stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2012 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 | |
kjellander@webrtc.org | 0fcaf99 | 2015-11-26 15:24:52 +0100 | [diff] [blame] | 11 | #include <algorithm> |
| 12 | |
Yves Gerey | 3e70781 | 2018-11-28 16:47:49 +0100 | [diff] [blame] | 13 | #include "system_wrappers/include/ntp_time.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 14 | #include "test/gtest.h" |
| 15 | #include "video/stream_synchronization.h" |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 16 | |
| 17 | namespace webrtc { |
| 18 | |
| 19 | // These correspond to the same constants defined in vie_sync_module.cc. |
| 20 | enum { kMaxVideoDiffMs = 80 }; |
| 21 | enum { kMaxAudioDiffMs = 80 }; |
| 22 | enum { kMaxDelay = 1500 }; |
| 23 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 24 | // Test constants. |
| 25 | enum { kDefaultAudioFrequency = 8000 }; |
| 26 | enum { kDefaultVideoFrequency = 90000 }; |
| 27 | const double kNtpFracPerMs = 4.294967296E6; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 28 | static const int kSmoothingFilter = 4 * 2; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 29 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 30 | class Time { |
| 31 | public: |
| 32 | explicit Time(int64_t offset) |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 33 | : kNtpJan1970(2208988800UL), time_now_ms_(offset) {} |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 34 | |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 35 | NtpTime GetNowNtp() const { |
| 36 | uint32_t ntp_secs = time_now_ms_ / 1000 + kNtpJan1970; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 37 | int64_t remainder_ms = time_now_ms_ % 1000; |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 38 | uint32_t ntp_frac = static_cast<uint32_t>( |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 39 | static_cast<double>(remainder_ms) * kNtpFracPerMs + 0.5); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 40 | return NtpTime(ntp_secs, ntp_frac); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 41 | } |
| 42 | |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 43 | uint32_t GetNowRtp(int frequency, uint32_t offset) const { |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 44 | return frequency * time_now_ms_ / 1000 + offset; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 45 | } |
| 46 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 47 | void IncreaseTimeMs(int64_t inc) { time_now_ms_ += inc; } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 48 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 49 | int64_t time_now_ms() const { return time_now_ms_; } |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 50 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 51 | private: |
| 52 | // January 1970, in NTP seconds. |
| 53 | const uint32_t kNtpJan1970; |
| 54 | int64_t time_now_ms_; |
| 55 | }; |
| 56 | |
| 57 | class StreamSynchronizationTest : public ::testing::Test { |
| 58 | protected: |
| 59 | virtual void SetUp() { |
| 60 | sync_ = new StreamSynchronization(0, 0); |
| 61 | send_time_ = new Time(kSendTimeOffsetMs); |
| 62 | receive_time_ = new Time(kReceiveTimeOffsetMs); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 63 | audio_clock_drift_ = 1.0; |
| 64 | video_clock_drift_ = 1.0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 65 | } |
| 66 | |
| 67 | virtual void TearDown() { |
| 68 | delete sync_; |
| 69 | delete send_time_; |
| 70 | delete receive_time_; |
| 71 | } |
| 72 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 73 | // Generates the necessary RTCP measurements and RTP timestamps and computes |
| 74 | // the audio and video delays needed to get the two streams in sync. |
| 75 | // |audio_delay_ms| and |video_delay_ms| are the number of milliseconds after |
| 76 | // capture which the frames are rendered. |
| 77 | // |current_audio_delay_ms| is the number of milliseconds which audio is |
| 78 | // currently being delayed by the receiver. |
| 79 | bool DelayedStreams(int audio_delay_ms, |
| 80 | int video_delay_ms, |
| 81 | int current_audio_delay_ms, |
| 82 | int* extra_audio_delay_ms, |
| 83 | int* total_video_delay_ms) { |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 84 | int audio_frequency = |
| 85 | static_cast<int>(kDefaultAudioFrequency * audio_clock_drift_ + 0.5); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 86 | int audio_offset = 0; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 87 | int video_frequency = |
| 88 | static_cast<int>(kDefaultVideoFrequency * video_clock_drift_ + 0.5); |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 89 | bool new_sr; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 90 | int video_offset = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 91 | StreamSynchronization::Measurements audio; |
| 92 | StreamSynchronization::Measurements video; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 93 | // Generate NTP/RTP timestamp pair for both streams corresponding to RTCP. |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 94 | NtpTime ntp_time = send_time_->GetNowNtp(); |
