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