1 /*
2 * Copyright 2018 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10 #include "test/scenario/scenario.h"
11
12 #include <atomic>
13
14 #include "api/test/network_emulation/create_cross_traffic.h"
15 #include "api/test/network_emulation/cross_traffic.h"
16 #include "test/field_trial.h"
17 #include "test/gtest.h"
18 #include "test/logging/memory_log_writer.h"
19 #include "test/scenario/stats_collection.h"
20
21 namespace webrtc {
22 namespace test {
TEST(ScenarioTest,StartsAndStopsWithoutErrors)23 TEST(ScenarioTest, StartsAndStopsWithoutErrors) {
24 std::atomic<bool> packet_received(false);
25 std::atomic<bool> bitrate_changed(false);
26 Scenario s;
27 CallClientConfig call_client_config;
28 call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
29 auto* alice = s.CreateClient("alice", call_client_config);
30 auto* bob = s.CreateClient("bob", call_client_config);
31 NetworkSimulationConfig network_config;
32 auto alice_net = s.CreateSimulationNode(network_config);
33 auto bob_net = s.CreateSimulationNode(network_config);
34 auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
35
36 VideoStreamConfig video_stream_config;
37 s.CreateVideoStream(route->forward(), video_stream_config);
38 s.CreateVideoStream(route->reverse(), video_stream_config);
39
40 AudioStreamConfig audio_stream_config;
41 audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6);
42 audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64);
43 audio_stream_config.encoder.allocate_bitrate = true;
44 audio_stream_config.stream.in_bandwidth_estimation = false;
45 s.CreateAudioStream(route->forward(), audio_stream_config);
46 s.CreateAudioStream(route->reverse(), audio_stream_config);
47
48 RandomWalkConfig cross_traffic_config;
49 s.net()->StartCrossTraffic(CreateRandomWalkCrossTraffic(
50 s.net()->CreateCrossTrafficRoute({alice_net}), cross_traffic_config));
51
52 s.NetworkDelayedAction({alice_net, bob_net}, 100,
53 [&packet_received] { packet_received = true; });
54 s.Every(TimeDelta::Millis(10), [alice, bob, &bitrate_changed] {
55 if (alice->GetStats().send_bandwidth_bps != 300000 &&
56 bob->GetStats().send_bandwidth_bps != 300000)
57 bitrate_changed = true;
58 });
59 s.RunUntil(TimeDelta::Seconds(2), TimeDelta::Millis(5),
60 [&bitrate_changed, &packet_received] {
61 return packet_received && bitrate_changed;
62 });
63 EXPECT_TRUE(packet_received);
64 EXPECT_TRUE(bitrate_changed);
65 }
66 namespace {
SetupVideoCall(Scenario & s,VideoQualityAnalyzer * analyzer)67 void SetupVideoCall(Scenario& s, VideoQualityAnalyzer* analyzer) {
68 CallClientConfig call_config;
69 auto* alice = s.CreateClient("alice", call_config);
70 auto* bob = s.CreateClient("bob", call_config);
71 NetworkSimulationConfig network_config;
72 network_config.bandwidth = DataRate::KilobitsPerSec(1000);
73 network_config.delay = TimeDelta::Millis(50);
74 auto alice_net = s.CreateSimulationNode(network_config);
75 auto bob_net = s.CreateSimulationNode(network_config);
76 auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
77 VideoStreamConfig video;
78 if (analyzer) {
79 video.source.capture = VideoStreamConfig::Source::Capture::kVideoFile;
80 video.source.video_file.name = "foreman_cif";
81 video.source.video_file.width = 352;
82 video.source.video_file.height = 288;
83 video.source.framerate = 30;
84 video.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
85 video.encoder.implementation =
86 VideoStreamConfig::Encoder::Implementation::kSoftware;
87 video.hooks.frame_pair_handlers = {analyzer->Handler()};
88 }
89 s.CreateVideoStream(route->forward(), video);
90 s.CreateAudioStream(route->forward(), AudioStreamConfig());
91 }
92 } // namespace
93
TEST(ScenarioTest,SimTimeEncoding)94 TEST(ScenarioTest, SimTimeEncoding) {
95 VideoQualityAnalyzerConfig analyzer_config;
96 analyzer_config.psnr_coverage = 0.1;
97 VideoQualityAnalyzer analyzer(analyzer_config);
98 {
99 Scenario s("scenario/encode_sim", false);
100 SetupVideoCall(s, &analyzer);
101 s.RunFor(TimeDelta::Seconds(2));
102 }
103 // Regression tests based on previous runs.