| 95 | uint32_t rtp_timestamp = |
| 96 | send_time_->GetNowRtp(audio_frequency, audio_offset); |
| 97 | EXPECT_TRUE(audio.rtp_to_ntp.UpdateMeasurements( |
| 98 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 99 | send_time_->IncreaseTimeMs(100); |
| 100 | receive_time_->IncreaseTimeMs(100); |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 101 | ntp_time = send_time_->GetNowNtp(); |
| 102 | rtp_timestamp = send_time_->GetNowRtp(video_frequency, video_offset); |
| 103 | EXPECT_TRUE(video.rtp_to_ntp.UpdateMeasurements( |
| 104 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 105 | send_time_->IncreaseTimeMs(900); |
| 106 | receive_time_->IncreaseTimeMs(900); |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 107 | ntp_time = send_time_->GetNowNtp(); |
| 108 | rtp_timestamp = send_time_->GetNowRtp(audio_frequency, audio_offset); |
| 109 | EXPECT_TRUE(audio.rtp_to_ntp.UpdateMeasurements( |
| 110 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 111 | send_time_->IncreaseTimeMs(100); |
| 112 | receive_time_->IncreaseTimeMs(100); |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 113 | ntp_time = send_time_->GetNowNtp(); |
| 114 | rtp_timestamp = send_time_->GetNowRtp(video_frequency, video_offset); |
| 115 | EXPECT_TRUE(video.rtp_to_ntp.UpdateMeasurements( |
| 116 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 117 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 118 | send_time_->IncreaseTimeMs(900); |
| 119 | receive_time_->IncreaseTimeMs(900); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 120 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 121 | // Capture an audio and a video frame at the same time. |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 122 | audio.latest_timestamp = |
| 123 | send_time_->GetNowRtp(audio_frequency, audio_offset); |
| 124 | video.latest_timestamp = |
| 125 | send_time_->GetNowRtp(video_frequency, video_offset); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 126 | |
| 127 | if (audio_delay_ms > video_delay_ms) { |
| 128 | // Audio later than video. |
| 129 | receive_time_->IncreaseTimeMs(video_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 130 | video.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 131 | receive_time_->IncreaseTimeMs(audio_delay_ms - video_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 132 | audio.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 133 | } else { |
| 134 | // Video later than audio. |
| 135 | receive_time_->IncreaseTimeMs(audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 136 | audio.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 137 | receive_time_->IncreaseTimeMs(video_delay_ms - audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 138 | video.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 139 | } |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 140 | int relative_delay_ms; |
| 141 | StreamSynchronization::ComputeRelativeDelay(audio, video, |
| 142 | &relative_delay_ms); |
| 143 | EXPECT_EQ(video_delay_ms - audio_delay_ms, relative_delay_ms); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 144 | return sync_->ComputeDelays(relative_delay_ms, current_audio_delay_ms, |
| 145 | extra_audio_delay_ms, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 146 | } |
| 147 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 148 | // Simulate audio playback 300 ms after capture and video rendering 100 ms |
| 149 | // after capture. Verify that the correct extra delays are calculated for |
| 150 | // audio and video, and that they change correctly when we simulate that |
| 151 | // NetEQ or the VCM adds more delay to the streams. |
| 152 | // TODO(holmer): This is currently wrong! We should simply change |
| 153 | // audio_delay_ms or video_delay_ms since those now include VCM and NetEQ |
| 154 | // delays. |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 155 | void BothDelayedAudioLaterTest(int base_target_delay) { |
| 156 | int current_audio_delay_ms = base_target_delay; |
| 157 | int audio_delay_ms = base_target_delay + 300; |
| 158 | int video_delay_ms = base_target_delay + 100; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 159 | int extra_audio_delay_ms = 0; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 160 | int total_video_delay_ms = base_target_delay; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 161 | int filtered_move = (audio_delay_ms - video_delay_ms) / kSmoothingFilter; |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 162 | const int kNeteqDelayIncrease = 50; |
| 163 | const int kNeteqDelayDecrease = 10; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 164 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 165 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 166 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 167 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 168 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 169 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 170 | current_audio_delay_ms = extra_audio_delay_ms; |
| 171 | |
| 172 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 173 | receive_time_->IncreaseTimeMs(1000 - |
