video/end_to_end_tests/bandwidth_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 "absl/memory/memory.h"
 #include "api/test/simulated_network.h"
 #include "api/video/builtin_video_bitrate_allocator_factory.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "call/fake_network_pipe.h"
 #include "call/simulated_network.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "rtc_base/rate_limiter.h"
 #include "system_wrappers/include/sleep.h"
 #include "test/call_test.h"
 #include "test/fake_encoder.h"
 #include "test/field_trial.h"
 #include "test/gtest.h"
 #include "test/rtcp_packet_parser.h"
 #include "test/rtp_rtcp_observer.h"
 #include "test/video_encoder_proxy_factory.h"

 namespace webrtc {
 namespace {
 enum : int {  // The first valid value is 1.
   kAbsSendTimeExtensionId = 1,
 };
 }  // namespace

 class BandwidthEndToEndTest : public test::CallTest {
  public:
   BandwidthEndToEndTest() = default;
 };

 TEST_F(BandwidthEndToEndTest, ReceiveStreamSendsRemb) {
   class RembObserver : public test::EndToEndTest {
    public:
     RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
       (*receive_configs)[0].rtp.remb = true;
       (*receive_configs)[0].rtp.transport_cc = false;
     }

     Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
       test::RtcpPacketParser parser;
       EXPECT_TRUE(parser.Parse(packet, length));

       if (parser.remb()->num_packets() > 0) {
         EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc());
         EXPECT_LT(0U, parser.remb()->bitrate_bps());
         EXPECT_EQ(1U, parser.remb()->ssrcs().size());
         EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]);
         observation_complete_.Set();
       }

       return SEND_PACKET;
     }
     void PerformTest() override {
       EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
                              "receiver RTCP REMB packet to be "
                              "sent.";
     }
   } test;

   RunBaseTest(&test);
 }

 class BandwidthStatsTest : public test::EndToEndTest {
  public:
   explicit BandwidthStatsTest(bool send_side_bwe)
       : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
         sender_call_(nullptr),
         receiver_call_(nullptr),
         has_seen_pacer_delay_(false),
         send_side_bwe_(send_side_bwe) {}

   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override {
     if (!send_side_bwe_) {
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
       (*receive_configs)[0].rtp.remb = true;
       (*receive_configs)[0].rtp.transport_cc = false;
     }
   }

   Action OnSendRtp(const uint8_t* packet, size_t length) override {
     Call::Stats sender_stats = sender_call_->GetStats();
     Call::Stats receiver_stats = receiver_call_->GetStats();
     if (!has_seen_pacer_delay_)
       has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
     if (sender_stats.send_bandwidth_bps > 0 && has_seen_pacer_delay_) {
       if (send_side_bwe_ || receiver_stats.recv_bandwidth_bps > 0)
         observation_complete_.Set();
     }
     return SEND_PACKET;
   }

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

   void PerformTest() override {
     EXPECT_TRUE(Wait()) << "Timed out while waiting for "
                            "non-zero bandwidth stats.";
   }

  private:
   Call* sender_call_;
   Call* receiver_call_;
   bool has_seen_pacer_delay_;
   const bool send_side_bwe_;
 };

 TEST_F(BandwidthEndToEndTest, VerifySendSideBweStats) {
   BandwidthStatsTest test(true);
   RunBaseTest(&test);
 }

 TEST_F(BandwidthEndToEndTest, VerifyRecvSideBweStats) {
   BandwidthStatsTest test(false);
   RunBaseTest(&test);
 }

 // Verifies that it's possible to limit the send BWE by sending a REMB.
 // This is verified by allowing the send BWE to ramp-up to >1000 kbps,
 // then have the test generate a REMB of 500 kbps and verify that the send BWE
 // is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the
 // test verifies that the send BWE ramps back up to exactly 1000 kbps.
 TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
   class BweObserver : public test::EndToEndTest {
    public:
     BweObserver()
         : EndToEndTest(kDefaultTimeoutMs),
           sender_call_(nullptr),
           clock_(Clock::GetRealTimeClock()),
           sender_ssrc_(0),
           remb_bitrate_bps_(1000000),
           receive_transport_(nullptr),
           poller_thread_(&BitrateStatsPollingThread,
                          this,
                          "BitrateStatsPollingThread"),
           state_(kWaitForFirstRampUp),
           retransmission_rate_limiter_(clock_, 1000) {}

