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 | |
| 11 | #include <math.h> |
| 12 | #include <algorithm> |
| 13 | |
| 14 | #include "gtest/gtest.h" |
| 15 | #include "video_engine/stream_synchronization.h" |
| 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; |
| 28 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 29 | class Time { |
| 30 | public: |
| 31 | explicit Time(int64_t offset) |
| 32 | : kNtpJan1970(2208988800UL), |
| 33 | time_now_ms_(offset) {} |
| 34 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 35 | synchronization::RtcpMeasurement GenerateRtcp(int frequency, |
| 36 | uint32_t offset) const { |
| 37 | synchronization::RtcpMeasurement rtcp; |
| 38 | NowNtp(&rtcp.ntp_secs, &rtcp.ntp_frac); |
| 39 | rtcp.rtp_timestamp = NowRtp(frequency, offset); |
| 40 | return rtcp; |
| 41 | } |
| 42 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 43 | void NowNtp(uint32_t* ntp_secs, uint32_t* ntp_frac) const { |
| 44 | *ntp_secs = time_now_ms_ / 1000 + kNtpJan1970; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 45 | int64_t remainder_ms = time_now_ms_ % 1000; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 46 | *ntp_frac = static_cast<uint32_t>( |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 47 | static_cast<double>(remainder_ms) * kNtpFracPerMs + 0.5); |
| 48 | } |
| 49 | |
| 50 | uint32_t NowRtp(int frequency, uint32_t offset) const { |
| 51 | return frequency * time_now_ms_ / 1000 + offset; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 52 | } |
| 53 | |
| 54 | void IncreaseTimeMs(int64_t inc) { |
| 55 | time_now_ms_ += inc; |
| 56 | } |
| 57 | |
| 58 | int64_t time_now_ms() const { |
| 59 | return time_now_ms_; |
| 60 | } |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 61 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 62 | private: |
| 63 | // January 1970, in NTP seconds. |
| 64 | const uint32_t kNtpJan1970; |
| 65 | int64_t time_now_ms_; |
| 66 | }; |
| 67 | |
| 68 | class StreamSynchronizationTest : public ::testing::Test { |
| 69 | protected: |
| 70 | virtual void SetUp() { |
| 71 | sync_ = new StreamSynchronization(0, 0); |
| 72 | send_time_ = new Time(kSendTimeOffsetMs); |
| 73 | receive_time_ = new Time(kReceiveTimeOffsetMs); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 74 | audio_clock_drift_ = 1.0; |
| 75 | video_clock_drift_ = 1.0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 76 | } |
| 77 | |
| 78 | virtual void TearDown() { |
| 79 | delete sync_; |
| 80 | delete send_time_; |
| 81 | delete receive_time_; |
| 82 | } |
| 83 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 84 | // Generates the necessary RTCP measurements and RTP timestamps and computes |
| 85 | // the audio and video delays needed to get the two streams in sync. |
| 86 | // |audio_delay_ms| and |video_delay_ms| are the number of milliseconds after |
| 87 | // capture which the frames are rendered. |
| 88 | // |current_audio_delay_ms| is the number of milliseconds which audio is |
| 89 | // currently being delayed by the receiver. |
| 90 | bool DelayedStreams(int audio_delay_ms, |
| 91 | int video_delay_ms, |
| 92 | int current_audio_delay_ms, |
| 93 | int* extra_audio_delay_ms, |
| 94 | int* total_video_delay_ms) { |
| 95 | int audio_frequency = static_cast<int>(kDefaultAudioFrequency * |
| 96 | audio_clock_drift_ + 0.5); |
| 97 | int audio_offset = 0; |
| 98 | int video_frequency = static_cast<int>(kDefaultVideoFrequency * |
| 99 | video_clock_drift_ + 0.5); |
| 100 | int video_offset = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 101 | StreamSynchronization::Measurements audio; |
| 102 | StreamSynchronization::Measurements video; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 103 | // Generate NTP/RTP timestamp pair for both streams corresponding to RTCP. |
| 104 | audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency, |
| 105 | audio_offset)); |
| 106 | send_time_->IncreaseTimeMs(100); |
| 107 | receive_time_->IncreaseTimeMs(100); |
| 108 | video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency, |
