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1 /* | |
2 * Copyright (c) 2013 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 #include <algorithm> | |
11 #include <sstream> | |
12 #include <string> | |
13 | |
14 #include "testing/gtest/include/gtest/gtest.h" | |
15 | |
16 #include "webrtc/base/checks.h" | |
17 #include "webrtc/base/scoped_ptr.h" | |
18 #include "webrtc/base/thread_annotations.h" | |
19 #include "webrtc/call.h" | |
20 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h" | |
21 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" | |
22 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" | |
23 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" | |
24 #include "webrtc/system_wrappers/interface/rtp_to_ntp.h" | |
25 #include "webrtc/test/call_test.h" | |
26 #include "webrtc/test/direct_transport.h" | |
27 #include "webrtc/test/encoder_settings.h" | |
28 #include "webrtc/test/fake_audio_device.h" | |
29 #include "webrtc/test/fake_decoder.h" | |
30 #include "webrtc/test/fake_encoder.h" | |
31 #include "webrtc/test/frame_generator.h" | |
32 #include "webrtc/test/frame_generator_capturer.h" | |
33 #include "webrtc/test/rtp_rtcp_observer.h" | |
34 #include "webrtc/test/testsupport/fileutils.h" | |
35 #include "webrtc/test/testsupport/perf_test.h" | |
36 #include "webrtc/video/transport_adapter.h" | |
37 #include "webrtc/voice_engine/include/voe_base.h" | |
38 #include "webrtc/voice_engine/include/voe_codec.h" | |
39 #include "webrtc/voice_engine/include/voe_network.h" | |
40 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h" | |
41 #include "webrtc/voice_engine/include/voe_video_sync.h" | |
42 | |
43 namespace webrtc { | |
44 | |
45 class CallPerfTest : public test::CallTest { | |
46 protected: | |
47 void TestAudioVideoSync(bool fec, bool create_audio_first); | |
48 | |
49 void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms); | |
50 | |
51 void TestMinTransmitBitrate(bool pad_to_min_bitrate); | |
52 | |
53 void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, | |
54 int threshold_ms, | |
55 int start_time_ms, | |
56 int run_time_ms); | |
57 }; | |
58 | |
59 class SyncRtcpObserver : public test::RtpRtcpObserver { | |
60 public: | |
61 explicit SyncRtcpObserver(const FakeNetworkPipe::Config& config) | |
62 : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs, config) {} | |
63 | |
64 Action OnSendRtcp(const uint8_t* packet, size_t length) override { | |
65 RTCPUtility::RTCPParserV2 parser(packet, length, true); | |
66 EXPECT_TRUE(parser.IsValid()); | |
67 | |
68 for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin(); | |
69 packet_type != RTCPUtility::RTCPPacketTypes::kInvalid; | |
70 packet_type = parser.Iterate()) { | |
71 if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) { | |
72 const RTCPUtility::RTCPPacket& packet = parser.Packet(); | |
73 RtcpMeasurement ntp_rtp_pair( | |
74 packet.SR.NTPMostSignificant, | |
75 packet.SR.NTPLeastSignificant, | |
76 packet.SR.RTPTimestamp); | |
77 StoreNtpRtpPair(ntp_rtp_pair); | |
78 } | |
79 } | |
80 return SEND_PACKET; | |
81 } | |
82 | |
83 int64_t RtpTimestampToNtp(uint32_t timestamp) const { | |
84 rtc::CritScope lock(&crit_); | |
85 int64_t timestamp_in_ms = -1; | |
86 if (ntp_rtp_pairs_.size() == 2) { | |
87 // TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the | |
88 // RTCP sender where it sends RTCP SR before any RTP packets, which leads | |
89 // to a bogus NTP/RTP mapping. | |
90 RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms); | |
91 return timestamp_in_ms; | |
92 } | |
93 return -1; | |
94 } | |
95 | |
96 private: | |
97 void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) { | |
98 rtc::CritScope lock(&crit_); | |
99 for (RtcpList::iterator it = ntp_rtp_pairs_.begin(); | |
100 it != ntp_rtp_pairs_.end(); | |
101 ++it) { | |
102 if (ntp_rtp_pair.ntp_secs == it->ntp_secs && | |
