| /* |
| * Copyright (c) 2015 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 <queue> |
| |
| #include "webrtc/base/format_macros.h" |
| #include "webrtc/base/timeutils.h" |
| #include "webrtc/system_wrappers/include/sleep.h" |
| #include "webrtc/test/gtest.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| #include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h" |
| |
| namespace { |
| const int kRttMs = 25; |
| |
| bool IsNear(int ref, int comp, int error) { |
| return (ref - comp <= error) && (comp - ref >= -error); |
| } |
| |
| void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) { |
| FILE* fid = fopen(silence_file.c_str(), "wb"); |
| int16_t zero = 0; |
| for (int i = 0; i < sample_rate_hz; ++i) { |
| // Write 1 second, but it does not matter since the file will be looped. |
| fwrite(&zero, sizeof(int16_t), 1, fid); |
| } |
| fclose(fid); |
| } |
| |
| } // namespace |
| |
| namespace voetest { |
| |
| TEST(VoeConferenceTest, RttAndStartNtpTime) { |
| struct Stats { |
| Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay) |
| : rtt_receiver_1_(rtt_receiver_1), |
| rtt_receiver_2_(rtt_receiver_2), |
| ntp_delay_(ntp_delay) { |
| } |
| int64_t rtt_receiver_1_; |
| int64_t rtt_receiver_2_; |
| int64_t ntp_delay_; |
| }; |
| |
| const std::string input_file = |
| webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
| const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile; |
| |
| const int kDelayMs = 987; |
| ConferenceTransport trans; |
| trans.SetRtt(kRttMs); |
| |
| unsigned int id_1 = trans.AddStream(input_file, kInputFormat); |
| unsigned int id_2 = trans.AddStream(input_file, kInputFormat); |
| |
| EXPECT_TRUE(trans.StartPlayout(id_1)); |
| // Start NTP time is the time when a stream is played out, rather than |
| // when it is added. |
| webrtc::SleepMs(kDelayMs); |
| EXPECT_TRUE(trans.StartPlayout(id_2)); |
| |
| const int kMaxRunTimeMs = 25000; |
| const int kNeedSuccessivePass = 3; |
| const int kStatsRequestIntervalMs = 1000; |
| const int kStatsBufferSize = 3; |
| |
| int64_t deadline = rtc::TimeAfter(kMaxRunTimeMs); |
| // Run the following up to |kMaxRunTimeMs| milliseconds. |
| int successive_pass = 0; |
| webrtc::CallStatistics stats_1; |
| webrtc::CallStatistics stats_2; |
| std::queue<Stats> stats_buffer; |
| |
| while (rtc::TimeMillis() < deadline && |
| successive_pass < kNeedSuccessivePass) { |
| webrtc::SleepMs(kStatsRequestIntervalMs); |
| |
| EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1)); |
| EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2)); |
| |
| // It is not easy to verify the NTP time directly. We verify it by testing |
| // the difference of two start NTP times. |
| int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ - |
| stats_1.capture_start_ntp_time_ms_; |
| |
| // For the checks of RTT and start NTP time, We allow 10% accuracy. |
| if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) && |
| IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) && |
| IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) { |
| successive_pass++; |
| } else { |
| successive_pass = 0; |
| } |
| if (stats_buffer.size() >= kStatsBufferSize) { |
| stats_buffer.pop(); |
| } |
| stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs, |
| captured_start_ntp_delay)); |
| } |
| |
| EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and" |
| " start NTP time estimate within 10% of the correct value over " |
| << kStatsRequestIntervalMs * kNeedSuccessivePass / 1000 |
| << " seconds."; |
| if (successive_pass < kNeedSuccessivePass) { |
| printf("The most recent values (RTT for receiver 1, RTT for receiver 2, " |
| "NTP delay between receiver 1 and 2) are (from oldest):\n"); |
| while (!stats_buffer.empty()) { |
| Stats stats = stats_buffer.front(); |
| printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_, |
| stats.rtt_receiver_2_, stats.ntp_delay_); |
| stats_buffer.pop(); |
| } |
| } |
| } |
| |
| |
| TEST(VoeConferenceTest, ReceivedPackets) { |
| const int kPackets = 50; |
| const int kPacketDurationMs = 20; // Correspond to Opus. |
| |
| const std::string input_file = |
| webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); |
| const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile; |
| |
| const std::string silence_file = |
| webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence"); |
| CreateSilenceFile(silence_file, 32000); |
| |
| { |
| ConferenceTransport trans; |
| // Add silence to stream 0, so that it will be filtered out. |
| unsigned int id_0 = trans.AddStream(silence_file, kInputFormat); |
| unsigned int id_1 = trans.AddStream(input_file, kInputFormat); |
| unsigned int id_2 = trans.AddStream(input_file, kInputFormat); |
| unsigned int id_3 = trans.AddStream(input_file, kInputFormat); |
| |
| EXPECT_TRUE(trans.StartPlayout(id_0)); |
| EXPECT_TRUE(trans.StartPlayout(id_1)); |
| EXPECT_TRUE(trans.StartPlayout(id_2)); |
| EXPECT_TRUE(trans.StartPlayout(id_3)); |
| |
| webrtc::SleepMs(kPacketDurationMs * kPackets); |
| |
| webrtc::CallStatistics stats_0; |
| webrtc::CallStatistics stats_1; |
| webrtc::CallStatistics stats_2; |
| webrtc::CallStatistics stats_3; |
| EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0)); |
| EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1)); |
| EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2)); |
| EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3)); |
| |
| // We expect stream 0 to be filtered out totally, but since it may join the |
| // call earlier than other streams and the beginning packets might have got |
| // through. So we only expect |packetsReceived| to be close to zero. |
| EXPECT_NEAR(stats_0.packetsReceived, 0, 2); |
| // We expect |packetsReceived| to match |kPackets|, but the actual value |
| // depends on the sleep timer. So we allow a small off from |kPackets|. |
| EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2); |
| EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2); |
| EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2); |
| } |
| |
| remove(silence_file.c_str()); |
| } |
| |
| } // namespace voetest |