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(kDefaultTimeoutMs),
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_NOTREACHED();
86 break;
87 }
88
89 return SEND_PACKET;
90 }
91
92 void ModifyVideoConfigs(
93 VideoSendStream::Config* send_config,
94 std::vector<VideoReceiveStream::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 width,
142 int height,
143 const VideoEncoderConfig& encoder_config) override {
144 std::vector<VideoStream> streams =
145 test::CreateVideoStreams(width, 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(kDefaultTimeoutMs),
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 = kDefaultTimeoutMs * 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 RTC_FROM_HERE, task_queue(),
288 [this, &observer, &send_transport, &receive_transport, &one_stream,
289 use_rtx]() {
290 CreateCalls();
291
292 send_transport = std::make_unique<test::PacketTransport>(
293 task_queue(), sender_call_.get(), &observer,
294 test::PacketTransport::kSender, payload_type_map_,
295 std::make_unique<FakeNetworkPipe>(
296 Clock::GetRealTimeClock(),
297 std::make_unique<SimulatedNetwork>(
298 BuiltInNetworkBehaviorConfig())));
299 receive_transport = std::make_unique<test::PacketTransport>(
300 task_queue(), nullptr, &observer, test::PacketTransport::kReceiver,
301 payload_type_map_,
302 std::make_unique<FakeNetworkPipe>(
303 Clock::GetRealTimeClock(),
304 std::make_unique<SimulatedNetwork>(
305 BuiltInNetworkBehaviorConfig())));
306 send_transport->SetReceiver(receiver_call_->Receiver());
307 receive_transport->SetReceiver(sender_call_->Receiver());
308
309 CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport.get());
310
311 if (use_rtx) {
312 for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
313 GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
314 }
315 GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
316 }
317
318 GetVideoEncoderConfig()->video_stream_factory =
319 new rtc::RefCountedObject<VideoStreamFactory>();
320 // Use the same total bitrates when sending a single stream to avoid
321 // lowering the bitrate estimate and requiring a subsequent rampup.
322 one_stream = GetVideoEncoderConfig()->Copy();
323 // one_stream.streams.resize(1);
324 one_stream.number_of_streams = 1;
325 CreateMatchingReceiveConfigs(receive_transport.get());
326
327 CreateVideoStreams();
328 CreateFrameGeneratorCapturer(30, 1280, 720);
329
330 Start();
331 });
332
333 EXPECT_TRUE(observer.Wait())
334 << "Timed out waiting for all SSRCs to send packets.";
335
336 // Test stream resetting more than once to make sure that the state doesn't
337 // get set once (this could be due to using std::map::insert for instance).
338 for (size_t i = 0; i < 3; ++i) {
339 SendTask(RTC_FROM_HERE, task_queue(), [&]() {
340 DestroyVideoSendStreams();
341
342 // Re-create VideoSendStream with only one stream.
343 CreateVideoSendStream(one_stream);
344 GetVideoSendStream()->Start();
345 if (provoke_rtcpsr_before_rtp) {
346 // Rapid Resync Request forces sending RTCP Sender Report back.
347 // Using this request speeds up this test because then there is no need
348 // to wait for a second for periodic Sender Report.
349 rtcp::RapidResyncRequest force_send_sr_back_request;
350 rtc::Buffer packet = force_send_sr_back_request.Build();
351 static_cast<webrtc::test::DirectTransport*>(receive_transport.get())
352 ->SendRtcp(packet.data(), packet.size());
353 }
354 CreateFrameGeneratorCapturer(30, 1280, 720);
355 });
356
357 observer.ResetExpectedSsrcs(1);
358 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
359
360 // Reconfigure back to use all streams.
361 SendTask(RTC_FROM_HERE, task_queue(), [this]() {
362 GetVideoSendStream()->ReconfigureVideoEncoder(
363 GetVideoEncoderConfig()->Copy());
364 });
365 observer.ResetExpectedSsrcs(kNumSimulcastStreams);
366 EXPECT_TRUE(observer.Wait())
367 << "Timed out waiting for all SSRCs to send packets.";
368
369 // Reconfigure down to one stream.
370 SendTask(RTC_FROM_HERE, task_queue(), [this, &one_stream]() {
371 GetVideoSendStream()->ReconfigureVideoEncoder(one_stream.Copy());
372 });
373 observer.ResetExpectedSsrcs(1);
374 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
375
376 // Reconfigure back to use all streams.
