|  | /* | 
|  | *  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 |