video/end_to_end_tests/network_state_tests.cc - src - 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 "system_wrappers/include/sleep.h"
 #include "test/call_test.h"
 #include "test/gtest.h"

 namespace webrtc {
 namespace {
 constexpr int kSilenceTimeoutMs = 2000;
 }

 class NetworkStateEndToEndTest : public test::CallTest {
  protected:
   class UnusedTransport : public Transport {
    private:
     bool SendRtp(const uint8_t* packet,
                  size_t length,
                  const PacketOptions& options) override {
       ADD_FAILURE() << "Unexpected RTP sent.";
       return false;
     }

     bool SendRtcp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected RTCP sent.";
       return false;
     }
   };
   class RequiredTransport : public Transport {
    public:
     RequiredTransport(bool rtp_required, bool rtcp_required)
         : need_rtp_(rtp_required), need_rtcp_(rtcp_required) {}
     ~RequiredTransport() {
       if (need_rtp_) {
         ADD_FAILURE() << "Expected RTP packet not sent.";
       }
       if (need_rtcp_) {
         ADD_FAILURE() << "Expected RTCP packet not sent.";
       }
     }

    private:
     bool SendRtp(const uint8_t* packet,
                  size_t length,
                  const PacketOptions& options) override {
       rtc::CritScope lock(&crit_);
       need_rtp_ = false;
       return true;
     }

     bool SendRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&crit_);
       need_rtcp_ = false;
       return true;
     }
     bool need_rtp_;
     bool need_rtcp_;
     rtc::CriticalSection crit_;
   };
   void VerifyNewVideoSendStreamsRespectNetworkState(
       MediaType network_to_bring_up,
       VideoEncoder* encoder,
       Transport* transport);
   void VerifyNewVideoReceiveStreamsRespectNetworkState(
       MediaType network_to_bring_up,
       Transport* transport);
 };

 void NetworkStateEndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
     MediaType network_to_bring_up,
     VideoEncoder* encoder,
     Transport* transport) {
   task_queue_.SendTask([this, network_to_bring_up, encoder, transport]() {
     CreateSenderCall(Call::Config(event_log_.get()));
     sender_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);

     CreateSendConfig(1, 0, 0, transport);
     video_send_config_.encoder_settings.encoder = encoder;
     CreateVideoStreams();
     CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
                                  kDefaultHeight);

     Start();
   });

   SleepMs(kSilenceTimeoutMs);

   task_queue_.SendTask([this]() {
     Stop();
     DestroyStreams();
     DestroyCalls();
   });
 }

 void NetworkStateEndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
     MediaType network_to_bring_up,
     Transport* transport) {
   std::unique_ptr<test::DirectTransport> sender_transport;

   task_queue_.SendTask([this, &sender_transport, network_to_bring_up,
                         transport]() {
     Call::Config config(event_log_.get());
     CreateCalls(config, config);
     receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
     sender_transport = rtc::MakeUnique<test::DirectTransport>(
         &task_queue_, sender_call_.get(), payload_type_map_);
     sender_transport->SetReceiver(receiver_call_->Receiver());
     CreateSendConfig(1, 0, 0, sender_transport.get());
     CreateMatchingReceiveConfigs(transport);
     CreateVideoStreams();
     CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
                                  kDefaultHeight);
     Start();
   });

   SleepMs(kSilenceTimeoutMs);

   task_queue_.SendTask([this, &sender_transport]() {
     Stop();
     DestroyStreams();
     sender_transport.reset();
     DestroyCalls();
   });
 }

 TEST_F(NetworkStateEndToEndTest, RespectsNetworkState) {
   // TODO(pbos): Remove accepted downtime packets etc. when signaling network
   // down blocks until no more packets will be sent.