     ~BweObserver() {}

     test::PacketTransport* CreateReceiveTransport(
         test::SingleThreadedTaskQueueForTesting* task_queue) override {
       receive_transport_ = new test::PacketTransport(
           task_queue, nullptr, this, test::PacketTransport::kReceiver,
           payload_type_map_,
           absl::make_unique<FakeNetworkPipe>(
               Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
                                              BuiltInNetworkBehaviorConfig())));
       return receive_transport_;
     }

     void ModifySenderBitrateConfig(
         BitrateConstraints* bitrate_config) override {
       // Set a high start bitrate to reduce the test completion time.
       bitrate_config->start_bitrate_bps = remb_bitrate_bps_;
     }

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
       sender_ssrc_ = send_config->rtp.ssrcs[0];

       encoder_config->max_bitrate_bps = 2000000;

       ASSERT_EQ(1u, receive_configs->size());
       RtpRtcp::Configuration config;
       config.receiver_only = true;
       config.clock = clock_;
       config.outgoing_transport = receive_transport_;
       config.retransmission_rate_limiter = &retransmission_rate_limiter_;
       rtp_rtcp_ = RtpRtcp::Create(config);
       rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
       rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
       rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
     }

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

     static void BitrateStatsPollingThread(void* obj) {
       static_cast<BweObserver*>(obj)->PollStats();
     }

     void PollStats() {
       do {
         if (sender_call_) {
           Call::Stats stats = sender_call_->GetStats();
           switch (state_) {
             case kWaitForFirstRampUp:
               if (stats.send_bandwidth_bps >= remb_bitrate_bps_) {
                 state_ = kWaitForRemb;
                 remb_bitrate_bps_ /= 2;
                 rtp_rtcp_->SetRemb(
                     remb_bitrate_bps_,
                     std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
                 rtp_rtcp_->SendRTCP(kRtcpRr);
               }
               break;

             case kWaitForRemb:
               if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
                 state_ = kWaitForSecondRampUp;
                 remb_bitrate_bps_ *= 2;
                 rtp_rtcp_->SetRemb(
                     remb_bitrate_bps_,
                     std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
                 rtp_rtcp_->SendRTCP(kRtcpRr);
               }
               break;

             case kWaitForSecondRampUp:
               if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
                 observation_complete_.Set();
               }
               break;
           }
         }
       } while (!stop_event_.Wait(1000));
     }

     void PerformTest() override {
       poller_thread_.Start();
       EXPECT_TRUE(Wait())
           << "Timed out while waiting for bitrate to change according to REMB.";
       stop_event_.Set();
       poller_thread_.Stop();
     }

    private:
     enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };

     Call* sender_call_;
     Clock* const clock_;
     uint32_t sender_ssrc_;
     int remb_bitrate_bps_;
     std::unique_ptr<RtpRtcp> rtp_rtcp_;
     test::PacketTransport* receive_transport_;
     rtc::Event stop_event_;
     rtc::PlatformThread poller_thread_;
     TestState state_;
     RateLimiter retransmission_rate_limiter_;
   } test;

   RunBaseTest(&test);
 }

 TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) {
   // If these fields trial are on, we get lower bitrates than expected by this
   // test, due to the packetization overhead and encoder pushback.
   webrtc::test::ScopedFieldTrials field_trials(
       std::string(field_trial::GetFieldTrialString()) +
       "WebRTC-SubtractPacketizationOverhead/Disabled/"
       "WebRTC-VideoRateControl/bitrate_adjuster:false/");
   class EncoderRateStatsTest : public test::EndToEndTest,
                                public test::FakeEncoder {
    public:
     explicit EncoderRateStatsTest(
         test::SingleThreadedTaskQueueForTesting* task_queue)
         : EndToEndTest(kDefaultTimeoutMs),
           FakeEncoder(Clock::GetRealTimeClock()),
           task_queue_(task_queue),
           send_stream_(nullptr),
           encoder_factory_(this),
           bitrate_allocator_factory_(
               CreateBuiltinVideoBitrateAllocatorFactory()),
           bitrate_kbps_(0) {}

