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/*
* 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;
for (const auto& substream : video->send()->GetStats().substreams) {
retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
}
EXPECT_GT(retransmit_packets, 0);
}
} // namespace test
} // namespace webrtc
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