   // Pacer will send from its packet list and then send required padding before
   // checking paused_ again. This should be enough for one round of pacing,
   // otherwise increase.
   static const int kNumAcceptedDowntimeRtp = 5;
   // A single RTCP may be in the pipeline.
   static const int kNumAcceptedDowntimeRtcp = 1;
   class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
    public:
     explicit NetworkStateTest(
         test::SingleThreadedTaskQueueForTesting* task_queue)
         : EndToEndTest(kDefaultTimeoutMs),
           FakeEncoder(Clock::GetRealTimeClock()),
           task_queue_(task_queue),
           encoded_frames_(false, false),
           packet_event_(false, false),
           sender_call_(nullptr),
           receiver_call_(nullptr),
           sender_state_(kNetworkUp),
           sender_rtp_(0),
           sender_padding_(0),
           sender_rtcp_(0),
           receiver_rtcp_(0),
           down_frames_(0) {}

     Action OnSendRtp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&test_crit_);
       RTPHeader header;
       EXPECT_TRUE(parser_->Parse(packet, length, &header));
       if (length == header.headerLength + header.paddingLength)
         ++sender_padding_;
       ++sender_rtp_;
       packet_event_.Set();
       return SEND_PACKET;
     }

     Action OnSendRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&test_crit_);
       ++sender_rtcp_;
       packet_event_.Set();
       return SEND_PACKET;
     }

     Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
       ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
       return SEND_PACKET;
     }

     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       rtc::CritScope lock(&test_crit_);
       ++receiver_rtcp_;
       packet_event_.Set();
       return SEND_PACKET;
     }

     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
       sender_call_ = sender_call;
       receiver_call_ = receiver_call;
     }

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder = this;
     }

     void PerformTest() override {
       EXPECT_TRUE(encoded_frames_.Wait(kDefaultTimeoutMs))
           << "No frames received by the encoder.";

       task_queue_->SendTask([this]() {
         // Wait for packets from both sender/receiver.
         WaitForPacketsOrSilence(false, false);

         // Sender-side network down for audio; there should be no effect on
         // video
         sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkDown);
         WaitForPacketsOrSilence(false, false);

         // Receiver-side network down for audio; no change expected
         receiver_call_->SignalChannelNetworkState(MediaType::AUDIO,
                                                   kNetworkDown);
         WaitForPacketsOrSilence(false, false);

         // Sender-side network down.
         sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkDown);
         {
           rtc::CritScope lock(&test_crit_);
           // After network goes down we shouldn't be encoding more frames.
           sender_state_ = kNetworkDown;
         }
         // Wait for receiver-packets and no sender packets.
         WaitForPacketsOrSilence(true, false);

         // Receiver-side network down.
         receiver_call_->SignalChannelNetworkState(MediaType::VIDEO,
                                                   kNetworkDown);
         WaitForPacketsOrSilence(true, true);

         // Network up for audio for both sides; video is still not expected to
         // start
         sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
         receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
         WaitForPacketsOrSilence(true, true);

         // Network back up again for both.
         {
           rtc::CritScope lock(&test_crit_);
           // It's OK to encode frames again, as we're about to bring up the
           // network.
           sender_state_ = kNetworkUp;
         }
         sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
         receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
         WaitForPacketsOrSilence(false, false);

         // TODO(skvlad): add tests to verify that the audio streams are stopped
         // when the network goes down for audio once the workaround in
         // paced_sender.cc is removed.
       });
     }

     int32_t Encode(const VideoFrame& input_image,
                    const CodecSpecificInfo* codec_specific_info,
                    const std::vector<FrameType>* frame_types) override {
       {
         rtc::CritScope lock(&test_crit_);
         if (sender_state_ == kNetworkDown) {
           ++down_frames_;
           EXPECT_LE(down_frames_, 1)
               << "Encoding more than one frame while network is down.";
           if (down_frames_ > 1)
             encoded_frames_.Set();
         } else {
           encoded_frames_.Set();
         }
       }
       return test::FakeEncoder::Encode(input_image, codec_specific_info,
                                        frame_types);
     }

