191 lines
7.4 KiB
C++
191 lines
7.4 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "test/scenario/scenario.h"
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#include <atomic>
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#include "test/field_trial.h"
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#include "test/gtest.h"
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#include "test/logging/memory_log_writer.h"
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#include "test/scenario/stats_collection.h"
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namespace webrtc {
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namespace test {
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TEST(ScenarioTest, StartsAndStopsWithoutErrors) {
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std::atomic<bool> packet_received(false);
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std::atomic<bool> bitrate_changed(false);
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Scenario s;
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CallClientConfig call_client_config;
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call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
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auto* alice = s.CreateClient("alice", call_client_config);
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auto* bob = s.CreateClient("bob", call_client_config);
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NetworkSimulationConfig network_config;
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auto alice_net = s.CreateSimulationNode(network_config);
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auto bob_net = s.CreateSimulationNode(network_config);
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auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
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VideoStreamConfig video_stream_config;
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s.CreateVideoStream(route->forward(), video_stream_config);
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s.CreateVideoStream(route->reverse(), video_stream_config);
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AudioStreamConfig audio_stream_config;
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audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6);
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audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64);
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audio_stream_config.encoder.allocate_bitrate = true;
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audio_stream_config.stream.in_bandwidth_estimation = false;
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s.CreateAudioStream(route->forward(), audio_stream_config);
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s.CreateAudioStream(route->reverse(), audio_stream_config);
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RandomWalkConfig cross_traffic_config;
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s.net()->CreateRandomWalkCrossTraffic(
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s.net()->CreateTrafficRoute({alice_net}), cross_traffic_config);
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s.NetworkDelayedAction({alice_net, bob_net}, 100,
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[&packet_received] { packet_received = true; });
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s.Every(TimeDelta::Millis(10), [alice, bob, &bitrate_changed] {
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if (alice->GetStats().send_bandwidth_bps != 300000 &&
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bob->GetStats().send_bandwidth_bps != 300000)
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bitrate_changed = true;
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});
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s.RunUntil(TimeDelta::Seconds(2), TimeDelta::Millis(5),
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[&bitrate_changed, &packet_received] {
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return packet_received && bitrate_changed;
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});
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EXPECT_TRUE(packet_received);
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EXPECT_TRUE(bitrate_changed);
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}
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namespace {
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void SetupVideoCall(Scenario& s, VideoQualityAnalyzer* analyzer) {
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CallClientConfig call_config;
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auto* alice = s.CreateClient("alice", call_config);
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auto* bob = s.CreateClient("bob", call_config);
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NetworkSimulationConfig network_config;
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network_config.bandwidth = DataRate::KilobitsPerSec(1000);
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network_config.delay = TimeDelta::Millis(50);
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auto alice_net = s.CreateSimulationNode(network_config);
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auto bob_net = s.CreateSimulationNode(network_config);
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auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
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VideoStreamConfig video;
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if (analyzer) {
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video.source.capture = VideoStreamConfig::Source::Capture::kVideoFile;
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video.source.video_file.name = "foreman_cif";
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video.source.video_file.width = 352;
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video.source.video_file.height = 288;
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video.source.framerate = 30;
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video.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
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video.encoder.implementation =
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VideoStreamConfig::Encoder::Implementation::kSoftware;
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video.hooks.frame_pair_handlers = {analyzer->Handler()};
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}
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s.CreateVideoStream(route->forward(), video);
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s.CreateAudioStream(route->forward(), AudioStreamConfig());
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}
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} // namespace
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TEST(ScenarioTest, SimTimeEncoding) {
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VideoQualityAnalyzerConfig analyzer_config;
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analyzer_config.psnr_coverage = 0.1;
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VideoQualityAnalyzer analyzer(analyzer_config);
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{
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Scenario s("scenario/encode_sim", false);
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SetupVideoCall(s, &analyzer);
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s.RunFor(TimeDelta::Seconds(2));
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}
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// Regression tests based on previous runs.
