1 /*
2 * Copyright 2018 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 <memory>
12
13 #include "api/test/simulated_network.h"
14 #include "call/fake_network_pipe.h"
15 #include "call/simulated_network.h"
16 #include "modules/include/module_common_types_public.h"
17 #include "modules/rtp_rtcp/source/rtp_packet.h"
18 #include "modules/video_coding/codecs/vp8/include/vp8.h"
19 #include "rtc_base/synchronization/mutex.h"
20 #include "rtc_base/task_queue_for_test.h"
21 #include "test/call_test.h"
22 #include "test/gtest.h"
23 #include "test/rtcp_packet_parser.h"
24
25 namespace webrtc {
26 namespace {
27 enum : int { // The first valid value is 1.
28 kTransportSequenceNumberExtensionId = 1,
29 };
30 } // namespace
31
32 class RtpRtcpEndToEndTest : public test::CallTest {
33 protected:
34 void RespectsRtcpMode(RtcpMode rtcp_mode);
35 void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp);
36 };
37
RespectsRtcpMode(RtcpMode rtcp_mode)38 void RtpRtcpEndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
39 static const int kNumCompoundRtcpPacketsToObserve = 10;
40 class RtcpModeObserver : public test::EndToEndTest {
41 public:
42 explicit RtcpModeObserver(RtcpMode rtcp_mode)
43 : EndToEndTest(kDefaultTimeout),
44 rtcp_mode_(rtcp_mode),
45 sent_rtp_(0),
46 sent_rtcp_(0) {}
47
48 private:
49 Action OnSendRtp(const uint8_t* packet, size_t length) override {
50 MutexLock lock(&mutex_);
51 if (++sent_rtp_ % 3 == 0)
52 return DROP_PACKET;
53
54 return SEND_PACKET;
55 }
56
57 Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
58 MutexLock lock(&mutex_);
59 ++sent_rtcp_;
60 test::RtcpPacketParser parser;
61 EXPECT_TRUE(parser.Parse(packet, length));
62
63 EXPECT_EQ(0, parser.sender_report()->num_packets());
64
65 switch (rtcp_mode_) {
66 case RtcpMode::kCompound:
67 // TODO(holmer): We shouldn't send transport feedback alone if
68 // compound RTCP is negotiated.
69 if (parser.receiver_report()->num_packets() == 0 &&
70 parser.transport_feedback()->num_packets() == 0) {
71 ADD_FAILURE() << "Received RTCP packet without receiver report for "
72 "RtcpMode::kCompound.";
73 observation_complete_.Set();
74 }
75
76 if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
77 observation_complete_.Set();
78
79 break;
80 case RtcpMode::kReducedSize:
81 if (parser.receiver_report()->num_packets() == 0)
82 observation_complete_.Set();
83 break;
84 case RtcpMode::kOff:
85 RTC_DCHECK_NOTREACHED();
86 break;
87 }
88
89 return SEND_PACKET;
90 }
91
92 void ModifyVideoConfigs(
93 VideoSendStream::Config* send_config,
94 std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
95 VideoEncoderConfig* encoder_config) override {
96 send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
97 (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
98 (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
99 }
100
101 void PerformTest() override {
102 EXPECT_TRUE(Wait())
103 << (rtcp_mode_ == RtcpMode::kCompound
104 ? "Timed out before observing enough compound packets."
105 : "Timed out before receiving a non-compound RTCP packet.");
106 }
107
108 RtcpMode rtcp_mode_;
109 Mutex mutex_;
110 // Must be protected since RTCP can be sent by both the process thread
111 // and the pacer thread.
