video/end_to_end_tests/retransmission_tests.cc - src.git - Git at Google

 /*
  *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */

 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/call_test.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
 #include "test/rtcp_packet_parser.h"

 namespace webrtc {
 class RetransmissionEndToEndTest
     : public test::CallTest,
       public testing::WithParamInterface<std::string> {
  public:
   RetransmissionEndToEndTest() : field_trial_(GetParam()) {}

   virtual ~RetransmissionEndToEndTest() {
     EXPECT_EQ(nullptr, video_send_stream_);
     EXPECT_TRUE(video_receive_streams_.empty());
   }

  protected:
   void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
   void ReceivesPliAndRecovers(int rtp_history_ms);

  private:
  private:
   test::ScopedFieldTrials field_trial_;
 };

 INSTANTIATE_TEST_CASE_P(RoundRobin,
                         RetransmissionEndToEndTest,
                         ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
                                           "WebRTC-RoundRobinPacing/Enabled/"));

 TEST_P(RetransmissionEndToEndTest, ReceivesAndRetransmitsNack) {
   static const int kNumberOfNacksToObserve = 2;
   static const int kLossBurstSize = 2;
   static const int kPacketsBetweenLossBursts = 9;
   class NackObserver : public test::EndToEndTest {
    public:
     NackObserver()
         : EndToEndTest(kLongTimeoutMs),
           sent_rtp_packets_(0),
           packets_left_to_drop_(0),
           nacks_left_(kNumberOfNacksToObserve) {}

    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));

       // Never drop retransmitted packets.
       if (dropped_packets_.find(header.sequenceNumber) !=
           dropped_packets_.end()) {
         retransmitted_packets_.insert(header.sequenceNumber);
         return SEND_PACKET;
       }

       if (nacks_left_ <= 0 &&
           retransmitted_packets_.size() == dropped_packets_.size()) {
         observation_complete_.Set();
       }

       ++sent_rtp_packets_;

       // Enough NACKs received, stop dropping packets.
       if (nacks_left_ <= 0)
         return SEND_PACKET;

       // Check if it's time for a new loss burst.
       if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
         packets_left_to_drop_ = kLossBurstSize;

       // Never drop padding packets as those won't be retransmitted.
       if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
         --packets_left_to_drop_;
         dropped_packets_.insert(header.sequenceNumber);
         return DROP_PACKET;
       }

       return SEND_PACKET;
     }

     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet, length));
       nacks_left_ -= parser.nack()->num_packets();
       return SEND_PACKET;
     }

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     }

     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out waiting for packets to be NACKed, retransmitted and "
              "rendered.";
     }

     rtc::CriticalSection crit_;
     std::set<uint16_t> dropped_packets_;
     std::set<uint16_t> retransmitted_packets_;
     uint64_t sent_rtp_packets_;
     int packets_left_to_drop_;
     int nacks_left_ RTC_GUARDED_BY(&crit_);
   } test;

   RunBaseTest(&test);
 }

 TEST_P(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
   class NackObserver : public test::EndToEndTest {
    public:
     NackObserver()
         : EndToEndTest(kLongTimeoutMs),
           local_ssrc_(0),
           remote_ssrc_(0),
           receive_transport_(nullptr) {}

    private:
     size_t GetNumVideoStreams() const override { return 0; }
     size_t GetNumAudioStreams() const override { return 1; }

     test::PacketTransport* CreateReceiveTransport(
         test::SingleThreadedTaskQueueForTesting* task_queue) override {
       test::PacketTransport* receive_transport = new test::PacketTransport(
           task_queue, nullptr, this, test::PacketTransport::kReceiver,
           payload_type_map_, FakeNetworkPipe::Config());
       receive_transport_ = receive_transport;
       return receive_transport;
     }

     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));

       if (!sequence_number_to_retransmit_) {
         sequence_number_to_retransmit_ = header.sequenceNumber;

         // Don't ask for retransmission straight away, may be deduped in pacer.
       } else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
         observation_complete_.Set();
       } else {
         // Send a NACK as often as necessary until retransmission is received.
         rtcp::Nack nack;
         nack.SetSenderSsrc(local_ssrc_);
         nack.SetMediaSsrc(remote_ssrc_);
         uint16_t nack_list[] = {*sequence_number_to_retransmit_};
         nack.SetPacketIds(nack_list, 1);
         rtc::Buffer buffer = nack.Build();

         EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
       }

       return SEND_PACKET;
     }

     void ModifyAudioConfigs(
         AudioSendStream::Config* send_config,
         std::vector<AudioReceiveStream::Config>* receive_configs) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
       remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
     }

     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out waiting for packets to be NACKed, retransmitted and "
              "rendered.";
     }

     uint32_t local_ssrc_;
     uint32_t remote_ssrc_;
     Transport* receive_transport_;
     rtc::Optional<uint16_t> sequence_number_to_retransmit_;
   } test;

   RunBaseTest(&test);
 }

 TEST_P(RetransmissionEndToEndTest,
        StopSendingKeyframeRequestsForInactiveStream) {
   class KeyframeRequestObserver : public test::EndToEndTest {
    public:
     explicit KeyframeRequestObserver(
         test::SingleThreadedTaskQueueForTesting* task_queue)
         : clock_(Clock::GetRealTimeClock()), task_queue_(task_queue) {}

     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       RTC_DCHECK_EQ(1, receive_streams.size());
       send_stream_ = send_stream;
       receive_stream_ = receive_streams[0];
     }

     void PerformTest() override {
       bool frame_decoded = false;
       int64_t start_time = clock_->TimeInMilliseconds();
       while (clock_->TimeInMilliseconds() - start_time <= 5000) {
         if (receive_stream_->GetStats().frames_decoded > 0) {
           frame_decoded = true;
           break;
         }
         SleepMs(100);
       }
       ASSERT_TRUE(frame_decoded);
       task_queue_->SendTask([this]() { send_stream_->Stop(); });
       SleepMs(10000);
       ASSERT_EQ(
           1U, receive_stream_->GetStats().rtcp_packet_type_counts.pli_packets);
     }

    private:
     Clock* clock_;
     VideoSendStream* send_stream_;
     VideoReceiveStream* receive_stream_;
     test::SingleThreadedTaskQueueForTesting* const task_queue_;
   } test(&task_queue_);

   RunBaseTest(&test);
 }

 void RetransmissionEndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
   static const int kPacketsToDrop = 1;

   class PliObserver : public test::EndToEndTest,
                       public rtc::VideoSinkInterface<VideoFrame> {
    public:
     explicit PliObserver(int rtp_history_ms)
         : EndToEndTest(kLongTimeoutMs),
           rtp_history_ms_(rtp_history_ms),
           nack_enabled_(rtp_history_ms > 0),
           highest_dropped_timestamp_(0),
           frames_to_drop_(0),
           received_pli_(false) {}

    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));

       // Drop all retransmitted packets to force a PLI.
       if (header.timestamp <= highest_dropped_timestamp_)
         return DROP_PACKET;

       if (frames_to_drop_ > 0) {
         highest_dropped_timestamp_ = header.timestamp;
         --frames_to_drop_;
         return DROP_PACKET;
       }

       return SEND_PACKET;
     }

     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet, length));
       if (!nack_enabled_)
         EXPECT_EQ(0, parser.nack()->num_packets());
       if (parser.pli()->num_packets() > 0)
         received_pli_ = true;
       return SEND_PACKET;
     }

     void OnFrame(const VideoFrame& video_frame) override {
       rtc::CritScope lock(&crit_);
       if (received_pli_ &&
           video_frame.timestamp() > highest_dropped_timestamp_) {
         observation_complete_.Set();
       }
       if (!received_pli_)
         frames_to_drop_ = kPacketsToDrop;
     }

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
       (*receive_configs)[0].renderer = this;
     }

     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out waiting for PLI to be "
                              "received and a frame to be "
                              "rendered afterwards.";
     }

     rtc::CriticalSection crit_;
     int rtp_history_ms_;
     bool nack_enabled_;
     uint32_t highest_dropped_timestamp_ RTC_GUARDED_BY(&crit_);
     int frames_to_drop_ RTC_GUARDED_BY(&crit_);
     bool received_pli_ RTC_GUARDED_BY(&crit_);
   } test(rtp_history_ms);

   RunBaseTest(&test);
 }

 TEST_P(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithNack) {
   ReceivesPliAndRecovers(1000);
 }

