Only use 95% of the link capacity if the true link capacity is found by probing.

webrtc/modules/congestion_controller/delay_based_bwe_unittest.cc

 /*
  *  Copyright (c) 2016 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.
  */
 
 #include "webrtc/test/gtest.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/congestion_controller/delay_based_bwe.h"
 #include "webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 constexpr int kNumProbesCluster0 = 5;
 constexpr int kNumProbesCluster1 = 8;
 const PacedPacketInfo kPacingInfo0(0, kNumProbesCluster0, 2000);
 const PacedPacketInfo kPacingInfo1(1, kNumProbesCluster1, 4000);
+constexpr float kTargetUtilizationFraction = 0.95f;
 }  // namespace
 
 TEST_F(DelayBasedBweTest, NoCrashEmptyFeedback) {
   std::vector<PacketFeedback> packet_feedback_vector;
   bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector,
                                                    rtc::Optional<uint32_t>());
 }
 
 TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
   std::vector<PacketFeedback> packet_feedback_vector;
@@ skipping 60 matching lines @@
   }
 
   EXPECT_FALSE(bitrate_observer_.updated());
 }
 
 TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrival) {
   int64_t now_ms = clock_.TimeInMilliseconds();
   uint16_t seq_num = 0;
   // First burst sent at 8 * 1000 / 5 = 1600 kbps.
   // Arriving at 8 * 1000 / 7 = 1142 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
   int64_t send_time_ms = 0;
   for (int i = 0; i < kNumProbesCluster1; ++i) {
     clock_.AdvanceTimeMilliseconds(7);
     send_time_ms += 5;
     now_ms = clock_.TimeInMilliseconds();
     IncomingFeedback(now_ms, send_time_ms, seq_num++, 1000, kPacingInfo1);
   }
 
   EXPECT_TRUE(bitrate_observer_.updated());
-  EXPECT_NEAR(bitrate_observer_.latest_bitrate(), 1140000u, 10000u);
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 1140000u, 10000u);
 }
 
 TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrivalHighBitrate) {
   int64_t now_ms = clock_.TimeInMilliseconds();
   uint16_t seq_num = 0;
   // Burst sent at 8 * 1000 / 1 = 8000 kbps.
   // Arriving at 8 * 1000 / 2 = 4000 kbps.
+  // Since the receive rate is significantly below the send rate, we expect to
+  // use 95% of the estimated capacity.
   int64_t send_time_ms = 0;
   for (int i = 0; i < kNumProbesCluster1; ++i) {
     clock_.AdvanceTimeMilliseconds(2);
     send_time_ms += 1;
     now_ms = clock_.TimeInMilliseconds();
     IncomingFeedback(now_ms, send_time_ms, seq_num++, 1000, kPacingInfo1);
   }
 
   EXPECT_TRUE(bitrate_observer_.updated());
-  EXPECT_NEAR(bitrate_observer_.latest_bitrate(), 4000000u, 10000u);
+  EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
+              kTargetUtilizationFraction * 4000000u, 10000u);
 }
 
 TEST_F(DelayBasedBweTest, GetExpectedBwePeriodMs) {
   int64_t default_interval_ms = bitrate_estimator_->GetExpectedBwePeriodMs();
   EXPECT_GT(default_interval_ms, 0);
   CapacityDropTestHelper(1, true, 333, 0);
   int64_t interval_ms = bitrate_estimator_->GetExpectedBwePeriodMs();
   EXPECT_GT(interval_ms, 0);
   EXPECT_NE(interval_ms, default_interval_ms);
 }
@@ skipping 38 matching lines @@
 
 TEST_F(DelayBasedBweTest, TestLongTimeoutAndWrap) {
   // Simulate a client leaving and rejoining the call after some multiple of
   // 64 seconds later. This will cause a zero difference in abs send times due
   // to the wrap, but a big difference in arrival time, if streams aren't
   // properly timed out.
   TestWrappingHelper(10 * 64);
 }
 
 }  // namespace webrtc