diff options
Diffstat (limited to 'third_party/libwebrtc/test/scenario/scenario_unittest.cc')
| -rw-r--r-- | third_party/libwebrtc/test/scenario/scenario_unittest.cc | 196 |
1 files changed, 196 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/scenario/scenario_unittest.cc b/third_party/libwebrtc/test/scenario/scenario_unittest.cc new file mode 100644 index 0000000000..6861151a2d --- /dev/null +++ b/third_party/libwebrtc/test/scenario/scenario_unittest.cc @@ -0,0 +1,196 @@ +/* + * Copyright 2018 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/scenario/scenario.h" + +#include <atomic> + +#include "api/test/network_emulation/create_cross_traffic.h" +#include "api/test/network_emulation/cross_traffic.h" +#include "test/field_trial.h" +#include "test/gtest.h" +#include "test/logging/memory_log_writer.h" +#include "test/scenario/stats_collection.h" + +namespace webrtc { +namespace test { +TEST(ScenarioTest, StartsAndStopsWithoutErrors) { + std::atomic<bool> packet_received(false); + std::atomic<bool> bitrate_changed(false); + Scenario s; + CallClientConfig call_client_config; + call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300); + auto* alice = s.CreateClient("alice", call_client_config); + auto* bob = s.CreateClient("bob", call_client_config); + NetworkSimulationConfig network_config; + auto alice_net = s.CreateSimulationNode(network_config); + auto bob_net = s.CreateSimulationNode(network_config); + auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net}); + + VideoStreamConfig video_stream_config; + s.CreateVideoStream(route->forward(), video_stream_config); + s.CreateVideoStream(route->reverse(), video_stream_config); + + AudioStreamConfig audio_stream_config; + audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6); + audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64); + audio_stream_config.encoder.allocate_bitrate = true; + audio_stream_config.stream.in_bandwidth_estimation = false; + s.CreateAudioStream(route->forward(), audio_stream_config); + s.CreateAudioStream(route->reverse(), audio_stream_config); + + RandomWalkConfig cross_traffic_config; + s.net()->StartCrossTraffic(CreateRandomWalkCrossTraffic( + s.net()->CreateCrossTrafficRoute({alice_net}), cross_traffic_config)); + + s.NetworkDelayedAction({alice_net, bob_net}, 100, + [&packet_received] { packet_received = true; }); + s.Every(TimeDelta::Millis(10), [alice, bob, &bitrate_changed] { + if (alice->GetStats().send_bandwidth_bps != 300000 && + bob->GetStats().send_bandwidth_bps != 300000) + bitrate_changed = true; + }); + s.RunUntil(TimeDelta::Seconds(2), TimeDelta::Millis(5), + [&bitrate_changed, &packet_received] { + return packet_received && bitrate_changed; + }); + EXPECT_TRUE(packet_received); + EXPECT_TRUE(bitrate_changed); +} +namespace { +void SetupVideoCall(Scenario& s, VideoQualityAnalyzer* analyzer) { + CallClientConfig call_config; + auto* alice = s.CreateClient("alice", call_config); + auto* bob = s.CreateClient("bob", call_config); + NetworkSimulationConfig network_config; + network_config.bandwidth = DataRate::KilobitsPerSec(1000); + network_config.delay = TimeDelta::Millis(50); + auto alice_net = s.CreateSimulationNode(network_config); + auto bob_net = s.CreateSimulationNode(network_config); + auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net}); + VideoStreamConfig video; + if (analyzer) { + video.source.capture = VideoStreamConfig::Source::Capture::kVideoFile; + video.source.video_file.name = "foreman_cif"; + video.source.video_file.width = 352; + video.source.video_file.height = 288; + video.source.framerate = 30; + video.