112 int sent_rtp_ RTC_GUARDED_BY(&mutex_);
113 int sent_rtcp_ RTC_GUARDED_BY(&mutex_);
114 } test(rtcp_mode);
115
116 RunBaseTest(&test);
117 }
118
TEST_F(RtpRtcpEndToEndTest,UsesRtcpCompoundMode)119 TEST_F(RtpRtcpEndToEndTest, UsesRtcpCompoundMode) {
120 RespectsRtcpMode(RtcpMode::kCompound);
121 }
122
TEST_F(RtpRtcpEndToEndTest,UsesRtcpReducedSizeMode)123 TEST_F(RtpRtcpEndToEndTest, UsesRtcpReducedSizeMode) {
124 RespectsRtcpMode(RtcpMode::kReducedSize);
125 }
126
TestRtpStatePreservation(bool use_rtx,bool provoke_rtcpsr_before_rtp)127 void RtpRtcpEndToEndTest::TestRtpStatePreservation(
128 bool use_rtx,
129 bool provoke_rtcpsr_before_rtp) {
130 // This test uses other VideoStream settings than the the default settings
131 // implemented in DefaultVideoStreamFactory. Therefore this test implements
132 // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
133 // in ModifyVideoConfigs.
134 class VideoStreamFactory
135 : public VideoEncoderConfig::VideoStreamFactoryInterface {
136 public:
137 VideoStreamFactory() {}
138
139 private:
140 std::vector<VideoStream> CreateEncoderStreams(
141 int frame_width,
142 int frame_height,
143 const VideoEncoderConfig& encoder_config) override {
144 std::vector<VideoStream> streams =
145 test::CreateVideoStreams(frame_width, frame_height, encoder_config);
146
147 if (encoder_config.number_of_streams > 1) {
148 // Lower bitrates so that all streams send initially.
149 RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
150 for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
151 streams[i].min_bitrate_bps = 10000;
152 streams[i].target_bitrate_bps = 15000;
153 streams[i].max_bitrate_bps = 20000;
154 }
155 } else {
156 // Use the same total bitrates when sending a single stream to avoid
157 // lowering
158 // the bitrate estimate and requiring a subsequent rampup.
159 streams[0].min_bitrate_bps = 3 * 10000;
160 streams[0].target_bitrate_bps = 3 * 15000;
161 streams[0].max_bitrate_bps = 3 * 20000;
162 }
163 return streams;
164 }
165 };
166
167 class RtpSequenceObserver : public test::RtpRtcpObserver {
168 public:
169 explicit RtpSequenceObserver(bool use_rtx)
170 : test::RtpRtcpObserver(kDefaultTimeout),
171 ssrcs_to_observe_(kNumSimulcastStreams) {
172 for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
173 ssrc_is_rtx_[kVideoSendSsrcs[i]] = false;
174 if (use_rtx)
175 ssrc_is_rtx_[kSendRtxSsrcs[i]] = true;
176 }
177 }
178
179 void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
180 MutexLock lock(&mutex_);
181 ssrc_observed_.clear();
182 ssrcs_to_observe_ = num_expected_ssrcs;
183 }
184
185 private:
186 void ValidateTimestampGap(uint32_t ssrc,
187 uint32_t timestamp,
188 bool only_padding)
189 RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_) {
190 static const int32_t kMaxTimestampGap = kDefaultTimeout.ms() * 90;
191 auto timestamp_it = last_observed_timestamp_.find(ssrc);
192 if (timestamp_it == last_observed_timestamp_.end()) {
193 EXPECT_FALSE(only_padding);
194 last_observed_timestamp_[ssrc] = timestamp;
195 } else {
196 // Verify timestamps are reasonably close.
197 uint32_t latest_observed = timestamp_it->second;
198 // Wraparound handling is unnecessary here as long as an int variable
199 // is used to store the result.