 TEST_P(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithoutNack) {
   ReceivesPliAndRecovers(0);
 }
 // This test drops second RTP packet with a marker bit set, makes sure it's
 // retransmitted and renders. Retransmission SSRCs are also checked.
 void RetransmissionEndToEndTest::DecodesRetransmittedFrame(bool enable_rtx,
                                                            bool enable_red) {
   static const int kDroppedFrameNumber = 10;
   class RetransmissionObserver : public test::EndToEndTest,
                                  public rtc::VideoSinkInterface<VideoFrame> {
    public:
     RetransmissionObserver(bool enable_rtx, bool enable_red)
         : EndToEndTest(kDefaultTimeoutMs),
           payload_type_(GetPayloadType(false, enable_red)),
           retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0]
                                           : kVideoSendSsrcs[0]),
           retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
           encoder_(VP8Encoder::Create()),
           marker_bits_observed_(0),
           retransmitted_timestamp_(0) {}

    private:
     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));

       // Ignore padding-only packets over RTX.
       if (header.payloadType != payload_type_) {
         EXPECT_EQ(retransmission_ssrc_, header.ssrc);
         if (length == header.headerLength + header.paddingLength)
           return SEND_PACKET;
       }

       if (header.timestamp == retransmitted_timestamp_) {
         EXPECT_EQ(retransmission_ssrc_, header.ssrc);
         EXPECT_EQ(retransmission_payload_type_, header.payloadType);
         return SEND_PACKET;
       }

       // Found the final packet of the frame to inflict loss to, drop this and
       // expect a retransmission.
       if (header.payloadType == payload_type_ && header.markerBit &&
           ++marker_bits_observed_ == kDroppedFrameNumber) {
         // This should be the only dropped packet.
         EXPECT_EQ(0u, retransmitted_timestamp_);
         retransmitted_timestamp_ = header.timestamp;
         if (std::find(rendered_timestamps_.begin(), rendered_timestamps_.end(),
                       retransmitted_timestamp_) != rendered_timestamps_.end()) {
           // Frame was rendered before last packet was scheduled for sending.
           // This is extremly rare but possible scenario because prober able to
           // resend packet before it was send.
           // TODO(danilchap): Remove this corner case when prober would not be
           // able to sneak in between packet saved to history for resending and
           // pacer notified about existance of that packet for sending.
           // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5540 for
           // details.
           observation_complete_.Set();
         }
         return DROP_PACKET;
       }

       return SEND_PACKET;
     }

     void OnFrame(const VideoFrame& frame) override {
       EXPECT_EQ(kVideoRotation_90, frame.rotation());
       {
         rtc::CritScope lock(&crit_);
         if (frame.timestamp() == retransmitted_timestamp_)
           observation_complete_.Set();
         rendered_timestamps_.push_back(frame.timestamp());
       }
       orig_renderer_->OnFrame(frame);
     }

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;

       // Insert ourselves into the rendering pipeline.
       RTC_DCHECK(!orig_renderer_);
       orig_renderer_ = (*receive_configs)[0].renderer;
       RTC_DCHECK(orig_renderer_);
       (*receive_configs)[0].disable_prerenderer_smoothing = true;
       (*receive_configs)[0].renderer = this;

       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;

       if (payload_type_ == kRedPayloadType) {
         send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
         send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
         if (retransmission_ssrc_ == kSendRtxSsrcs[0])
           send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
         (*receive_configs)[0].rtp.ulpfec_payload_type =
             send_config->rtp.ulpfec.ulpfec_payload_type;
         (*receive_configs)[0].rtp.red_payload_type =
             send_config->rtp.ulpfec.red_payload_type;
       }

       if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
         send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
         send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
         (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
         (*receive_configs)[0]
             .rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType)
                                                   ? kRtxRedPayloadType
                                                   : kSendRtxPayloadType] =
             payload_type_;
       }
       // Configure encoding and decoding with VP8, since generic packetization
       // doesn't support FEC with NACK.
       RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
       send_config->encoder_settings.encoder = encoder_.get();
       send_config->encoder_settings.payload_name = "VP8";
       (*receive_configs)[0].decoders[0].payload_name = "VP8";
     }

     void OnFrameGeneratorCapturerCreated(
         test::FrameGeneratorCapturer* frame_generator_capturer) override {
       frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
     }

