blob: b4b31cc62a6427641bd88fe569fa5e826dc90cc1 [file] [log] [blame]
/*
* 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/stats_collection.h"
#include "test/gtest.h"
#include "test/scenario/scenario.h"
namespace webrtc {
namespace test {
namespace {
void CreateAnalyzedStream(Scenario* s,
NetworkSimulationConfig network_config,
VideoQualityAnalyzer* analyzer,
CallStatsCollectors* collectors) {
VideoStreamConfig config;
config.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
config.encoder.implementation =
VideoStreamConfig::Encoder::Implementation::kSoftware;
config.hooks.frame_pair_handlers = {analyzer->Handler()};
auto* caller = s->CreateClient("caller", CallClientConfig());
auto route =
s->CreateRoutes(caller, {s->CreateSimulationNode(network_config)},
s->CreateClient("callee", CallClientConfig()),
{s->CreateSimulationNode(NetworkSimulationConfig())});
auto* video = s->CreateVideoStream(route->forward(), config);
auto* audio = s->CreateAudioStream(route->forward(), AudioStreamConfig());
if (collectors) {
s->Every(TimeDelta::seconds(1), [=] {
collectors->call.AddStats(caller->GetStats());
collectors->audio_receive.AddStats(audio->receive()->GetStats());
collectors->video_send.AddStats(video->send()->GetStats(), s->Now());
collectors->video_receive.AddStats(video->receive()->GetStats());
});
}
}
} // namespace
TEST(ScenarioAnalyzerTest, PsnrIsHighWhenNetworkIsGood) {
VideoQualityAnalyzer analyzer;
CallStatsCollectors stats;
{
Scenario s;
NetworkSimulationConfig good_network;
good_network.bandwidth = DataRate::kbps(1000);
CreateAnalyzedStream(&s, good_network, &analyzer, &stats);
s.RunFor(TimeDelta::seconds(3));
}
// This is a change detecting test, the targets are based on previous runs and
// might change due to changes in configuration and encoder etc. The main
// purpose is to show how the stats can be used. To avoid being overly
// sensistive to change, the ranges are chosen to be quite large.
EXPECT_NEAR(analyzer.stats().psnr.Mean(), 43, 10);
EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 700, 300);
EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 500, 200);
EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 40, 20);
}
TEST(ScenarioAnalyzerTest, PsnrIsLowWhenNetworkIsBad) {
VideoQualityAnalyzer analyzer;
CallStatsCollectors stats;
{
Scenario s;
NetworkSimulationConfig bad_network;
bad_network.bandwidth = DataRate::kbps(100);
bad_network.loss_rate = 0.02;
CreateAnalyzedStream(&s, bad_network, &analyzer, &stats);
s.RunFor(TimeDelta::seconds(3));
}
// This is a change detecting test, the targets are based on previous runs and
// might change due to changes in configuration and encoder etc.
EXPECT_NEAR(analyzer.stats().psnr.Mean(), 16, 10);
EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 75, 50);
EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 100, 50);
EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 45, 20);
}
} // namespace test
} // namespace webrtc