200 int32_t timestamp_gap = timestamp - latest_observed;
201 EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap)
202 << "Gap in timestamps (" << latest_observed << " -> " << timestamp
203 << ") too large for SSRC: " << ssrc << ".";
204 timestamp_it->second = timestamp;
205 }
206 }
207
208 Action OnSendRtp(const uint8_t* packet, size_t length) override {
209 RtpPacket rtp_packet;
210 EXPECT_TRUE(rtp_packet.Parse(packet, length));
211 const uint32_t ssrc = rtp_packet.Ssrc();
212 const int64_t sequence_number =
213 seq_numbers_unwrapper_.Unwrap(rtp_packet.SequenceNumber());
214 const uint32_t timestamp = rtp_packet.Timestamp();
215 const bool only_padding = rtp_packet.payload_size() == 0;
216
217 EXPECT_TRUE(ssrc_is_rtx_.find(ssrc) != ssrc_is_rtx_.end())
218 << "Received SSRC that wasn't configured: " << ssrc;
219
220 static const int64_t kMaxSequenceNumberGap = 100;
221 std::list<int64_t>* seq_numbers = &last_observed_seq_numbers_[ssrc];
222 if (seq_numbers->empty()) {
223 seq_numbers->push_back(sequence_number);
224 } else {
225 // We shouldn't get replays of previous sequence numbers.
226 for (int64_t observed : *seq_numbers) {
227 EXPECT_NE(observed, sequence_number)
228 << "Received sequence number " << sequence_number << " for SSRC "
229 << ssrc << " 2nd time.";
230 }
231 // Verify sequence numbers are reasonably close.
232 int64_t latest_observed = seq_numbers->back();
233 int64_t sequence_number_gap = sequence_number - latest_observed;
234 EXPECT_LE(std::abs(sequence_number_gap), kMaxSequenceNumberGap)
235 << "Gap in sequence numbers (" << latest_observed << " -> "
236 << sequence_number << ") too large for SSRC: " << ssrc << ".";
237 seq_numbers->push_back(sequence_number);
238 if (seq_numbers->size() >= kMaxSequenceNumberGap) {
239 seq_numbers->pop_front();
240 }
241 }
242
243 if (!ssrc_is_rtx_[ssrc]) {
244 MutexLock lock(&mutex_);
245 ValidateTimestampGap(ssrc, timestamp, only_padding);
246
247 // Wait for media packets on all ssrcs.
248 if (!ssrc_observed_[ssrc] && !only_padding) {
249 ssrc_observed_[ssrc] = true;
250 if (--ssrcs_to_observe_ == 0)
251 observation_complete_.Set();
252 }
253 }
254
255 return SEND_PACKET;
256 }
257
258 Action OnSendRtcp(const uint8_t* packet, size_t length) override {
259 test::RtcpPacketParser rtcp_parser;
260 rtcp_parser.Parse(packet, length);
261 if (rtcp_parser.sender_report()->num_packets() > 0) {
262 uint32_t ssrc = rtcp_parser.sender_report()->sender_ssrc();
263 uint32_t rtcp_timestamp = rtcp_parser.sender_report()->rtp_timestamp();
264
265 MutexLock lock(&mutex_);
266 ValidateTimestampGap(ssrc, rtcp_timestamp, false);
267 }
268 return SEND_PACKET;
269 }
270
271 SequenceNumberUnwrapper seq_numbers_unwrapper_;
272 std::map<uint32_t, std::list<int64_t>> last_observed_seq_numbers_;
273 std::map<uint32_t, uint32_t> last_observed_timestamp_;
274 std::map<uint32_t, bool> ssrc_is_rtx_;
275
276 Mutex mutex_;
277 size_t ssrcs_to_observe_ RTC_GUARDED_BY(mutex_);
278 std::map<uint32_t, bool> ssrc_observed_ RTC_GUARDED_BY(mutex_);
279 } observer(use_rtx);
280
281 std::unique_ptr<test::PacketTransport> send_transport;
282 std::unique_ptr<test::PacketTransport> receive_transport;
283
284 VideoEncoderConfig one_stream;
285
286 SendTask(
287 task_queue(), [this, &observer, &send_transport, &receive_transport,
288 &one_stream, use_rtx]() {
289 CreateCalls();
290
291 send_transport = std::make_unique<test::PacketTransport>(
292 task_queue(), sender_call_.get(), &observer,
293 test::PacketTransport::kSender, payload_type_map_,
294 std::make_unique<FakeNetworkPipe>(
295 Clock::GetRealTimeClock(),
296 std::make_unique<SimulatedNetwork>(
297 BuiltInNetworkBehaviorConfig())));
298 receive_transport = std::make_unique<test::PacketTransport>(
299 task_queue(), nullptr, &observer, test::PacketTransport::kReceiver,
300 payload_type_map_,
301 std::make_unique<FakeNetworkPipe>(
302 Clock::GetRealTimeClock(),
303 std::make_unique<SimulatedNetwork>(
304 BuiltInNetworkBehaviorConfig())));
305 send_transport->SetReceiver(receiver_call_->Receiver());
306 receive_transport->SetReceiver(sender_call_->Receiver());
307
308 CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport.get());
309
310 if (use_rtx) {
311 for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
312 GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
313 }
314 GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
315 }
316
317 GetVideoEncoderConfig()->video_stream_factory =
318 rtc::make_ref_counted<VideoStreamFactory>();
319 // Use the same total bitrates when sending a single stream to avoid
320 // lowering the bitrate estimate and requiring a subsequent rampup.
