call/bitrate_estimator_tests.cc - src.git - Git at Google

 /*
  *  Copyright (c) 2013 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 <functional>
 #include <list>
 #include <memory>
 #include <string>

 #include "call/call.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/thread_annotations.h"
 #include "test/call_test.h"
 #include "test/direct_transport.h"
 #include "test/encoder_settings.h"
 #include "test/fake_decoder.h"
 #include "test/fake_encoder.h"
 #include "test/frame_generator_capturer.h"
 #include "test/gtest.h"

 namespace webrtc {
 namespace {
 // Note: If you consider to re-use this class, think twice and instead consider
 // writing tests that don't depend on the logging system.
 class LogObserver {
  public:
   LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }

   ~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }

   void PushExpectedLogLine(const std::string& expected_log_line) {
     callback_.PushExpectedLogLine(expected_log_line);
   }

   bool Wait() { return callback_.Wait(); }

  private:
   class Callback : public rtc::LogSink {
    public:
     Callback() : done_(false, false) {}

     void OnLogMessage(const std::string& message) override {
       rtc::CritScope lock(&crit_sect_);
       // Ignore log lines that are due to missing AST extensions, these are
       // logged when we switch back from AST to TOF until the wrapping bitrate
       // estimator gives up on using AST.
       if (message.find("BitrateEstimator") != std::string::npos &&
           message.find("packet is missing") == std::string::npos) {
         received_log_lines_.push_back(message);
       }

       int num_popped = 0;
       while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
         std::string a = received_log_lines_.front();
         std::string b = expected_log_lines_.front();
         received_log_lines_.pop_front();
         expected_log_lines_.pop_front();
         num_popped++;
         EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
       }
       if (expected_log_lines_.size() <= 0) {
         if (num_popped > 0) {
           done_.Set();
         }
         return;
       }
     }

     bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }

     void PushExpectedLogLine(const std::string& expected_log_line) {
       rtc::CritScope lock(&crit_sect_);
       expected_log_lines_.push_back(expected_log_line);
     }

    private:
     typedef std::list<std::string> Strings;
     rtc::CriticalSection crit_sect_;
     Strings received_log_lines_ RTC_GUARDED_BY(crit_sect_);
     Strings expected_log_lines_ RTC_GUARDED_BY(crit_sect_);
     rtc::Event done_;
   };

   Callback callback_;
 };
 }  // namespace

 static const int kTOFExtensionId = 4;
 static const int kASTExtensionId = 5;

 class BitrateEstimatorTest : public test::CallTest {
  public:
   BitrateEstimatorTest() : receive_config_(nullptr) {}

   virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }

   virtual void SetUp() {
     task_queue_.SendTask([this]() {
       Call::Config config(event_log_.get());
       receiver_call_.reset(Call::Create(config));
       sender_call_.reset(Call::Create(config));

       send_transport_.reset(new test::DirectTransport(
           &task_queue_, sender_call_.get(), payload_type_map_));
       send_transport_->SetReceiver(receiver_call_->Receiver());
       receive_transport_.reset(new test::DirectTransport(
           &task_queue_, receiver_call_.get(), payload_type_map_));
       receive_transport_->SetReceiver(sender_call_->Receiver());

       video_send_config_ = VideoSendStream::Config(send_transport_.get());
       video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
       // Encoders will be set separately per stream.
       video_send_config_.encoder_settings.encoder = nullptr;
       video_send_config_.encoder_settings.payload_name = "FAKE";
       video_send_config_.encoder_settings.payload_type =
           kFakeVideoSendPayloadType;
       test::FillEncoderConfiguration(1, &video_encoder_config_);

       receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
       // receive_config_.decoders will be set by every stream separately.
       receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
       receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
       receive_config_.rtp.remb = true;
       receive_config_.rtp.extensions.push_back(
           RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
       receive_config_.rtp.extensions.push_back(
           RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
     });
   }

   virtual void TearDown() {
     task_queue_.SendTask([this]() {
       std::for_each(streams_.begin(), streams_.end(),
                     std::mem_fun(&Stream::StopSending));

       while (!streams_.empty()) {
         delete streams_.back();
         streams_.pop_back();
       }

       send_transport_.reset();
       receive_transport_.reset();

       receiver_call_.reset();
       sender_call_.reset();
     });
   }

  protected:
   friend class Stream;

   class Stream {
    public:
     explicit Stream(BitrateEstimatorTest* test)
         : test_(test),
           is_sending_receiving_(false),
           send_stream_(nullptr),
           frame_generator_capturer_(),
           fake_encoder_(Clock::GetRealTimeClock()),
           fake_decoder_() {
       test_->video_send_config_.rtp.ssrcs[0]++;
       test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
       send_stream_ = test_->sender_call_->CreateVideoSendStream(
           test_->video_send_config_.Copy(),
           test_->video_encoder_config_.Copy());
       RTC_DCHECK_EQ(1, test_->video_encoder_config_.number_of_streams);
       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
           kDefaultWidth, kDefaultHeight, kDefaultFramerate,
           Clock::GetRealTimeClock()));
       send_stream_->SetSource(
           frame_generator_capturer_.get(),
           VideoSendStream::DegradationPreference::kMaintainFramerate);
       send_stream_->Start();
       frame_generator_capturer_->Start();