    private:
     void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
       int64_t initial_time_ms = clock_->TimeInMilliseconds();
       int initial_sender_rtp;
       int initial_sender_rtcp;
       int initial_receiver_rtcp;
       {
         rtc::CritScope lock(&test_crit_);
         initial_sender_rtp = sender_rtp_;
         initial_sender_rtcp = sender_rtcp_;
         initial_receiver_rtcp = receiver_rtcp_;
       }
       bool sender_done = false;
       bool receiver_done = false;
       while (!sender_done || !receiver_done) {
         packet_event_.Wait(kSilenceTimeoutMs);
         int64_t time_now_ms = clock_->TimeInMilliseconds();
         rtc::CritScope lock(&test_crit_);
         if (sender_down) {
           ASSERT_LE(sender_rtp_ - initial_sender_rtp - sender_padding_,
                     kNumAcceptedDowntimeRtp)
               << "RTP sent during sender-side downtime.";
           ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
                     kNumAcceptedDowntimeRtcp)
               << "RTCP sent during sender-side downtime.";
           if (time_now_ms - initial_time_ms >=
               static_cast<int64_t>(kSilenceTimeoutMs)) {
             sender_done = true;
           }
         } else {
           if (sender_rtp_ > initial_sender_rtp + kNumAcceptedDowntimeRtp)
             sender_done = true;
         }
         if (receiver_down) {
           ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
                     kNumAcceptedDowntimeRtcp)
               << "RTCP sent during receiver-side downtime.";
           if (time_now_ms - initial_time_ms >=
               static_cast<int64_t>(kSilenceTimeoutMs)) {
             receiver_done = true;
           }
         } else {
           if (receiver_rtcp_ > initial_receiver_rtcp + kNumAcceptedDowntimeRtcp)
             receiver_done = true;
         }
       }
     }

     test::SingleThreadedTaskQueueForTesting* const task_queue_;
     rtc::CriticalSection test_crit_;
     rtc::Event encoded_frames_;
     rtc::Event packet_event_;
     Call* sender_call_;
     Call* receiver_call_;
     NetworkState sender_state_ RTC_GUARDED_BY(test_crit_);
     int sender_rtp_ RTC_GUARDED_BY(test_crit_);
     int sender_padding_ RTC_GUARDED_BY(test_crit_);
     int sender_rtcp_ RTC_GUARDED_BY(test_crit_);
     int receiver_rtcp_ RTC_GUARDED_BY(test_crit_);
     int down_frames_ RTC_GUARDED_BY(test_crit_);
   } test(&task_queue_);

   RunBaseTest(&test);
 }

 TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
   class UnusedEncoder : public test::FakeEncoder {
    public:
     UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}

     int32_t InitEncode(const VideoCodec* config,
                        int32_t number_of_cores,
                        size_t max_payload_size) override {
       EXPECT_GT(config->startBitrate, 0u);
       return 0;
     }
     int32_t Encode(const VideoFrame& input_image,
                    const CodecSpecificInfo* codec_specific_info,
                    const std::vector<FrameType>* frame_types) override {
       ADD_FAILURE() << "Unexpected frame encode.";
       return test::FakeEncoder::Encode(input_image, codec_specific_info,
                                        frame_types);
     }
   };

   UnusedEncoder unused_encoder;
   UnusedTransport unused_transport;
   VerifyNewVideoSendStreamsRespectNetworkState(
       MediaType::AUDIO, &unused_encoder, &unused_transport);
 }

 TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
   class RequiredEncoder : public test::FakeEncoder {
    public:
     RequiredEncoder()
         : FakeEncoder(Clock::GetRealTimeClock()), encoded_frame_(false) {}
     ~RequiredEncoder() {
       if (!encoded_frame_) {
         ADD_FAILURE() << "Didn't encode an expected frame";
       }
     }
     int32_t Encode(const VideoFrame& input_image,
                    const CodecSpecificInfo* codec_specific_info,
                    const std::vector<FrameType>* frame_types) override {
       encoded_frame_ = true;
       return test::FakeEncoder::Encode(input_image, codec_specific_info,
                                        frame_types);
     }

    private:
     bool encoded_frame_;
   };

   RequiredTransport required_transport(true /*rtp*/, false /*rtcp*/);
   RequiredEncoder required_encoder;
   VerifyNewVideoSendStreamsRespectNetworkState(
       MediaType::VIDEO, &required_encoder, &required_transport);
 }

 TEST_F(NetworkStateEndToEndTest,
        NewVideoReceiveStreamsRespectVideoNetworkDown) {
   UnusedTransport transport;
   VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
 }

 TEST_F(NetworkStateEndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown) {
   RequiredTransport transport(false /*rtp*/, true /*rtcp*/);
   VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
 }