| 174 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 175 | // Simulate base_target_delay minimum delay in the VCM. |
| 176 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 177 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 178 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 179 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 180 | EXPECT_EQ(base_target_delay + 2 * filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 181 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 182 | current_audio_delay_ms = extra_audio_delay_ms; |
| 183 | |
| 184 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 185 | receive_time_->IncreaseTimeMs(1000 - |
| 186 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 187 | // Simulate base_target_delay minimum delay in the VCM. |
| 188 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 189 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 190 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 191 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 192 | EXPECT_EQ(base_target_delay + 3 * filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 193 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 194 | |
| 195 | // Simulate that NetEQ introduces some audio delay. |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 196 | current_audio_delay_ms = base_target_delay + kNeteqDelayIncrease; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 197 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 198 | receive_time_->IncreaseTimeMs(1000 - |
| 199 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 200 | // Simulate base_target_delay minimum delay in the VCM. |
| 201 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 202 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 203 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 204 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 205 | filtered_move = 3 * filtered_move + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 206 | (kNeteqDelayIncrease + audio_delay_ms - video_delay_ms) / |
| 207 | kSmoothingFilter; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 208 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 209 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 210 | |
| 211 | // Simulate that NetEQ reduces its delay. |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 212 | current_audio_delay_ms = base_target_delay + kNeteqDelayDecrease; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 213 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 214 | receive_time_->IncreaseTimeMs(1000 - |
| 215 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 216 | // Simulate base_target_delay minimum delay in the VCM. |
| 217 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 218 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 219 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 220 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 221 | |
| 222 | filtered_move = filtered_move + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 223 | (kNeteqDelayDecrease + audio_delay_ms - video_delay_ms) / |
| 224 | kSmoothingFilter; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 225 | |
| 226 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 227 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
| 228 | } |
| 229 | |
| 230 | void BothDelayedVideoLaterTest(int base_target_delay) { |
| 231 | int current_audio_delay_ms = base_target_delay; |
| 232 | int audio_delay_ms = base_target_delay + 100; |
| 233 | int video_delay_ms = base_target_delay + 300; |
| 234 | int extra_audio_delay_ms = 0; |
| 235 | int total_video_delay_ms = base_target_delay; |
| 236 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 237 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 238 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 239 | &total_video_delay_ms)); |
| 240 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 241 | // The audio delay is not allowed to change more than this in 1 second. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 242 | EXPECT_GE(base_target_delay + kMaxAudioDiffMs, extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 243 | current_audio_delay_ms = extra_audio_delay_ms; |
| 244 | int current_extra_delay_ms = extra_audio_delay_ms; |
| 245 | |
| 246 | send_time_->IncreaseTimeMs(1000); |
| 247 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 248 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 249 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 250 | &total_video_delay_ms)); |
| 251 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 252 | // The audio delay is not allowed to change more than the half of the |
| 253 | // required change in delay. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 254 | EXPECT_EQ(current_extra_delay_ms + |
| 255 | MaxAudioDelayIncrease( |
| 256 | current_audio_delay_ms, |
| 257 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 258 | extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 259 | current_audio_delay_ms = extra_audio_delay_ms; |
| 260 | current_extra_delay_ms = extra_audio_delay_ms; |