321 one_stream = GetVideoEncoderConfig()->Copy();
322 // one_stream.streams.resize(1);
323 one_stream.number_of_streams = 1;
324 CreateMatchingReceiveConfigs(receive_transport.get());
325
326 CreateVideoStreams();
327 CreateFrameGeneratorCapturer(30, 1280, 720);
328
329 Start();
330 });
331
332 EXPECT_TRUE(observer.Wait())
333 << "Timed out waiting for all SSRCs to send packets.";
334
335 // Test stream resetting more than once to make sure that the state doesn't
336 // get set once (this could be due to using std::map::insert for instance).
337 for (size_t i = 0; i < 3; ++i) {
338 SendTask(task_queue(), [&]() {
339 DestroyVideoSendStreams();
340
341 // Re-create VideoSendStream with only one stream.
342 CreateVideoSendStream(one_stream);
343 GetVideoSendStream()->Start();
344 if (provoke_rtcpsr_before_rtp) {
345 // Rapid Resync Request forces sending RTCP Sender Report back.
346 // Using this request speeds up this test because then there is no need
347 // to wait for a second for periodic Sender Report.
348 rtcp::RapidResyncRequest force_send_sr_back_request;
349 rtc::Buffer packet = force_send_sr_back_request.Build();
350 static_cast<webrtc::test::DirectTransport*>(receive_transport.get())
351 ->SendRtcp(packet.data(), packet.size());
352 }
353 CreateFrameGeneratorCapturer(30, 1280, 720);
354 });
355
356 observer.ResetExpectedSsrcs(1);
357 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
358
359 // Reconfigure back to use all streams.
360 SendTask(task_queue(), [this]() {
361 GetVideoSendStream()->ReconfigureVideoEncoder(
362 GetVideoEncoderConfig()->Copy());
363 });
364 observer.ResetExpectedSsrcs(kNumSimulcastStreams);
365 EXPECT_TRUE(observer.Wait())
366 << "Timed out waiting for all SSRCs to send packets.";
367
368 // Reconfigure down to one stream.
369 SendTask(task_queue(), [this, &one_stream]() {
370 GetVideoSendStream()->ReconfigureVideoEncoder(one_stream.Copy());
371 });
372 observer.ResetExpectedSsrcs(1);
373 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
374
375 // Reconfigure back to use all streams.
376 SendTask(task_queue(), [this]() {
377 GetVideoSendStream()->ReconfigureVideoEncoder(
378 GetVideoEncoderConfig()->Copy());
379 });
380 observer.ResetExpectedSsrcs(kNumSimulcastStreams);
381 EXPECT_TRUE(observer.Wait())
382 << "Timed out waiting for all SSRCs to send packets.";
383 }
384
385 SendTask(task_queue(), [this, &send_transport, &receive_transport]() {
386 Stop();
387 DestroyStreams();
388 send_transport.reset();
389 receive_transport.reset();
390 DestroyCalls();
391 });
392 }
393
TEST_F(RtpRtcpEndToEndTest,RestartingSendStreamPreservesRtpState)394 TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpState) {
395 TestRtpStatePreservation(false, false);
396 }
397
TEST_F(RtpRtcpEndToEndTest,RestartingSendStreamPreservesRtpStatesWithRtx)398 TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
399 TestRtpStatePreservation(true, false);
400 }
401
TEST_F(RtpRtcpEndToEndTest,RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced)402 TEST_F(RtpRtcpEndToEndTest,
403 RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
404 TestRtpStatePreservation(true, true);
405 }
406
407 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=9648.
