test/scenario/stats_collection_unittest.cc

/*
 *  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