377 SendTask(RTC_FROM_HERE, task_queue(), [this]() {
378 GetVideoSendStream()->ReconfigureVideoEncoder(
379 GetVideoEncoderConfig()->Copy());
380 });
381 observer.ResetExpectedSsrcs(kNumSimulcastStreams);
382 EXPECT_TRUE(observer.Wait())
383 << "Timed out waiting for all SSRCs to send packets.";
384 }
385
386 SendTask(RTC_FROM_HERE, task_queue(),
387 [this, &send_transport, &receive_transport]() {
388 Stop();
389 DestroyStreams();
390 send_transport.reset();
391 receive_transport.reset();
392 DestroyCalls();
393 });
394 }
395
TEST_F(RtpRtcpEndToEndTest,RestartingSendStreamPreservesRtpState)396 TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpState) {
397 TestRtpStatePreservation(false, false);
398 }
399
TEST_F(RtpRtcpEndToEndTest,RestartingSendStreamPreservesRtpStatesWithRtx)400 TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
401 TestRtpStatePreservation(true, false);
402 }
403
TEST_F(RtpRtcpEndToEndTest,RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced)404 TEST_F(RtpRtcpEndToEndTest,
405 RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
406 TestRtpStatePreservation(true, true);
407 }
408
409 // See https://bugs.chromium.org/p/webrtc/issues/detail?id=9648.
TEST_F(RtpRtcpEndToEndTest,DISABLED_TestFlexfecRtpStatePreservation)410 TEST_F(RtpRtcpEndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
411 class RtpSequenceObserver : public test::RtpRtcpObserver {
412 public:
413 RtpSequenceObserver()
414 : test::RtpRtcpObserver(kDefaultTimeoutMs),
415 num_flexfec_packets_sent_(0) {}
416
417 void ResetPacketCount() {
418 MutexLock lock(&mutex_);
419 num_flexfec_packets_sent_ = 0;
420 }
421
422 private:
423 Action OnSendRtp(const uint8_t* packet, size_t length) override {
424 MutexLock lock(&mutex_);
425
426 RtpPacket rtp_packet;
427 EXPECT_TRUE(rtp_packet.Parse(packet, length));
428 const uint16_t sequence_number = rtp_packet.SequenceNumber();
429 const uint32_t timestamp = rtp_packet.Timestamp();
430 const uint32_t ssrc = rtp_packet.Ssrc();
431
432 if (ssrc == kVideoSendSsrcs[0] || ssrc == kSendRtxSsrcs[0]) {
433 return SEND_PACKET;
434 }
435 EXPECT_EQ(kFlexfecSendSsrc, ssrc) << "Unknown SSRC sent.";
436
437 ++num_flexfec_packets_sent_;
438
439 // If this is the first packet, we have nothing to compare to.
440 if (!last_observed_sequence_number_) {
441 last_observed_sequence_number_.emplace(sequence_number);
442 last_observed_timestamp_.emplace(timestamp);
443
444 return SEND_PACKET;
445 }
446
447 // Verify continuity and monotonicity of RTP sequence numbers.
448 EXPECT_EQ(static_cast<uint16_t>(*last_observed_sequence_number_ + 1),
449 sequence_number);
450 last_observed_sequence_number_.emplace(sequence_number);
451
452 // Timestamps should be non-decreasing...
453 const bool timestamp_is_same_or_newer =
454 timestamp == *last_observed_timestamp_ ||
455 IsNewerTimestamp(timestamp, *last_observed_timestamp_);
456 EXPECT_TRUE(timestamp_is_same_or_newer);
457 // ...but reasonably close in time.
458 const int k10SecondsInRtpTimestampBase = 10 * kVideoPayloadTypeFrequency;
459 EXPECT_TRUE(IsNewerTimestamp(
460 *last_observed_timestamp_ + k10SecondsInRtpTimestampBase, timestamp));
461 last_observed_timestamp_.emplace(timestamp);
462
463 // Pass test when enough packets have been let through.
