| /* |
| * 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(2500); |
| |
| 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 |