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 | |
Jonas Olsson | a4d8737 | 2019-07-05 19:08:33 +0200 | [diff] [blame] | 11 | #include "video/stream_synchronization.h" |
| 12 | |
kjellander@webrtc.org | 0fcaf99 | 2015-11-26 15:24:52 +0100 | [diff] [blame] | 13 | #include <algorithm> |
| 14 | |
Yves Gerey | 3e70781 | 2018-11-28 16:47:49 +0100 | [diff] [blame] | 15 | #include "system_wrappers/include/ntp_time.h" |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 16 | #include "test/gtest.h" |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 17 | |
| 18 | namespace webrtc { |
| 19 | |
| 20 | // These correspond to the same constants defined in vie_sync_module.cc. |
| 21 | enum { kMaxVideoDiffMs = 80 }; |
| 22 | enum { kMaxAudioDiffMs = 80 }; |
| 23 | enum { kMaxDelay = 1500 }; |
| 24 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 25 | // Test constants. |
| 26 | enum { kDefaultAudioFrequency = 8000 }; |
| 27 | enum { kDefaultVideoFrequency = 90000 }; |
| 28 | const double kNtpFracPerMs = 4.294967296E6; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 29 | static const int kSmoothingFilter = 4 * 2; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 30 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 31 | class Time { |
| 32 | public: |
| 33 | explicit Time(int64_t offset) |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 34 | : kNtpJan1970(2208988800UL), time_now_ms_(offset) {} |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 35 | |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 36 | NtpTime GetNowNtp() const { |
| 37 | uint32_t ntp_secs = time_now_ms_ / 1000 + kNtpJan1970; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 38 | int64_t remainder_ms = time_now_ms_ % 1000; |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 39 | uint32_t ntp_frac = static_cast<uint32_t>( |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 40 | static_cast<double>(remainder_ms) * kNtpFracPerMs + 0.5); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 41 | return NtpTime(ntp_secs, ntp_frac); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 42 | } |
| 43 | |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 44 | uint32_t GetNowRtp(int frequency, uint32_t offset) const { |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 45 | return frequency * time_now_ms_ / 1000 + offset; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 46 | } |
| 47 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 48 | void IncreaseTimeMs(int64_t inc) { time_now_ms_ += inc; } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 49 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 50 | int64_t time_now_ms() const { return time_now_ms_; } |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 51 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 52 | private: |
| 53 | // January 1970, in NTP seconds. |
| 54 | const uint32_t kNtpJan1970; |
| 55 | int64_t time_now_ms_; |
| 56 | }; |
| 57 | |
| 58 | class StreamSynchronizationTest : public ::testing::Test { |
| 59 | protected: |
| 60 | virtual void SetUp() { |
| 61 | sync_ = new StreamSynchronization(0, 0); |
| 62 | send_time_ = new Time(kSendTimeOffsetMs); |
| 63 | receive_time_ = new Time(kReceiveTimeOffsetMs); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 64 | audio_clock_drift_ = 1.0; |
| 65 | video_clock_drift_ = 1.0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 66 | } |
| 67 | |
| 68 | virtual void TearDown() { |
| 69 | delete sync_; |
| 70 | delete send_time_; |
| 71 | delete receive_time_; |
| 72 | } |
| 73 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 74 | // Generates the necessary RTCP measurements and RTP timestamps and computes |
| 75 | // the audio and video delays needed to get the two streams in sync. |
| 76 | // |audio_delay_ms| and |video_delay_ms| are the number of milliseconds after |
| 77 | // capture which the frames are rendered. |
| 78 | // |current_audio_delay_ms| is the number of milliseconds which audio is |
| 79 | // currently being delayed by the receiver. |
| 80 | bool DelayedStreams(int audio_delay_ms, |
| 81 | int video_delay_ms, |
| 82 | int current_audio_delay_ms, |
| 83 | int* extra_audio_delay_ms, |
| 84 | int* total_video_delay_ms) { |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 85 | int audio_frequency = |
| 86 | static_cast<int>(kDefaultAudioFrequency * audio_clock_drift_ + 0.5); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 87 | int audio_offset = 0; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 88 | int video_frequency = |
| 89 | static_cast<int>(kDefaultVideoFrequency * video_clock_drift_ + 0.5); |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 90 | bool new_sr; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 91 | int video_offset = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 92 | StreamSynchronization::Measurements audio; |
| 93 | StreamSynchronization::Measurements video; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 94 | // Generate NTP/RTP timestamp pair for both streams corresponding to RTCP. |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 95 | NtpTime ntp_time = send_time_->GetNowNtp(); |
| 96 | uint32_t rtp_timestamp = |
| 97 | send_time_->GetNowRtp(audio_frequency, audio_offset); |
| 98 | EXPECT_TRUE(audio.rtp_to_ntp.UpdateMeasurements( |
| 99 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 100 | send_time_->IncreaseTimeMs(100); |
| 101 | receive_time_->IncreaseTimeMs(100); |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 102 | ntp_time = send_time_->GetNowNtp(); |
| 103 | rtp_timestamp = send_time_->GetNowRtp(video_frequency, video_offset); |
| 104 | EXPECT_TRUE(video.rtp_to_ntp.UpdateMeasurements( |
| 105 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 106 | send_time_->IncreaseTimeMs(900); |
| 107 | receive_time_->IncreaseTimeMs(900); |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 108 | ntp_time = send_time_->GetNowNtp(); |
| 109 | rtp_timestamp = send_time_->GetNowRtp(audio_frequency, audio_offset); |
| 110 | EXPECT_TRUE(audio.rtp_to_ntp.UpdateMeasurements( |
| 111 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 112 | send_time_->IncreaseTimeMs(100); |
| 113 | receive_time_->IncreaseTimeMs(100); |
asapersson | fe50b4d | 2016-12-22 07:53:51 -0800 | [diff] [blame] | 114 | ntp_time = send_time_->GetNowNtp(); |
| 115 | rtp_timestamp = send_time_->GetNowRtp(video_frequency, video_offset); |
| 116 | EXPECT_TRUE(video.rtp_to_ntp.UpdateMeasurements( |
| 117 | ntp_time.seconds(), ntp_time.fractions(), rtp_timestamp, &new_sr)); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 118 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 119 | send_time_->IncreaseTimeMs(900); |
| 120 | receive_time_->IncreaseTimeMs(900); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 121 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 122 | // Capture an audio and a video frame at the same time. |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 123 | audio.latest_timestamp = |
| 124 | send_time_->GetNowRtp(audio_frequency, audio_offset); |
| 125 | video.latest_timestamp = |
| 126 | send_time_->GetNowRtp(video_frequency, video_offset); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 127 | |
| 128 | if (audio_delay_ms > video_delay_ms) { |
| 129 | // Audio later than video. |
| 130 | receive_time_->IncreaseTimeMs(video_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 131 | video.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 132 | receive_time_->IncreaseTimeMs(audio_delay_ms - video_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 133 | audio.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 134 | } else { |
| 135 | // Video later than audio. |
| 136 | receive_time_->IncreaseTimeMs(audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 137 | audio.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 138 | receive_time_->IncreaseTimeMs(video_delay_ms - audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 139 | video.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 140 | } |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 141 | int relative_delay_ms; |
| 142 | StreamSynchronization::ComputeRelativeDelay(audio, video, |
| 143 | &relative_delay_ms); |
| 144 | EXPECT_EQ(video_delay_ms - audio_delay_ms, relative_delay_ms); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 145 | return sync_->ComputeDelays(relative_delay_ms, current_audio_delay_ms, |
| 146 | extra_audio_delay_ms, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 147 | } |
| 148 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 149 | // Simulate audio playback 300 ms after capture and video rendering 100 ms |
| 150 | // after capture. Verify that the correct extra delays are calculated for |
| 151 | // audio and video, and that they change correctly when we simulate that |
| 152 | // NetEQ or the VCM adds more delay to the streams. |
| 153 | // TODO(holmer): This is currently wrong! We should simply change |
| 154 | // audio_delay_ms or video_delay_ms since those now include VCM and NetEQ |
| 155 | // delays. |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 156 | void BothDelayedAudioLaterTest(int base_target_delay) { |
| 157 | int current_audio_delay_ms = base_target_delay; |
| 158 | int audio_delay_ms = base_target_delay + 300; |
| 159 | int video_delay_ms = base_target_delay + 100; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 160 | int extra_audio_delay_ms = 0; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 161 | int total_video_delay_ms = base_target_delay; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 162 | int filtered_move = (audio_delay_ms - video_delay_ms) / kSmoothingFilter; |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 163 | const int kNeteqDelayIncrease = 50; |
| 164 | const int kNeteqDelayDecrease = 10; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 165 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 166 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 167 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 168 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 169 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 170 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 171 | current_audio_delay_ms = extra_audio_delay_ms; |
| 172 | |
| 173 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 174 | receive_time_->IncreaseTimeMs(1000 - |
| 175 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 176 | // Simulate base_target_delay minimum delay in the VCM. |
| 177 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 178 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 179 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 180 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 181 | 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] | 182 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 183 | current_audio_delay_ms = extra_audio_delay_ms; |
| 184 | |
| 185 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 186 | receive_time_->IncreaseTimeMs(1000 - |
| 187 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 188 | // Simulate base_target_delay minimum delay in the VCM. |
| 189 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 190 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 191 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 192 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 193 | 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] | 194 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 195 | |
| 196 | // Simulate that NetEQ introduces some audio delay. |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 197 | current_audio_delay_ms = base_target_delay + kNeteqDelayIncrease; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 198 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 199 | receive_time_->IncreaseTimeMs(1000 - |
| 200 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 201 | // Simulate base_target_delay minimum delay in the VCM. |
| 202 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 203 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 204 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 205 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 206 | filtered_move = 3 * filtered_move + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 207 | (kNeteqDelayIncrease + audio_delay_ms - video_delay_ms) / |
| 208 | kSmoothingFilter; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 209 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 210 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 211 | |
| 212 | // Simulate that NetEQ reduces its delay. |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 213 | current_audio_delay_ms = base_target_delay + kNeteqDelayDecrease; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 214 | send_time_->IncreaseTimeMs(1000); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 215 | receive_time_->IncreaseTimeMs(1000 - |
| 216 | std::max(audio_delay_ms, video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 217 | // Simulate base_target_delay minimum delay in the VCM. |
| 218 | total_video_delay_ms = base_target_delay; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 219 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 220 | current_audio_delay_ms, &extra_audio_delay_ms, |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 221 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 222 | |
| 223 | filtered_move = filtered_move + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 224 | (kNeteqDelayDecrease + audio_delay_ms - video_delay_ms) / |
| 225 | kSmoothingFilter; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 226 | |
| 227 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 228 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
| 229 | } |
| 230 | |
| 231 | void BothDelayedVideoLaterTest(int base_target_delay) { |
| 232 | int current_audio_delay_ms = base_target_delay; |
| 233 | int audio_delay_ms = base_target_delay + 100; |
| 234 | int video_delay_ms = base_target_delay + 300; |
| 235 | int extra_audio_delay_ms = 0; |
| 236 | int total_video_delay_ms = base_target_delay; |
| 237 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 238 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 239 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 240 | &total_video_delay_ms)); |
| 241 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 242 | // 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] | 243 | EXPECT_GE(base_target_delay + kMaxAudioDiffMs, extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 244 | current_audio_delay_ms = extra_audio_delay_ms; |
| 245 | int current_extra_delay_ms = extra_audio_delay_ms; |
| 246 | |
| 247 | send_time_->IncreaseTimeMs(1000); |
| 248 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 249 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 250 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 251 | &total_video_delay_ms)); |
| 252 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 253 | // The audio delay is not allowed to change more than the half of the |
| 254 | // required change in delay. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 255 | EXPECT_EQ(current_extra_delay_ms + |
| 256 | MaxAudioDelayIncrease( |
| 257 | current_audio_delay_ms, |
| 258 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 259 | extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 260 | current_audio_delay_ms = extra_audio_delay_ms; |
| 261 | current_extra_delay_ms = extra_audio_delay_ms; |
| 262 | |
| 263 | send_time_->IncreaseTimeMs(1000); |
| 264 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 265 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 266 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 267 | &total_video_delay_ms)); |
| 268 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 269 | // The audio delay is not allowed to change more than the half of the |
| 270 | // required change in delay. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 271 | EXPECT_EQ(current_extra_delay_ms + |
| 272 | MaxAudioDelayIncrease( |
| 273 | current_audio_delay_ms, |
| 274 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 275 | extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 276 | current_extra_delay_ms = extra_audio_delay_ms; |
| 277 | |
| 278 | // Simulate that NetEQ for some reason reduced the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 279 | current_audio_delay_ms = base_target_delay + 10; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 280 | send_time_->IncreaseTimeMs(1000); |
| 281 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 282 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 283 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 284 | &total_video_delay_ms)); |
| 285 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 286 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 287 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 288 | // here to try to stay in sync. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 289 | EXPECT_EQ(current_extra_delay_ms + |
| 290 | MaxAudioDelayIncrease( |
| 291 | current_audio_delay_ms, |
| 292 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 293 | extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 294 | current_extra_delay_ms = extra_audio_delay_ms; |
| 295 | |
| 296 | // Simulate that NetEQ for some reason significantly increased the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 297 | current_audio_delay_ms = base_target_delay + 350; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 298 | send_time_->IncreaseTimeMs(1000); |
| 299 | receive_time_->IncreaseTimeMs(800); |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 300 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms, |
| 301 | current_audio_delay_ms, &extra_audio_delay_ms, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 302 | &total_video_delay_ms)); |
| 303 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 304 | // The audio delay is not allowed to change more than the half of the |
| 305 | // required change in delay. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 306 | EXPECT_EQ(current_extra_delay_ms + |
| 307 | MaxAudioDelayIncrease( |
| 308 | current_audio_delay_ms, |
| 309 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 310 | extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 311 | } |
| 312 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 313 | int MaxAudioDelayIncrease(int current_audio_delay_ms, int delay_ms) { |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 314 | return std::min((delay_ms - current_audio_delay_ms) / kSmoothingFilter, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 315 | static_cast<int>(kMaxAudioDiffMs)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 316 | } |
| 317 | |
| 318 | int MaxAudioDelayDecrease(int current_audio_delay_ms, int delay_ms) { |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 319 | return std::max((delay_ms - current_audio_delay_ms) / kSmoothingFilter, |
| 320 | -kMaxAudioDiffMs); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 321 | } |
| 322 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 323 | enum { kSendTimeOffsetMs = 98765 }; |
| 324 | enum { kReceiveTimeOffsetMs = 43210 }; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 325 | |
| 326 | StreamSynchronization* sync_; |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 327 | Time* send_time_; // The simulated clock at the sender. |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 328 | Time* receive_time_; // The simulated clock at the receiver. |
| 329 | double audio_clock_drift_; |
| 330 | double video_clock_drift_; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 331 | }; |
| 332 | |
| 333 | TEST_F(StreamSynchronizationTest, NoDelay) { |
| 334 | uint32_t current_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 335 | int extra_audio_delay_ms = 0; |
| 336 | int total_video_delay_ms = 0; |
| 337 | |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 338 | EXPECT_FALSE(DelayedStreams(0, 0, current_audio_delay_ms, |
| 339 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 340 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 341 | EXPECT_EQ(0, total_video_delay_ms); |
| 342 | } |
| 343 | |
| 344 | TEST_F(StreamSynchronizationTest, VideoDelay) { |
| 345 | uint32_t current_audio_delay_ms = 0; |
| 346 | int delay_ms = 200; |
| 347 | int extra_audio_delay_ms = 0; |
| 348 | int total_video_delay_ms = 0; |
| 349 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 350 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 351 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 352 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 353 | // 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] | 354 | EXPECT_EQ(delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 355 | |
| 356 | send_time_->IncreaseTimeMs(1000); |
| 357 | receive_time_->IncreaseTimeMs(800); |
| 358 | // Simulate 0 minimum delay in the VCM. |
| 359 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 360 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 361 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 362 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 363 | // 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] | 364 | EXPECT_EQ(2 * delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 365 | |
| 366 | send_time_->IncreaseTimeMs(1000); |
| 367 | receive_time_->IncreaseTimeMs(800); |
| 368 | // Simulate 0 minimum delay in the VCM. |
| 369 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 370 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 371 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 372 | EXPECT_EQ(0, extra_audio_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 373 | EXPECT_EQ(3 * delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 374 | } |
| 375 | |
| 376 | TEST_F(StreamSynchronizationTest, AudioDelay) { |
| 377 | int current_audio_delay_ms = 0; |
| 378 | int delay_ms = 200; |
| 379 | int extra_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 380 | int total_video_delay_ms = 0; |
| 381 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 382 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 383 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 384 | EXPECT_EQ(0, total_video_delay_ms); |
| 385 | // 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] | 386 | EXPECT_EQ(delay_ms / kSmoothingFilter, extra_audio_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 387 | current_audio_delay_ms = extra_audio_delay_ms; |
andrew@webrtc.org | d7a71d0 | 2012-08-01 01:40:02 +0000 | [diff] [blame] | 388 | int current_extra_delay_ms = extra_audio_delay_ms; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 389 | |
| 390 | send_time_->IncreaseTimeMs(1000); |
| 391 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 392 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 393 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 394 | EXPECT_EQ(0, total_video_delay_ms); |
| 395 | // The audio delay is not allowed to change more than the half of the required |
| 396 | // change in delay. |
| 397 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 398 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 399 | extra_audio_delay_ms); |
| 400 | current_audio_delay_ms = extra_audio_delay_ms; |
| 401 | current_extra_delay_ms = extra_audio_delay_ms; |
| 402 | |
| 403 | send_time_->IncreaseTimeMs(1000); |
| 404 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 405 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 406 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 407 | EXPECT_EQ(0, total_video_delay_ms); |
| 408 | // The audio delay is not allowed to change more than the half of the required |
| 409 | // change in delay. |
| 410 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 411 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 412 | extra_audio_delay_ms); |
| 413 | current_extra_delay_ms = extra_audio_delay_ms; |
| 414 | |
| 415 | // Simulate that NetEQ for some reason reduced the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 416 | current_audio_delay_ms = 10; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 417 | send_time_->IncreaseTimeMs(1000); |
| 418 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 419 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 420 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 421 | EXPECT_EQ(0, total_video_delay_ms); |
| 422 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 423 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 424 | // here to try to |
| 425 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 426 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 427 | extra_audio_delay_ms); |
| 428 | current_extra_delay_ms = extra_audio_delay_ms; |
| 429 | |
| 430 | // Simulate that NetEQ for some reason significantly increased the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 431 | current_audio_delay_ms = 350; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 432 | send_time_->IncreaseTimeMs(1000); |
| 433 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 434 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 435 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 436 | EXPECT_EQ(0, total_video_delay_ms); |
| 437 | // The audio delay is not allowed to change more than the half of the required |
| 438 | // change in delay. |
| 439 | EXPECT_EQ(current_extra_delay_ms + |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 440 | MaxAudioDelayDecrease(current_audio_delay_ms, delay_ms), |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 441 | extra_audio_delay_ms); |
| 442 | } |
| 443 | |
| 444 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) { |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 445 | BothDelayedVideoLaterTest(0); |
| 446 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 447 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 448 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterAudioClockDrift) { |
| 449 | audio_clock_drift_ = 1.05; |
| 450 | BothDelayedVideoLaterTest(0); |
| 451 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 452 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 453 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterVideoClockDrift) { |
| 454 | video_clock_drift_ = 1.05; |
| 455 | BothDelayedVideoLaterTest(0); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 456 | } |
| 457 | |
| 458 | TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 459 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 460 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 461 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 462 | TEST_F(StreamSynchronizationTest, BothDelayedAudioClockDrift) { |
| 463 | audio_clock_drift_ = 1.05; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 464 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 465 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 466 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 467 | TEST_F(StreamSynchronizationTest, BothDelayedVideoClockDrift) { |
| 468 | video_clock_drift_ = 1.05; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 469 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 470 | } |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 471 | |
| 472 | TEST_F(StreamSynchronizationTest, BaseDelay) { |
| 473 | int base_target_delay_ms = 2000; |
| 474 | int current_audio_delay_ms = 2000; |
| 475 | int extra_audio_delay_ms = 0; |
| 476 | int total_video_delay_ms = base_target_delay_ms; |
| 477 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 478 | // We are in sync don't change. |
| 479 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 480 | current_audio_delay_ms, &extra_audio_delay_ms, |
| 481 | &total_video_delay_ms)); |
mikhal@webrtc.org | 0d8d010 | 2013-02-22 19:30:44 +0000 | [diff] [blame] | 482 | // Triggering another call with the same values. Delay should not be modified. |
| 483 | base_target_delay_ms = 2000; |
| 484 | current_audio_delay_ms = base_target_delay_ms; |
| 485 | total_video_delay_ms = base_target_delay_ms; |
| 486 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 487 | // We are in sync don't change. |
| 488 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 489 | current_audio_delay_ms, &extra_audio_delay_ms, |
| 490 | &total_video_delay_ms)); |
mikhal@webrtc.org | 0d8d010 | 2013-02-22 19:30:44 +0000 | [diff] [blame] | 491 | // Changing delay value - intended to test this module only. In practice it |
| 492 | // would take VoE time to adapt. |
| 493 | base_target_delay_ms = 5000; |
| 494 | current_audio_delay_ms = base_target_delay_ms; |
| 495 | total_video_delay_ms = base_target_delay_ms; |
| 496 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 497 | // We are in sync don't change. |
| 498 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame] | 499 | current_audio_delay_ms, &extra_audio_delay_ms, |
| 500 | &total_video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 501 | } |
| 502 | |
| 503 | TEST_F(StreamSynchronizationTest, BothDelayedAudioLaterWithBaseDelay) { |
| 504 | int base_target_delay_ms = 3000; |
| 505 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 506 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 507 | } |
| 508 | |
| 509 | TEST_F(StreamSynchronizationTest, BothDelayedAudioClockDriftWithBaseDelay) { |
| 510 | int base_target_delay_ms = 3000; |
| 511 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 512 | audio_clock_drift_ = 1.05; |
| 513 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 514 | } |
| 515 | |
| 516 | TEST_F(StreamSynchronizationTest, BothDelayedVideoClockDriftWithBaseDelay) { |
| 517 | int base_target_delay_ms = 3000; |
| 518 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 519 | video_clock_drift_ = 1.05; |
| 520 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 521 | } |
| 522 | |
| 523 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterWithBaseDelay) { |
| 524 | int base_target_delay_ms = 2000; |
| 525 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 526 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 527 | } |
| 528 | |
| 529 | TEST_F(StreamSynchronizationTest, |
| 530 | BothDelayedVideoLaterAudioClockDriftWithBaseDelay) { |
| 531 | int base_target_delay_ms = 2000; |
| 532 | audio_clock_drift_ = 1.05; |
| 533 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 534 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 535 | } |
| 536 | |
| 537 | TEST_F(StreamSynchronizationTest, |
| 538 | BothDelayedVideoLaterVideoClockDriftWithBaseDelay) { |
| 539 | int base_target_delay_ms = 2000; |
| 540 | video_clock_drift_ = 1.05; |
| 541 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 542 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 543 | } |
| 544 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 545 | } // namespace webrtc |