       VideoReceiveStream::Decoder decoder;
       decoder.decoder = &fake_decoder_;
       decoder.payload_type =
           test_->video_send_config_.encoder_settings.payload_type;
       decoder.payload_name =
           test_->video_send_config_.encoder_settings.payload_name;
       test_->receive_config_.decoders.clear();
       test_->receive_config_.decoders.push_back(decoder);
       test_->receive_config_.rtp.remote_ssrc =
           test_->video_send_config_.rtp.ssrcs[0];
       test_->receive_config_.rtp.local_ssrc++;
       test_->receive_config_.renderer = &test->fake_renderer_;
       video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
           test_->receive_config_.Copy());
       video_receive_stream_->Start();
       is_sending_receiving_ = true;
     }

     ~Stream() {
       EXPECT_FALSE(is_sending_receiving_);
       test_->sender_call_->DestroyVideoSendStream(send_stream_);
       frame_generator_capturer_.reset(nullptr);
       send_stream_ = nullptr;
       if (video_receive_stream_) {
         test_->receiver_call_->DestroyVideoReceiveStream(video_receive_stream_);
         video_receive_stream_ = nullptr;
       }
     }

     void StopSending() {
       if (is_sending_receiving_) {
         frame_generator_capturer_->Stop();
         send_stream_->Stop();
         if (video_receive_stream_) {
           video_receive_stream_->Stop();
         }
         is_sending_receiving_ = false;
       }
     }

    private:
     BitrateEstimatorTest* test_;
     bool is_sending_receiving_;
     VideoSendStream* send_stream_;
     VideoReceiveStream* video_receive_stream_;
     std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };

   LogObserver receiver_log_;
   std::unique_ptr<test::DirectTransport> send_transport_;
   std::unique_ptr<test::DirectTransport> receive_transport_;
   std::unique_ptr<Call> sender_call_;
   std::unique_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };

 static const char* kAbsSendTimeLog =
     "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
 static const char* kSingleStreamLog =
     "RemoteBitrateEstimatorSingleStream: Instantiating.";

 TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
   task_queue_.SendTask([this]() {
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     streams_.push_back(new Stream(this));
   });
   EXPECT_TRUE(receiver_log_.Wait());
 }

 TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
   task_queue_.SendTask([this]() {
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
     streams_.push_back(new Stream(this));
   });
   EXPECT_TRUE(receiver_log_.Wait());
 }

 TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
   task_queue_.SendTask([this]() {
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     streams_.push_back(new Stream(this));
   });
   EXPECT_TRUE(receiver_log_.Wait());

   task_queue_.SendTask([this]() {
     video_send_config_.rtp.extensions[0] =
         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
     receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
     streams_.push_back(new Stream(this));
   });
   EXPECT_TRUE(receiver_log_.Wait());
 }

 // This test is flaky. See webrtc:5790.
 TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
   task_queue_.SendTask([this]() {
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
     streams_.push_back(new Stream(this));
   });
   EXPECT_TRUE(receiver_log_.Wait());

   task_queue_.SendTask([this]() {
     video_send_config_.rtp.extensions[0] =
         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
     receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
     streams_.push_back(new Stream(this));
   });
   EXPECT_TRUE(receiver_log_.Wait());

   task_queue_.SendTask([this]() {
     video_send_config_.rtp.extensions[0] =
         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
     receiver_log_.PushExpectedLogLine(
         "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
     streams_.push_back(new Stream(this));
     streams_[0]->StopSending();
     streams_[1]->StopSending();
   });
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc
	/*
	* Copyright (c) 2013 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 <functional>
	#include <list>
	#include <memory>
	#include <string>

	#include "call/call.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/event.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/thread_annotations.h"
	#include "test/call_test.h"
	#include "test/direct_transport.h"
	#include "test/encoder_settings.h"
	#include "test/fake_decoder.h"
	#include "test/fake_encoder.h"
	#include "test/frame_generator_capturer.h"
	#include "test/gtest.h"

	namespace webrtc {
	namespace {
	// Note: If you consider to re-use this class, think twice and instead consider
	// writing tests that don't depend on the logging system.
	class LogObserver {
	public:
	LogObserver() { rtc::LogMessage::AddLogToStream(&callback_, rtc::LS_INFO); }