| 261 | |
| 262 | send_time_->IncreaseTimeMs(1000); |
| 263 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 264 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 265 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 266 | &total_video_delay_ms)); |
| 267 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 268 | // The audio delay is not allowed to change more than the half of the |
| 269 | // required change in delay. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 270 | EXPECT_EQ(current_extra_delay_ms + |
| 271 | MaxAudioDelayIncrease( |
| 272 | current_audio_delay_ms, |
| 273 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 274 | extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 275 | current_extra_delay_ms = extra_audio_delay_ms; |
| 276 | |
| 277 | // Simulate that NetEQ for some reason reduced the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 278 | current_audio_delay_ms = base_target_delay + 10; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 279 | send_time_->IncreaseTimeMs(1000); |
| 280 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 281 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 282 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 283 | &total_video_delay_ms)); |
| 284 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 285 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 286 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 287 | // here to try to stay in sync. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 288 | EXPECT_EQ(current_extra_delay_ms + |
| 289 | MaxAudioDelayIncrease( |
| 290 | current_audio_delay_ms, |
| 291 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 292 | extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 293 | current_extra_delay_ms = extra_audio_delay_ms; |
| 294 | |
| 295 | // Simulate that NetEQ for some reason significantly increased the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 296 | current_audio_delay_ms = base_target_delay + 350; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 297 | send_time_->IncreaseTimeMs(1000); |
| 298 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 299 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 300 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 301 | &total_video_delay_ms)); |
| 302 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 303 | // The audio delay is not allowed to change more than the half of the |
| 304 | // required change in delay. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 305 | EXPECT_EQ(current_extra_delay_ms + |
| 306 | MaxAudioDelayIncrease( |
| 307 | current_audio_delay_ms, |
| 308 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 309 | extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 310 | } |
| 311 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 312 | int MaxAudioDelayIncrease(int current_audio_delay_ms, int delay_ms) { |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 313 | return std::min((delay_ms - current_audio_delay_ms) / kSmoothingFilter, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 314 | static_cast<int>(kMaxAudioDiffMs)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 315 | } |
| 316 | |
| 317 | int MaxAudioDelayDecrease(int current_audio_delay_ms, int delay_ms) { |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 318 | return std::max((delay_ms - current_audio_delay_ms) / kSmoothingFilter, |
| 319 | -kMaxAudioDiffMs); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 320 | } |
| 321 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 322 | enum { kSendTimeOffsetMs = 98765 }; |
| 323 | enum { kReceiveTimeOffsetMs = 43210 }; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 324 | |
| 325 | StreamSynchronization* sync_; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 326 | Time* send_time_; // The simulated clock at the sender. |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 327 | Time* receive_time_; // The simulated clock at the receiver. |
| 328 | double audio_clock_drift_; |
| 329 | double video_clock_drift_; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 330 | }; |
| 331 | |
| 332 | TEST_F(StreamSynchronizationTest, NoDelay) { |
| 333 | uint32_t current_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 334 | int extra_audio_delay_ms = 0; |
| 335 | int total_video_delay_ms = 0; |
| 336 | |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 337 | EXPECT_FALSE(DelayedStreams(0, 0, current_audio_delay_ms, |
| 338 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 339 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 340 | EXPECT_EQ(0, total_video_delay_ms); |
| 341 | } |
| 342 | |
| 343 | TEST_F(StreamSynchronizationTest, VideoDelay) { |
| 344 | uint32_t current_audio_delay_ms = 0; |
| 345 | int delay_ms = 200; |
| 346 | int extra_audio_delay_ms = 0; |
| 347 | int total_video_delay_ms = 0; |
| 348 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 349 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 350 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 351 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 352 | // The video delay is not allowed to change more than this in 1 second. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 353 | EXPECT_EQ(delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 354 | |