     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
     }

     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->encoder_settings.encoder_factory = &encoder_factory_;
       send_config->encoder_settings.bitrate_allocator_factory =
           bitrate_allocator_factory_.get();
       RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
     }

     int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
                               uint32_t framerate) override {
       // Make sure not to trigger on any default zero bitrates.
       if (rate_allocation.get_sum_bps() == 0)
         return 0;
       rtc::CritScope lock(&crit_);
       bitrate_kbps_ = rate_allocation.get_sum_kbps();
       observation_complete_.Set();
       return 0;
     }

     void PerformTest() override {
       ASSERT_TRUE(Wait())
           << "Timed out while waiting for encoder SetRates() call.";

       task_queue_->SendTask([this]() {
         WaitForEncoderTargetBitrateMatchStats();
         send_stream_->Stop();
         WaitForStatsReportZeroTargetBitrate();
         send_stream_->Start();
         WaitForEncoderTargetBitrateMatchStats();
       });
     }

     void WaitForEncoderTargetBitrateMatchStats() {
       for (int i = 0; i < kDefaultTimeoutMs; ++i) {
         VideoSendStream::Stats stats = send_stream_->GetStats();
         {
           rtc::CritScope lock(&crit_);
           if ((stats.target_media_bitrate_bps + 500) / 1000 ==
               static_cast<int>(bitrate_kbps_)) {
             return;
           }
         }
         SleepMs(1);
       }
       FAIL()
           << "Timed out waiting for stats reporting the currently set bitrate.";
     }

     void WaitForStatsReportZeroTargetBitrate() {
       for (int i = 0; i < kDefaultTimeoutMs; ++i) {
         if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
           return;
         }
         SleepMs(1);
       }
       FAIL() << "Timed out waiting for stats reporting zero bitrate.";
     }

    private:
     test::SingleThreadedTaskQueueForTesting* const task_queue_;
     rtc::CriticalSection crit_;
     VideoSendStream* send_stream_;
     test::VideoEncoderProxyFactory encoder_factory_;
     std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
     uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
   } test(&task_queue_);

   RunBaseTest(&test);
 }
 }  // 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 "absl/memory/memory.h"
	#include "api/test/simulated_network.h"
	#include "api/video/builtin_video_bitrate_allocator_factory.h"
	#include "api/video/video_bitrate_allocation.h"
	#include "call/fake_network_pipe.h"
	#include "call/simulated_network.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp.h"
	#include "rtc_base/rate_limiter.h"
	#include "system_wrappers/include/sleep.h"
	#include "test/call_test.h"
	#include "test/fake_encoder.h"
	#include "test/field_trial.h"
	#include "test/gtest.h"
	#include "test/rtcp_packet_parser.h"
	#include "test/rtp_rtcp_observer.h"
	#include "test/video_encoder_proxy_factory.h"

	namespace webrtc {
	namespace {
	enum : int { // The first valid value is 1.
	kAbsSendTimeExtensionId = 1,
	};
	} // namespace

	class BandwidthEndToEndTest : public test::CallTest {
	public:
	BandwidthEndToEndTest() = default;
	};

	TEST_F(BandwidthEndToEndTest, ReceiveStreamSendsRemb) {
	class RembObserver : public test::EndToEndTest {
	public:
	RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	send_config->rtp.extensions.clear();
	send_config->rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
	(*receive_configs)[0].rtp.remb = true;
	(*receive_configs)[0].rtp.transport_cc = false;
	}

	Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
	test::RtcpPacketParser parser;
	EXPECT_TRUE(parser.Parse(packet, length));

	if (parser.remb()->num_packets() > 0) {
	EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc());
	EXPECT_LT(0U, parser.remb()->bitrate_bps());
	EXPECT_EQ(1U, parser.remb()->ssrcs().size());
	EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]);
	observation_complete_.Set();
	}

	return SEND_PACKET;
	}
	void PerformTest() override {
	EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
	"receiver RTCP REMB packet to be "
	"sent.";
	}
	} test;

	RunBaseTest(&test);
	}

	class BandwidthStatsTest : public test::EndToEndTest {
	public:
	explicit BandwidthStatsTest(bool send_side_bwe)
	: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
	sender_call_(nullptr),
	receiver_call_(nullptr),
	has_seen_pacer_delay_(false),
	send_side_bwe_(send_side_bwe) {}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	if (!send_side_bwe_) {
	send_config->rtp.extensions.clear();
	send_config->rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
	(*receive_configs)[0].rtp.remb = true;
	(*receive_configs)[0].rtp.transport_cc = false;
	}
	}