     void PerformTest() override {
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for retransmission to render.";
     }

     int GetPayloadType(bool use_rtx, bool use_fec) {
       if (use_fec) {
         if (use_rtx)
           return kRtxRedPayloadType;
         return kRedPayloadType;
       }
       if (use_rtx)
         return kSendRtxPayloadType;
       return kFakeVideoSendPayloadType;
     }

     rtc::CriticalSection crit_;
     rtc::VideoSinkInterface<VideoFrame>* orig_renderer_ = nullptr;
     const int payload_type_;
     const uint32_t retransmission_ssrc_;
     const int retransmission_payload_type_;
     std::unique_ptr<VideoEncoder> encoder_;
     const std::string payload_name_;
     int marker_bits_observed_;
     uint32_t retransmitted_timestamp_ RTC_GUARDED_BY(&crit_);
     std::vector<uint32_t> rendered_timestamps_ RTC_GUARDED_BY(&crit_);
   } test(enable_rtx, enable_red);

   RunBaseTest(&test);
 }

 TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrame) {
   DecodesRetransmittedFrame(false, false);
 }

 TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameOverRtx) {
   DecodesRetransmittedFrame(true, false);
 }

 TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRed) {
   DecodesRetransmittedFrame(false, true);
 }

 TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
   DecodesRetransmittedFrame(true, true);
 }

 }  // namespace webrtc
	/*
	* Copyright 2018 The WebRTC project authors. All Rights Reserved.
	*
	* Use of this source code is governed by a BSD-style license
	* that can be found in the LICENSE file in the root of the source
	* tree. An additional intellectual property rights grant can be found
	* in the file PATENTS. All contributing project authors may
	* be found in the AUTHORS file in the root of the source tree.
	*/

	#include "modules/video_coding/codecs/vp8/include/vp8.h"
	#include "system_wrappers/include/sleep.h"
	#include "test/call_test.h"
	#include "test/field_trial.h"
	#include "test/gtest.h"
	#include "test/rtcp_packet_parser.h"

	namespace webrtc {
	class RetransmissionEndToEndTest
	: public test::CallTest,
	public testing::WithParamInterface<std::string> {
	public:
	RetransmissionEndToEndTest() : field_trial_(GetParam()) {}

	virtual ~RetransmissionEndToEndTest() {
	EXPECT_EQ(nullptr, video_send_stream_);
	EXPECT_TRUE(video_receive_streams_.empty());
	}

	protected:
	void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
	void ReceivesPliAndRecovers(int rtp_history_ms);

	private:
	private:
	test::ScopedFieldTrials field_trial_;
	};

	INSTANTIATE_TEST_CASE_P(RoundRobin,
	RetransmissionEndToEndTest,
	::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
	"WebRTC-RoundRobinPacing/Enabled/"));

	TEST_P(RetransmissionEndToEndTest, ReceivesAndRetransmitsNack) {
	static const int kNumberOfNacksToObserve = 2;
	static const int kLossBurstSize = 2;
	static const int kPacketsBetweenLossBursts = 9;
	class NackObserver : public test::EndToEndTest {
	public:
	NackObserver()
	: EndToEndTest(kLongTimeoutMs),
	sent_rtp_packets_(0),
	packets_left_to_drop_(0),
	nacks_left_(kNumberOfNacksToObserve) {}

	private:
	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&crit_);
	RTPHeader header;
	EXPECT_TRUE(parser_->Parse(packet, length, &header));

	// Never drop retransmitted packets.
	if (dropped_packets_.find(header.sequenceNumber) !=
	dropped_packets_.end()) {
	retransmitted_packets_.insert(header.sequenceNumber);
	return SEND_PACKET;
	}

	if (nacks_left_ <= 0 &&
	retransmitted_packets_.size() == dropped_packets_.size()) {
	observation_complete_.Set();
	}

	++sent_rtp_packets_;

	// Enough NACKs received, stop dropping packets.
	if (nacks_left_ <= 0)
	return SEND_PACKET;

	// Check if it's time for a new loss burst.
	if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
	packets_left_to_drop_ = kLossBurstSize;