103 ntp_rtp_pair.ntp_frac == it->ntp_frac) { | |
104 // This RTCP has already been added to the list. | |
105 return; | |
106 } | |
107 } | |
108 // We need two RTCP SR reports to map between RTP and NTP. More than two | |
109 // will not improve the mapping. | |
110 if (ntp_rtp_pairs_.size() == 2) { | |
111 ntp_rtp_pairs_.pop_back(); | |
112 } | |
113 ntp_rtp_pairs_.push_front(ntp_rtp_pair); | |
114 } | |
115 | |
116 mutable rtc::CriticalSection crit_; | |
117 RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_); | |
118 }; | |
119 | |
120 class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer { | |
121 static const int kInSyncThresholdMs = 50; | |
122 static const int kStartupTimeMs = 2000; | |
123 static const int kMinRunTimeMs = 30000; | |
124 | |
125 public: | |
126 VideoRtcpAndSyncObserver(Clock* clock, | |
127 int voe_channel, | |
128 VoEVideoSync* voe_sync, | |
129 SyncRtcpObserver* audio_observer) | |
130 : SyncRtcpObserver(FakeNetworkPipe::Config()), | |
131 clock_(clock), | |
132 voe_channel_(voe_channel), | |
133 voe_sync_(voe_sync), | |
134 audio_observer_(audio_observer), | |
135 creation_time_ms_(clock_->TimeInMilliseconds()), | |
136 first_time_in_sync_(-1) {} | |
137 | |
138 void RenderFrame(const VideoFrame& video_frame, | |
139 int time_to_render_ms) override { | |
140 int64_t now_ms = clock_->TimeInMilliseconds(); | |
141 uint32_t playout_timestamp = 0; | |
142 if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0) | |
143 return; | |
144 int64_t latest_audio_ntp = | |
145 audio_observer_->RtpTimestampToNtp(playout_timestamp); | |
146 int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp()); | |
147 if (latest_audio_ntp < 0 || latest_video_ntp < 0) | |
148 return; | |
149 int time_until_render_ms = | |
150 std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms)); | |
151 latest_video_ntp += time_until_render_ms; | |
152 int64_t stream_offset = latest_audio_ntp - latest_video_ntp; | |
153 std::stringstream ss; | |
154 ss << stream_offset; | |
155 webrtc::test::PrintResult("stream_offset", | |
156 "", | |
157 "synchronization", | |
158 ss.str(), | |
159 "ms", | |
160 false); | |
161 int64_t time_since_creation = now_ms - creation_time_ms_; | |
162 // During the first couple of seconds audio and video can falsely be | |
163 // estimated as being synchronized. We don't want to trigger on those. | |
164 if (time_since_creation < kStartupTimeMs) | |
165 return; | |
166 if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) { | |
167 if (first_time_in_sync_ == -1) { | |
168 first_time_in_sync_ = now_ms; | |
169 webrtc::test::PrintResult("sync_convergence_time", | |
170 "", | |
171 "synchronization", | |
172 time_since_creation, | |
173 "ms", | |
174 false); | |
175 } | |
176 if (time_since_creation > kMinRunTimeMs) | |
177 observation_complete_->Set(); | |
178 } | |
179 } | |
180 | |
181 bool IsTextureSupported() const override { return false; } | |
182 | |
183 private: | |
184 Clock* const clock_; | |
185 int voe_channel_; | |
186 VoEVideoSync* voe_sync_; | |
187 SyncRtcpObserver* audio_observer_; | |
188 int64_t creation_time_ms_; | |
189 int64_t first_time_in_sync_; | |
190 }; | |
191 | |
192 void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) { | |
193 const char* kSyncGroup = "av_sync"; | |
194 class AudioPacketReceiver : public PacketReceiver { | |
195 public: | |
196 AudioPacketReceiver(int channel, VoENetwork* voe_network) | |
197 : channel_(channel), | |
198 voe_network_(voe_network), | |
199 parser_(RtpHeaderParser::Create()) {} | |
200 DeliveryStatus DeliverPacket(MediaType media_type, | |
201 const uint8_t* packet, | |
202 size_t length, | |
203 const PacketTime& packet_time) override { | |
204 EXPECT_TRUE(media_type == MediaType::ANY || | |
205 media_type == MediaType::AUDIO); | |
206 int ret; | |
207 if (parser_->IsRtcp(packet, length)) { | |
