| /* |
| * 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 <atomic> |
| |
| #include "test/gtest.h" |
| #include "test/logging/memory_log_writer.h" |
| #include "test/scenario/scenario.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::kbps(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::kbps(6); |
| audio_stream_config.encoder.max_rate = DataRate::kbps(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()->CreateRandomWalkCrossTraffic( |
| s.net()->CreateTrafficRoute({alice_net}), cross_traffic_config); |
| |
| s.NetworkDelayedAction({alice_net, bob_net}, 100, |
| [&packet_received] { packet_received = true; }); |
| s.Every(TimeDelta::ms(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::ms(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::kbps(1000); |
| network_config.delay = TimeDelta::ms(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 |
| |
| // TODO(bugs.webrtc.org/10515): Remove this when performance has been improved. |
| #if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG) |
| #define MAYBE_SimTimeEncoding DISABLED_SimTimeEncoding |
| #else |
| #define MAYBE_SimTimeEncoding SimTimeEncoding |
| #endif |
| TEST(ScenarioTest, MAYBE_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(60)); |
| } |
| // Regression tests based on previous runs. |
| EXPECT_EQ(analyzer.stats().lost_count, 0); |
| EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 38, 2); |
| } |
| |
| // 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(10)); |
| } |
| // Regression tests based on previous runs. |
| EXPECT_LT(analyzer.stats().lost_count, 2); |
| EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 38, 10); |
| } |
| |
| TEST(ScenarioTest, SimTimeFakeing) { |
| Scenario s("scenario/encode_sim", false); |
| SetupVideoCall(s, nullptr); |
| s.RunFor(TimeDelta::seconds(10)); |
| } |
| |
| 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); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |