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 | |
pbos@webrtc.org | f5d4cb1 | 2013-05-17 13:44:48 +0000 | [diff] [blame] | 11 | #include <math.h> |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 12 | |
kjellander@webrtc.org | 0fcaf99 | 2015-11-26 15:24:52 +0100 | [diff] [blame] | 13 | #include <algorithm> |
| 14 | |
kwiberg | 77eab70 | 2016-09-28 17:42:01 -0700 | [diff] [blame] | 15 | #include "webrtc/test/gtest.h" |
Peter Boström | 7623ce4 | 2015-12-09 12:13:30 +0100 | [diff] [blame] | 16 | #include "webrtc/video/stream_synchronization.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) |
| 34 | : kNtpJan1970(2208988800UL), |
| 35 | time_now_ms_(offset) {} |
| 36 | |
wu@webrtc.org | 66773a0 | 2014-05-07 17:09:44 +0000 | [diff] [blame] | 37 | RtcpMeasurement GenerateRtcp(int frequency, uint32_t offset) const { |
| 38 | RtcpMeasurement rtcp; |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 39 | rtcp.ntp_time = GetNowNtp(); |
| 40 | rtcp.rtp_timestamp = GetNowRtp(frequency, offset); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 41 | return rtcp; |
| 42 | } |
| 43 | |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 44 | NtpTime GetNowNtp() const { |
| 45 | uint32_t ntp_secs = time_now_ms_ / 1000 + kNtpJan1970; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 46 | int64_t remainder_ms = time_now_ms_ % 1000; |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 47 | uint32_t ntp_frac = static_cast<uint32_t>( |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 48 | static_cast<double>(remainder_ms) * kNtpFracPerMs + 0.5); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 49 | return NtpTime(ntp_secs, ntp_frac); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 50 | } |
| 51 | |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 52 | uint32_t GetNowRtp(int frequency, uint32_t offset) const { |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 53 | return frequency * time_now_ms_ / 1000 + offset; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 54 | } |
| 55 | |
| 56 | void IncreaseTimeMs(int64_t inc) { |
| 57 | time_now_ms_ += inc; |
| 58 | } |
| 59 | |
| 60 | int64_t time_now_ms() const { |
| 61 | return time_now_ms_; |
| 62 | } |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 63 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 64 | private: |
| 65 | // January 1970, in NTP seconds. |
| 66 | const uint32_t kNtpJan1970; |
| 67 | int64_t time_now_ms_; |
| 68 | }; |
| 69 | |
| 70 | class StreamSynchronizationTest : public ::testing::Test { |
| 71 | protected: |
| 72 | virtual void SetUp() { |
| 73 | sync_ = new StreamSynchronization(0, 0); |
| 74 | send_time_ = new Time(kSendTimeOffsetMs); |
| 75 | receive_time_ = new Time(kReceiveTimeOffsetMs); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 76 | audio_clock_drift_ = 1.0; |
| 77 | video_clock_drift_ = 1.0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 78 | } |
| 79 | |
| 80 | virtual void TearDown() { |
| 81 | delete sync_; |
| 82 | delete send_time_; |
| 83 | delete receive_time_; |
| 84 | } |
| 85 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 86 | // Generates the necessary RTCP measurements and RTP timestamps and computes |
| 87 | // the audio and video delays needed to get the two streams in sync. |
| 88 | // |audio_delay_ms| and |video_delay_ms| are the number of milliseconds after |
| 89 | // capture which the frames are rendered. |
| 90 | // |current_audio_delay_ms| is the number of milliseconds which audio is |
| 91 | // currently being delayed by the receiver. |
| 92 | bool DelayedStreams(int audio_delay_ms, |
| 93 | int video_delay_ms, |
| 94 | int current_audio_delay_ms, |
| 95 | int* extra_audio_delay_ms, |
| 96 | int* total_video_delay_ms) { |
| 97 | int audio_frequency = static_cast<int>(kDefaultAudioFrequency * |
| 98 | audio_clock_drift_ + 0.5); |
| 99 | int audio_offset = 0; |
| 100 | int video_frequency = static_cast<int>(kDefaultVideoFrequency * |
| 101 | video_clock_drift_ + 0.5); |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 102 | bool new_sr; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 103 | int video_offset = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 104 | StreamSynchronization::Measurements audio; |
| 105 | StreamSynchronization::Measurements video; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 106 | // Generate NTP/RTP timestamp pair for both streams corresponding to RTCP. |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 107 | RtcpMeasurement rtcp = |
| 108 | send_time_->GenerateRtcp(audio_frequency, audio_offset); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 109 | EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(), |
| 110 | rtcp.ntp_time.fractions(), rtcp.rtp_timestamp, |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 111 | &audio.rtcp, &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 | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 114 | rtcp = send_time_->GenerateRtcp(video_frequency, video_offset); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 115 | EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(), |
| 116 | rtcp.ntp_time.fractions(), rtcp.rtp_timestamp, |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 117 | &video.rtcp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 118 | send_time_->IncreaseTimeMs(900); |
| 119 | receive_time_->IncreaseTimeMs(900); |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 120 | rtcp = send_time_->GenerateRtcp(audio_frequency, audio_offset); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 121 | EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(), |
| 122 | rtcp.ntp_time.fractions(), rtcp.rtp_timestamp, |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 123 | &audio.rtcp, &new_sr)); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 124 | send_time_->IncreaseTimeMs(100); |
| 125 | receive_time_->IncreaseTimeMs(100); |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 126 | rtcp = send_time_->GenerateRtcp(video_frequency, video_offset); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 127 | EXPECT_TRUE(UpdateRtcpList(rtcp.ntp_time.seconds(), |
| 128 | rtcp.ntp_time.fractions(), rtcp.rtp_timestamp, |
asapersson | de9e5ff | 2016-11-02 07:14:03 -0700 | [diff] [blame] | 129 | &video.rtcp, &new_sr)); |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 130 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 131 | send_time_->IncreaseTimeMs(900); |
| 132 | receive_time_->IncreaseTimeMs(900); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 133 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 134 | // Capture an audio and a video frame at the same time. |
asapersson | b7e7b49 | 2016-11-17 02:27:14 -0800 | [diff] [blame] | 135 | audio.latest_timestamp = |
| 136 | send_time_->GetNowRtp(audio_frequency, audio_offset); |
| 137 | video.latest_timestamp = |
| 138 | send_time_->GetNowRtp(video_frequency, video_offset); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 139 | |
| 140 | if (audio_delay_ms > video_delay_ms) { |
| 141 | // Audio later than video. |
| 142 | receive_time_->IncreaseTimeMs(video_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 143 | video.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 144 | receive_time_->IncreaseTimeMs(audio_delay_ms - video_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 145 | audio.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 146 | } else { |
| 147 | // Video later than audio. |
| 148 | receive_time_->IncreaseTimeMs(audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 149 | audio.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 150 | receive_time_->IncreaseTimeMs(video_delay_ms - audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 151 | video.latest_receive_time_ms = receive_time_->time_now_ms(); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 152 | } |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 153 | int relative_delay_ms; |
| 154 | StreamSynchronization::ComputeRelativeDelay(audio, video, |
| 155 | &relative_delay_ms); |
| 156 | EXPECT_EQ(video_delay_ms - audio_delay_ms, relative_delay_ms); |
| 157 | return sync_->ComputeDelays(relative_delay_ms, |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 158 | current_audio_delay_ms, |
| 159 | extra_audio_delay_ms, |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 160 | total_video_delay_ms); |
| 161 | } |
| 162 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 163 | // Simulate audio playback 300 ms after capture and video rendering 100 ms |
| 164 | // after capture. Verify that the correct extra delays are calculated for |
| 165 | // audio and video, and that they change correctly when we simulate that |
| 166 | // NetEQ or the VCM adds more delay to the streams. |
| 167 | // TODO(holmer): This is currently wrong! We should simply change |
| 168 | // audio_delay_ms or video_delay_ms since those now include VCM and NetEQ |
| 169 | // delays. |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 170 | void BothDelayedAudioLaterTest(int base_target_delay) { |
| 171 | int current_audio_delay_ms = base_target_delay; |
| 172 | int audio_delay_ms = base_target_delay + 300; |
| 173 | int video_delay_ms = base_target_delay + 100; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 174 | int extra_audio_delay_ms = 0; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 175 | int total_video_delay_ms = base_target_delay; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 176 | int filtered_move = (audio_delay_ms - video_delay_ms) / kSmoothingFilter; |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 177 | const int kNeteqDelayIncrease = 50; |
| 178 | const int kNeteqDelayDecrease = 10; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 179 | |
| 180 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 181 | video_delay_ms, |
| 182 | current_audio_delay_ms, |
| 183 | &extra_audio_delay_ms, |
| 184 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 185 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 186 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 187 | current_audio_delay_ms = extra_audio_delay_ms; |
| 188 | |
| 189 | send_time_->IncreaseTimeMs(1000); |
| 190 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 191 | video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 192 | // Simulate base_target_delay minimum delay in the VCM. |
| 193 | total_video_delay_ms = base_target_delay; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 194 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 195 | video_delay_ms, |
| 196 | current_audio_delay_ms, |
| 197 | &extra_audio_delay_ms, |
| 198 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 199 | 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] | 200 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 201 | current_audio_delay_ms = extra_audio_delay_ms; |
| 202 | |
| 203 | send_time_->IncreaseTimeMs(1000); |
| 204 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 205 | video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 206 | // Simulate base_target_delay minimum delay in the VCM. |
| 207 | total_video_delay_ms = base_target_delay; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 208 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 209 | video_delay_ms, |
| 210 | current_audio_delay_ms, |
| 211 | &extra_audio_delay_ms, |
| 212 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 213 | 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] | 214 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 215 | |
| 216 | // Simulate that NetEQ introduces some audio delay. |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 217 | current_audio_delay_ms = base_target_delay + kNeteqDelayIncrease; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 218 | send_time_->IncreaseTimeMs(1000); |
| 219 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 220 | video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 221 | // Simulate base_target_delay minimum delay in the VCM. |
| 222 | total_video_delay_ms = base_target_delay; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 223 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 224 | video_delay_ms, |
| 225 | current_audio_delay_ms, |
| 226 | &extra_audio_delay_ms, |
| 227 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 228 | filtered_move = 3 * filtered_move + |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 229 | (kNeteqDelayIncrease + audio_delay_ms - video_delay_ms) / |
| 230 | kSmoothingFilter; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 231 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 232 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 233 | |
| 234 | // Simulate that NetEQ reduces its delay. |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 235 | current_audio_delay_ms = base_target_delay + kNeteqDelayDecrease; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 236 | send_time_->IncreaseTimeMs(1000); |
| 237 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 238 | video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 239 | // Simulate base_target_delay minimum delay in the VCM. |
| 240 | total_video_delay_ms = base_target_delay; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 241 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 242 | video_delay_ms, |
| 243 | current_audio_delay_ms, |
| 244 | &extra_audio_delay_ms, |
| 245 | &total_video_delay_ms)); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 246 | |
| 247 | filtered_move = filtered_move + |
pwestin@webrtc.org | 4e545b3 | 2013-04-26 15:23:34 +0000 | [diff] [blame] | 248 | (kNeteqDelayDecrease + audio_delay_ms - video_delay_ms) / |
| 249 | kSmoothingFilter; |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 250 | |
| 251 | EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 252 | EXPECT_EQ(base_target_delay, extra_audio_delay_ms); |
| 253 | } |
| 254 | |
| 255 | void BothDelayedVideoLaterTest(int base_target_delay) { |
| 256 | int current_audio_delay_ms = base_target_delay; |
| 257 | int audio_delay_ms = base_target_delay + 100; |
| 258 | int video_delay_ms = base_target_delay + 300; |
| 259 | int extra_audio_delay_ms = 0; |
| 260 | int total_video_delay_ms = base_target_delay; |
| 261 | |
| 262 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 263 | video_delay_ms, |
| 264 | current_audio_delay_ms, |
| 265 | &extra_audio_delay_ms, |
| 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 this in 1 second. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 269 | EXPECT_GE(base_target_delay + kMaxAudioDiffMs, extra_audio_delay_ms); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 270 | current_audio_delay_ms = extra_audio_delay_ms; |
| 271 | int current_extra_delay_ms = extra_audio_delay_ms; |
| 272 | |
| 273 | send_time_->IncreaseTimeMs(1000); |
| 274 | receive_time_->IncreaseTimeMs(800); |
| 275 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 276 | video_delay_ms, |
| 277 | current_audio_delay_ms, |
| 278 | &extra_audio_delay_ms, |
| 279 | &total_video_delay_ms)); |
| 280 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 281 | // The audio delay is not allowed to change more than the half of the |
| 282 | // required change in delay. |
| 283 | EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease( |
| 284 | current_audio_delay_ms, |
| 285 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 286 | extra_audio_delay_ms); |
| 287 | current_audio_delay_ms = extra_audio_delay_ms; |
| 288 | current_extra_delay_ms = extra_audio_delay_ms; |
| 289 | |
| 290 | send_time_->IncreaseTimeMs(1000); |
| 291 | receive_time_->IncreaseTimeMs(800); |
| 292 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 293 | video_delay_ms, |
| 294 | current_audio_delay_ms, |
| 295 | &extra_audio_delay_ms, |
| 296 | &total_video_delay_ms)); |
| 297 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 298 | // The audio delay is not allowed to change more than the half of the |
| 299 | // required change in delay. |
| 300 | EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease( |
| 301 | current_audio_delay_ms, |
| 302 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 303 | extra_audio_delay_ms); |
| 304 | current_extra_delay_ms = extra_audio_delay_ms; |
| 305 | |
| 306 | // Simulate that NetEQ for some reason reduced the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 307 | current_audio_delay_ms = base_target_delay + 10; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 308 | send_time_->IncreaseTimeMs(1000); |
| 309 | receive_time_->IncreaseTimeMs(800); |
| 310 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 311 | video_delay_ms, |
| 312 | current_audio_delay_ms, |
| 313 | &extra_audio_delay_ms, |
| 314 | &total_video_delay_ms)); |
| 315 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 316 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 317 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 318 | // here to try to stay in sync. |
| 319 | EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease( |
| 320 | current_audio_delay_ms, |
| 321 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 322 | extra_audio_delay_ms); |
| 323 | current_extra_delay_ms = extra_audio_delay_ms; |
| 324 | |
| 325 | // Simulate that NetEQ for some reason significantly increased the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 326 | current_audio_delay_ms = base_target_delay + 350; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 327 | send_time_->IncreaseTimeMs(1000); |
| 328 | receive_time_->IncreaseTimeMs(800); |
| 329 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 330 | video_delay_ms, |
| 331 | current_audio_delay_ms, |
| 332 | &extra_audio_delay_ms, |
| 333 | &total_video_delay_ms)); |
| 334 | EXPECT_EQ(base_target_delay, total_video_delay_ms); |
| 335 | // The audio delay is not allowed to change more than the half of the |
| 336 | // required change in delay. |
| 337 | EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease( |
| 338 | current_audio_delay_ms, |
| 339 | base_target_delay + video_delay_ms - audio_delay_ms), |
| 340 | extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 341 | } |
| 342 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 343 | int MaxAudioDelayIncrease(int current_audio_delay_ms, int delay_ms) { |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 344 | return std::min((delay_ms - current_audio_delay_ms) / kSmoothingFilter, |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 345 | static_cast<int>(kMaxAudioDiffMs)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 346 | } |
| 347 | |
| 348 | int MaxAudioDelayDecrease(int current_audio_delay_ms, int delay_ms) { |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 349 | return std::max((delay_ms - current_audio_delay_ms) / kSmoothingFilter, |
| 350 | -kMaxAudioDiffMs); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 351 | } |
| 352 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 353 | enum { kSendTimeOffsetMs = 98765 }; |
| 354 | enum { kReceiveTimeOffsetMs = 43210 }; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 355 | |
| 356 | StreamSynchronization* sync_; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 357 | Time* send_time_; // The simulated clock at the sender. |
| 358 | Time* receive_time_; // The simulated clock at the receiver. |
| 359 | double audio_clock_drift_; |
| 360 | double video_clock_drift_; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 361 | }; |
| 362 | |
| 363 | TEST_F(StreamSynchronizationTest, NoDelay) { |
| 364 | uint32_t current_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 365 | int extra_audio_delay_ms = 0; |
| 366 | int total_video_delay_ms = 0; |
| 367 | |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 368 | EXPECT_FALSE(DelayedStreams(0, 0, current_audio_delay_ms, |
| 369 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 370 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 371 | EXPECT_EQ(0, total_video_delay_ms); |
| 372 | } |
| 373 | |
| 374 | TEST_F(StreamSynchronizationTest, VideoDelay) { |
| 375 | uint32_t current_audio_delay_ms = 0; |
| 376 | int delay_ms = 200; |
| 377 | int extra_audio_delay_ms = 0; |
| 378 | int total_video_delay_ms = 0; |
| 379 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 380 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 381 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 382 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 383 | // 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] | 384 | EXPECT_EQ(delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 385 | |
| 386 | send_time_->IncreaseTimeMs(1000); |
| 387 | receive_time_->IncreaseTimeMs(800); |
| 388 | // Simulate 0 minimum delay in the VCM. |
| 389 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 390 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 391 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 392 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 393 | // 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] | 394 | EXPECT_EQ(2 * delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 395 | |
| 396 | send_time_->IncreaseTimeMs(1000); |
| 397 | receive_time_->IncreaseTimeMs(800); |
| 398 | // Simulate 0 minimum delay in the VCM. |
| 399 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 400 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 401 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 402 | EXPECT_EQ(0, extra_audio_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 403 | EXPECT_EQ(3 * delay_ms / kSmoothingFilter, total_video_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 404 | } |
| 405 | |
| 406 | TEST_F(StreamSynchronizationTest, AudioDelay) { |
| 407 | int current_audio_delay_ms = 0; |
| 408 | int delay_ms = 200; |
| 409 | int extra_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 410 | int total_video_delay_ms = 0; |
| 411 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 412 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 413 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 414 | EXPECT_EQ(0, total_video_delay_ms); |
| 415 | // 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] | 416 | EXPECT_EQ(delay_ms / kSmoothingFilter, extra_audio_delay_ms); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 417 | current_audio_delay_ms = extra_audio_delay_ms; |
andrew@webrtc.org | d7a71d0 | 2012-08-01 01:40:02 +0000 | [diff] [blame] | 418 | int current_extra_delay_ms = extra_audio_delay_ms; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 419 | |
| 420 | send_time_->IncreaseTimeMs(1000); |
| 421 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 422 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 423 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 424 | EXPECT_EQ(0, total_video_delay_ms); |
| 425 | // The audio delay is not allowed to change more than the half of the required |
| 426 | // change in delay. |
| 427 | EXPECT_EQ(current_extra_delay_ms + |
| 428 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
| 429 | extra_audio_delay_ms); |
| 430 | current_audio_delay_ms = extra_audio_delay_ms; |
| 431 | current_extra_delay_ms = extra_audio_delay_ms; |
| 432 | |
| 433 | send_time_->IncreaseTimeMs(1000); |
| 434 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 435 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 436 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 437 | EXPECT_EQ(0, total_video_delay_ms); |
| 438 | // The audio delay is not allowed to change more than the half of the required |
| 439 | // change in delay. |
| 440 | EXPECT_EQ(current_extra_delay_ms + |
| 441 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
| 442 | extra_audio_delay_ms); |
| 443 | current_extra_delay_ms = extra_audio_delay_ms; |
| 444 | |
| 445 | // Simulate that NetEQ for some reason reduced the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 446 | current_audio_delay_ms = 10; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 447 | send_time_->IncreaseTimeMs(1000); |
| 448 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 449 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 450 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 451 | EXPECT_EQ(0, total_video_delay_ms); |
| 452 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 453 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 454 | // here to try to |
| 455 | EXPECT_EQ(current_extra_delay_ms + |
| 456 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
| 457 | extra_audio_delay_ms); |
| 458 | current_extra_delay_ms = extra_audio_delay_ms; |
| 459 | |
| 460 | // Simulate that NetEQ for some reason significantly increased the delay. |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 461 | current_audio_delay_ms = 350; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 462 | send_time_->IncreaseTimeMs(1000); |
| 463 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 464 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 465 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 466 | EXPECT_EQ(0, total_video_delay_ms); |
| 467 | // The audio delay is not allowed to change more than the half of the required |
| 468 | // change in delay. |
| 469 | EXPECT_EQ(current_extra_delay_ms + |
| 470 | MaxAudioDelayDecrease(current_audio_delay_ms, delay_ms), |
| 471 | extra_audio_delay_ms); |
| 472 | } |
| 473 | |
| 474 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) { |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 475 | BothDelayedVideoLaterTest(0); |
| 476 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 477 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 478 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterAudioClockDrift) { |
| 479 | audio_clock_drift_ = 1.05; |
| 480 | BothDelayedVideoLaterTest(0); |
| 481 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 482 | |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 483 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterVideoClockDrift) { |
| 484 | video_clock_drift_ = 1.05; |
| 485 | BothDelayedVideoLaterTest(0); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 486 | } |
| 487 | |
| 488 | TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 489 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 490 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 491 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 492 | TEST_F(StreamSynchronizationTest, BothDelayedAudioClockDrift) { |
| 493 | audio_clock_drift_ = 1.05; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 494 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 495 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 496 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 497 | TEST_F(StreamSynchronizationTest, BothDelayedVideoClockDrift) { |
| 498 | video_clock_drift_ = 1.05; |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 499 | BothDelayedAudioLaterTest(0); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 500 | } |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 501 | |
| 502 | TEST_F(StreamSynchronizationTest, BaseDelay) { |
| 503 | int base_target_delay_ms = 2000; |
| 504 | int current_audio_delay_ms = 2000; |
| 505 | int extra_audio_delay_ms = 0; |
| 506 | int total_video_delay_ms = base_target_delay_ms; |
| 507 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 508 | // We are in sync don't change. |
| 509 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
| 510 | current_audio_delay_ms, |
| 511 | &extra_audio_delay_ms, &total_video_delay_ms)); |
mikhal@webrtc.org | 0d8d010 | 2013-02-22 19:30:44 +0000 | [diff] [blame] | 512 | // Triggering another call with the same values. Delay should not be modified. |
| 513 | base_target_delay_ms = 2000; |
| 514 | current_audio_delay_ms = base_target_delay_ms; |
| 515 | total_video_delay_ms = base_target_delay_ms; |
| 516 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 517 | // We are in sync don't change. |
| 518 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
| 519 | current_audio_delay_ms, |
| 520 | &extra_audio_delay_ms, &total_video_delay_ms)); |
mikhal@webrtc.org | 0d8d010 | 2013-02-22 19:30:44 +0000 | [diff] [blame] | 521 | // Changing delay value - intended to test this module only. In practice it |
| 522 | // would take VoE time to adapt. |
| 523 | base_target_delay_ms = 5000; |
| 524 | current_audio_delay_ms = base_target_delay_ms; |
| 525 | total_video_delay_ms = base_target_delay_ms; |
| 526 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
pwestin@webrtc.org | 6311733 | 2013-04-22 18:57:14 +0000 | [diff] [blame] | 527 | // We are in sync don't change. |
| 528 | EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms, |
| 529 | current_audio_delay_ms, |
| 530 | &extra_audio_delay_ms, &total_video_delay_ms)); |
mikhal@webrtc.org | ef9f76a | 2013-02-15 23:22:18 +0000 | [diff] [blame] | 531 | } |
| 532 | |
| 533 | TEST_F(StreamSynchronizationTest, BothDelayedAudioLaterWithBaseDelay) { |
| 534 | int base_target_delay_ms = 3000; |
| 535 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 536 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 537 | } |
| 538 | |
| 539 | TEST_F(StreamSynchronizationTest, BothDelayedAudioClockDriftWithBaseDelay) { |
| 540 | int base_target_delay_ms = 3000; |
| 541 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 542 | audio_clock_drift_ = 1.05; |
| 543 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 544 | } |
| 545 | |
| 546 | TEST_F(StreamSynchronizationTest, BothDelayedVideoClockDriftWithBaseDelay) { |
| 547 | int base_target_delay_ms = 3000; |
| 548 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 549 | video_clock_drift_ = 1.05; |
| 550 | BothDelayedAudioLaterTest(base_target_delay_ms); |
| 551 | } |
| 552 | |
| 553 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLaterWithBaseDelay) { |
| 554 | int base_target_delay_ms = 2000; |
| 555 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 556 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 557 | } |
| 558 | |
| 559 | TEST_F(StreamSynchronizationTest, |
| 560 | BothDelayedVideoLaterAudioClockDriftWithBaseDelay) { |
| 561 | int base_target_delay_ms = 2000; |
| 562 | audio_clock_drift_ = 1.05; |
| 563 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 564 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 565 | } |
| 566 | |
| 567 | TEST_F(StreamSynchronizationTest, |
| 568 | BothDelayedVideoLaterVideoClockDriftWithBaseDelay) { |
| 569 | int base_target_delay_ms = 2000; |
| 570 | video_clock_drift_ = 1.05; |
| 571 | sync_->SetTargetBufferingDelay(base_target_delay_ms); |
| 572 | BothDelayedVideoLaterTest(base_target_delay_ms); |
| 573 | } |
| 574 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 575 | } // namespace webrtc |