| 355 | send_time_->IncreaseTimeMs(1000); |
| 356 | receive_time_->IncreaseTimeMs(800); |
| 357 | // Simulate 0 minimum delay in the VCM. |
| 358 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 359 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 360 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 361 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 362 | // The video delay is not allowed to change more than this in 1 second. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 363 | EXPECT_EQ(2 * delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 364 | |
| 365 | send_time_->IncreaseTimeMs(1000); |
| 366 | receive_time_->IncreaseTimeMs(800); |
| 367 | // Simulate 0 minimum delay in the VCM. |
| 368 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 369 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 370 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 371 | EXPECT_EQ(0, extra_audio_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 372 | EXPECT_EQ(3 * delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 373 | } |
| 374 | |
| 375 | TEST_F(StreamSynchronizationTest, AudioDelay) { |
| 376 | int current_audio_delay_ms = 0; |
| 377 | int delay_ms = 200; |
| 378 | int extra_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 379 | int total_video_delay_ms = 0; |
| 380 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 381 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 382 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 383 | EXPECT_EQ(0, total_video_delay_ms); |
| 384 | // The audio delay is not allowed to change more than this in 1 second. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 385 | EXPECT_EQ(delay_ms / kSmoothingFilter, extra_audio_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 386 | current_audio_delay_ms = extra_audio_delay_ms; |
andrew@webrtc.org | d7a71d0 | 2012-08-01 01:40:02 +0000 | [diff] [blame] | 387 | int current_extra_delay_ms = extra_audio_delay_ms; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 388 | |
| 389 | send_time_->IncreaseTimeMs(1000); |
| 390 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 391 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 392 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 393 | EXPECT_EQ(0, total_video_delay_ms); |
| 394 | // The audio delay is not allowed to change more than the half of the required |
| 395 | // change in delay. |
| 396 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 397 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 398 | extra_audio_delay_ms); |
| 399 | current_audio_delay_ms = extra_audio_delay_ms; |
| 400 | current_extra_delay_ms = extra_audio_delay_ms; |
| 401 | |
| 402 | send_time_->IncreaseTimeMs(1000); |
| 403 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 404 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 405 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 406 | EXPECT_EQ(0, total_video_delay_ms); |
| 407 | // The audio delay is not allowed to change more than the half of the required |
| 408 | // change in delay. |
| 409 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 410 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 411 | extra_audio_delay_ms); |
| 412 | current_extra_delay_ms = extra_audio_delay_ms; |
| 413 | |
| 414 | // Simulate that NetEQ for some reason reduced the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 415 | current_audio_delay_ms = 10; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 416 | send_time_->IncreaseTimeMs(1000); |
| 417 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 418 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 419 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 420 | EXPECT_EQ(0, total_video_delay_ms); |
| 421 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 422 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 423 | // here to try to |
| 424 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 425 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 426 | extra_audio_delay_ms); |
| 427 | current_extra_delay_ms = extra_audio_delay_ms; |
| 428 | |
| 429 | // Simulate that NetEQ for some reason significantly increased the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 430 | current_audio_delay_ms = 350; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 431 | send_time_->IncreaseTimeMs(1000); |
| 432 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 433 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 434 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 435 | EXPECT_EQ(0, total_video_delay_ms); |
| 436 | // The audio delay is not allowed to change more than the half of the required |
| 437 | // change in delay. |
| 438 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 439 | MaxAudioDelayDecrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 440 | extra_audio_delay_ms); |
| 441 | } |
| 442 | |
