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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/test/scenario/probing_test.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/scenario/probing_test.cc')
-rw-r--r-- | third_party/libwebrtc/test/scenario/probing_test.cc | 135 |
1 files changed, 135 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/scenario/probing_test.cc b/third_party/libwebrtc/test/scenario/probing_test.cc new file mode 100644 index 0000000000..86653ced9b --- /dev/null +++ b/third_party/libwebrtc/test/scenario/probing_test.cc @@ -0,0 +1,135 @@ +/* + * 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 |