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8; + video.encoder.implementation = + VideoStreamConfig::Encoder::Implementation::kSoftware; + video.hooks.frame_pair_handlers = {analyzer->Handler()}; + } + s.CreateVideoStream(route->forward(), video); + s.CreateAudioStream(route->forward(), AudioStreamConfig()); +} +} // namespace + +TEST(ScenarioTest, SimTimeEncoding) { + VideoQualityAnalyzerConfig analyzer_config; + analyzer_config.psnr_coverage = 0.1; + VideoQualityAnalyzer analyzer(analyzer_config); + { + Scenario s("scenario/encode_sim", false); + SetupVideoCall(s, &analyzer); + s.RunFor(TimeDelta::Seconds(2)); + } + // Regression tests based on previous runs. + EXPECT_EQ(analyzer.stats().lost_count, 0); + EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 38, 5); +} + +// TODO(bugs.webrtc.org/10515): Remove this when performance has been improved. +#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG) +#define MAYBE_RealTimeEncoding DISABLED_RealTimeEncoding +#else +#define MAYBE_RealTimeEncoding RealTimeEncoding +#endif +TEST(ScenarioTest, MAYBE_RealTimeEncoding) { + VideoQualityAnalyzerConfig analyzer_config; + analyzer_config.psnr_coverage = 0.1; + VideoQualityAnalyzer analyzer(analyzer_config); + { + Scenario s("scenario/encode_real", true); + SetupVideoCall(s, &analyzer); + s.RunFor(TimeDelta::Seconds(2)); + } + // Regression tests based on previous runs. + EXPECT_LT(analyzer.stats().lost_count, 2); + // This far below expected but ensures that we get something. + EXPECT_GT(analyzer.stats().psnr_with_freeze.Mean(), 10); +} + +TEST(ScenarioTest, SimTimeFakeing) { + Scenario s("scenario/encode_sim", false); + SetupVideoCall(s, nullptr); + s.RunFor(TimeDelta::Seconds(2)); +} + +TEST(ScenarioTest, WritesToRtcEventLog) { + MemoryLogStorage storage; + { + Scenario s(storage.CreateFactory(), false); + SetupVideoCall(s, nullptr); + s.RunFor(TimeDelta::Seconds(1)); + } + auto logs = storage.logs(); + // We expect that a rtc event log has been created and that it has some data. + EXPECT_GE(storage.logs().at("alice.rtc.dat").size(), 1u); +} + +TEST(ScenarioTest, + RetransmitsVideoPacketsInAudioAndVideoCallWithSendSideBweAndLoss) { + // Make sure audio packets are included in transport feedback. + test::ScopedFieldTrials override_field_trials( + "WebRTC-Audio-ABWENoTWCC/Disabled/"); + + Scenario s; + CallClientConfig call_client_config; + call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300); + auto* alice = s.CreateClient("alice", call_client_config); + auto* bob = s.CreateClient("bob", call_client_config); + NetworkSimulationConfig network_config; + // Add some loss and delay. + network_config.delay = TimeDelta::Millis(200); + network_config.loss_rate = 0.05; + auto alice_net = s.CreateSimulationNode(network_config); + auto bob_net = s.CreateSimulationNode(network_config); + auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net}); + + // First add an audio stream, then a video stream. + // Needed to make sure audio RTP module is selected first when sending + // transport feedback message. + AudioStreamConfig audio_stream_config; + audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6); + audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64); + audio_stream_config.encoder.allocate_bitrate = true; + audio_stream_config.stream.in_bandwidth_estimation = true; + s.CreateAudioStream(route->forward(), audio_stream_config); + s.CreateAudioStream(route->reverse(), audio_stream_config); + + VideoStreamConfig video_stream_config; + auto video = s.CreateVideoStream(route->forward(), video_stream_config); + s.CreateVideoStream(route->reverse(), video_stream_config); + + // Run for 10 seconds. + s.RunFor(TimeDelta::Seconds(10)); + // Make sure retransmissions have happened. + int retransmit_packets = 0; + + VideoSendStream::Stats stats; + alice->SendTask([&]() { stats = video->send()->GetStats(); }); + + for (const auto& substream : stats.substreams) { + retransmit_packets += substream.second.rtp_stats.retransmitted.packets; + } + EXPECT_GT(retransmit_packets, 0); +} + +} // namespace test +} // namespace webrtc |