	~LogObserver() { rtc::LogMessage::RemoveLogToStream(&callback_); }

	void PushExpectedLogLine(const std::string& expected_log_line) {
	callback_.PushExpectedLogLine(expected_log_line);
	}

	bool Wait() { return callback_.Wait(); }

	private:
	class Callback : public rtc::LogSink {
	public:
	Callback() : done_(false, false) {}

	void OnLogMessage(const std::string& message) override {
	rtc::CritScope lock(&crit_sect_);
	// Ignore log lines that are due to missing AST extensions, these are
	// logged when we switch back from AST to TOF until the wrapping bitrate
	// estimator gives up on using AST.
	if (message.find("BitrateEstimator") != std::string::npos &&
	message.find("packet is missing") == std::string::npos) {
	received_log_lines_.push_back(message);
	}

	int num_popped = 0;
	while (!received_log_lines_.empty() && !expected_log_lines_.empty()) {
	std::string a = received_log_lines_.front();
	std::string b = expected_log_lines_.front();
	received_log_lines_.pop_front();
	expected_log_lines_.pop_front();
	num_popped++;
	EXPECT_TRUE(a.find(b) != std::string::npos) << a << " != " << b;
	}
	if (expected_log_lines_.size() <= 0) {
	if (num_popped > 0) {
	done_.Set();
	}
	return;
	}
	}

	bool Wait() { return done_.Wait(test::CallTest::kDefaultTimeoutMs); }

	void PushExpectedLogLine(const std::string& expected_log_line) {
	rtc::CritScope lock(&crit_sect_);
	expected_log_lines_.push_back(expected_log_line);
	}

	private:
	typedef std::list<std::string> Strings;
	rtc::CriticalSection crit_sect_;
	Strings received_log_lines_ RTC_GUARDED_BY(crit_sect_);
	Strings expected_log_lines_ RTC_GUARDED_BY(crit_sect_);
	rtc::Event done_;
	};

	Callback callback_;
	};
	} // namespace

	static const int kTOFExtensionId = 4;
	static const int kASTExtensionId = 5;

	class BitrateEstimatorTest : public test::CallTest {
	public:
	BitrateEstimatorTest() : receive_config_(nullptr) {}

	virtual ~BitrateEstimatorTest() { EXPECT_TRUE(streams_.empty()); }

	virtual void SetUp() {
	task_queue_.SendTask([this]() {
	Call::Config config(event_log_.get());
	receiver_call_.reset(Call::Create(config));
	sender_call_.reset(Call::Create(config));

	send_transport_.reset(new test::DirectTransport(
	&task_queue_, sender_call_.get(), payload_type_map_));
	send_transport_->SetReceiver(receiver_call_->Receiver());
	receive_transport_.reset(new test::DirectTransport(
	&task_queue_, receiver_call_.get(), payload_type_map_));
	receive_transport_->SetReceiver(sender_call_->Receiver());

	video_send_config_ = VideoSendStream::Config(send_transport_.get());
	video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
	// Encoders will be set separately per stream.
	video_send_config_.encoder_settings.encoder = nullptr;
	video_send_config_.encoder_settings.payload_name = "FAKE";
	video_send_config_.encoder_settings.payload_type =
	kFakeVideoSendPayloadType;
	test::FillEncoderConfiguration(1, &video_encoder_config_);

	receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
	// receive_config_.decoders will be set by every stream separately.
	receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
	receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
	receive_config_.rtp.remb = true;
	receive_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
	receive_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
	});
	}

	virtual void TearDown() {
	task_queue_.SendTask([this]() {
	std::for_each(streams_.begin(), streams_.end(),
	std::mem_fun(&Stream::StopSending));

	while (!streams_.empty()) {
	delete streams_.back();
	streams_.pop_back();
	}

	send_transport_.reset();
	receive_transport_.reset();

	receiver_call_.reset();
	sender_call_.reset();
	});
	}

	protected:
	friend class Stream;

	class Stream {
	public:
	explicit Stream(BitrateEstimatorTest* test)
	: test_(test),
	is_sending_receiving_(false),
	send_stream_(nullptr),
	frame_generator_capturer_(),
	fake_encoder_(Clock::GetRealTimeClock()),
	fake_decoder_() {
	test_->video_send_config_.rtp.ssrcs[0]++;
	test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
	send_stream_ = test_->sender_call_->CreateVideoSendStream(
	test_->video_send_config_.Copy(),
	test_->video_encoder_config_.Copy());
	RTC_DCHECK_EQ(1, test_->video_encoder_config_.number_of_streams);
	frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
	kDefaultWidth, kDefaultHeight, kDefaultFramerate,
	Clock::GetRealTimeClock()));
	send_stream_->SetSource(
	frame_generator_capturer_.get(),
	VideoSendStream::DegradationPreference::kMaintainFramerate);
	send_stream_->Start();
	frame_generator_capturer_->Start();

