blob: 11aab07c0bec2e3ab28f397da3aacb3bc4b978d3 [file] [log] [blame]
/*
* Copyright (c) 2019 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/gtest.h"
#include "test/peer_scenario/peer_scenario.h"
namespace webrtc {
namespace test {
TEST(PeerScenarioQualityTest, PsnrIsCollected) {
VideoQualityAnalyzerConfig analyzer_config;
analyzer_config.thread = rtc::Thread::Current();
VideoQualityAnalyzer analyzer(analyzer_config);
PeerScenario s(*test_info_);
auto caller = s.CreateClient(PeerScenarioClient::Config());
auto callee = s.CreateClient(PeerScenarioClient::Config());
PeerScenarioClient::VideoSendTrackConfig video_conf;
video_conf.generator.squares_video->framerate = 20;
auto video = caller->CreateVideo("VIDEO", video_conf);
auto link_builder = s.net()->NodeBuilder().delay_ms(100).capacity_kbps(600);
s.AttachVideoQualityAnalyzer(&analyzer, video.track, callee);
s.SimpleConnection(caller, callee, {link_builder.Build().node},
{link_builder.Build().node});
s.ProcessMessages(TimeDelta::seconds(2));
// We expect ca 40 frames to be produced, but to avoid flakiness on slow
// machines we only test for 10.
EXPECT_GT(analyzer.stats().render.count, 10);
EXPECT_GT(analyzer.stats().psnr_with_freeze.Mean(), 20);
}
} // namespace test
} // namespace webrtc