	// Never drop padding packets as those won't be retransmitted.
	if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
	--packets_left_to_drop_;
	dropped_packets_.insert(header.sequenceNumber);
	return DROP_PACKET;
	}

	return SEND_PACKET;
	}

	Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&crit_);
	test::RtcpPacketParser parser;
	EXPECT_TRUE(parser.Parse(packet, length));
	nacks_left_ -= parser.nack()->num_packets();
	return SEND_PACKET;
	}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
	(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
	}

	void PerformTest() override {
	EXPECT_TRUE(Wait())
	<< "Timed out waiting for packets to be NACKed, retransmitted and "
	"rendered.";
	}

	rtc::CriticalSection crit_;
	std::set<uint16_t> dropped_packets_;
	std::set<uint16_t> retransmitted_packets_;
	uint64_t sent_rtp_packets_;
	int packets_left_to_drop_;
	int nacks_left_ RTC_GUARDED_BY(&crit_);
	} test;

	RunBaseTest(&test);
	}

	TEST_P(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
	class NackObserver : public test::EndToEndTest {
	public:
	NackObserver()
	: EndToEndTest(kLongTimeoutMs),
	local_ssrc_(0),
	remote_ssrc_(0),
	receive_transport_(nullptr) {}

	private:
	size_t GetNumVideoStreams() const override { return 0; }
	size_t GetNumAudioStreams() const override { return 1; }

	test::PacketTransport* CreateReceiveTransport(
	test::SingleThreadedTaskQueueForTesting* task_queue) override {
	test::PacketTransport* receive_transport = new test::PacketTransport(
	task_queue, nullptr, this, test::PacketTransport::kReceiver,
	payload_type_map_, FakeNetworkPipe::Config());
	receive_transport_ = receive_transport;
	return receive_transport;
	}

	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	RTPHeader header;
	EXPECT_TRUE(parser_->Parse(packet, length, &header));

	if (!sequence_number_to_retransmit_) {
	sequence_number_to_retransmit_ = header.sequenceNumber;

	// Don't ask for retransmission straight away, may be deduped in pacer.
	} else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
	observation_complete_.Set();
	} else {
	// Send a NACK as often as necessary until retransmission is received.
	rtcp::Nack nack;
	nack.SetSenderSsrc(local_ssrc_);
	nack.SetMediaSsrc(remote_ssrc_);
	uint16_t nack_list[] = {*sequence_number_to_retransmit_};
	nack.SetPacketIds(nack_list, 1);
	rtc::Buffer buffer = nack.Build();

	EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
	}

	return SEND_PACKET;
	}

	void ModifyAudioConfigs(
	AudioSendStream::Config* send_config,
	std::vector<AudioReceiveStream::Config>* receive_configs) override {
	send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
	(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
	local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
	remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
	}

	void PerformTest() override {
	EXPECT_TRUE(Wait())
	<< "Timed out waiting for packets to be NACKed, retransmitted and "
	"rendered.";
	}

	uint32_t local_ssrc_;
	uint32_t remote_ssrc_;
	Transport* receive_transport_;
	rtc::Optional<uint16_t> sequence_number_to_retransmit_;
	} test;

	RunBaseTest(&test);
	}

	TEST_P(RetransmissionEndToEndTest,
	StopSendingKeyframeRequestsForInactiveStream) {
	class KeyframeRequestObserver : public test::EndToEndTest {
	public:
	explicit KeyframeRequestObserver(
	test::SingleThreadedTaskQueueForTesting* task_queue)
	: clock_(Clock::GetRealTimeClock()), task_queue_(task_queue) {}

	void OnVideoStreamsCreated(
	VideoSendStream* send_stream,
	const std::vector<VideoReceiveStream*>& receive_streams) override {
	RTC_DCHECK_EQ(1, receive_streams.size());
	send_stream_ = send_stream;
	receive_stream_ = receive_streams[0];
	}

	void PerformTest() override {
	bool frame_decoded = false;
	int64_t start_time = clock_->TimeInMilliseconds();
	while (clock_->TimeInMilliseconds() - start_time <= 5000) {
	if (receive_stream_->GetStats().frames_decoded > 0) {
	frame_decoded = true;
	break;
	}
	SleepMs(100);
	}
	ASSERT_TRUE(frame_decoded);
	task_queue_->SendTask([this]() { send_stream_->Stop(); });
	SleepMs(10000);
	ASSERT_EQ(
	1U, receive_stream_->GetStats().rtcp_packet_type_counts.pli_packets);
	}

	private:
	Clock* clock_;
	VideoSendStream* send_stream_;
	VideoReceiveStream* receive_stream_;
	test::SingleThreadedTaskQueueForTesting* const task_queue_;
	} test(&task_queue_);

