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diff --git a/third_party/libwebrtc/test/scenario/probing_test.cc b/third_party/libwebrtc/test/scenario/probing_test.cc
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+++ b/third_party/libwebrtc/test/scenario/probing_test.cc
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+/*
+ *  Copyright 2020 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 "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+namespace webrtc {
+namespace test {
+
+TEST(ProbingTest, InitialProbingRampsUpTargetRateWhenNetworkIsGood) {
+  Scenario s;
+  NetworkSimulationConfig good_network;
+  good_network.bandwidth = DataRate::KilobitsPerSec(2000);
+
+  VideoStreamConfig video_config;
+  video_config.encoder.codec =
+      VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+  CallClientConfig send_config;
+  auto* caller = s.CreateClient("caller", send_config);
+  auto* callee = s.CreateClient("callee", CallClientConfig());
+  auto route =
+      s.CreateRoutes(caller, {s.CreateSimulationNode(good_network)}, callee,
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), video_config);
+
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_GE(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            3 * send_config.transport.rates.start_rate);
+}
+
+TEST(ProbingTest, MidCallProbingRampupTriggeredByUpdatedBitrateConstraints) {
+  Scenario s;
+
+  const DataRate kStartRate = DataRate::KilobitsPerSec(300);
+  const DataRate kConstrainedRate = DataRate::KilobitsPerSec(100);
+  const DataRate kHighRate = DataRate::KilobitsPerSec(1500);
+
+  VideoStreamConfig video_config;
+  video_config.encoder.codec =
+      VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+  CallClientConfig send_call_config;
+  send_call_config.transport.rates.start_rate = kStartRate;
+  send_call_config.transport.rates.max_rate = kHighRate * 2;
+  auto* caller = s.CreateClient("caller", send_call_config);
+  auto* callee = s.CreateClient("callee", CallClientConfig());
+  auto route = s.CreateRoutes(
+      caller, {s.CreateSimulationNode(NetworkSimulationConfig())}, callee,
+      {s.CreateSimulationNode(NetworkSimulationConfig())});
+  s.CreateVideoStream(route->forward(), video_config);
+
+  // Wait until initial probing rampup is done and then set a low max bitrate.
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_GE(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            5 * send_call_config.transport.rates.start_rate);
+  BitrateConstraints bitrate_config;
+  bitrate_config.max_bitrate_bps = kConstrainedRate.bps();
+  caller->UpdateBitrateConstraints(bitrate_config);
+
+  // Wait until the low send bitrate has taken effect, and then set a much
+  // higher max bitrate.
+  s.RunFor(TimeDelta::Seconds(2));
+  EXPECT_LT(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            kConstrainedRate * 1.1);
+  bitrate_config.max_bitrate_bps = 2 * kHighRate.bps();
+  caller->UpdateBitrateConstraints(bitrate_config);
+
+  // Check that the max send bitrate is reached quicker than would be possible
+  // with simple AIMD rate control.
+  s.RunFor(TimeDelta::Seconds(1));
+  EXPECT_GE(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            kHighRate);
+}
+
+TEST(ProbingTest, ProbesRampsUpWhenVideoEncoderConfigChanges) {
+  Scenario s;
+  const DataRate kStartRate = DataRate::KilobitsPerSec(50);
+  const DataRate kHdRate = DataRate::KilobitsPerSec(3250);
+
+  // Set up 3-layer simulcast.
+  VideoStreamConfig video_config;
+  video_config.encoder.codec =
+      VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+  video_config.encoder.simulcast_streams = {webrtc::ScalabilityMode::kL1T3,
+                                            webrtc::ScalabilityMode::kL1T3,
+                                            webrtc::ScalabilityMode::kL1T3};
+  video_config.source.generator.width = 1280;
+  video_config.source.generator.height = 720;
+
+  CallClientConfig send_call_config;
+  send_call_config.transport.rates.start_rate = kStartRate;
+  send_call_config.transport.rates.max_rate = kHdRate * 2;
+  auto* caller = s.CreateClient("caller", send_call_config);
+  auto* callee = s.CreateClient("callee", CallClientConfig());
+  auto send_net =
+      s.CreateMutableSimulationNode([&](NetworkSimulationConfig* c) {
+        c->bandwidth = DataRate::KilobitsPerSec(200);
+      });
+  auto route =
+      s.CreateRoutes(caller, {send_net->node()}, callee,
+                     {s.CreateSimulationNode(NetworkSimulationConfig())});
+  auto* video_stream = s.CreateVideoStream(route->forward(), video_config);
+
+  // Only QVGA enabled initially. Run until initial probing is done and BWE
+  // has settled.
+  video_stream->send()->UpdateActiveLayers({true, false, false});
+  s.RunFor(TimeDelta::Seconds(2));
+
+  // Remove network constraints and run for a while more, BWE should be much
+  // less than required HD rate.
+  send_net->UpdateConfig([&](NetworkSimulationConfig* c) {
+    c->bandwidth = DataRate::PlusInfinity();
+  });
+  s.RunFor(TimeDelta::Seconds(2));
+
+  DataRate bandwidth =
+      DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps);
+  EXPECT_LT(bandwidth, kHdRate / 4);
+
+  // Enable all layers, triggering a probe.
+  video_stream->send()->UpdateActiveLayers({true, true, true});
+
+  // Run for a short while and verify BWE has ramped up fast.
+  s.RunFor(TimeDelta::Seconds(2));
+  EXPECT_GT(DataRate::BitsPerSec(caller->GetStats().send_bandwidth_bps),
+            kHdRate);
+}
+
+}  // namespace test
+}  // namespace webrtc