	Action OnSendRtp(const uint8_t* packet, size_t length) override {
	Call::Stats sender_stats = sender_call_->GetStats();
	Call::Stats receiver_stats = receiver_call_->GetStats();
	if (!has_seen_pacer_delay_)
	has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
	if (sender_stats.send_bandwidth_bps > 0 && has_seen_pacer_delay_) {
	if (send_side_bwe_ \|\| receiver_stats.recv_bandwidth_bps > 0)
	observation_complete_.Set();
	}
	return SEND_PACKET;
	}

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

	void PerformTest() override {
	EXPECT_TRUE(Wait()) << "Timed out while waiting for "
	"non-zero bandwidth stats.";
	}

	private:
	Call* sender_call_;
	Call* receiver_call_;
	bool has_seen_pacer_delay_;
	const bool send_side_bwe_;
	};

	TEST_F(BandwidthEndToEndTest, VerifySendSideBweStats) {
	BandwidthStatsTest test(true);
	RunBaseTest(&test);
	}

	TEST_F(BandwidthEndToEndTest, VerifyRecvSideBweStats) {
	BandwidthStatsTest test(false);
	RunBaseTest(&test);
	}

	// Verifies that it's possible to limit the send BWE by sending a REMB.
	// This is verified by allowing the send BWE to ramp-up to >1000 kbps,
	// then have the test generate a REMB of 500 kbps and verify that the send BWE
	// is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the
	// test verifies that the send BWE ramps back up to exactly 1000 kbps.
	TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
	class BweObserver : public test::EndToEndTest {
	public:
	BweObserver()
	: EndToEndTest(kDefaultTimeoutMs),
	sender_call_(nullptr),
	clock_(Clock::GetRealTimeClock()),
	sender_ssrc_(0),
	remb_bitrate_bps_(1000000),
	receive_transport_(nullptr),
	poller_thread_(&BitrateStatsPollingThread,
	this,
	"BitrateStatsPollingThread"),
	state_(kWaitForFirstRampUp),
	retransmission_rate_limiter_(clock_, 1000) {}

	~BweObserver() {}

	test::PacketTransport* CreateReceiveTransport(
	test::SingleThreadedTaskQueueForTesting* task_queue) override {
	receive_transport_ = new test::PacketTransport(
	task_queue, nullptr, this, test::PacketTransport::kReceiver,
	payload_type_map_,
	absl::make_unique<FakeNetworkPipe>(
	Clock::GetRealTimeClock(), absl::make_unique<SimulatedNetwork>(
	BuiltInNetworkBehaviorConfig())));
	return receive_transport_;
	}

	void ModifySenderBitrateConfig(
	BitrateConstraints* bitrate_config) override {
	// Set a high start bitrate to reduce the test completion time.
	bitrate_config->start_bitrate_bps = remb_bitrate_bps_;
	}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
	sender_ssrc_ = send_config->rtp.ssrcs[0];

	encoder_config->max_bitrate_bps = 2000000;

	ASSERT_EQ(1u, receive_configs->size());
	RtpRtcp::Configuration config;
	config.receiver_only = true;
	config.clock = clock_;
	config.outgoing_transport = receive_transport_;
	config.retransmission_rate_limiter = &retransmission_rate_limiter_;
	rtp_rtcp_ = RtpRtcp::Create(config);
	rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
	rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
	rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
	}

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

	static void BitrateStatsPollingThread(void* obj) {
	static_cast<BweObserver*>(obj)->PollStats();
	}

	void PollStats() {
	do {
	if (sender_call_) {
	Call::Stats stats = sender_call_->GetStats();
	switch (state_) {
	case kWaitForFirstRampUp:
	if (stats.send_bandwidth_bps >= remb_bitrate_bps_) {
	state_ = kWaitForRemb;
	remb_bitrate_bps_ /= 2;
	rtp_rtcp_->SetRemb(
	remb_bitrate_bps_,
	std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
	rtp_rtcp_->SendRTCP(kRtcpRr);
	}
	break;

	case kWaitForRemb:
	if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
	state_ = kWaitForSecondRampUp;
	remb_bitrate_bps_ *= 2;
	rtp_rtcp_->SetRemb(
	remb_bitrate_bps_,
	std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
	rtp_rtcp_->SendRTCP(kRtcpRr);
	}
	break;

	case kWaitForSecondRampUp:
	if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
	observation_complete_.Set();
	}
	break;
	}
	}
	} while (!stop_event_.Wait(1000));
	}

	void PerformTest() override {
	poller_thread_.Start();
	EXPECT_TRUE(Wait())
	<< "Timed out while waiting for bitrate to change according to REMB.";
	stop_event_.Set();
	poller_thread_.Stop();
	}

	private:
	enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };

	Call* sender_call_;
	Clock* const clock_;
	uint32_t sender_ssrc_;
	int remb_bitrate_bps_;
	std::unique_ptr<RtpRtcp> rtp_rtcp_;
	test::PacketTransport* receive_transport_;
	rtc::Event stop_event_;
	rtc::PlatformThread poller_thread_;
	TestState state_;
	RateLimiter retransmission_rate_limiter_;
	} test;

	RunBaseTest(&test);
	}

	TEST_F(BandwidthEndToEndTest, ReportsSetEncoderRates) {
	// If these fields trial are on, we get lower bitrates than expected by this
	// test, due to the packetization overhead and encoder pushback.
	webrtc::test::ScopedFieldTrials field_trials(
	std::string(field_trial::GetFieldTrialString()) +
	"WebRTC-SubtractPacketizationOverhead/Disabled/"
	"WebRTC-VideoRateControl/bitrate_adjuster:false/");
	class EncoderRateStatsTest : public test::EndToEndTest,
	public test::FakeEncoder {
	public:
	explicit EncoderRateStatsTest(
	test::SingleThreadedTaskQueueForTesting* task_queue)
	: EndToEndTest(kDefaultTimeoutMs),
	FakeEncoder(Clock::GetRealTimeClock()),
	task_queue_(task_queue),
	send_stream_(nullptr),
	encoder_factory_(this),
	bitrate_allocator_factory_(
	CreateBuiltinVideoBitrateAllocatorFactory()),
	bitrate_kbps_(0) {}

	void OnVideoStreamsCreated(
	VideoSendStream* send_stream,
	const std::vector<VideoReceiveStream*>& receive_streams) override {
	send_stream_ = send_stream;
	}

	void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config) override {
	send_config->encoder_settings.encoder_factory = &encoder_factory_;
	send_config->encoder_settings.bitrate_allocator_factory =
	bitrate_allocator_factory_.get();
	RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
	}

	int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
	uint32_t framerate) override {
	// Make sure not to trigger on any default zero bitrates.
	if (rate_allocation.get_sum_bps() == 0)
	return 0;
	rtc::CritScope lock(&crit_);
	bitrate_kbps_ = rate_allocation.get_sum_kbps();
	observation_complete_.Set();
	return 0;
	}

	void PerformTest() override {
	ASSERT_TRUE(Wait())
	<< "Timed out while waiting for encoder SetRates() call.";

	task_queue_->SendTask([this]() {
	WaitForEncoderTargetBitrateMatchStats();
	send_stream_->Stop();
	WaitForStatsReportZeroTargetBitrate();
	send_stream_->Start();
	WaitForEncoderTargetBitrateMatchStats();
	});
	}

	void WaitForEncoderTargetBitrateMatchStats() {
	for (int i = 0; i < kDefaultTimeoutMs; ++i) {
	VideoSendStream::Stats stats = send_stream_->GetStats();
	{
	rtc::CritScope lock(&crit_);
	if ((stats.target_media_bitrate_bps + 500) / 1000 ==
	static_cast<int>(bitrate_kbps_)) {
	return;
	}
	}
	SleepMs(1);
	}
	FAIL()
	<< "Timed out waiting for stats reporting the currently set bitrate.";
	}

	void WaitForStatsReportZeroTargetBitrate() {
	for (int i = 0; i < kDefaultTimeoutMs; ++i) {
	if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
	return;
	}
	SleepMs(1);
	}
	FAIL() << "Timed out waiting for stats reporting zero bitrate.";
	}

	private:
	test::SingleThreadedTaskQueueForTesting* const task_queue_;
	rtc::CriticalSection crit_;
	VideoSendStream* send_stream_;
	test::VideoEncoderProxyFactory encoder_factory_;
	std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
	uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
	} test(&task_queue_);

	RunBaseTest(&test);
	}
	} // namespace webrtc