 }  // 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 "system_wrappers/include/sleep.h"
	#include "test/call_test.h"
	#include "test/gtest.h"

	namespace webrtc {
	namespace {
	constexpr int kSilenceTimeoutMs = 2000;
	}

	class NetworkStateEndToEndTest : public test::CallTest {
	protected:
	class UnusedTransport : public Transport {
	private:
	bool SendRtp(const uint8_t* packet,
	size_t length,
	const PacketOptions& options) override {
	ADD_FAILURE() << "Unexpected RTP sent.";
	return false;
	}

	bool SendRtcp(const uint8_t* packet, size_t length) override {
	ADD_FAILURE() << "Unexpected RTCP sent.";
	return false;
	}
	};
	class RequiredTransport : public Transport {
	public:
	RequiredTransport(bool rtp_required, bool rtcp_required)
	: need_rtp_(rtp_required), need_rtcp_(rtcp_required) {}
	~RequiredTransport() {
	if (need_rtp_) {
	ADD_FAILURE() << "Expected RTP packet not sent.";
	}
	if (need_rtcp_) {
	ADD_FAILURE() << "Expected RTCP packet not sent.";
	}
	}

	private:
	bool SendRtp(const uint8_t* packet,
	size_t length,
	const PacketOptions& options) override {
	rtc::CritScope lock(&crit_);
	need_rtp_ = false;
	return true;
	}

	bool SendRtcp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&crit_);
	need_rtcp_ = false;
	return true;
	}
	bool need_rtp_;
	bool need_rtcp_;
	rtc::CriticalSection crit_;
	};
	void VerifyNewVideoSendStreamsRespectNetworkState(
	MediaType network_to_bring_up,
	VideoEncoder* encoder,
	Transport* transport);
	void VerifyNewVideoReceiveStreamsRespectNetworkState(
	MediaType network_to_bring_up,
	Transport* transport);
	};

	void NetworkStateEndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
	MediaType network_to_bring_up,
	VideoEncoder* encoder,
	Transport* transport) {
	task_queue_.SendTask([this, network_to_bring_up, encoder, transport]() {
	CreateSenderCall(Call::Config(event_log_.get()));
	sender_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);

	CreateSendConfig(1, 0, 0, transport);
	video_send_config_.encoder_settings.encoder = encoder;
	CreateVideoStreams();
	CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
	kDefaultHeight);

	Start();
	});

	SleepMs(kSilenceTimeoutMs);

	task_queue_.SendTask([this]() {
	Stop();
	DestroyStreams();
	DestroyCalls();
	});
	}

	void NetworkStateEndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
	MediaType network_to_bring_up,
	Transport* transport) {
	std::unique_ptr<test::DirectTransport> sender_transport;

	task_queue_.SendTask([this, &sender_transport, network_to_bring_up,
	transport]() {
	Call::Config config(event_log_.get());
	CreateCalls(config, config);
	receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
	sender_transport = rtc::MakeUnique<test::DirectTransport>(
	&task_queue_, sender_call_.get(), payload_type_map_);
	sender_transport->SetReceiver(receiver_call_->Receiver());
	CreateSendConfig(1, 0, 0, sender_transport.get());
	CreateMatchingReceiveConfigs(transport);
	CreateVideoStreams();
	CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
	kDefaultHeight);
	Start();
	});

	SleepMs(kSilenceTimeoutMs);

	task_queue_.SendTask([this, &sender_transport]() {
	Stop();
	DestroyStreams();
	sender_transport.reset();
	DestroyCalls();
	});
	}

	TEST_F(NetworkStateEndToEndTest, RespectsNetworkState) {
	// TODO(pbos): Remove accepted downtime packets etc. when signaling network
	// down blocks until no more packets will be sent.

