forked from JumpingYang001/webrtc
-
Notifications
You must be signed in to change notification settings - Fork 0
/
Copy pathcall_perf_tests.cc
1039 lines (920 loc) · 41.1 KB
/
call_perf_tests.cc
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <algorithm>
#include <limits>
#include <memory>
#include <string>
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video_codecs/video_encoder.h"
#include "api/video_codecs/video_encoder_config.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/metrics.h"
#include "test/call_test.h"
#include "test/direct_transport.h"
#include "test/drifting_clock.h"
#include "test/encoder_settings.h"
#include "test/fake_encoder.h"
#include "test/field_trial.h"
#include "test/frame_generator_capturer.h"
#include "test/gtest.h"
#include "test/null_transport.h"
#include "test/rtp_header_parser.h"
#include "test/rtp_rtcp_observer.h"
#include "test/testsupport/file_utils.h"
#include "test/testsupport/perf_test.h"
#include "test/video_encoder_proxy_factory.h"
#include "video/transport_adapter.h"
using webrtc::test::DriftingClock;
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kTransportSequenceNumberExtensionId = 1,
};
} // namespace
class CallPerfTest : public test::CallTest {
public:
CallPerfTest() {
RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
}
protected:
enum class FecMode { kOn, kOff };
enum class CreateOrder { kAudioFirst, kVideoFirst };
void TestAudioVideoSync(FecMode fec,
CreateOrder create_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed,
const std::string& test_label);
void TestMinTransmitBitrate(bool pad_to_min_bitrate);
void TestCaptureNtpTime(const BuiltInNetworkBehaviorConfig& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms);
void TestMinAudioVideoBitrate(int test_bitrate_from,
int test_bitrate_to,
int test_bitrate_step,
int min_bwe,
int start_bwe,
int max_bwe);
};
class VideoRtcpAndSyncObserver : public test::RtpRtcpObserver,
public rtc::VideoSinkInterface<VideoFrame> {
static const int kInSyncThresholdMs = 50;
static const int kStartupTimeMs = 2000;
static const int kMinRunTimeMs = 30000;
public:
explicit VideoRtcpAndSyncObserver(TaskQueueBase* task_queue,
Clock* clock,
const std::string& test_label)
: test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs),
clock_(clock),
test_label_(test_label),
creation_time_ms_(clock_->TimeInMilliseconds()),
task_queue_(task_queue) {}
void OnFrame(const VideoFrame& video_frame) override {
task_queue_->PostTask(ToQueuedTask([this]() { CheckStats(); }));
}
void CheckStats() {
if (!receive_stream_)
return;
VideoReceiveStream::Stats stats = receive_stream_->GetStats();
if (stats.sync_offset_ms == std::numeric_limits<int>::max())
return;
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t time_since_creation = now_ms - creation_time_ms_;
// During the first couple of seconds audio and video can falsely be
// estimated as being synchronized. We don't want to trigger on those.
if (time_since_creation < kStartupTimeMs)
return;
if (std::abs(stats.sync_offset_ms) < kInSyncThresholdMs) {
if (first_time_in_sync_ == -1) {
first_time_in_sync_ = now_ms;
webrtc::test::PrintResult("sync_convergence_time", test_label_,
"synchronization", time_since_creation, "ms",
false);
}
if (time_since_creation > kMinRunTimeMs)
observation_complete_.Set();
}
if (first_time_in_sync_ != -1)
sync_offset_ms_list_.push_back(stats.sync_offset_ms);
}
void set_receive_stream(VideoReceiveStream* receive_stream) {
RTC_DCHECK_EQ(task_queue_, TaskQueueBase::Current());
// Note that receive_stream may be nullptr.