TEST_F(RtpRtcpEndToEndTest,DISABLED_TestFlexfecRtpStatePreservation)408 TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
409 class RtpSequenceObserver : public test::RtpRtcpObserver {
410 public:
411 RtpSequenceObserver()
412 : test::RtpRtcpObserver(kDefaultTimeout),
413 num_flexfec_packets_sent_(0) {}
414
415 void ResetPacketCount() {
416 MutexLock lock(&mutex_);
417 num_flexfec_packets_sent_ = 0;
418 }
419
420 private:
421 Action OnSendRtp(const uint8_t* packet, size_t length) override {
422 MutexLock lock(&mutex_);
423
424 RtpPacket rtp_packet;
425 EXPECT_TRUE(rtp_packet.Parse(packet, length));
426 const uint16_t sequence_number = rtp_packet.SequenceNumber();
427 const uint32_t timestamp = rtp_packet.Timestamp();
428 const uint32_t ssrc = rtp_packet.Ssrc();
429
430 if (ssrc == kVideoSendSsrcs[0] || ssrc == kSendRtxSsrcs[0]) {
431 return SEND_PACKET;
432 }
433 EXPECT_EQ(kFlexfecSendSsrc, ssrc) << "Unknown SSRC sent.";
434
435 ++num_flexfec_packets_sent_;
436
437 // If this is the first packet, we have nothing to compare to.
438 if (!last_observed_sequence_number_) {
439 last_observed_sequence_number_.emplace(sequence_number);
440 last_observed_timestamp_.emplace(timestamp);
441
442 return SEND_PACKET;
443 }
444
445 // Verify continuity and monotonicity of RTP sequence numbers.
446 EXPECT_EQ(static_cast<uint16_t>(*last_observed_sequence_number_ + 1),
447 sequence_number);
448 last_observed_sequence_number_.emplace(sequence_number);
449
450 // Timestamps should be non-decreasing...
451 const bool timestamp_is_same_or_newer =
452 timestamp == *last_observed_timestamp_ ||
453 IsNewerTimestamp(timestamp, *last_observed_timestamp_);
454 EXPECT_TRUE(timestamp_is_same_or_newer);
455 // ...but reasonably close in time.
456 const int k10SecondsInRtpTimestampBase = 10 * kVideoPayloadTypeFrequency;
457 EXPECT_TRUE(IsNewerTimestamp(
458 *last_observed_timestamp_ + k10SecondsInRtpTimestampBase, timestamp));
459 last_observed_timestamp_.emplace(timestamp);
460
461 // Pass test when enough packets have been let through.
