Increase the send-time history to 60 seconds.

This helps us avoid time-outs on really bad networks with long queues.
Also adding periodic logging of the fake network pipe's queue in milliseconds.

BUG=webrtc:5079

Review-Url: https://codereview.webrtc.org/2687013005
Cr-Commit-Position: refs/heads/master@{#16532}

webrtc/modules/congestion_controller/transport_feedback_adapter.cc

@@ -24,7 +24,7 @@
 namespace webrtc {
 
 const int64_t kNoTimestamp = -1;
-const int64_t kSendTimeHistoryWindowMs = 10000;
+const int64_t kSendTimeHistoryWindowMs = 60000;
 const int64_t kBaseTimestampScaleFactor =
     rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
 const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);

webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc

@@ -13,6 +13,7 @@
 #include <vector>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/transport_feedback_adapter.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@@ -128,6 +129,72 @@ TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
   ComparePacketVectors(packets, adapter_->GetTransportFeedbackVector());
 }
 
+TEST_F(TransportFeedbackAdapterTest, Timeout) {
+  const int64_t kFeedbackTimeoutMs = 60001;
+  {
+    std::vector<PacketInfo> packets;
+    packets.push_back(PacketInfo(100, 200, 0, 1500, 0));
+    packets.push_back(PacketInfo(110, 210, 1, 1500, 0));
+    packets.push_back(PacketInfo(120, 220, 2, 1500, 0));
+    packets.push_back(PacketInfo(130, 230, 3, 1500, 1));
+    packets.push_back(PacketInfo(140, 240, 4, 1500, 1));
+
+    for (const PacketInfo& packet : packets)
+      OnSentPacket(packet);
+
+    rtcp::TransportFeedback feedback;
+    feedback.SetBase(packets[0].sequence_number,
+                     packets[0].arrival_time_ms * 1000);
+
+    for (const PacketInfo& packet : packets) {
+      EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
+                                             packet.arrival_time_ms * 1000));
+    }
+
+    feedback.Build();
+
+    clock_.AdvanceTimeMilliseconds(kFeedbackTimeoutMs);
+    PacketInfo later_packet(kFeedbackTimeoutMs + 140, kFeedbackTimeoutMs + 240,
+                            5, 1500, 1);
+    OnSentPacket(later_packet);
+
+    adapter_->OnTransportFeedback(feedback);
+    EXPECT_EQ(0, rtc::checked_cast<int>(
+                     adapter_->GetTransportFeedbackVector().size()));
+  }
+
+  {
+    std::vector<PacketInfo> packets;
+    packets.push_back(PacketInfo(100, 200, 0, 1500, 0));
+    packets.push_back(PacketInfo(110, 210, 1, 1500, 0));
+    packets.push_back(PacketInfo(120, 220, 2, 1500, 0));
+    packets.push_back(PacketInfo(130, 230, 3, 1500, 1));
+    packets.push_back(PacketInfo(140, 240, 4, 1500, 1));
+
+    for (const PacketInfo& packet : packets)
+      OnSentPacket(packet);
+
+    rtcp::TransportFeedback feedback;
+    feedback.SetBase(packets[0].sequence_number,
+                     packets[0].arrival_time_ms * 1000);
+
+    for (const PacketInfo& packet : packets) {
+      EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
+                                             packet.arrival_time_ms * 1000));
+    }
+
+    feedback.Build();
+
+    clock_.AdvanceTimeMilliseconds(kFeedbackTimeoutMs - 1);
+    PacketInfo later_packet(kFeedbackTimeoutMs + 140, kFeedbackTimeoutMs + 240,
+                            5, 1500, 1);
+    OnSentPacket(later_packet);
+
+    adapter_->OnTransportFeedback(feedback);
+    ComparePacketVectors(packets, adapter_->GetTransportFeedbackVector());
+  }
+}
+
 TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
   std::vector<PacketInfo> packets;
   packets.push_back(PacketInfo(100, 200, 0, 1500, 1));

webrtc/test/fake_network_pipe.cc

@@ -17,6 +17,7 @@
 #include <algorithm>
 #include <cmath>
 
+#include "webrtc/base/logging.h"
 #include "webrtc/call/call.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
@@ -37,7 +38,8 @@ FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
       sent_packets_(0),
       total_packet_delay_(0),
       bursting_(false),
-      next_process_time_(clock_->TimeInMilliseconds()) {
+      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
@@ -134,6 +136,14 @@ void FakeNetworkPipe::Process() {
   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()) {

webrtc/test/fake_network_pipe.h

@@ -146,6 +146,8 @@ class FakeNetworkPipe {
 
   int64_t next_process_time_;
 
+  int64_t last_log_time_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
 };