	RunBaseTest(&test);
	}

	void RetransmissionEndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
	static const int kPacketsToDrop = 1;

	class PliObserver : public test::EndToEndTest,
	public rtc::VideoSinkInterface<VideoFrame> {
	public:
	explicit PliObserver(int rtp_history_ms)
	: EndToEndTest(kLongTimeoutMs),
	rtp_history_ms_(rtp_history_ms),
	nack_enabled_(rtp_history_ms > 0),
	highest_dropped_timestamp_(0),
	frames_to_drop_(0),
	received_pli_(false) {}

	private:
	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&crit_);
	RTPHeader header;
	EXPECT_TRUE(parser_->Parse(packet, length, &header));

	// Drop all retransmitted packets to force a PLI.
	if (header.timestamp <= highest_dropped_timestamp_)
	return DROP_PACKET;

	if (frames_to_drop_ > 0) {
	highest_dropped_timestamp_ = header.timestamp;
	--frames_to_drop_;
	return DROP_PACKET;
	}

	return SEND_PACKET;
	}

	Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&crit_);
	test::RtcpPacketParser parser;
	EXPECT_TRUE(parser.Parse(packet, length));
	if (!nack_enabled_)
	EXPECT_EQ(0, parser.nack()->num_packets());
	if (parser.pli()->num_packets() > 0)
	received_pli_ = true;
	return SEND_PACKET;
	}

	void OnFrame(const VideoFrame& video_frame) override {
	rtc::CritScope lock(&crit_);
	if (received_pli_ &&
	video_frame.timestamp() > highest_dropped_timestamp_) {
	observation_complete_.Set();
	}
	if (!received_pli_)
	frames_to_drop_ = kPacketsToDrop;
	}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
	(*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
	(*receive_configs)[0].renderer = this;
	}

	void PerformTest() override {
	EXPECT_TRUE(Wait()) << "Timed out waiting for PLI to be "
	"received and a frame to be "
	"rendered afterwards.";
	}

	rtc::CriticalSection crit_;
	int rtp_history_ms_;
	bool nack_enabled_;
	uint32_t highest_dropped_timestamp_ RTC_GUARDED_BY(&crit_);
	int frames_to_drop_ RTC_GUARDED_BY(&crit_);
	bool received_pli_ RTC_GUARDED_BY(&crit_);
	} test(rtp_history_ms);

	RunBaseTest(&test);
	}

	TEST_P(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithNack) {
	ReceivesPliAndRecovers(1000);
	}

	TEST_P(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithoutNack) {
	ReceivesPliAndRecovers(0);
	}
	// This test drops second RTP packet with a marker bit set, makes sure it's
	// retransmitted and renders. Retransmission SSRCs are also checked.
	void RetransmissionEndToEndTest::DecodesRetransmittedFrame(bool enable_rtx,
	bool enable_red) {
	static const int kDroppedFrameNumber = 10;
	class RetransmissionObserver : public test::EndToEndTest,
	public rtc::VideoSinkInterface<VideoFrame> {
	public:
	RetransmissionObserver(bool enable_rtx, bool enable_red)
	: EndToEndTest(kDefaultTimeoutMs),
	payload_type_(GetPayloadType(false, enable_red)),
	retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0]
	: kVideoSendSsrcs[0]),
	retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
	encoder_(VP8Encoder::Create()),
	marker_bits_observed_(0),
	retransmitted_timestamp_(0) {}

	private:
	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&crit_);
	RTPHeader header;
	EXPECT_TRUE(parser_->Parse(packet, length, &header));

	// Ignore padding-only packets over RTX.
	if (header.payloadType != payload_type_) {
	EXPECT_EQ(retransmission_ssrc_, header.ssrc);
	if (length == header.headerLength + header.paddingLength)
	return SEND_PACKET;
	}

	if (header.timestamp == retransmitted_timestamp_) {
	EXPECT_EQ(retransmission_ssrc_, header.ssrc);
	EXPECT_EQ(retransmission_payload_type_, header.payloadType);
	return SEND_PACKET;
	}