462 if (num_flexfec_packets_sent_ >= 10) {
463 observation_complete_.Set();
464 }
465
466 return SEND_PACKET;
467 }
468
469 absl::optional<uint16_t> last_observed_sequence_number_
470 RTC_GUARDED_BY(mutex_);
471 absl::optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(mutex_);
472 size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(mutex_);
473 Mutex mutex_;
474 } observer;
475
476 static constexpr int kFrameMaxWidth = 320;
477 static constexpr int kFrameMaxHeight = 180;
478 static constexpr int kFrameRate = 15;
479
480 std::unique_ptr<test::PacketTransport> send_transport;
481 std::unique_ptr<test::PacketTransport> receive_transport;
482 test::FunctionVideoEncoderFactory encoder_factory(
483 []() { return VP8Encoder::Create(); });
484
485 SendTask(task_queue(), [&]() {
486 CreateCalls();
487
488 BuiltInNetworkBehaviorConfig lossy_delayed_link;
489 lossy_delayed_link.loss_percent = 2;
490 lossy_delayed_link.queue_delay_ms = 50;
491
492 send_transport = std::make_unique<test::PacketTransport>(
493 task_queue(), sender_call_.get(), &observer,
494 test::PacketTransport::kSender, payload_type_map_,
495 std::make_unique<FakeNetworkPipe>(
496 Clock::GetRealTimeClock(),
497 std::make_unique<SimulatedNetwork>(lossy_delayed_link)));
498 send_transport->SetReceiver(receiver_call_->Receiver());
499
500 BuiltInNetworkBehaviorConfig flawless_link;
501 receive_transport = std::make_unique<test::PacketTransport>(
502 task_queue(), nullptr, &observer, test::PacketTransport::kReceiver,
503 payload_type_map_,
504 std::make_unique<FakeNetworkPipe>(
505 Clock::GetRealTimeClock(),
506 std::make_unique<SimulatedNetwork>(flawless_link)));
507 receive_transport->SetReceiver(sender_call_->Receiver());
508
509 // For reduced flakyness, we use a real VP8 encoder together with NACK
510 // and RTX.
511 const int kNumVideoStreams = 1;
512 const int kNumFlexfecStreams = 1;
513 CreateSendConfig(kNumVideoStreams, 0, kNumFlexfecStreams,
514 send_transport.get());
515
516 GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
517 GetVideoSendConfig()->rtp.payload_name = "VP8";
518 GetVideoSendConfig()->rtp.payload_type = kVideoSendPayloadType;
519 GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
520 GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
521 GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
522 GetVideoEncoderConfig()->codec_type = kVideoCodecVP8;
523
524 CreateMatchingReceiveConfigs(receive_transport.get());
525 video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
526 video_receive_configs_[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
527 video_receive_configs_[0]
528 .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
529 kVideoSendPayloadType;
530
531 // The matching FlexFEC receive config is not created by
532 // CreateMatchingReceiveConfigs since this is not a test::BaseTest.
533 // Set up the receive config manually instead.
534 FlexfecReceiveStream::Config flexfec_receive_config(
535 receive_transport.get());
536 flexfec_receive_config.payload_type =
537 GetVideoSendConfig()->rtp.flexfec.payload_type;
538 flexfec_receive_config.rtp.remote_ssrc =
539 GetVideoSendConfig()->rtp.flexfec.ssrc;
540 flexfec_receive_config.protected_media_ssrcs =
541 GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs;
542 flexfec_receive_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
543 flexfec_receive_config.rtp.transport_cc = true;
544 flexfec_receive_config.rtp.extensions.emplace_back(
545 RtpExtension::kTransportSequenceNumberUri,
546 kTransportSequenceNumberExtensionId);
547 flexfec_receive_configs_.push_back(flexfec_receive_config);
548
549 CreateFlexfecStreams();
550 CreateVideoStreams();
551
552 // RTCP might be disabled if the network is "down".
553 sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
554 receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
555
556 CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
557
558 Start();
559 });
560
561 // Initial test.
562 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
563
564 SendTask(task_queue(), [this, &observer]() {
565 // Ensure monotonicity when the VideoSendStream is restarted.
566 Stop();
567 observer.ResetPacketCount();
568 Start();
569 });
570
571 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
572
573 SendTask(task_queue(), [this, &observer]() {
574 // Ensure monotonicity when the VideoSendStream is recreated.
575 DestroyVideoSendStreams();
576 observer.ResetPacketCount();
577 CreateVideoSendStreams();
578 GetVideoSendStream()->Start();
579 CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
580 });
581
582 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
583
584 // Cleanup.
585 SendTask(task_queue(), [this, &send_transport, &receive_transport]() {
586 Stop();
587 DestroyStreams();
588 send_transport.reset();
589 receive_transport.reset();
590 DestroyCalls();
591 });
592 }
593 } // namespace webrtc
594