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1 /*
2  *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 #include "test/scenario/stats_collection.h"
11 
12 #include "test/gtest.h"
13 #include "test/scenario/scenario.h"
14 
15 namespace webrtc {
16 namespace test {
17 namespace {
CreateAnalyzedStream(Scenario * s,NetworkSimulationConfig network_config,VideoQualityAnalyzer * analyzer,CallStatsCollectors * collectors)18 void CreateAnalyzedStream(Scenario* s,
19                           NetworkSimulationConfig network_config,
20                           VideoQualityAnalyzer* analyzer,
21                           CallStatsCollectors* collectors) {
22   VideoStreamConfig config;
23   config.encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
24   config.encoder.implementation =
25       VideoStreamConfig::Encoder::Implementation::kSoftware;
26   config.hooks.frame_pair_handlers = {analyzer->Handler()};
27   auto* caller = s->CreateClient("caller", CallClientConfig());
28   auto* callee = s->CreateClient("callee", CallClientConfig());
29   auto route =
30       s->CreateRoutes(caller, {s->CreateSimulationNode(network_config)}, callee,
31                       {s->CreateSimulationNode(NetworkSimulationConfig())});
32   VideoStreamPair* video = s->CreateVideoStream(route->forward(), config);
33   auto* audio = s->CreateAudioStream(route->forward(), AudioStreamConfig());
34   s->Every(TimeDelta::Seconds(1), [=] {
35     collectors->call.AddStats(caller->GetStats());
36     collectors->video_send.AddStats(video->send()->GetStats(), s->Now());
37     collectors->audio_receive.AddStats(audio->receive()->GetStats());
38 
39     // Querying the video stats from within the expected runtime environment
40     // (i.e. the TQ that belongs to the CallClient, not the Scenario TQ that
41     // we're currently on).
42     VideoReceiveStream::Stats video_receive_stats;
43     auto* video_stream = video->receive();
44     callee->SendTask([&video_stream, &video_receive_stats]() {
45       video_receive_stats = video_stream->GetStats();
46     });
47     collectors->video_receive.AddStats(video_receive_stats);
48   });
49 }
50 }  // namespace
51 
TEST(ScenarioAnalyzerTest,PsnrIsHighWhenNetworkIsGood)52 TEST(ScenarioAnalyzerTest, PsnrIsHighWhenNetworkIsGood) {
53   VideoQualityAnalyzer analyzer;
54   CallStatsCollectors stats;
55   {
56     Scenario s;
57     NetworkSimulationConfig good_network;
58     good_network.bandwidth = DataRate::KilobitsPerSec(1000);
59     CreateAnalyzedStream(&s, good_network, &analyzer, &stats);
60     s.RunFor(TimeDelta::Seconds(3));
61   }
62   // This is a change detecting test, the targets are based on previous runs and
63   // might change due to changes in configuration and encoder etc. The main
64   // purpose is to show how the stats can be used. To avoid being overly
65   // sensistive to change, the ranges are chosen to be quite large.
66   EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 43, 10);
67   EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 700, 300);
68   EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 500, 200);
69   EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
70   EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 40, 20);
71 }
72 
TEST(ScenarioAnalyzerTest,PsnrIsLowWhenNetworkIsBad)73 TEST(ScenarioAnalyzerTest, PsnrIsLowWhenNetworkIsBad) {
74   VideoQualityAnalyzer analyzer;
75   CallStatsCollectors stats;
76   {
77     Scenario s;
78     NetworkSimulationConfig bad_network;
79     bad_network.bandwidth = DataRate::KilobitsPerSec(100);
80     bad_network.loss_rate = 0.02;
81     CreateAnalyzedStream(&s, bad_network, &analyzer, &stats);
82     s.RunFor(TimeDelta::Seconds(3));
83   }
84   // This is a change detecting test, the targets are based on previous runs and
85   // might change due to changes in configuration and encoder etc.
86   EXPECT_NEAR(analyzer.stats().psnr_with_freeze.Mean(), 20, 10);
87   EXPECT_NEAR(stats.call.stats().target_rate.Mean().kbps(), 75, 50);
88   EXPECT_NEAR(stats.video_send.stats().media_bitrate.Mean().kbps(), 100, 50);
89   EXPECT_NEAR(stats.video_receive.stats().resolution.Mean(), 180, 10);
90   EXPECT_NEAR(stats.audio_receive.stats().jitter_buffer.Mean().ms(), 200, 150);
91 }
92 
TEST(ScenarioAnalyzerTest,CountsCapturedButNotRendered)93 TEST(ScenarioAnalyzerTest, CountsCapturedButNotRendered) {
94   VideoQualityAnalyzer analyzer;
95   CallStatsCollectors stats;
96   {
97     Scenario s;
98     NetworkSimulationConfig long_delays;
99     long_delays.delay = TimeDelta::Seconds(5);
100     CreateAnalyzedStream(&s, long_delays, &analyzer, &stats);
101     // Enough time to send frames but not enough to deliver.
102     s.RunFor(TimeDelta::Millis(100));
103   }
104   EXPECT_GE(analyzer.stats().capture.count, 1);
105   EXPECT_EQ(analyzer.stats().render.count, 0);
106 }
107 }  // namespace test
108 }  // namespace webrtc
109