limit_bytes)
+    : TcpSender(listener, flow_id, offset_ms) {
+  send_limit_bytes_ = send_limit_bytes;

[stefan-webrtc, 2015/07/06 08:24:51]

Set in initializer list.

[magalhaesc, 2015/07/06 09:28:07]

Done.

+}
+
 void TcpSender::RunFor(int64_t time_ms, Packets* in_out) {
   if (clock_.TimeInMilliseconds() + time_ms < offset_ms_) {
     clock_.AdvanceTimeMilliseconds(time_ms);
+    if (running_) {
+      running_ = false;
+    }
     return;
   }
+
+  if (!running_) {
+    running_ = true;
+  }
+
   int64_t start_time_ms = clock_.TimeInMilliseconds();
   BWE_TEST_LOGGING_CONTEXT("Sender");
   BWE_TEST_LOGGING_CONTEXT(*flow_ids().begin());
 
   std::list<FeedbackPacket*> feedbacks = GetFeedbackPackets(
       in_out, clock_.TimeInMilliseconds() + time_ms, *flow_ids().begin());
   // The number of packets which are sent in during time_ms depends on the
   // number of packets in_flight_ and the max number of packets in flight
   // (cwnd_). Therefore SendPackets() isn't directly dependent on time_ms.
   for (FeedbackPacket* fb : feedbacks) {
@@ skipping 72 matching lines @@
 void TcpSender::HandleLoss() {
   if (clock_.TimeInMilliseconds() - last_reduction_time_ms_ < last_rtt_ms_)
     return;
   last_reduction_time_ms_ = clock_.TimeInMilliseconds();
   ssthresh_ = std::max(static_cast<int>(in_flight_.size() / 2), 2);
   cwnd_ = ssthresh_;
 }
 
 Packets TcpSender::GeneratePackets(size_t num_packets) {
   Packets generated;
+
+  UpdateSendBitrateEstimate(num_packets);
+
   for (size_t i = 0; i < num_packets; ++i) {
-    generated.push_back(new MediaPacket(*flow_ids().begin(),
-                                        1000 * clock_.TimeInMilliseconds(),
-                                        1200, next_sequence_number_++));
+    if ((total_sent_bytes_ + kPacketSizeBytes) > send_limit_bytes_) {
+      if (running_) {
+        running_ = false;
+      }
+      break;
+    }
+    generated.push_back(
+        new MediaPacket(*flow_ids().begin(), 1000 * clock_.TimeInMilliseconds(),
+                        kPacketSizeBytes, next_sequence_number_++));
     generated.back()->set_sender_timestamp_us(
         1000 * clock_.TimeInMilliseconds());
+
+    total_sent_bytes_ += kPacketSizeBytes;
   }
+
   return generated;
 }
+
+void TcpSender::UpdateSendBitrateEstimate(size_t num_packets) {
+  const int kTimeWindowMs = 500;
+  num_recent_sent_packets_ += num_packets;
+
+  int64_t delta_ms = clock_.TimeInMilliseconds() - last_generated_packets_ms_;
+  if (delta_ms >= kTimeWindowMs) {
+    bitrate_kbps_ =
+        static_cast<uint32_t>(8 * num_recent_sent_packets_ * kPacketSizeBytes) /
+        delta_ms;
+    last_generated_packets_ms_ = clock_.TimeInMilliseconds();
+    num_recent_sent_packets_ = 0;
+  }
+}
+
+uint32_t TcpSender::TargetBitrateKbps() {
+  return bitrate_kbps_;
+}
+
 }  // namespace bwe
 }  // namespace testing
 }  // namespace webrtc