| 109 | video_offset)); |
| 110 | send_time_->IncreaseTimeMs(900); |
| 111 | receive_time_->IncreaseTimeMs(900); |
| 112 | audio.rtcp.push_front(send_time_->GenerateRtcp(audio_frequency, |
| 113 | audio_offset)); |
| 114 | send_time_->IncreaseTimeMs(100); |
| 115 | receive_time_->IncreaseTimeMs(100); |
| 116 | video.rtcp.push_front(send_time_->GenerateRtcp(video_frequency, |
| 117 | video_offset)); |
| 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. |
| 122 | audio.latest_timestamp = send_time_->NowRtp(audio_frequency, |
| 123 | audio_offset); |
| 124 | video.latest_timestamp = send_time_->NowRtp(video_frequency, |
| 125 | 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); |
| 144 | return sync_->ComputeDelays(relative_delay_ms, |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 145 | current_audio_delay_ms, |
| 146 | extra_audio_delay_ms, |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 147 | total_video_delay_ms); |
| 148 | } |
| 149 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 150 | // Simulate audio playback 300 ms after capture and video rendering 100 ms |
| 151 | // after capture. Verify that the correct extra delays are calculated for |
| 152 | // audio and video, and that they change correctly when we simulate that |
| 153 | // NetEQ or the VCM adds more delay to the streams. |
| 154 | // TODO(holmer): This is currently wrong! We should simply change |
| 155 | // audio_delay_ms or video_delay_ms since those now include VCM and NetEQ |
| 156 | // delays. |
| 157 | void BothDelayedAudioLaterTest() { |
| 158 | int current_audio_delay_ms = 0; |
| 159 | int audio_delay_ms = 300; |
| 160 | int video_delay_ms = 100; |
| 161 | int extra_audio_delay_ms = 0; |
| 162 | int total_video_delay_ms = 0; |
| 163 | |
| 164 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 165 | video_delay_ms, |
| 166 | current_audio_delay_ms, |
| 167 | &extra_audio_delay_ms, |
| 168 | &total_video_delay_ms)); |
| 169 | EXPECT_EQ(kMaxVideoDiffMs, total_video_delay_ms); |
| 170 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 171 | current_audio_delay_ms = extra_audio_delay_ms; |
| 172 | |
| 173 | send_time_->IncreaseTimeMs(1000); |
| 174 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 175 | video_delay_ms)); |
| 176 | // Simulate 0 minimum delay in the VCM. |
| 177 | total_video_delay_ms = 0; |
| 178 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 179 | video_delay_ms, |
| 180 | current_audio_delay_ms, |
| 181 | &extra_audio_delay_ms, |
| 182 | &total_video_delay_ms)); |
| 183 | EXPECT_EQ(2 * kMaxVideoDiffMs, total_video_delay_ms); |
| 184 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 185 | current_audio_delay_ms = extra_audio_delay_ms; |
| 186 | |
| 187 | send_time_->IncreaseTimeMs(1000); |
| 188 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 189 | video_delay_ms)); |
| 190 | // Simulate 0 minimum delay in the VCM. |
| 191 | total_video_delay_ms = 0; |
| 192 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 193 | video_delay_ms, |
| 194 | current_audio_delay_ms, |
| 195 | &extra_audio_delay_ms, |
| 196 | &total_video_delay_ms)); |
| 197 | EXPECT_EQ(audio_delay_ms - video_delay_ms, total_video_delay_ms); |
| 198 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 199 | |
| 200 | // Simulate that NetEQ introduces some audio delay. |
| 201 | current_audio_delay_ms = 50; |
| 202 | send_time_->IncreaseTimeMs(1000); |
| 203 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 204 | video_delay_ms)); |
| 205 | // Simulate 0 minimum delay in the VCM. |
| 206 | total_video_delay_ms = 0; |
| 207 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 208 | video_delay_ms, |
| 209 | current_audio_delay_ms, |
| 210 | &extra_audio_delay_ms, |
| 211 | &total_video_delay_ms)); |
| 212 | EXPECT_EQ(audio_delay_ms - video_delay_ms + current_audio_delay_ms, |
| 213 | total_video_delay_ms); |
| 214 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 215 | |
| 216 | // Simulate that NetEQ reduces its delay. |
| 217 | current_audio_delay_ms = 10; |
