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