464 if (num_flexfec_packets_sent_ >= 10) {
465 observation_complete_.Set();
466 }
467
468 return SEND_PACKET;
469 }
470
471 absl::optional<uint16_t> last_observed_sequence_number_
472 RTC_GUARDED_BY(mutex_);
473 absl::optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(mutex_);
474 size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(mutex_);
475 Mutex mutex_;
476 } observer;
477
478 static constexpr int kFrameMaxWidth = 320;
479 static constexpr int kFrameMaxHeight = 180;
480 static constexpr int kFrameRate = 15;
481
482 std::unique_ptr<test::PacketTransport> send_transport;
483 std::unique_ptr<test::PacketTransport> receive_transport;
484 test::FunctionVideoEncoderFactory encoder_factory(
485 []() { return VP8Encoder::Create(); });
486
487 SendTask(RTC_FROM_HERE, task_queue(), [&]() {
488 CreateCalls();
489
490 BuiltInNetworkBehaviorConfig lossy_delayed_link;
491 lossy_delayed_link.loss_percent = 2;
492 lossy_delayed_link.queue_delay_ms = 50;
493
494 send_transport = std::make_unique<test::PacketTransport>(
495 task_queue(), sender_call_.get(), &observer,
496 test::PacketTransport::kSender, payload_type_map_,
497 std::make_unique<FakeNetworkPipe>(
498 Clock::GetRealTimeClock(),
499 std::make_unique<SimulatedNetwork>(lossy_delayed_link)));
500 send_transport->SetReceiver(receiver_call_->Receiver());
501
502 BuiltInNetworkBehaviorConfig flawless_link;
503 receive_transport = std::make_unique<test::PacketTransport>(
504 task_queue(), nullptr, &observer, test::PacketTransport::kReceiver,
505 payload_type_map_,
506 std::make_unique<FakeNetworkPipe>(
507 Clock::GetRealTimeClock(),
508 std::make_unique<SimulatedNetwork>(flawless_link)));
509 receive_transport->SetReceiver(sender_call_->Receiver());
510
511 // For reduced flakyness, we use a real VP8 encoder together with NACK
512 // and RTX.
513 const int kNumVideoStreams = 1;
514 const int kNumFlexfecStreams = 1;
515 CreateSendConfig(kNumVideoStreams, 0, kNumFlexfecStreams,
516 send_transport.get());
517
518 GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
519 GetVideoSendConfig()->rtp.payload_name = "VP8";
520 GetVideoSendConfig()->rtp.payload_type = kVideoSendPayloadType;
521 GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
522 GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
523 GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
524 GetVideoEncoderConfig()->codec_type = kVideoCodecVP8;
525
526 CreateMatchingReceiveConfigs(receive_transport.get());
527 video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
528 video_receive_configs_[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
529 video_receive_configs_[0]
530 .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
531 kVideoSendPayloadType;
532
533 // The matching FlexFEC receive config is not created by
534 // CreateMatchingReceiveConfigs since this is not a test::BaseTest.
535 // Set up the receive config manually instead.
536 FlexfecReceiveStream::Config flexfec_receive_config(
537 receive_transport.get());
538 flexfec_receive_config.payload_type =
539 GetVideoSendConfig()->rtp.flexfec.payload_type;
540 flexfec_receive_config.remote_ssrc = GetVideoSendConfig()->rtp.flexfec.ssrc;
541 flexfec_receive_config.protected_media_ssrcs =
542 GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs;
543 flexfec_receive_config.local_ssrc = kReceiverLocalVideoSsrc;
544 flexfec_receive_config.transport_cc = true;
545 flexfec_receive_config.rtp_header_extensions.emplace_back(
546 RtpExtension::kTransportSequenceNumberUri,
547 kTransportSequenceNumberExtensionId);
548 flexfec_receive_configs_.push_back(flexfec_receive_config);
549
550 CreateFlexfecStreams();
551 CreateVideoStreams();
552
553 // RTCP might be disabled if the network is "down".
554 sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
555 receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
556
557 CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
558
559 Start();
560 });
561
562 // Initial test.
563 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
564
565 SendTask(RTC_FROM_HERE, task_queue(), [this, &observer]() {
566 // Ensure monotonicity when the VideoSendStream is restarted.
567 Stop();
568 observer.ResetPacketCount();
569 Start();
570 });
571
572 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
573
574 SendTask(RTC_FROM_HERE, task_queue(), [this, &observer]() {
575 // Ensure monotonicity when the VideoSendStream is recreated.
576 DestroyVideoSendStreams();
577 observer.ResetPacketCount();
578 CreateVideoSendStreams();
579 GetVideoSendStream()->Start();
580 CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
581 });
582
583 EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
584
585 // Cleanup.
586 SendTask(RTC_FROM_HERE, task_queue(),
587 [this, &send_transport, &receive_transport]() {
588 Stop();
589 DestroyStreams();
590 send_transport.reset();
591 receive_transport.reset();
592 DestroyCalls();
593 });
594 }
595 } // namespace webrtc
596