104 EXPECT_EQ(analyzer.stats().lost_count, 0);
105 EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 38, 5);
106 }
107
108 // TODO(bugs.webrtc.org/10515): Remove this when performance has been improved.
109 #if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
110 #define MAYBE_RealTimeEncoding DISABLED_RealTimeEncoding
111 #else
112 #define MAYBE_RealTimeEncoding RealTimeEncoding
113 #endif
TEST(ScenarioTest,MAYBE_RealTimeEncoding)114 TEST(ScenarioTest, MAYBE_RealTimeEncoding) {
115 VideoQualityAnalyzerConfig analyzer_config;
116 analyzer_config.psnr_coverage = 0.1;
117 VideoQualityAnalyzer analyzer(analyzer_config);
118 {
119 Scenario s("scenario/encode_real", true);
120 SetupVideoCall(s, &analyzer);
121 s.RunFor(TimeDelta::Seconds(2));
122 }
123 // Regression tests based on previous runs.
124 EXPECT_LT(analyzer.stats().lost_count, 2);
125 // This far below expected but ensures that we get something.
126 EXPECT_GT(analyzer.stats().psnr_with_freeze.Mean(), 10);
127 }
128
TEST(ScenarioTest,SimTimeFakeing)129 TEST(ScenarioTest, SimTimeFakeing) {
130 Scenario s("scenario/encode_sim", false);
131 SetupVideoCall(s, nullptr);
132 s.RunFor(TimeDelta::Seconds(2));
133 }
134
TEST(ScenarioTest,WritesToRtcEventLog)135 TEST(ScenarioTest, WritesToRtcEventLog) {
136 MemoryLogStorage storage;
137 {
138 Scenario s(storage.CreateFactory(), false);
139 SetupVideoCall(s, nullptr);
140 s.RunFor(TimeDelta::Seconds(1));
141 }
142 auto logs = storage.logs();
143 // We expect that a rtc event log has been created and that it has some data.
144 EXPECT_GE(storage.logs().at("alice.rtc.dat").size(), 1u);
145 }
146
TEST(ScenarioTest,RetransmitsVideoPacketsInAudioAndVideoCallWithSendSideBweAndLoss)147 TEST(ScenarioTest,
148 RetransmitsVideoPacketsInAudioAndVideoCallWithSendSideBweAndLoss) {
149 // Make sure audio packets are included in transport feedback.
150 test::ScopedFieldTrials override_field_trials(
151 "WebRTC-Audio-ABWENoTWCC/Disabled/");
152
153 Scenario s;
154 CallClientConfig call_client_config;
155 call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
156 auto* alice = s.CreateClient("alice", call_client_config);
157 auto* bob = s.CreateClient("bob", call_client_config);
158 NetworkSimulationConfig network_config;
159 // Add some loss and delay.
160 network_config.delay = TimeDelta::Millis(200);
161 network_config.loss_rate = 0.05;
162 auto alice_net = s.CreateSimulationNode(network_config);
163 auto bob_net = s.CreateSimulationNode(network_config);
164 auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
165
166 // First add an audio stream, then a video stream.
167 // Needed to make sure audio RTP module is selected first when sending
168 // transport feedback message.
169 AudioStreamConfig audio_stream_config;
170 audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6);
171 audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64);
172 audio_stream_config.encoder.allocate_bitrate = true;
173 audio_stream_config.stream.in_bandwidth_estimation = true;
174 s.CreateAudioStream(route->forward(), audio_stream_config);
175 s.CreateAudioStream(route->reverse(), audio_stream_config);
176
177 VideoStreamConfig video_stream_config;
178 auto video = s.CreateVideoStream(route->forward(), video_stream_config);
179 s.CreateVideoStream(route->reverse(), video_stream_config);
180
181 // Run for 10 seconds.
182 s.RunFor(TimeDelta::Seconds(10));
183 // Make sure retransmissions have happened.
184 int retransmit_packets = 0;
185
186 VideoSendStream::Stats stats;
187 alice->SendTask([&]() { stats = video->send()->GetStats(); });
188
189 for (const auto& substream : stats.substreams) {
190 retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
191 }
192 EXPECT_GT(retransmit_packets, 0);
193 }
194
195 } // namespace test
196 } // namespace webrtc
197