| 218 | send_time_->IncreaseTimeMs(1000); |
| 219 | receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms, |
| 220 | video_delay_ms)); |
| 221 | // Simulate 0 minimum delay in the VCM. |
| 222 | total_video_delay_ms = 0; |
| 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)); |
| 228 | EXPECT_EQ(audio_delay_ms - video_delay_ms + current_audio_delay_ms, |
| 229 | total_video_delay_ms); |
| 230 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 231 | } |
| 232 | |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 233 | int MaxAudioDelayIncrease(int current_audio_delay_ms, int delay_ms) { |
| 234 | return std::min((delay_ms - current_audio_delay_ms) / 2, |
| 235 | static_cast<int>(kMaxAudioDiffMs)); |
| 236 | } |
| 237 | |
| 238 | int MaxAudioDelayDecrease(int current_audio_delay_ms, int delay_ms) { |
| 239 | return std::max((delay_ms - current_audio_delay_ms) / 2, -kMaxAudioDiffMs); |
| 240 | } |
| 241 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 242 | enum { kSendTimeOffsetMs = 98765 }; |
| 243 | enum { kReceiveTimeOffsetMs = 43210 }; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 244 | |
| 245 | StreamSynchronization* sync_; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 246 | Time* send_time_; // The simulated clock at the sender. |
| 247 | Time* receive_time_; // The simulated clock at the receiver. |
| 248 | double audio_clock_drift_; |
| 249 | double video_clock_drift_; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 250 | }; |
| 251 | |
| 252 | TEST_F(StreamSynchronizationTest, NoDelay) { |
| 253 | uint32_t current_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 254 | int extra_audio_delay_ms = 0; |
| 255 | int total_video_delay_ms = 0; |
| 256 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 257 | EXPECT_TRUE(DelayedStreams(0, 0, current_audio_delay_ms, |
| 258 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 259 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 260 | EXPECT_EQ(0, total_video_delay_ms); |
| 261 | } |
| 262 | |
| 263 | TEST_F(StreamSynchronizationTest, VideoDelay) { |
| 264 | uint32_t current_audio_delay_ms = 0; |
| 265 | int delay_ms = 200; |
| 266 | int extra_audio_delay_ms = 0; |
| 267 | int total_video_delay_ms = 0; |
| 268 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 269 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 270 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 271 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 272 | // The video delay is not allowed to change more than this in 1 second. |
| 273 | EXPECT_EQ(kMaxVideoDiffMs, total_video_delay_ms); |
| 274 | |
| 275 | send_time_->IncreaseTimeMs(1000); |
| 276 | receive_time_->IncreaseTimeMs(800); |
| 277 | // Simulate 0 minimum delay in the VCM. |
| 278 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 279 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 280 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 281 | EXPECT_EQ(0, extra_audio_delay_ms); |
| 282 | // The video delay is not allowed to change more than this in 1 second. |
| 283 | EXPECT_EQ(2*kMaxVideoDiffMs, total_video_delay_ms); |
| 284 | |
| 285 | send_time_->IncreaseTimeMs(1000); |
| 286 | receive_time_->IncreaseTimeMs(800); |
| 287 | // Simulate 0 minimum delay in the VCM. |
| 288 | total_video_delay_ms = 0; |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 289 | EXPECT_TRUE(DelayedStreams(delay_ms, 0, current_audio_delay_ms, |
| 290 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 291 | EXPECT_EQ(0, extra_audio_delay_ms); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 292 | // Enough time should have elapsed for the requested total video delay to be |
| 293 | // equal to the relative delay between audio and video, i.e., we are in sync. |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 294 | EXPECT_EQ(delay_ms, total_video_delay_ms); |
| 295 | } |
| 296 | |
| 297 | TEST_F(StreamSynchronizationTest, AudioDelay) { |
| 298 | int current_audio_delay_ms = 0; |
| 299 | int delay_ms = 200; |