| 443 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) { |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 444 | BothDelayedVideoLaterTest(0); |
| 445 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 446 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 447 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterAudioClockDrift) { |
| 448 | audio_clock_drift_ = 1.05; |
| 449 | BothDelayedVideoLaterTest(0); |
| 450 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 451 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 452 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterVideoClockDrift) { |
| 453 | video_clock_drift_ = 1.05; |
| 454 | BothDelayedVideoLaterTest(0); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 455 | } |
| 456 | |
| 457 | TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 458 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 459 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 460 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 461 | TEST_F(StreamSynchronizationTest, BothDelayedAudioClockDrift) { |
| 462 | audio_clock_drift_ = 1.05; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 463 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 464 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 465 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 466 | TEST_F(StreamSynchronizationTest, BothDelayedVideoClockDrift) { |
| 467 | video_clock_drift_ = 1.05; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 468 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 469 | } |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 470 | |
| 471 | TEST_F(StreamSynchronizationTest, BaseDelay) { |
| 472 | int base_target_delay_ms = 2000; |
| 473 | int current_audio_delay_ms = 2000; |
| 474 | int extra_audio_delay_ms = 0; |
| 475 | int total_video_delay_ms = base_target_delay_ms; |
| 476 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 477 | // We are in sync don't change. |
| 478 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 479 | current_audio_delay_ms, &extra_audio_delay_ms, |
| 480 | &total_video_delay_ms)); |
mikhal@webrtc.org | 0d8d010 | 2013-02-22 19:30:44 +0000 | [diff] [blame] | 481 | // Triggering another call with the same values. Delay should not be modified. |
| 482 | base_target_delay_ms = 2000; |
| 483 | current_audio_delay_ms = base_target_delay_ms; |
| 484 | total_video_delay_ms = base_target_delay_ms; |
| 485 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 486 | // We are in sync don't change. |
| 487 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 488 | current_audio_delay_ms, &extra_audio_delay_ms, |
| 489 | &total_video_delay_ms)); |
mikhal@webrtc.org | 0d8d010 | 2013-02-22 19:30:44 +0000 | [diff] [blame] | 490 | // Changing delay value - intended to test this module only. In practice it |
| 491 | // would take VoE time to adapt. |
| 492 | base_target_delay_ms = 5000; |
| 493 | current_audio_delay_ms = base_target_delay_ms; |
| 494 | total_video_delay_ms = base_target_delay_ms; |
| 495 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 496 | // We are in sync don't change. |
| 497 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 498 | current_audio_delay_ms, &extra_audio_delay_ms, |
| 499 | &total_video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 500 | } |
| 501 | |
| 502 | TEST_F(StreamSynchronizationTest, BothDelayedAudioLaterWithBaseDelay) { |
| 503 | int base_target_delay_ms = 3000; |
| 504 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 505 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 506 | } |
| 507 | |
| 508 | TEST_F(StreamSynchronizationTest, BothDelayedAudioClockDriftWithBaseDelay) { |
| 509 | int base_target_delay_ms = 3000; |
| 510 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 511 | audio_clock_drift_ = 1.05; |
| 512 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 513 | } |
| 514 | |
| 515 | TEST_F(StreamSynchronizationTest, BothDelayedVideoClockDriftWithBaseDelay) { |
| 516 | int base_target_delay_ms = 3000; |
| 517 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 518 | video_clock_drift_ = 1.05; |
| 519 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 520 | } |
| 521 | |
| 522 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterWithBaseDelay) { |
| 523 | int base_target_delay_ms = 2000; |
| 524 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 525 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 526 | } |
| 527 | |
| 528 | TEST_F(StreamSynchronizationTest, |
| 529 | BothDelayedVideoLaterAudioClockDriftWithBaseDelay) { |
| 530 | int base_target_delay_ms = 2000; |
| 531 | audio_clock_drift_ = 1.05; |
| 532 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 533 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 534 | } |
| 535 | |
| 536 | TEST_F(StreamSynchronizationTest, |
| 537 | BothDelayedVideoLaterVideoClockDriftWithBaseDelay) { |
| 538 | int base_target_delay_ms = 2000; |
| 539 | video_clock_drift_ = 1.05; |
| 540 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 541 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 542 | } |
| 543 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 544 | } // namespace webrtc |