208 ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length); | |
209 } else { | |
210 ret = voe_network_->ReceivedRTPPacket(channel_, packet, length, | |
211 PacketTime()); | |
212 } | |
213 return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR; | |
214 } | |
215 | |
216 private: | |
217 int channel_; | |
218 VoENetwork* voe_network_; | |
219 rtc::scoped_ptr<RtpHeaderParser> parser_; | |
220 }; | |
221 | |
222 VoiceEngine* voice_engine = VoiceEngine::Create(); | |
223 VoEBase* voe_base = VoEBase::GetInterface(voice_engine); | |
224 VoECodec* voe_codec = VoECodec::GetInterface(voice_engine); | |
225 VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine); | |
226 VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine); | |
227 const std::string audio_filename = | |
228 test::ResourcePath("voice_engine/audio_long16", "pcm"); | |
229 ASSERT_STRNE("", audio_filename.c_str()); | |
230 test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), | |
231 audio_filename); | |
232 EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr)); | |
233 int channel = voe_base->CreateChannel(); | |
234 | |
235 FakeNetworkPipe::Config net_config; | |
236 net_config.queue_delay_ms = 500; | |
237 net_config.loss_percent = 5; | |
238 SyncRtcpObserver audio_observer(net_config); | |
239 VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), | |
240 channel, | |
241 voe_sync, | |
242 &audio_observer); | |
243 | |
244 Call::Config receiver_config; | |
245 receiver_config.voice_engine = voice_engine; | |
246 CreateCalls(Call::Config(), receiver_config); | |
247 | |
248 CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000}; | |
249 EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac)); | |
250 | |
251 AudioPacketReceiver voe_packet_receiver(channel, voe_network); | |
252 audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver); | |
253 | |
254 internal::TransportAdapter transport_adapter(audio_observer.SendTransport()); | |
255 transport_adapter.Enable(); | |
256 EXPECT_EQ(0, | |
257 voe_network->RegisterExternalTransport(channel, transport_adapter)); | |
258 | |
259 observer.SetReceivers(receiver_call_->Receiver(), sender_call_->Receiver()); | |
260 | |
261 test::FakeDecoder fake_decoder; | |
262 | |
263 CreateSendConfig(1, observer.SendTransport()); | |
264 CreateMatchingReceiveConfigs(observer.ReceiveTransport()); | |
265 | |
266 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; | |
267 if (fec) { | |
268 send_config_.rtp.fec.red_payload_type = kRedPayloadType; | |
269 send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; | |
270 receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType; | |
271 receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType; | |
272 } | |
273 receive_configs_[0].rtp.nack.rtp_history_ms = 1000; | |
274 receive_configs_[0].renderer = &observer; | |
275 receive_configs_[0].sync_group = kSyncGroup; | |
276 | |
277 AudioReceiveStream::Config audio_config; | |
278 audio_config.voe_channel_id = channel; | |
279 audio_config.sync_group = kSyncGroup; | |
280 | |
281 AudioReceiveStream* audio_receive_stream = nullptr; | |
282 | |
283 if (create_audio_first) { | |
284 audio_receive_stream = | |
285 receiver_call_->CreateAudioReceiveStream(audio_config); | |
286 CreateStreams(); | |
287 } else { | |
288 CreateStreams(); | |
289 audio_receive_stream = | |
290 receiver_call_->CreateAudioReceiveStream(audio_config); | |
291 } | |
292 | |
293 CreateFrameGeneratorCapturer(); | |
294 | |
295 Start(); | |
296 | |
297 fake_audio_device.Start(); | |
298 EXPECT_EQ(0, voe_base->StartPlayout(channel)); | |
299 EXPECT_EQ(0, voe_base->StartReceive(channel)); | |
300 EXPECT_EQ(0, voe_base->StartSend(channel)); | |
301 | |
302 EXPECT_EQ(kEventSignaled, observer.Wait()) | |
303 << "Timed out while waiting for audio and video to be synchronized."; | |
304 | |
305 EXPECT_EQ(0, voe_base->StopSend(channel)); | |
306 EXPECT_EQ(0, voe_base->StopReceive(channel)); | |
307 EXPECT_EQ(0, voe_base->StopPlayout(channel)); | |
308 fake_audio_device.Stop(); | |
309 | |
310 Stop(); | |
311 observer.StopSending(); | |
312 audio_observer.StopSending(); | |
313 | |
314 voe_base->DeleteChannel(channel); | |
315 voe_base->Release(); | |
316 voe_codec->Release(); | |
317 voe_network->Release(); | |
318 voe_sync->Release(); | |
319 | |
320 DestroyStreams(); | |
321 | |
322 receiver_call_->DestroyAudioReceiveStream(audio_receive_stream); | |
323 | |
324 VoiceEngine::Delete(voice_engine); | |
325 } | |
326 | |
327 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) { | |
328 TestAudioVideoSync(false, true); | |
329 } | |
330 | |
331 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) { | |
332 TestAudioVideoSync(false, false); | |
333 } | |
334 | |
335 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) { | |
336 TestAudioVideoSync(true, false); | |
337 } | |
338 | |
339 void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config, | |
340 int threshold_ms, | |
341 int start_time_ms, | |
342 int run_time_ms) { | |
343 class CaptureNtpTimeObserver : public test::EndToEndTest, | |
344 public VideoRenderer { | |
345 public: | |
346 CaptureNtpTimeObserver(const FakeNetworkPipe::Config& config, | |
347 int threshold_ms, | |
348 int start_time_ms, | |
349 int run_time_ms) | |
350 : EndToEndTest(kLongTimeoutMs, config), | |
351 clock_(Clock::GetRealTimeClock()), | |
352 threshold_ms_(threshold_ms), | |
353 start_time_ms_(start_time_ms), | |
354 run_time_ms_(run_time_ms), | |
355 creation_time_ms_(clock_->TimeInMilliseconds()), | |
356 capturer_(nullptr), | |
357 rtp_start_timestamp_set_(false), | |
358 rtp_start_timestamp_(0) {} | |
359 | |
360 private: | |
361 void RenderFrame(const VideoFrame& video_frame, | |
362 int time_to_render_ms) override { | |
363 if (video_frame.ntp_time_ms() <= 0) { | |
364 // Haven't got enough RTCP SR in order to calculate the capture ntp | |
365 // time. | |
366 return; | |
367 } | |
368 | |
369 int64_t now_ms = clock_->TimeInMilliseconds(); | |
370 int64_t time_since_creation = now_ms - creation_time_ms_; | |
371 if (time_since_creation < start_time_ms_) { | |
372 // Wait for |start_time_ms_| before start measuring. | |
373 return; | |
374 } | |
375 | |
376 if (time_since_creation > run_time_ms_) { | |
377 observation_complete_->Set(); | |
378 } | |
379 | |
380 FrameCaptureTimeList::iterator iter = | |
381 capture_time_list_.find(video_frame.timestamp()); | |
382 EXPECT_TRUE(iter != capture_time_list_.end()); | |
383 | |
384 // The real capture time has been wrapped to uint32_t before converted | |
385 // to rtp timestamp in the sender side. So here we convert the estimated | |
386 // capture time to a uint32_t 90k timestamp also for comparing. | |
387 uint32_t estimated_capture_timestamp = | |
388 90 * static_cast<uint32_t>(video_frame.ntp_time_ms()); | |
389 uint32_t real_capture_timestamp = iter->second; | |
390 int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp; | |
391 time_offset_ms = time_offset_ms / 90; | |
392 std::stringstream ss; | |
393 ss << time_offset_ms; | |
394 | |
395 webrtc::test::PrintResult( | |
396 "capture_ntp_time", "", "real - estimated", ss.str(), "ms", true); | |
397 EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_); | |
398 } | |
399 | |
400 bool IsTextureSupported() const override { return false; } | |
401 | |
402 virtual Action OnSendRtp(const uint8_t* packet, size_t length) { | |
403 RTPHeader header; | |
404 EXPECT_TRUE(parser_->Parse(packet, length, &header)); | |
405 | |
406 if (!rtp_start_timestamp_set_) { | |
407 // Calculate the rtp timestamp offset in order to calculate the real | |
408 // capture time. | |
409 uint32_t first_capture_timestamp = | |
410 90 * static_cast<uint32_t>(capturer_->first_frame_capture_time()); | |
411 rtp_start_timestamp_ = header.timestamp - first_capture_timestamp; | |
412 rtp_start_timestamp_set_ = true; | |
413 } | |
414 | |
415 uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_; | |
416 capture_time_list_.insert( | |
417 capture_time_list_.end(), | |
418 std::make_pair(header.timestamp, capture_timestamp)); | |
419 return SEND_PACKET; | |
420 } | |
421 | |
422 void OnFrameGeneratorCapturerCreated( | |
423 test::FrameGeneratorCapturer* frame_generator_capturer) override { | |
424 capturer_ = frame_generator_capturer; | |
425 } | |
426 | |
427 void ModifyConfigs(VideoSendStream::Config* send_config, | |
428 std::vector<VideoReceiveStream::Config>* receive_configs, | |
429 VideoEncoderConfig* encoder_config) override { | |
430 (*receive_configs)[0].renderer = this; | |
431 // Enable the receiver side rtt calculation. | |
432 (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true; | |
433 } | |
434 | |
435 void PerformTest() override { | |
436 EXPECT_EQ(kEventSignaled, Wait()) << "Timed out while waiting for " | |
437 "estimated capture NTP time to be " | |
438 "within bounds."; | |
439 } | |
440 | |
441 Clock* clock_; | |
442 int threshold_ms_; | |
443 int start_time_ms_; | |
444 int run_time_ms_; | |
445 int64_t creation_time_ms_; | |
446 test::FrameGeneratorCapturer* capturer_; | |
447 bool rtp_start_timestamp_set_; | |
448 uint32_t rtp_start_timestamp_; | |
449 typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList; | |
450 FrameCaptureTimeList capture_time_list_; | |
451 } test(net_config, threshold_ms, start_time_ms, run_time_ms); | |
452 | |
453 RunBaseTest(&test); | |
454 } | |
455 | |
456 TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) { | |
457 FakeNetworkPipe::Config net_config; | |
458 net_config.queue_delay_ms = 100; | |
459 // TODO(wu): lower the threshold as the calculation/estimatation becomes more | |
460 // accurate. | |
461 const int kThresholdMs = 100; | |
462 const int kStartTimeMs = 10000; | |
463 const int kRunTimeMs = 20000; | |
464 TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs); | |
465 } | |
466 | |
467 TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) { | |
468 FakeNetworkPipe::Config net_config; | |
469 net_config.queue_delay_ms = 100; | |
470 net_config.delay_standard_deviation_ms = 10; | |
471 // TODO(wu): lower the threshold as the calculation/estimatation becomes more | |
472 // accurate. | |
473 const int kThresholdMs = 100; | |
474 const int kStartTimeMs = 10000; | |
475 const int kRunTimeMs = 20000; | |
476 TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs); | |
477 } | |
478 | |
479 void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load, | |
480 int encode_delay_ms) { | |
481 class LoadObserver : public test::SendTest, public webrtc::LoadObserver { | |
482 public: | |
483 LoadObserver(LoadObserver::Load tested_load, int encode_delay_ms) | |
484 : SendTest(kLongTimeoutMs), | |
485 tested_load_(tested_load), | |
486 encoder_(Clock::GetRealTimeClock(), encode_delay_ms) {} | |
487 | |
488 void OnLoadUpdate(Load load) override { | |
489 if (load == tested_load_) | |
490 observation_complete_->Set(); | |
491 } | |
492 | |
493 void ModifyConfigs(VideoSendStream::Config* send_config, | |
494 std::vector<VideoReceiveStream::Config>* receive_configs, | |
495 VideoEncoderConfig* encoder_config) override { | |
496 send_config->overuse_callback = this; | |
497 send_config->encoder_settings.encoder = &encoder_; | |
498 } | |
499 | |
500 void PerformTest() override { | |
501 EXPECT_EQ(kEventSignaled, Wait()) | |
502 << "Timed out before receiving an overuse callback."; | |
503 } | |
504 | |
505 LoadObserver::Load tested_load_; | |
506 test::DelayedEncoder encoder_; | |
507 } test(tested_load, encode_delay_ms); | |
508 | |
509 RunBaseTest(&test); | |
510 } | |
511 | |
512 TEST_F(CallPerfTest, ReceivesCpuUnderuse) { | |
513 const int kEncodeDelayMs = 2; | |
514 TestCpuOveruse(LoadObserver::kUnderuse, kEncodeDelayMs); | |
515 } | |
516 | |
517 TEST_F(CallPerfTest, ReceivesCpuOveruse) { | |
518 const int kEncodeDelayMs = 35; | |
519 TestCpuOveruse(LoadObserver::kOveruse, kEncodeDelayMs); | |
520 } | |
521 | |
522 void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) { | |
523 static const int kMaxEncodeBitrateKbps = 30; | |
524 static const int kMinTransmitBitrateBps = 150000; | |
525 static const int kMinAcceptableTransmitBitrate = 130; | |
526 static const int kMaxAcceptableTransmitBitrate = 170; | |
527 static const int kNumBitrateObservationsInRange = 100; | |
528 static const int kAcceptableBitrateErrorMargin = 15; // +- 7 | |
529 class BitrateObserver : public test::EndToEndTest, public PacketReceiver { | |
530 public: | |
531 explicit BitrateObserver(bool using_min_transmit_bitrate) | |
532 : EndToEndTest(kLongTimeoutMs), | |
533 send_stream_(nullptr), | |
534 send_transport_receiver_(nullptr), | |
535 pad_to_min_bitrate_(using_min_transmit_bitrate), | |
536 num_bitrate_observations_in_range_(0) {} | |
537 | |
538 private: | |
539 void SetReceivers(PacketReceiver* send_transport_receiver, | |
540 PacketReceiver* receive_transport_receiver) override { | |
541 send_transport_receiver_ = send_transport_receiver; | |
542 test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver); | |
543 } | |
544 | |
545 DeliveryStatus DeliverPacket(MediaType media_type, | |
546 const uint8_t* packet, | |
547 size_t length, | |
548 const PacketTime& packet_time) override { | |
549 VideoSendStream::Stats stats = send_stream_->GetStats(); | |
550 if (stats.substreams.size() > 0) { | |
551 RTC_DCHECK_EQ(1u, stats.substreams.size()); | |
552 int bitrate_kbps = | |
553 stats.substreams.begin()->second.total_bitrate_bps / 1000; | |
554 if (bitrate_kbps > 0) { | |
555 test::PrintResult( | |
556 "bitrate_stats_", | |
557 (pad_to_min_bitrate_ ? "min_transmit_bitrate" | |
558 : "without_min_transmit_bitrate"), | |
559 "bitrate_kbps", | |
560 static_cast<size_t>(bitrate_kbps), | |
561 "kbps", | |
562 false); | |
563 if (pad_to_min_bitrate_) { | |
564 if (bitrate_kbps > kMinAcceptableTransmitBitrate && | |
565 bitrate_kbps < kMaxAcceptableTransmitBitrate) { | |
566 ++num_bitrate_observations_in_range_; | |
567 } | |
568 } else { | |
569 // Expect bitrate stats to roughly match the max encode bitrate. | |
570 if (bitrate_kbps > (kMaxEncodeBitrateKbps - | |
571 kAcceptableBitrateErrorMargin / 2) && | |
572 bitrate_kbps < (kMaxEncodeBitrateKbps + | |
573 kAcceptableBitrateErrorMargin / 2)) { | |
574 ++num_bitrate_observations_in_range_; | |
575 } | |
576 } | |
577 if (num_bitrate_observations_in_range_ == | |
578 kNumBitrateObservationsInRange) | |
579 observation_complete_->Set(); | |
580 } | |
581 } | |
582 return send_transport_receiver_->DeliverPacket(media_type, packet, length, | |
583 packet_time); | |
584 } | |
585 | |
586 void OnStreamsCreated( | |
587 VideoSendStream* send_stream, | |
588 const std::vector<VideoReceiveStream*>& receive_streams) override { | |
589 send_stream_ = send_stream; | |
590 } | |
591 | |
592 void ModifyConfigs(VideoSendStream::Config* send_config, | |
593 std::vector<VideoReceiveStream::Config>* receive_configs, | |
594 VideoEncoderConfig* encoder_config) override { | |
595 if (pad_to_min_bitrate_) { | |
596 encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps; | |
597 } else { | |
598 RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps); | |
599 } | |
600 } | |
601 | |
602 void PerformTest() override { | |
603 EXPECT_EQ(kEventSignaled, Wait()) | |
604 << "Timeout while waiting for send-bitrate stats."; | |
605 } | |
606 | |
607 VideoSendStream* send_stream_; | |
608 PacketReceiver* send_transport_receiver_; | |
609 const bool pad_to_min_bitrate_; | |
610 int num_bitrate_observations_in_range_; | |
611 } test(pad_to_min_bitrate); | |
612 | |