	// Pacer will send from its packet list and then send required padding before
	// checking paused_ again. This should be enough for one round of pacing,
	// otherwise increase.
	static const int kNumAcceptedDowntimeRtp = 5;
	// A single RTCP may be in the pipeline.
	static const int kNumAcceptedDowntimeRtcp = 1;
	class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
	public:
	explicit NetworkStateTest(
	test::SingleThreadedTaskQueueForTesting* task_queue)
	: EndToEndTest(kDefaultTimeoutMs),
	FakeEncoder(Clock::GetRealTimeClock()),
	task_queue_(task_queue),
	encoded_frames_(false, false),
	packet_event_(false, false),
	sender_call_(nullptr),
	receiver_call_(nullptr),
	sender_state_(kNetworkUp),
	sender_rtp_(0),
	sender_padding_(0),
	sender_rtcp_(0),
	receiver_rtcp_(0),
	down_frames_(0) {}

	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&test_crit_);
	RTPHeader header;
	EXPECT_TRUE(parser_->Parse(packet, length, &header));
	if (length == header.headerLength + header.paddingLength)
	++sender_padding_;
	++sender_rtp_;
	packet_event_.Set();
	return SEND_PACKET;
	}

	Action OnSendRtcp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&test_crit_);
	++sender_rtcp_;
	packet_event_.Set();
	return SEND_PACKET;
	}

	Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
	ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
	return SEND_PACKET;
	}

	Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
	rtc::CritScope lock(&test_crit_);
	++receiver_rtcp_;
	packet_event_.Set();
	return SEND_PACKET;
	}

	void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
	sender_call_ = sender_call;
	receiver_call_ = receiver_call;
	}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	send_config->encoder_settings.encoder = this;
	}

	void PerformTest() override {
	EXPECT_TRUE(encoded_frames_.Wait(kDefaultTimeoutMs))
	<< "No frames received by the encoder.";

	task_queue_->SendTask([this]() {
	// Wait for packets from both sender/receiver.
	WaitForPacketsOrSilence(false, false);

	// Sender-side network down for audio; there should be no effect on
	// video
	sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkDown);
	WaitForPacketsOrSilence(false, false);

	// Receiver-side network down for audio; no change expected
	receiver_call_->SignalChannelNetworkState(MediaType::AUDIO,
	kNetworkDown);
	WaitForPacketsOrSilence(false, false);

	// Sender-side network down.
	sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkDown);
	{
	rtc::CritScope lock(&test_crit_);
	// After network goes down we shouldn't be encoding more frames.
	sender_state_ = kNetworkDown;
	}
	// Wait for receiver-packets and no sender packets.
	WaitForPacketsOrSilence(true, false);

	// Receiver-side network down.
	receiver_call_->SignalChannelNetworkState(MediaType::VIDEO,
	kNetworkDown);
	WaitForPacketsOrSilence(true, true);

	// Network up for audio for both sides; video is still not expected to
	// start
	sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
	receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
	WaitForPacketsOrSilence(true, true);

	// Network back up again for both.
	{
	rtc::CritScope lock(&test_crit_);
	// It's OK to encode frames again, as we're about to bring up the
	// network.
	sender_state_ = kNetworkUp;
	}
	sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
	receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
	WaitForPacketsOrSilence(false, false);

	// TODO(skvlad): add tests to verify that the audio streams are stopped
	// when the network goes down for audio once the workaround in
	// paced_sender.cc is removed.
	});
	}

	int32_t Encode(const VideoFrame& input_image,
	const CodecSpecificInfo* codec_specific_info,
	const std::vector<FrameType>* frame_types) override {
	{
	rtc::CritScope lock(&test_crit_);
	if (sender_state_ == kNetworkDown) {
	++down_frames_;
	EXPECT_LE(down_frames_, 1)
	<< "Encoding more than one frame while network is down.";
	if (down_frames_ > 1)
	encoded_frames_.Set();
	} else {
	encoded_frames_.Set();
	}
	}
	return test::FakeEncoder::Encode(input_image, codec_specific_info,
	frame_types);
	}