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EXPECT_EQ(analyzer.stats().lost_count, 0);
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EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 38, 5);
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}
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// TODO(bugs.webrtc.org/10515): Remove this when performance has been improved.
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#if defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_ARM64) && !defined(NDEBUG)
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#define MAYBE_RealTimeEncoding DISABLED_RealTimeEncoding
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#else
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#define MAYBE_RealTimeEncoding RealTimeEncoding
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#endif
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TEST(ScenarioTest, MAYBE_RealTimeEncoding) {
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VideoQualityAnalyzerConfig analyzer_config;
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analyzer_config.psnr_coverage = 0.1;
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VideoQualityAnalyzer analyzer(analyzer_config);
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{
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Scenario s("scenario/encode_real", true);
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SetupVideoCall(s, &analyzer);
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s.RunFor(TimeDelta::Seconds(2));
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}
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// Regression tests based on previous runs.
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EXPECT_LT(analyzer.stats().lost_count, 2);
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// This far below expected but ensures that we get something.
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EXPECT_GT(analyzer.stats().psnr_with_freeze.Mean(), 10);
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}
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TEST(ScenarioTest, SimTimeFakeing) {
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Scenario s("scenario/encode_sim", false);
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SetupVideoCall(s, nullptr);
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s.RunFor(TimeDelta::Seconds(2));
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}
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TEST(ScenarioTest, WritesToRtcEventLog) {
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MemoryLogStorage storage;
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{
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Scenario s(storage.CreateFactory(), false);
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SetupVideoCall(s, nullptr);
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s.RunFor(TimeDelta::Seconds(1));
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}
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auto logs = storage.logs();
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// We expect that a rtc event log has been created and that it has some data.
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EXPECT_GE(storage.logs().at("alice.rtc.dat").size(), 1u);
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}
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TEST(ScenarioTest,
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RetransmitsVideoPacketsInAudioAndVideoCallWithSendSideBweAndLoss) {
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// Make sure audio packets are included in transport feedback.
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test::ScopedFieldTrials override_field_trials(
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"WebRTC-Audio-SendSideBwe/Enabled/WebRTC-Audio-ABWENoTWCC/Disabled/");
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Scenario s;
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CallClientConfig call_client_config;
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call_client_config.transport.rates.start_rate = DataRate::KilobitsPerSec(300);
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auto* alice = s.CreateClient("alice", call_client_config);
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auto* bob = s.CreateClient("bob", call_client_config);
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NetworkSimulationConfig network_config;
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// Add some loss and delay.
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network_config.delay = TimeDelta::Millis(200);
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network_config.loss_rate = 0.05;
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auto alice_net = s.CreateSimulationNode(network_config);
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auto bob_net = s.CreateSimulationNode(network_config);
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auto route = s.CreateRoutes(alice, {alice_net}, bob, {bob_net});
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// First add an audio stream, then a video stream.
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// Needed to make sure audio RTP module is selected first when sending
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// transport feedback message.
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AudioStreamConfig audio_stream_config;
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audio_stream_config.encoder.min_rate = DataRate::KilobitsPerSec(6);
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audio_stream_config.encoder.max_rate = DataRate::KilobitsPerSec(64);
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audio_stream_config.encoder.allocate_bitrate = true;
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audio_stream_config.stream.in_bandwidth_estimation = true;
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s.CreateAudioStream(route->forward(), audio_stream_config);
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s.CreateAudioStream(route->reverse(), audio_stream_config);
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VideoStreamConfig video_stream_config;
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auto video = s.CreateVideoStream(route->forward(), video_stream_config);
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s.CreateVideoStream(route->reverse(), video_stream_config);
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// Run for 10 seconds.
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s.RunFor(TimeDelta::Seconds(10));
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// Make sure retransmissions have happened.
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int retransmit_packets = 0;
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for (const auto& substream : video->send()->GetStats().substreams) {
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retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
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}
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EXPECT_GT(retransmit_packets, 0);
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}
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} // namespace test
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} // namespace webrtc
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