613 fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps); | |
614 RunBaseTest(&test); | |
615 } | |
616 | |
617 TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); } | |
618 | |
619 TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) { | |
620 TestMinTransmitBitrate(false); | |
621 } | |
622 | |
623 TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) { | |
624 static const uint32_t kInitialBitrateKbps = 400; | |
625 static const uint32_t kReconfigureThresholdKbps = 600; | |
626 static const uint32_t kPermittedReconfiguredBitrateDiffKbps = 100; | |
627 | |
628 class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder { | |
629 public: | |
630 BitrateObserver() | |
631 : EndToEndTest(kDefaultTimeoutMs), | |
632 FakeEncoder(Clock::GetRealTimeClock()), | |
633 time_to_reconfigure_(webrtc::EventWrapper::Create()), | |
634 encoder_inits_(0), | |
635 last_set_bitrate_(0), | |
636 send_stream_(nullptr) {} | |
637 | |
638 int32_t InitEncode(const VideoCodec* config, | |
639 int32_t number_of_cores, | |
640 size_t max_payload_size) override { | |
641 if (encoder_inits_ == 0) { | |
642 EXPECT_EQ(kInitialBitrateKbps, config->startBitrate) | |
643 << "Encoder not initialized at expected bitrate."; | |
644 } | |
645 ++encoder_inits_; | |
646 if (encoder_inits_ == 2) { | |
647 EXPECT_GE(last_set_bitrate_, kReconfigureThresholdKbps); | |
648 EXPECT_NEAR(config->startBitrate, | |
649 last_set_bitrate_, | |
650 kPermittedReconfiguredBitrateDiffKbps) | |
651 << "Encoder reconfigured with bitrate too far away from last set."; | |
652 observation_complete_->Set(); | |
653 } | |
654 return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size); | |
655 } | |
656 | |
657 int32_t SetRates(uint32_t new_target_bitrate_kbps, | |
658 uint32_t framerate) override { | |
659 last_set_bitrate_ = new_target_bitrate_kbps; | |
660 if (encoder_inits_ == 1 && | |
661 new_target_bitrate_kbps > kReconfigureThresholdKbps) { | |
662 time_to_reconfigure_->Set(); | |
663 } | |
664 return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate); | |
665 } | |
666 | |
667 Call::Config GetSenderCallConfig() override { | |
668 Call::Config config = EndToEndTest::GetSenderCallConfig(); | |
669 config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000; | |
670 return config; | |
671 } | |
672 | |
673 void ModifyConfigs(VideoSendStream::Config* send_config, | |
674 std::vector<VideoReceiveStream::Config>* receive_configs, | |
675 VideoEncoderConfig* encoder_config) override { | |
676 send_config->encoder_settings.encoder = this; | |
677 encoder_config->streams[0].min_bitrate_bps = 50000; | |
678 encoder_config->streams[0].target_bitrate_bps = | |
679 encoder_config->streams[0].max_bitrate_bps = 2000000; | |
680 | |
681 encoder_config_ = *encoder_config; | |
682 } | |
683 | |
684 void OnStreamsCreated( | |
685 VideoSendStream* send_stream, | |
686 const std::vector<VideoReceiveStream*>& receive_streams) override { | |
687 send_stream_ = send_stream; | |
688 } | |
689 | |
690 void PerformTest() override { | |
691 ASSERT_EQ(kEventSignaled, time_to_reconfigure_->Wait(kDefaultTimeoutMs)) | |
692 << "Timed out before receiving an initial high bitrate."; | |
693 encoder_config_.streams[0].width *= 2; | |
694 encoder_config_.streams[0].height *= 2; | |
695 EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_)); | |
696 EXPECT_EQ(kEventSignaled, Wait()) | |
697 << "Timed out while waiting for a couple of high bitrate estimates " | |
698 "after reconfiguring the send stream."; | |
699 } | |
700 | |
701 private: | |
702 rtc::scoped_ptr<webrtc::EventWrapper> time_to_reconfigure_; | |
703 int encoder_inits_; | |
704 uint32_t last_set_bitrate_; | |
705 VideoSendStream* send_stream_; | |
706 VideoEncoderConfig encoder_config_; | |
707 } test; | |
708 | |
709 RunBaseTest(&test); | |
710 } | |
711 | |
712 } // namespace webrtc | |
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