	// Found the final packet of the frame to inflict loss to, drop this and
	// expect a retransmission.
	if (header.payloadType == payload_type_ && header.markerBit &&
	++marker_bits_observed_ == kDroppedFrameNumber) {
	// This should be the only dropped packet.
	EXPECT_EQ(0u, retransmitted_timestamp_);
	retransmitted_timestamp_ = header.timestamp;
	if (std::find(rendered_timestamps_.begin(), rendered_timestamps_.end(),
	retransmitted_timestamp_) != rendered_timestamps_.end()) {
	// Frame was rendered before last packet was scheduled for sending.
	// This is extremly rare but possible scenario because prober able to
	// resend packet before it was send.
	// TODO(danilchap): Remove this corner case when prober would not be
	// able to sneak in between packet saved to history for resending and
	// pacer notified about existance of that packet for sending.
	// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5540 for
	// details.
	observation_complete_.Set();
	}
	return DROP_PACKET;
	}

	return SEND_PACKET;
	}

	void OnFrame(const VideoFrame& frame) override {
	EXPECT_EQ(kVideoRotation_90, frame.rotation());
	{
	rtc::CritScope lock(&crit_);
	if (frame.timestamp() == retransmitted_timestamp_)
	observation_complete_.Set();
	rendered_timestamps_.push_back(frame.timestamp());
	}
	orig_renderer_->OnFrame(frame);
	}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;

	// Insert ourselves into the rendering pipeline.
	RTC_DCHECK(!orig_renderer_);
	orig_renderer_ = (*receive_configs)[0].renderer;
	RTC_DCHECK(orig_renderer_);
	(*receive_configs)[0].disable_prerenderer_smoothing = true;
	(*receive_configs)[0].renderer = this;

	(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;

	if (payload_type_ == kRedPayloadType) {
	send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
	send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
	if (retransmission_ssrc_ == kSendRtxSsrcs[0])
	send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
	(*receive_configs)[0].rtp.ulpfec_payload_type =
	send_config->rtp.ulpfec.ulpfec_payload_type;
	(*receive_configs)[0].rtp.red_payload_type =
	send_config->rtp.ulpfec.red_payload_type;
	}

	if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
	send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
	send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
	(*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
	(*receive_configs)[0]
	.rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType)
	? kRtxRedPayloadType
	: kSendRtxPayloadType] =
	payload_type_;
	}
	// Configure encoding and decoding with VP8, since generic packetization
	// doesn't support FEC with NACK.
	RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
	send_config->encoder_settings.encoder = encoder_.get();
	send_config->encoder_settings.payload_name = "VP8";
	(*receive_configs)[0].decoders[0].payload_name = "VP8";
	}

	void OnFrameGeneratorCapturerCreated(
	test::FrameGeneratorCapturer* frame_generator_capturer) override {
	frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
	}

	void PerformTest() override {
	EXPECT_TRUE(Wait())
	<< "Timed out while waiting for retransmission to render.";
	}

	int GetPayloadType(bool use_rtx, bool use_fec) {
	if (use_fec) {
	if (use_rtx)
	return kRtxRedPayloadType;
	return kRedPayloadType;
	}
	if (use_rtx)
	return kSendRtxPayloadType;
	return kFakeVideoSendPayloadType;
	}

	rtc::CriticalSection crit_;
	rtc::VideoSinkInterface<VideoFrame>* orig_renderer_ = nullptr;
	const int payload_type_;
	const uint32_t retransmission_ssrc_;
	const int retransmission_payload_type_;
	std::unique_ptr<VideoEncoder> encoder_;
	const std::string payload_name_;
	int marker_bits_observed_;
	uint32_t retransmitted_timestamp_ RTC_GUARDED_BY(&crit_);
	std::vector<uint32_t> rendered_timestamps_ RTC_GUARDED_BY(&crit_);
	} test(enable_rtx, enable_red);

	RunBaseTest(&test);
	}

	TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrame) {
	DecodesRetransmittedFrame(false, false);
	}

	TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameOverRtx) {
	DecodesRetransmittedFrame(true, false);
	}

	TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRed) {
	DecodesRetransmittedFrame(false, true);
	}

	TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
	DecodesRetransmittedFrame(true, true);
	}

	} // namespace webrtc