1 files changed, 114 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/scenario/stats_collection_unittest.cc b/third_party/libwebrtc/test/scenario/stats_collection_unittest.cc
new file mode 100644
index 0000000000..9f46f10073
--- /dev/null
+++ b/third_party/libwebrtc/test/scenario/stats_collection_unittest.cc
@@ -0,0 +1,114 @@
+/*
+ *  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* callee = s->CreateClient("callee", CallClientConfig());
+  auto route =
+      s->CreateRoutes(caller, {s->CreateSimulationNode(network_config)}, callee,
+                      {s->CreateSimulationNode(NetworkSimulationConfig())});
+  VideoStreamPair* video = s->CreateVideoStream(route->forward(), config);
+  auto* audio = s->CreateAudioStream(route->forward(), AudioStreamConfig());
+  s->Every(TimeDelta::Seconds(1), [=] {
+    collectors->call.AddStats(caller->GetStats());
+
+    VideoSendStream::Stats send_stats;
+    caller->SendTask([&]() { send_stats = video->send()->GetStats(); });
+    collectors->video_send.AddStats(send_stats, s->Now());
+
+    AudioReceiveStreamInterface::Stats receive_stats;
+    caller->SendTask([&]() { receive_stats = audio->receive()->GetStats(); });
+    collectors->audio_receive.AddStats(receive_stats);
+
+    // Querying the video stats from within the expected runtime environment
+    // (i.e. the TQ that belongs to the CallClient, not the Scenario TQ that
+    // we're currently on).
+    VideoReceiveStreamInterface::Stats video_receive_stats;
+    auto* video_stream = video->receive();
+    callee->SendTask([&video_stream, &video_receive_stats]() {
+      video_receive_stats = video_stream->GetStats();
+    });
+    collectors->video_receive.AddStats(video_receive_stats);
+  });
+}
+}  // namespace
+
+TEST(ScenarioAnalyzerTest, PsnrIsHighWhenNetworkIsGood) {
+  VideoQualityAnalyzer analyzer;
+  CallStatsCollectors stats;
+  {
+    Scenario s;
+    NetworkSimulationConfig good_network;
+    good_network.bandwidth = DataRate::KilobitsPerSec(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_with_freeze.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::KilobitsPerSec(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_with_freeze.Mean(), 20, 10);
+  EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 75, 50);
+  EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 70, 30);
+  EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
+  EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 250, 200);
+}
+
+TEST(ScenarioAnalyzerTest, CountsCapturedButNotRendered) {
+  VideoQualityAnalyzer analyzer;
+  CallStatsCollectors stats;
+  {
+    Scenario s;
+    NetworkSimulationConfig long_delays;
+    long_delays.delay = TimeDelta::Seconds(5);
+    CreateAnalyzedStream(&s, long_delays, &analyzer, &stats);
+    // Enough time to send frames but not enough to deliver.
+    s.RunFor(TimeDelta::Millis(100));
+  }
+  EXPECT_GE(analyzer.stats().capture.count, 1);
+  EXPECT_EQ(analyzer.stats().render.count, 0);
+}
+}  // namespace test
+}  // namespace webrtc