receive_stream_ = receive_stream;
}
void PrintResults() {
test::PrintResultList("stream_offset", test_label_, "synchronization",
sync_offset_ms_list_, "ms", false);
}
private:
Clock* const clock_;
std::string test_label_;
const int64_t creation_time_ms_;
int64_t first_time_in_sync_ = -1;
VideoReceiveStream* receive_stream_ = nullptr;
std::vector<double> sync_offset_ms_list_;
TaskQueueBase* const task_queue_;
};
void CallPerfTest::TestAudioVideoSync(FecMode fec,
CreateOrder create_first,
float video_ntp_speed,
float video_rtp_speed,
float audio_rtp_speed,
const std::string& test_label) {
const char* kSyncGroup = "av_sync";
const uint32_t kAudioSendSsrc = 1234;
const uint32_t kAudioRecvSsrc = 5678;
BuiltInNetworkBehaviorConfig audio_net_config;
audio_net_config.queue_delay_ms = 500;
audio_net_config.loss_percent = 5;
auto observer = std::make_unique<VideoRtcpAndSyncObserver>(
task_queue(), Clock::GetRealTimeClock(), test_label);
std::map<uint8_t, MediaType> audio_pt_map;
std::map<uint8_t, MediaType> video_pt_map;
std::unique_ptr<test::PacketTransport> audio_send_transport;
std::unique_ptr<test::PacketTransport> video_send_transport;
std::unique_ptr<test::PacketTransport> receive_transport;
AudioSendStream* audio_send_stream;
AudioReceiveStream* audio_receive_stream;
std::unique_ptr<DriftingClock> drifting_clock;
SendTask(RTC_FROM_HERE, task_queue(), [&]() {
metrics::Reset();
rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device =
TestAudioDeviceModule::Create(
task_queue_factory_.get(),
TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000),
TestAudioDeviceModule::CreateDiscardRenderer(48000),
audio_rtp_speed);
EXPECT_EQ(0, fake_audio_device->Init());
AudioState::Config send_audio_state_config;
send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
send_audio_state_config.audio_processing =
AudioProcessingBuilder().Create();
send_audio_state_config.audio_device_module = fake_audio_device;
Call::Config sender_config(send_event_log_.get());
auto audio_state = AudioState::Create(send_audio_state_config);
fake_audio_device->RegisterAudioCallback(audio_state->audio_transport());
sender_config.audio_state = audio_state;
Call::Config receiver_config(recv_event_log_.get());
receiver_config.audio_state = audio_state;
CreateCalls(sender_config, receiver_config);
std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
std::inserter(audio_pt_map, audio_pt_map.end()),
[](const std::pair<const uint8_t, MediaType>& pair) {
return pair.second == MediaType::AUDIO;
});
std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
std::inserter(video_pt_map, video_pt_map.end()),
[](const std::pair<const uint8_t, MediaType>& pair) {
return pair.second == MediaType::VIDEO;
});
audio_send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), observer.get(),
test::PacketTransport::kSender, audio_pt_map,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(audio_net_config)));
audio_send_transport->SetReceiver(receiver_call_->Receiver());
video_send_transport = std::make_unique<test::PacketTransport>(
task_queue(), sender_call_.get(), observer.get(),
test::PacketTransport::kSender, video_pt_map,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())));
video_send_transport->SetReceiver(receiver_call_->Receiver());
receive_transport = std::make_unique<test::PacketTransport>(
task_queue(), receiver_call_.get(), observer.get(),
test::PacketTransport::kReceiver, payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())));
receive_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, video_send_transport.get());
CreateMatchingReceiveConfigs(receive_transport.get());
AudioSendStream::Config audio_send_config(audio_send_transport.get());
audio_send_config.rtp.ssrc = kAudioSendSsrc;
audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
kAudioSendPayloadType, {"ISAC", 16000, 1});
audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config);
GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (fec == FecMode::kOn) {
GetVideoSendConfig()->rtp.ulpfec.red_payload_type = kRedPayloadType;
GetVideoSendConfig()->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
video_receive_configs_[0].rtp.red_payload_type = kRedPayloadType;
video_receive_configs_[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
}
video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
video_receive_configs_[0].renderer = observer.get();
video_receive_configs_[0].sync_group = kSyncGroup;
AudioReceiveStream::Config audio_recv_config;
audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
audio_recv_config.rtcp_send_transport = receive_transport.get();
audio_recv_config.sync_group = kSyncGroup;
audio_recv_config.decoder_factory = audio_decoder_factory_;
audio_recv_config.decoder_map = {
{kAudioSendPayloadType, {"ISAC", 16000, 1}}};
if (create_first == CreateOrder::kAudioFirst) {
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
CreateVideoStreams();
} else {
CreateVideoStreams();
audio_receive_stream =
receiver_call_->CreateAudioReceiveStream(audio_recv_config);
}
EXPECT_EQ(1u, video_receive_streams_.size());
observer->set_receive_stream(video_receive_streams_[0]);
drifting_clock = std::make_unique<DriftingClock>(clock_, video_ntp_speed);
CreateFrameGeneratorCapturerWithDrift(drifting_clock.get(), video_rtp_speed,
kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
audio_send_stream->Start();
audio_receive_stream->Start();
});
EXPECT_TRUE(observer->Wait())
<< "Timed out while waiting for audio and video to be synchronized.";
SendTask(RTC_FROM_HERE, task_queue(), [&]() {
// Clear the pointer to the receive stream since it will now be deleted.
