987ef48258
The decision to route audio packets to a separate overuse detector is off by default and requires the field trial WebRTC-Bwe-SeparateAudioPackets/enabled,packet_threshold:10,time_threshold:1000ms/ The parameters control the threshold for switching over to the audio overuse detector if we stop receiving feedback for video. Bug: webrtc:10932 Change-Id: Icdde35bc7a98b18b1a344bd2d620a890fd9421d9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168342 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30694}
47 lines
1.8 KiB
C++
47 lines
1.8 KiB
C++
/*
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* Copyright (c) 2019 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/gtest.h"
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#include "test/peer_scenario/peer_scenario.h"
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#include "test/peer_scenario/peer_scenario_client.h"
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namespace webrtc {
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namespace test {
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#if defined(WEBRTC_WIN)
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#define MAYBE_PsnrIsCollected DISABLED_PsnrIsCollected
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#else
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#define MAYBE_PsnrIsCollected PsnrIsCollected
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#endif
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TEST(PeerScenarioQualityTest, MAYBE_PsnrIsCollected) {
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VideoQualityAnalyzer analyzer;
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{
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PeerScenario s(*test_info_);
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auto caller = s.CreateClient(PeerScenarioClient::Config());
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auto callee = s.CreateClient(PeerScenarioClient::Config());
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PeerScenarioClient::VideoSendTrackConfig video_conf;
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video_conf.generator.squares_video->framerate = 20;
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auto video = caller->CreateVideo("VIDEO", video_conf);
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auto link_builder = s.net()->NodeBuilder().delay_ms(100).capacity_kbps(600);
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s.AttachVideoQualityAnalyzer(&analyzer, video.track, callee);
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s.SimpleConnection(caller, callee, {link_builder.Build().node},
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{link_builder.Build().node});
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s.ProcessMessages(TimeDelta::Seconds(2));
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// Exit scope to ensure that there's no pending tasks reporting to analyzer.
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}
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// We expect ca 40 frames to be produced, but to avoid flakiness on slow
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// machines we only test for 10.
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EXPECT_GT(analyzer.stats().render.count, 10);
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EXPECT_GT(analyzer.stats().psnr_with_freeze.Mean(), 20);
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}
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} // namespace test
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} // namespace webrtc
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