1 /*
2 * Copyright (c) 2015 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
11 #include <queue>
12
13 #include "testing/gtest/include/gtest/gtest.h"
14 #include "webrtc/base/format_macros.h"
15 #include "webrtc/base/timeutils.h"
16 #include "webrtc/system_wrappers/include/sleep.h"
17 #include "webrtc/test/testsupport/fileutils.h"
18 #include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
19
20 namespace {
21 const int kRttMs = 25;
22
IsNear(int ref,int comp,int error)23 bool IsNear(int ref, int comp, int error) {
24 return (ref - comp <= error) && (comp - ref >= -error);
25 }
26
CreateSilenceFile(const std::string & silence_file,int sample_rate_hz)27 void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) {
28 FILE* fid = fopen(silence_file.c_str(), "wb");
29 int16_t zero = 0;
30 for (int i = 0; i < sample_rate_hz; ++i) {
31 // Write 1 second, but it does not matter since the file will be looped.
32 fwrite(&zero, sizeof(int16_t), 1, fid);
33 }
34 fclose(fid);
35 }
36
37 } // namespace
38
39 namespace voetest {
40
TEST(VoeConferenceTest,RttAndStartNtpTime)41 TEST(VoeConferenceTest, RttAndStartNtpTime) {
42 struct Stats {
43 Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay)
44 : rtt_receiver_1_(rtt_receiver_1),
45 rtt_receiver_2_(rtt_receiver_2),
46 ntp_delay_(ntp_delay) {
47 }
48 int64_t rtt_receiver_1_;
49 int64_t rtt_receiver_2_;
50 int64_t ntp_delay_;
51 };
52
53 const std::string input_file =
54 webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
55 const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
56
57 const int kDelayMs = 987;
58 ConferenceTransport trans;
59 trans.SetRtt(kRttMs);
60
61 unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
62 unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
63
64 EXPECT_TRUE(trans.StartPlayout(id_1));
65 // Start NTP time is the time when a stream is played out, rather than
66 // when it is added.
67 webrtc::SleepMs(kDelayMs);
68 EXPECT_TRUE(trans.StartPlayout(id_2));
69
70 const int kMaxRunTimeMs = 25000;
71 const int kNeedSuccessivePass = 3;
72 const int kStatsRequestIntervalMs = 1000;
73 const int kStatsBufferSize = 3;
74
75 uint32_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
76 // Run the following up to |kMaxRunTimeMs| milliseconds.
77 int successive_pass = 0;
78 webrtc::CallStatistics stats_1;
79 webrtc::CallStatistics stats_2;
80 std::queue<Stats> stats_buffer;
81
82 while (rtc::TimeIsLater(rtc::Time(), deadline) &&
83 successive_pass < kNeedSuccessivePass) {
84 webrtc::SleepMs(kStatsRequestIntervalMs);
85
86 EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
87 EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
88
89 // It is not easy to verify the NTP time directly. We verify it by testing
90 // the difference of two start NTP times.
91 int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
92 stats_1.capture_start_ntp_time_ms_;
93
94 // For the checks of RTT and start NTP time, We allow 10% accuracy.
95 if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
96 IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
97 IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) {
98 successive_pass++;
99 } else {
100 successive_pass = 0;
101 }
102 if (stats_buffer.size() >= kStatsBufferSize) {
103 stats_buffer.pop();
104 }
105 stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
106 captured_start_ntp_delay));
107 }
108
109 EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and"
110 " start NTP time estimate within 10% of the correct value over "
111 << kStatsRequestIntervalMs * kNeedSuccessivePass / 1000
112 << " seconds.";
113 if (successive_pass < kNeedSuccessivePass) {
114 printf("The most recent values (RTT for receiver 1, RTT for receiver 2, "
115 "NTP delay between receiver 1 and 2) are (from oldest):\n");
116 while (!stats_buffer.empty()) {
117 Stats stats = stats_buffer.front();
118 printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_,
119 stats.rtt_receiver_2_, stats.ntp_delay_);
120 stats_buffer.pop();
121 }
122 }
123 }
124
125
TEST(VoeConferenceTest,ReceivedPackets)126 TEST(VoeConferenceTest, ReceivedPackets) {
127 const int kPackets = 50;
128 const int kPacketDurationMs = 20; // Correspond to Opus.
129
130 const std::string input_file =
131 webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
132 const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
133
134 const std::string silence_file =
135 webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
136 CreateSilenceFile(silence_file, 32000);
137
138 {
139 ConferenceTransport trans;
140 // Add silence to stream 0, so that it will be filtered out.
141 unsigned int id_0 = trans.AddStream(silence_file, kInputFormat);
142 unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
143 unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
144 unsigned int id_3 = trans.AddStream(input_file, kInputFormat);
145
146 EXPECT_TRUE(trans.StartPlayout(id_0));
147 EXPECT_TRUE(trans.StartPlayout(id_1));
148 EXPECT_TRUE(trans.StartPlayout(id_2));
149 EXPECT_TRUE(trans.StartPlayout(id_3));
150
151 webrtc::SleepMs(kPacketDurationMs * kPackets);
152
153 webrtc::CallStatistics stats_0;
154 webrtc::CallStatistics stats_1;
155 webrtc::CallStatistics stats_2;
156 webrtc::CallStatistics stats_3;
157 EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0));
158 EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
159 EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
160 EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3));
161
162 // We expect stream 0 to be filtered out totally, but since it may join the
163 // call earlier than other streams and the beginning packets might have got
164 // through. So we only expect |packetsReceived| to be close to zero.
165 EXPECT_NEAR(stats_0.packetsReceived, 0, 2);
166 // We expect |packetsReceived| to match |kPackets|, but the actual value
167 // depends on the sleep timer. So we allow a small off from |kPackets|.
168 EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
169 EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
170 EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
171 }
172
173 remove(silence_file.c_str());
174 }
175
176 } // namespace voetest
177