| 300 | int extra_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 301 | int total_video_delay_ms = 0; |
| 302 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 303 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 304 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 305 | EXPECT_EQ(0, total_video_delay_ms); |
| 306 | // The audio delay is not allowed to change more than this in 1 second. |
| 307 | EXPECT_EQ(kMaxAudioDiffMs, extra_audio_delay_ms); |
| 308 | current_audio_delay_ms = extra_audio_delay_ms; |
andrew@webrtc.org | d7a71d0 | 2012-08-01 01:40:02 +0000 | [diff] [blame] | 309 | int current_extra_delay_ms = extra_audio_delay_ms; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 310 | |
| 311 | send_time_->IncreaseTimeMs(1000); |
| 312 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 313 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 314 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 315 | EXPECT_EQ(0, total_video_delay_ms); |
| 316 | // The audio delay is not allowed to change more than the half of the required |
| 317 | // change in delay. |
| 318 | EXPECT_EQ(current_extra_delay_ms + |
| 319 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
| 320 | extra_audio_delay_ms); |
| 321 | current_audio_delay_ms = extra_audio_delay_ms; |
| 322 | current_extra_delay_ms = extra_audio_delay_ms; |
| 323 | |
| 324 | send_time_->IncreaseTimeMs(1000); |
| 325 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 326 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 327 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 328 | EXPECT_EQ(0, total_video_delay_ms); |
| 329 | // The audio delay is not allowed to change more than the half of the required |
| 330 | // change in delay. |
| 331 | EXPECT_EQ(current_extra_delay_ms + |
| 332 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
| 333 | extra_audio_delay_ms); |
| 334 | current_extra_delay_ms = extra_audio_delay_ms; |
| 335 | |
| 336 | // Simulate that NetEQ for some reason reduced the delay. |
| 337 | current_audio_delay_ms = 170; |
| 338 | send_time_->IncreaseTimeMs(1000); |
| 339 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 340 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 341 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 342 | EXPECT_EQ(0, total_video_delay_ms); |
| 343 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 344 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 345 | // here to try to |
| 346 | EXPECT_EQ(current_extra_delay_ms + |
| 347 | MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms), |
| 348 | extra_audio_delay_ms); |
| 349 | current_extra_delay_ms = extra_audio_delay_ms; |
| 350 | |
| 351 | // Simulate that NetEQ for some reason significantly increased the delay. |
| 352 | current_audio_delay_ms = 250; |
| 353 | send_time_->IncreaseTimeMs(1000); |
| 354 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 355 | EXPECT_TRUE(DelayedStreams(0, delay_ms, current_audio_delay_ms, |
| 356 | &extra_audio_delay_ms, &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 357 | EXPECT_EQ(0, total_video_delay_ms); |
| 358 | // The audio delay is not allowed to change more than the half of the required |
| 359 | // change in delay. |
| 360 | EXPECT_EQ(current_extra_delay_ms + |
| 361 | MaxAudioDelayDecrease(current_audio_delay_ms, delay_ms), |
| 362 | extra_audio_delay_ms); |
| 363 | } |
| 364 | |
| 365 | TEST_F(StreamSynchronizationTest, BothDelayedVideoLater) { |
| 366 | int current_audio_delay_ms = 0; |
| 367 | int audio_delay_ms = 100; |
| 368 | int video_delay_ms = 300; |
| 369 | int extra_audio_delay_ms = 0; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 370 | int total_video_delay_ms = 0; |
| 371 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 372 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 373 | video_delay_ms, |
| 374 | current_audio_delay_ms, |
| 375 | &extra_audio_delay_ms, |
| 376 | &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 377 | EXPECT_EQ(0, total_video_delay_ms); |
