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cad3e0e2fa
This is search and replace change: find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::Bytes<\(.*\)>()/DataSize::Bytes(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataSize::bytes/DataSize::Bytes/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BitsPerSec<\(.*\)>()/DataRate::BitsPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::BytesPerSec<\(.*\)>()/DataRate::BytesPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::KilobitsPerSec<\(.*\)>()/DataRate::KilobitsPerSec(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::bps/DataRate::BitsPerSec/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/DataRate::kbps/DataRate::KilobitsPerSec/g" git cl format Bug: webrtc:9709 Change-Id: I65aaca69474ba038c1fe2dd8dc30d3f8e7b94c29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168647 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30545}
99 lines
4.2 KiB
C++
99 lines
4.2 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/stats_collection.h"
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#include "test/gtest.h"
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#include "test/scenario/scenario.h"
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namespace webrtc {
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namespace test {
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namespace {
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void CreateAnalyzedStream(Scenario* s,
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NetworkSimulationConfig network_config,
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VideoQualityAnalyzer* analyzer,
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CallStatsCollectors* collectors) {
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VideoStreamConfig config;
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config.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
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config.encoder.implementation =
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VideoStreamConfig::Encoder::Implementation::kSoftware;
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config.hooks.frame_pair_handlers = {analyzer->Handler()};
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auto* caller = s->CreateClient("caller", CallClientConfig());
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auto route =
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s->CreateRoutes(caller, {s->CreateSimulationNode(network_config)},
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s->CreateClient("callee", CallClientConfig()),
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{s->CreateSimulationNode(NetworkSimulationConfig())});
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auto* video = s->CreateVideoStream(route->forward(), config);
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auto* audio = s->CreateAudioStream(route->forward(), AudioStreamConfig());
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s->Every(TimeDelta::Seconds(1), [=] {
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collectors->call.AddStats(caller->GetStats());
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collectors->audio_receive.AddStats(audio->receive()->GetStats());
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collectors->video_send.AddStats(video->send()->GetStats(), s->Now());
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collectors->video_receive.AddStats(video->receive()->GetStats());
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});
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}
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} // namespace
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TEST(ScenarioAnalyzerTest, PsnrIsHighWhenNetworkIsGood) {
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VideoQualityAnalyzer analyzer;
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CallStatsCollectors stats;
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{
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Scenario s;
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NetworkSimulationConfig good_network;
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good_network.bandwidth = DataRate::KilobitsPerSec(1000);
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CreateAnalyzedStream(&s, good_network, &analyzer, &stats);
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s.RunFor(TimeDelta::Seconds(3));
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}
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// This is a change detecting test, the targets are based on previous runs and
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// might change due to changes in configuration and encoder etc. The main
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// purpose is to show how the stats can be used. To avoid being overly
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// sensistive to change, the ranges are chosen to be quite large.
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EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 43, 10);
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EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 700, 300);
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EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 500, 200);
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EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
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EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 40, 20);
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}
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TEST(ScenarioAnalyzerTest, PsnrIsLowWhenNetworkIsBad) {
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VideoQualityAnalyzer analyzer;
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CallStatsCollectors stats;
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{
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Scenario s;
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NetworkSimulationConfig bad_network;
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bad_network.bandwidth = DataRate::KilobitsPerSec(100);
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bad_network.loss_rate = 0.02;
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CreateAnalyzedStream(&s, bad_network, &analyzer, &stats);
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s.RunFor(TimeDelta::Seconds(3));
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}
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// This is a change detecting test, the targets are based on previous runs and
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// might change due to changes in configuration and encoder etc.
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EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 16, 10);
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EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 75, 50);
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EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 100, 50);
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EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
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EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 200, 150);
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}
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TEST(ScenarioAnalyzerTest, CountsCapturedButNotRendered) {
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VideoQualityAnalyzer analyzer;
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CallStatsCollectors stats;
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{
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Scenario s;
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NetworkSimulationConfig long_delays;
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long_delays.delay = TimeDelta::Seconds(5);
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CreateAnalyzedStream(&s, long_delays, &analyzer, &stats);
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// Enough time to send frames but not enough to deliver.
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s.RunFor(TimeDelta::Millis(100));
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}
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EXPECT_GE(analyzer.stats().capture.count, 1);
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EXPECT_EQ(analyzer.stats().render.count, 0);
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}
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} // namespace test
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} // namespace webrtc
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