9d69cbeabf
This changes the default behavior to use pacing factor of 1.1x instead of 2.5x, it also sets libvpx rate controler as trusted, turns on the encoder pushback mechanism and sets spatial hysteresis to 1.2. The unused "dynamic rate" settings in libvpx is removed. The new settings matches what has been used in chromium since 2019. If needed, the legacy behavior can be enabled using the field trial WebRTC-VideoRateControl. Bug: webrtc:10155 Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32477}
109 lines
4.5 KiB
C++
109 lines
4.5 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* callee = s->CreateClient("callee", CallClientConfig());
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auto route =
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s->CreateRoutes(caller, {s->CreateSimulationNode(network_config)}, callee,
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{s->CreateSimulationNode(NetworkSimulationConfig())});
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VideoStreamPair* 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->video_send.AddStats(video->send()->GetStats(), s->Now());
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collectors->audio_receive.AddStats(audio->receive()->GetStats());
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// Querying the video stats from within the expected runtime environment
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// (i.e. the TQ that belongs to the CallClient, not the Scenario TQ that
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// we're currently on).
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VideoReceiveStream::Stats video_receive_stats;
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auto* video_stream = video->receive();
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callee->SendTask([&video_stream, &video_receive_stats]() {
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video_receive_stats = video_stream->GetStats();
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});
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collectors->video_receive.AddStats(video_receive_stats);
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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(), 20, 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(), 250, 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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