| 378 | // The audio delay is not allowed to change more than this in 1 second. |
| 379 | EXPECT_EQ(kMaxAudioDiffMs, extra_audio_delay_ms); |
| 380 | current_audio_delay_ms = extra_audio_delay_ms; |
andrew@webrtc.org | d7a71d0 | 2012-08-01 01:40:02 +0000 | [diff] [blame] | 381 | int current_extra_delay_ms = extra_audio_delay_ms; |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 382 | |
| 383 | send_time_->IncreaseTimeMs(1000); |
| 384 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 385 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 386 | video_delay_ms, |
| 387 | current_audio_delay_ms, |
| 388 | &extra_audio_delay_ms, |
| 389 | &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 390 | EXPECT_EQ(0, total_video_delay_ms); |
| 391 | // The audio delay is not allowed to change more than the half of the required |
| 392 | // change in delay. |
| 393 | EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease( |
| 394 | current_audio_delay_ms, video_delay_ms - audio_delay_ms), |
| 395 | extra_audio_delay_ms); |
| 396 | current_audio_delay_ms = extra_audio_delay_ms; |
| 397 | current_extra_delay_ms = extra_audio_delay_ms; |
| 398 | |
| 399 | send_time_->IncreaseTimeMs(1000); |
| 400 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 401 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 402 | video_delay_ms, |
| 403 | current_audio_delay_ms, |
| 404 | &extra_audio_delay_ms, |
| 405 | &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 + MaxAudioDelayIncrease( |
| 410 | current_audio_delay_ms, video_delay_ms - audio_delay_ms), |
| 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. |
| 415 | current_audio_delay_ms = 170; |
| 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(audio_delay_ms, |
| 419 | video_delay_ms, |
| 420 | current_audio_delay_ms, |
| 421 | &extra_audio_delay_ms, |
| 422 | &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 423 | EXPECT_EQ(0, total_video_delay_ms); |
| 424 | // Since we only can ask NetEQ for a certain amount of extra delay, and |
| 425 | // we only measure the total NetEQ delay, we will ask for additional delay |
| 426 | // here to try to stay in sync. |
| 427 | EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease( |
| 428 | current_audio_delay_ms, video_delay_ms - audio_delay_ms), |
| 429 | extra_audio_delay_ms); |
| 430 | current_extra_delay_ms = extra_audio_delay_ms; |
| 431 | |
| 432 | // Simulate that NetEQ for some reason significantly increased the delay. |
| 433 | current_audio_delay_ms = 250; |
| 434 | send_time_->IncreaseTimeMs(1000); |
| 435 | receive_time_->IncreaseTimeMs(800); |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 436 | EXPECT_TRUE(DelayedStreams(audio_delay_ms, |
| 437 | video_delay_ms, |
| 438 | current_audio_delay_ms, |
| 439 | &extra_audio_delay_ms, |
| 440 | &total_video_delay_ms)); |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 441 | EXPECT_EQ(0, total_video_delay_ms); |
| 442 | // The audio delay is not allowed to change more than the half of the required |
| 443 | // change in delay. |
| 444 | EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease( |
| 445 | current_audio_delay_ms, video_delay_ms - audio_delay_ms), |
| 446 | extra_audio_delay_ms); |
| 447 | } |
| 448 | |
| 449 | TEST_F(StreamSynchronizationTest, BothDelayedAudioLater) { |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 450 | BothDelayedAudioLaterTest(); |
| 451 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 452 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 453 | TEST_F(StreamSynchronizationTest, BothDelayedAudioClockDrift) { |
| 454 | audio_clock_drift_ = 1.05; |
| 455 | BothDelayedAudioLaterTest(); |
| 456 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 457 | |
stefan@webrtc.org | 7c3523c | 2012-09-11 07:00:42 +0000 | [diff] [blame] | 458 | TEST_F(StreamSynchronizationTest, BothDelayedVideoClockDrift) { |
| 459 | video_clock_drift_ = 1.05; |
| 460 | BothDelayedAudioLaterTest(); |
| 461 | } |
stefan@webrtc.org | 5f28498 | 2012-06-28 07:51:16 +0000 | [diff] [blame] | 462 | } // namespace webrtc |