observer->set_receive_stream(nullptr);
audio_send_stream->Stop();
audio_receive_stream->Stop();
Stop();
DestroyStreams();
sender_call_->DestroyAudioSendStream(audio_send_stream);
receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
DestroyCalls();
// Call may post periodic rtcp packet to the transport on the process
// thread, thus transport should be destroyed after the call objects.
// Though transports keep pointers to the call objects, transports handle
// packets on the task_queue() and thus wouldn't create a race while current
// destruction happens in the same task as destruction of the call objects.
video_send_transport.reset();
audio_send_transport.reset();
receive_transport.reset();
});
observer->PrintResults();
// In quick test synchronization may not be achieved in time.
if (!field_trial::IsEnabled("WebRTC-QuickPerfTest")) {
// TODO(bugs.webrtc.org/10417): Reenable this for iOS
#if !defined(WEBRTC_IOS)
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
#endif
}
task_queue()->PostTask(
ToQueuedTask([to_delete = observer.release()]() { delete to_delete; }));
}
TEST_F(CallPerfTest, Synchronization_PlaysOutAudioAndVideoWithoutClockDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::kNoDrift, DriftingClock::kNoDrift,
DriftingClock::kNoDrift, "_video_no_drift");
}
TEST_F(CallPerfTest, Synchronization_PlaysOutAudioAndVideoWithVideoNtpDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::PercentsFaster(10.0f),
DriftingClock::kNoDrift, DriftingClock::kNoDrift,
"_video_ntp_drift");
}
TEST_F(CallPerfTest,
Synchronization_PlaysOutAudioAndVideoWithAudioFasterThanVideoDrift) {
TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsSlower(30.0f),
DriftingClock::PercentsFaster(30.0f), "_audio_faster");
}
TEST_F(CallPerfTest,
Synchronization_PlaysOutAudioAndVideoWithVideoFasterThanAudioDrift) {
TestAudioVideoSync(FecMode::kOn, CreateOrder::kVideoFirst,
DriftingClock::kNoDrift,
DriftingClock::PercentsFaster(30.0f),
DriftingClock::PercentsSlower(30.0f), "_video_faster");
}
void CallPerfTest::TestCaptureNtpTime(
const BuiltInNetworkBehaviorConfig& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms) {
class CaptureNtpTimeObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
CaptureNtpTimeObserver(const BuiltInNetworkBehaviorConfig& net_config,
int threshold_ms,
int start_time_ms,
int run_time_ms)
: EndToEndTest(kLongTimeoutMs),
net_config_(net_config),
clock_(Clock::GetRealTimeClock()),
threshold_ms_(threshold_ms),
start_time_ms_(start_time_ms),
run_time_ms_(run_time_ms),
creation_time_ms_(clock_->TimeInMilliseconds()),
capturer_(nullptr),
rtp_start_timestamp_set_(false),
rtp_start_timestamp_(0) {}
private:
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) override {
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(net_config_)));
}
std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
TaskQueueBase* task_queue) override {
return std::make_unique<test::PacketTransport>(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
payload_type_map_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(net_config_)));
}
void OnFrame(const VideoFrame& video_frame) override {
MutexLock lock(&mutex_);
if (video_frame.ntp_time_ms() <= 0) {
// Haven't got enough RTCP SR in order to calculate the capture ntp
// time.
return;
}
int64_t now_ms = clock_->TimeInMilliseconds();
int64_t time_since_creation = now_ms - creation_time_ms_;
if (time_since_creation < start_time_ms_) {
// Wait for |start_time_ms_| before start measuring.
return;
}
if (time_since_creation > run_time_ms_) {
observation_complete_.Set();
}
FrameCaptureTimeList::iterator iter =
capture_time_list_.find(video_frame.timestamp());
EXPECT_TRUE(iter != capture_time_list_.end());
// The real capture time has been wrapped to uint32_t before converted
// to rtp timestamp in the sender side. So here we convert the estimated
// capture time to a uint32_t 90k timestamp also for comparing.