	private:
	void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
	int64_t initial_time_ms = clock_->TimeInMilliseconds();
	int initial_sender_rtp;
	int initial_sender_rtcp;
	int initial_receiver_rtcp;
	{
	rtc::CritScope lock(&test_crit_);
	initial_sender_rtp = sender_rtp_;
	initial_sender_rtcp = sender_rtcp_;
	initial_receiver_rtcp = receiver_rtcp_;
	}
	bool sender_done = false;
	bool receiver_done = false;
	while (!sender_done \|\| !receiver_done) {
	packet_event_.Wait(kSilenceTimeoutMs);
	int64_t time_now_ms = clock_->TimeInMilliseconds();
	rtc::CritScope lock(&test_crit_);
	if (sender_down) {
	ASSERT_LE(sender_rtp_ - initial_sender_rtp - sender_padding_,
	kNumAcceptedDowntimeRtp)
	<< "RTP sent during sender-side downtime.";
	ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
	kNumAcceptedDowntimeRtcp)
	<< "RTCP sent during sender-side downtime.";
	if (time_now_ms - initial_time_ms >=
	static_cast<int64_t>(kSilenceTimeoutMs)) {
	sender_done = true;
	}
	} else {
	if (sender_rtp_ > initial_sender_rtp + kNumAcceptedDowntimeRtp)
	sender_done = true;
	}
	if (receiver_down) {
	ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
	kNumAcceptedDowntimeRtcp)
	<< "RTCP sent during receiver-side downtime.";
	if (time_now_ms - initial_time_ms >=
	static_cast<int64_t>(kSilenceTimeoutMs)) {
	receiver_done = true;
	}
	} else {
	if (receiver_rtcp_ > initial_receiver_rtcp + kNumAcceptedDowntimeRtcp)
	receiver_done = true;
	}
	}
	}

	test::SingleThreadedTaskQueueForTesting* const task_queue_;
	rtc::CriticalSection test_crit_;
	rtc::Event encoded_frames_;
	rtc::Event packet_event_;
	Call* sender_call_;
	Call* receiver_call_;
	NetworkState sender_state_ RTC_GUARDED_BY(test_crit_);
	int sender_rtp_ RTC_GUARDED_BY(test_crit_);
	int sender_padding_ RTC_GUARDED_BY(test_crit_);
	int sender_rtcp_ RTC_GUARDED_BY(test_crit_);
	int receiver_rtcp_ RTC_GUARDED_BY(test_crit_);
	int down_frames_ RTC_GUARDED_BY(test_crit_);
	} test(&task_queue_);

	RunBaseTest(&test);
	}

	TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
	class UnusedEncoder : public test::FakeEncoder {
	public:
	UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}

	int32_t InitEncode(const VideoCodec* config,
	int32_t number_of_cores,
	size_t max_payload_size) override {
	EXPECT_GT(config->startBitrate, 0u);
	return 0;
	}
	int32_t Encode(const VideoFrame& input_image,
	const CodecSpecificInfo* codec_specific_info,
	const std::vector<FrameType>* frame_types) override {
	ADD_FAILURE() << "Unexpected frame encode.";
	return test::FakeEncoder::Encode(input_image, codec_specific_info,
	frame_types);
	}
	};

	UnusedEncoder unused_encoder;
	UnusedTransport unused_transport;
	VerifyNewVideoSendStreamsRespectNetworkState(
	MediaType::AUDIO, &unused_encoder, &unused_transport);
	}

	TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
	class RequiredEncoder : public test::FakeEncoder {
	public:
	RequiredEncoder()
	: FakeEncoder(Clock::GetRealTimeClock()), encoded_frame_(false) {}
	~RequiredEncoder() {
	if (!encoded_frame_) {
	ADD_FAILURE() << "Didn't encode an expected frame";
	}
	}
	int32_t Encode(const VideoFrame& input_image,
	const CodecSpecificInfo* codec_specific_info,
	const std::vector<FrameType>* frame_types) override {
	encoded_frame_ = true;
	return test::FakeEncoder::Encode(input_image, codec_specific_info,
	frame_types);
	}

	private:
	bool encoded_frame_;
	};

	RequiredTransport required_transport(true /rtp/, false /rtcp/);
	RequiredEncoder required_encoder;
	VerifyNewVideoSendStreamsRespectNetworkState(
	MediaType::VIDEO, &required_encoder, &required_transport);
	}

	TEST_F(NetworkStateEndToEndTest,
	NewVideoReceiveStreamsRespectVideoNetworkDown) {
	UnusedTransport transport;
	VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
	}

	TEST_F(NetworkStateEndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown) {
	RequiredTransport transport(false /rtp/, true /rtcp/);
	VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
	}

	} // namespace webrtc