	VideoReceiveStream::Decoder decoder;
	decoder.decoder = &fake_decoder_;
	decoder.payload_type =
	test_->video_send_config_.encoder_settings.payload_type;
	decoder.payload_name =
	test_->video_send_config_.encoder_settings.payload_name;
	test_->receive_config_.decoders.clear();
	test_->receive_config_.decoders.push_back(decoder);
	test_->receive_config_.rtp.remote_ssrc =
	test_->video_send_config_.rtp.ssrcs[0];
	test_->receive_config_.rtp.local_ssrc++;
	test_->receive_config_.renderer = &test->fake_renderer_;
	video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
	test_->receive_config_.Copy());
	video_receive_stream_->Start();
	is_sending_receiving_ = true;
	}

	~Stream() {
	EXPECT_FALSE(is_sending_receiving_);
	test_->sender_call_->DestroyVideoSendStream(send_stream_);
	frame_generator_capturer_.reset(nullptr);
	send_stream_ = nullptr;
	if (video_receive_stream_) {
	test_->receiver_call_->DestroyVideoReceiveStream(video_receive_stream_);
	video_receive_stream_ = nullptr;
	}
	}

	void StopSending() {
	if (is_sending_receiving_) {
	frame_generator_capturer_->Stop();
	send_stream_->Stop();
	if (video_receive_stream_) {
	video_receive_stream_->Stop();
	}
	is_sending_receiving_ = false;
	}
	}

	private:
	BitrateEstimatorTest* test_;
	bool is_sending_receiving_;
	VideoSendStream* send_stream_;
	VideoReceiveStream* video_receive_stream_;
	std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
	test::FakeEncoder fake_encoder_;
	test::FakeDecoder fake_decoder_;
	};

	LogObserver receiver_log_;
	std::unique_ptr<test::DirectTransport> send_transport_;
	std::unique_ptr<test::DirectTransport> receive_transport_;
	std::unique_ptr<Call> sender_call_;
	std::unique_ptr<Call> receiver_call_;
	VideoReceiveStream::Config receive_config_;
	std::vector<Stream*> streams_;
	};

	static const char* kAbsSendTimeLog =
	"RemoteBitrateEstimatorAbsSendTime: Instantiating.";
	static const char* kSingleStreamLog =
	"RemoteBitrateEstimatorSingleStream: Instantiating.";

	TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
	task_queue_.SendTask([this]() {
	video_send_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	streams_.push_back(new Stream(this));
	});
	EXPECT_TRUE(receiver_log_.Wait());
	}

	TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
	task_queue_.SendTask([this]() {
	video_send_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
	receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
	streams_.push_back(new Stream(this));
	});
	EXPECT_TRUE(receiver_log_.Wait());
	}

	TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
	task_queue_.SendTask([this]() {
	video_send_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	streams_.push_back(new Stream(this));
	});
	EXPECT_TRUE(receiver_log_.Wait());

	task_queue_.SendTask([this]() {
	video_send_config_.rtp.extensions[0] =
	RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
	receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
	receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
	streams_.push_back(new Stream(this));
	});
	EXPECT_TRUE(receiver_log_.Wait());
	}

	// This test is flaky. See webrtc:5790.
	TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
	task_queue_.SendTask([this]() {
	video_send_config_.rtp.extensions.push_back(
	RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
	receiver_log_.PushExpectedLogLine(kSingleStreamLog);
	streams_.push_back(new Stream(this));
	});
	EXPECT_TRUE(receiver_log_.Wait());

	task_queue_.SendTask([this]() {
	video_send_config_.rtp.extensions[0] =
	RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
	receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
	receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
	streams_.push_back(new Stream(this));
	});
	EXPECT_TRUE(receiver_log_.Wait());

	task_queue_.SendTask([this]() {
	video_send_config_.rtp.extensions[0] =
	RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
	receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
	receiver_log_.PushExpectedLogLine(
	"WrappingBitrateEstimator: Switching to transmission time offset RBE.");
	streams_.push_back(new Stream(this));
	streams_[0]->StopSending();
	streams_[1]->StopSending();
	});
	EXPECT_TRUE(receiver_log_.Wait());
	}
	} // namespace webrtc