uint32_t estimated_capture_timestamp =
90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
uint32_t real_capture_timestamp = iter->second;
int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
time_offset_ms = time_offset_ms / 90;
time_offset_ms_list_.push_back(time_offset_ms);
EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
if (!rtp_start_timestamp_set_) {
// Calculate the rtp timestamp offset in order to calculate the real
// capture time.
uint32_t first_capture_timestamp =
90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
rtp_start_timestamp_ = rtp_packet.Timestamp() - first_capture_timestamp;
rtp_start_timestamp_set_ = true;
}
uint32_t capture_timestamp =
rtp_packet.Timestamp() - rtp_start_timestamp_;
capture_time_list_.insert(
capture_time_list_.end(),
std::make_pair(rtp_packet.Timestamp(), capture_timestamp));
return SEND_PACKET;
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
capturer_ = frame_generator_capturer;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
(*receive_configs)[0].renderer = this;
// Enable the receiver side rtt calculation.
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for "
"estimated capture NTP time to be "
"within bounds.";
test::PrintResultList("capture_ntp_time", "", "real - estimated",
time_offset_ms_list_, "ms", true);
}
Mutex mutex_;
const BuiltInNetworkBehaviorConfig net_config_;
Clock* const clock_;
int threshold_ms_;
int start_time_ms_;
int run_time_ms_;
int64_t creation_time_ms_;
test::FrameGeneratorCapturer* capturer_;
bool rtp_start_timestamp_set_;
uint32_t rtp_start_timestamp_;
typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
FrameCaptureTimeList capture_time_list_ RTC_GUARDED_BY(&mutex_);
std::vector<double> time_offset_ms_list_;
} test(net_config, threshold_ms, start_time_ms, run_time_ms);
RunBaseTest(&test);
}
// Flaky tests, disabled on Mac and Windows due to webrtc:8291.
#if !(defined(WEBRTC_MAC) || defined(WEBRTC_WIN))
TEST_F(CallPerfTest, Real_Estimated_CaptureNtpTimeWithNetworkDelay) {
BuiltInNetworkBehaviorConfig net_config;
net_config.queue_delay_ms = 100;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
TEST_F(CallPerfTest, Real_Estimated_CaptureNtpTimeWithNetworkJitter) {
BuiltInNetworkBehaviorConfig net_config;
net_config.queue_delay_ms = 100;
net_config.delay_standard_deviation_ms = 10;
// TODO(wu): lower the threshold as the calculation/estimatation becomes more
// accurate.
const int kThresholdMs = 100;
const int kStartTimeMs = 10000;
const int kRunTimeMs = 20000;
TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
}
#endif
TEST_F(CallPerfTest, ReceivesCpuOveruseAndUnderuse) {
// Minimal normal usage at the start, then 30s overuse to allow filter to
// settle, and then 80s underuse to allow plenty of time for rampup again.
test::ScopedFieldTrials fake_overuse_settings(
"WebRTC-ForceSimulatedOveruseIntervalMs/1-30000-80000/");
class LoadObserver : public test::SendTest,
public test::FrameGeneratorCapturer::SinkWantsObserver {
public:
LoadObserver() : SendTest(kLongTimeoutMs), test_phase_(TestPhase::kInit) {}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_capturer->SetSinkWantsObserver(this);
// Set a high initial resolution to be sure that we can scale down.
frame_generator_capturer->ChangeResolution(1920, 1080);
}
// OnSinkWantsChanged is called when FrameGeneratorCapturer::AddOrUpdateSink
// is called.
// TODO(sprang): Add integration test for maintain-framerate mode?
void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override {
// The sink wants can change either because an adaptation happened (i.e.
// the pixels or frame rate changed) or for other reasons, such as encoded
// resolutions being communicated (happens whenever we capture a new frame
// size). In this test, we only care about adaptations.
bool did_adapt =
last_wants_.max_pixel_count != wants.max_pixel_count ||
last_wants_.target_pixel_count != wants.target_pixel_count ||
last_wants_.max_framerate_fps != wants.max_framerate_fps;
last_wants_ = wants;
if (!did_adapt) {
return;
}
// At kStart expect CPU overuse. Then expect CPU underuse when the encoder
// delay has been decreased.
switch (test_phase_) {
case TestPhase::kInit:
// Max framerate should be set initially.
if (wants.max_framerate_fps != std::numeric_limits<int>::max() &&
wants.max_pixel_count == std::numeric_limits<int>::max()) {
test_phase_ = TestPhase::kStart;
} else {
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
break;
case TestPhase::kStart:
if (wants.max_pixel_count < std::numeric_limits<int>::max()) {
// On adapting down, VideoStreamEncoder::VideoSourceProxy will set
// only the max pixel count, leaving the target unset.
test_phase_ = TestPhase::kAdaptedDown;
} else {
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
break;
case TestPhase::kAdaptedDown:
// On adapting up, the adaptation counter will again be at zero, and
// so all constraints will be reset.
if (wants.max_pixel_count == std::numeric_limits<int>::max() &&
!wants.target_pixel_count) {
test_phase_ = TestPhase::kAdaptedUp;
observation_complete_.Set();
} else {
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
break;
case TestPhase::kAdaptedUp:
ADD_FAILURE() << "Got unexpected adaptation request, max res = "
<< wants.max_pixel_count << ", target res = "
<< wants.target_pixel_count.value_or(-1)
<< ", max fps = " << wants.max_framerate_fps;
}
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
}
enum class TestPhase {
kInit,
kStart,
kAdaptedDown,
kAdaptedUp
} test_phase_;
private:
rtc::VideoSinkWants last_wants_;
} test;
RunBaseTest(&test);
}
void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
static const int kMaxEncodeBitrateKbps = 30;
static const int kMinTransmitBitrateBps = 150000;
static const int kMinAcceptableTransmitBitrate = 130;
static const int kMaxAcceptableTransmitBitrate = 170;
static const int kNumBitrateObservationsInRange = 100;
static const int kAcceptableBitrateErrorMargin = 15; // +- 7
class BitrateObserver : public test::EndToEndTest {
public:
explicit BitrateObserver(bool using_min_transmit_bitrate)
: EndToEndTest(kLongTimeoutMs),
send_stream_(nullptr),
converged_(false),
pad_to_min_bitrate_(using_min_transmit_bitrate),
min_acceptable_bitrate_(using_min_transmit_bitrate
? kMinAcceptableTransmitBitrate
: (kMaxEncodeBitrateKbps -
kAcceptableBitrateErrorMargin / 2)),
max_acceptable_bitrate_(using_min_transmit_bitrate
? kMaxAcceptableTransmitBitrate
: (kMaxEncodeBitrateKbps +
kAcceptableBitrateErrorMargin / 2)),
num_bitrate_observations_in_range_(0) {}
private:
// TODO(holmer): Run this with a timer instead of once per packet.
Action OnSendRtp(const uint8_t* packet, size_t length) override {
VideoSendStream::Stats stats = send_stream_->GetStats();
if (!stats.substreams.empty()) {
RTC_DCHECK_EQ(1, stats.substreams.size());
int bitrate_kbps =
stats.substreams.begin()->second.total_bitrate_bps / 1000;
if (bitrate_kbps > min_acceptable_bitrate_ &&
bitrate_kbps < max_acceptable_bitrate_) {
converged_ = true;
++num_bitrate_observations_in_range_;
if (num_bitrate_observations_in_range_ ==
kNumBitrateObservationsInRange)
observation_complete_.Set();
}
if (converged_)
bitrate_kbps_list_.push_back(bitrate_kbps);
}
return SEND_PACKET;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (pad_to_min_bitrate_) {
encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
} else {
RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
}
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
test::PrintResultList(
"bitrate_stats_",
(pad_to_min_bitrate_ ? "min_transmit_bitrate"
: "without_min_transmit_bitrate"),
"bitrate_kbps", bitrate_kbps_list_, "kbps", false);
}
VideoSendStream* send_stream_;
bool converged_;
const bool pad_to_min_bitrate_;
const int min_acceptable_bitrate_;
const int max_acceptable_bitrate_;
int num_bitrate_observations_in_range_;
std::vector<double> bitrate_kbps_list_;
} test(pad_to_min_bitrate);
fake_encoder_max_bitrate_ = kMaxEncodeBitrateKbps;
RunBaseTest(&test);
}
TEST_F(CallPerfTest, Bitrate_Kbps_PadsToMinTransmitBitrate) {
TestMinTransmitBitrate(true);
}
TEST_F(CallPerfTest, Bitrate_Kbps_NoPadWithoutMinTransmitBitrate) {
TestMinTransmitBitrate(false);
}
// TODO(bugs.webrtc.org/8878)
#if defined(WEBRTC_MAC)
#define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
DISABLED_KeepsHighBitrateWhenReconfiguringSender
#else
#define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
KeepsHighBitrateWhenReconfiguringSender
#endif
TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
static const uint32_t kInitialBitrateKbps = 400;
static const uint32_t kReconfigureThresholdKbps = 600;
// We get lower bitrate than expected by this test if the following field
// trial is enabled.
test::ScopedFieldTrials field_trials(
"WebRTC-SendSideBwe-WithOverhead/Disabled/");
class VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
VideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
streams[0].min_bitrate_bps = 50000;
streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000;
return streams;
}
};
class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder {
public:
BitrateObserver()
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
encoder_inits_(0),
last_set_bitrate_kbps_(0),
send_stream_(nullptr),
frame_generator_(nullptr),
encoder_factory_(this),
bitrate_allocator_factory_(
CreateBuiltinVideoBitrateAllocatorFactory()) {}
int32_t InitEncode(const VideoCodec* config,
const VideoEncoder::Settings& settings) override {
++encoder_inits_;
if (encoder_inits_ == 1) {
// First time initialization. Frame size is known.
// |expected_bitrate| is affected by bandwidth estimation before the
// first frame arrives to the encoder.
uint32_t expected_bitrate = last_set_bitrate_kbps_ > 0
? last_set_bitrate_kbps_
: kInitialBitrateKbps;
EXPECT_EQ(expected_bitrate, config->startBitrate)
<< "Encoder not initialized at expected bitrate.";
EXPECT_EQ(kDefaultWidth, config->width);
EXPECT_EQ(kDefaultHeight, config->height);
} else if (encoder_inits_ == 2) {
EXPECT_EQ(2 * kDefaultWidth, config->width);
EXPECT_EQ(2 * kDefaultHeight, config->height);
EXPECT_GE(last_set_bitrate_kbps_, kReconfigureThresholdKbps);
EXPECT_GT(config->startBitrate, kReconfigureThresholdKbps)
<< "Encoder reconfigured with bitrate too far away from last set.";
observation_complete_.Set();
}
return FakeEncoder::InitEncode(config, settings);
}
void SetRates(const RateControlParameters& parameters) override {
last_set_bitrate_kbps_ = parameters.bitrate.get_sum_kbps();
if (encoder_inits_ == 1 &&
parameters.bitrate.get_sum_kbps() > kReconfigureThresholdKbps) {
time_to_reconfigure_.Set();
}
FakeEncoder::SetRates(parameters);
}
void ModifySenderBitrateConfig(
BitrateConstraints* bitrate_config) override {
bitrate_config->start_bitrate_bps = kInitialBitrateKbps * 1000;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
send_config->encoder_settings.bitrate_allocator_factory =
bitrate_allocator_factory_.get();
encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000;
encoder_config->video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>();
encoder_config_ = encoder_config->Copy();
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_ = frame_generator_capturer;
}
void PerformTest() override {
ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
<< "Timed out before receiving an initial high bitrate.";
frame_generator_->ChangeResolution(kDefaultWidth * 2, kDefaultHeight * 2);
send_stream_->ReconfigureVideoEncoder(encoder_config_.Copy());
EXPECT_TRUE(Wait())
<< "Timed out while waiting for a couple of high bitrate estimates "
"after reconfiguring the send stream.";
}
private:
rtc::Event time_to_reconfigure_;
int encoder_inits_;
uint32_t last_set_bitrate_kbps_;
VideoSendStream* send_stream_;
test::FrameGeneratorCapturer* frame_generator_;
test::VideoEncoderProxyFactory encoder_factory_;
std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
VideoEncoderConfig encoder_config_;
} test;
RunBaseTest(&test);
}
// Discovers the minimal supported audio+video bitrate. The test bitrate is
// considered supported if Rtt does not go above 400ms with the network
// contrained to the test bitrate.
//
// |test_bitrate_from test_bitrate_to| bitrate constraint range
// |test_bitrate_step| bitrate constraint update step during the test
// |min_bwe max_bwe| BWE range
// |start_bwe| initial BWE
void CallPerfTest::TestMinAudioVideoBitrate(int test_bitrate_from,
int test_bitrate_to,
int test_bitrate_step,
int min_bwe,
int start_bwe,
int max_bwe) {
static const std::string kAudioTrackId = "audio_track_0";
static constexpr int kOpusBitrateFbBps = 32000;
static constexpr int kBitrateStabilizationMs = 10000;
static constexpr int kBitrateMeasurements = 10;
static constexpr int kBitrateMeasurementMs = 1000;
static constexpr int kShortDelayMs = 10;
static constexpr int kMinGoodRttMs = 400;
class MinVideoAndAudioBitrateTester : public test::EndToEndTest {
public:
MinVideoAndAudioBitrateTester(int test_bitrate_from,
int test_bitrate_to,
int test_bitrate_step,
int min_bwe,
int start_bwe,
int max_bwe,
TaskQueueBase* task_queue)
: EndToEndTest(),
test_bitrate_from_(test_bitrate_from),
test_bitrate_to_(test_bitrate_to),
test_bitrate_step_(test_bitrate_step),
min_bwe_(min_bwe),
start_bwe_(start_bwe),
max_bwe_(max_bwe),
task_queue_(task_queue) {}
protected:
BuiltInNetworkBehaviorConfig GetFakeNetworkPipeConfig() {
BuiltInNetworkBehaviorConfig pipe_config;
pipe_config.link_capacity_kbps = test_bitrate_from_;
return pipe_config;
}
std::unique_ptr<test::PacketTransport> CreateSendTransport(
TaskQueueBase* task_queue,
Call* sender_call) override {
auto network =
std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig());
send_simulated_network_ = network.get();
return std::make_unique<test::PacketTransport>(
task_queue, sender_call, this, test::PacketTransport::kSender,
test::CallTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network)));
}
std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
TaskQueueBase* task_queue) override {
auto network =
std::make_unique<SimulatedNetwork>(GetFakeNetworkPipeConfig());
receive_simulated_network_ = network.get();
return std::make_unique<test::PacketTransport>(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
test::CallTest::payload_type_map_,
std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
std::move(network)));
}
void PerformTest() override {
// Quick test mode, just to exercise all the code paths without actually
// caring about performance measurements.
const bool quick_perf_test =
field_trial::IsEnabled("WebRTC-QuickPerfTest");
int last_passed_test_bitrate = -1;
for (int test_bitrate = test_bitrate_from_;
test_bitrate_from_ < test_bitrate_to_
? test_bitrate <= test_bitrate_to_
: test_bitrate >= test_bitrate_to_;
test_bitrate += test_bitrate_step_) {
BuiltInNetworkBehaviorConfig pipe_config;
pipe_config.link_capacity_kbps = test_bitrate;
send_simulated_network_->SetConfig(pipe_config);
receive_simulated_network_->SetConfig(pipe_config);
rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs
: kBitrateStabilizationMs);
int64_t avg_rtt = 0;
for (int i = 0; i < kBitrateMeasurements; i++) {
Call::Stats call_stats;
SendTask(RTC_FROM_HERE, task_queue_, [this, &call_stats]() {
call_stats = sender_call_->GetStats();
});
avg_rtt += call_stats.rtt_ms;
rtc::Thread::SleepMs(quick_perf_test ? kShortDelayMs
: kBitrateMeasurementMs);
}
avg_rtt = avg_rtt / kBitrateMeasurements;
if (avg_rtt > kMinGoodRttMs) {
break;
} else {
last_passed_test_bitrate = test_bitrate;
}
}
EXPECT_GT(last_passed_test_bitrate, -1)
<< "Minimum supported bitrate out of the test scope";
webrtc::test::PrintResult("min_test_bitrate_", "", "min_bitrate",
last_passed_test_bitrate, "kbps", false);
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
BitrateConstraints bitrate_config;
bitrate_config.min_bitrate_bps = min_bwe_;
bitrate_config.start_bitrate_bps = start_bwe_;
bitrate_config.max_bitrate_bps = max_bwe_;
sender_call->GetTransportControllerSend()->SetSdpBitrateParameters(
bitrate_config);
}