test/call_test.h - src.git - Git at Google

 /*
  *  Copyright (c) 2014 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.
  */
 #ifndef TEST_CALL_TEST_H_
 #define TEST_CALL_TEST_H_

 #include <memory>
 #include <vector>

 #include "api/test/video/function_video_decoder_factory.h"
 #include "api/test/video/function_video_encoder_factory.h"
 #include "api/video/video_bitrate_allocator_factory.h"
 #include "call/call.h"
 #include "call/rtp_transport_controller_send.h"
 #include "logging/rtc_event_log/rtc_event_log.h"
 #include "modules/audio_device/include/test_audio_device.h"
 #include "test/encoder_settings.h"
 #include "test/fake_decoder.h"
 #include "test/fake_videorenderer.h"
 #include "test/fake_vp8_encoder.h"
 #include "test/frame_generator_capturer.h"
 #include "test/rtp_rtcp_observer.h"
 #include "test/single_threaded_task_queue.h"

 namespace webrtc {
 namespace test {

 class BaseTest;

 class CallTest : public ::testing::Test {
  public:
   CallTest();
   virtual ~CallTest();

   static constexpr size_t kNumSsrcs = 6;
   static const int kNumSimulcastStreams = 3;
   static const int kDefaultWidth = 320;
   static const int kDefaultHeight = 180;
   static const int kDefaultFramerate = 30;
   static const int kDefaultTimeoutMs;
   static const int kLongTimeoutMs;
   enum classPayloadTypes : uint8_t {
     kSendRtxPayloadType = 98,
     kRtxRedPayloadType = 99,
     kVideoSendPayloadType = 100,
     kAudioSendPayloadType = 103,
     kRedPayloadType = 118,
     kUlpfecPayloadType = 119,
     kFlexfecPayloadType = 120,
     kPayloadTypeH264 = 122,
     kPayloadTypeVP8 = 123,
     kPayloadTypeVP9 = 124,
     kFakeVideoSendPayloadType = 125,
   };
   static const uint32_t kSendRtxSsrcs[kNumSsrcs];
   static const uint32_t kVideoSendSsrcs[kNumSsrcs];
   static const uint32_t kAudioSendSsrc;
   static const uint32_t kFlexfecSendSsrc;
   static const uint32_t kReceiverLocalVideoSsrc;
   static const uint32_t kReceiverLocalAudioSsrc;
   static const int kNackRtpHistoryMs;
   static const uint8_t kDefaultKeepalivePayloadType;
   static const std::map<uint8_t, MediaType> payload_type_map_;

  protected:
   // RunBaseTest overwrites the audio_state of the send and receive Call configs
   // to simplify test code.
   void RunBaseTest(BaseTest* test);

   void CreateCalls();
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall();
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
   void DestroyCalls();

   void CreateVideoSendConfig(VideoSendStream::Config* video_config,
                              size_t num_video_streams,
                              size_t num_used_ssrcs,
                              Transport* send_transport);
   void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
                                     size_t num_flexfec_streams,
                                     Transport* send_transport);
   void SetAudioConfig(const AudioSendStream::Config& config);

   void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
   void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
   void SetReceiveUlpFecConfig(VideoReceiveStream::Config* receive_config);
   void CreateSendConfig(size_t num_video_streams,
                         size_t num_audio_streams,
                         size_t num_flexfec_streams,
                         Transport* send_transport);

   void CreateMatchingVideoReceiveConfigs(
       const VideoSendStream::Config& video_send_config,
       Transport* rtcp_send_transport);
   void CreateMatchingVideoReceiveConfigs(
       const VideoSendStream::Config& video_send_config,
       Transport* rtcp_send_transport,
       bool send_side_bwe,
       VideoDecoderFactory* decoder_factory,
       absl::optional<size_t> decode_sub_stream,
       bool receiver_reference_time_report,
       int rtp_history_ms);
   void AddMatchingVideoReceiveConfigs(
       std::vector<VideoReceiveStream::Config>* receive_configs,
       const VideoSendStream::Config& video_send_config,
       Transport* rtcp_send_transport,
       bool send_side_bwe,
       VideoDecoderFactory* decoder_factory,
       absl::optional<size_t> decode_sub_stream,
       bool receiver_reference_time_report,
       int rtp_history_ms);

   void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
   void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
   static AudioReceiveStream::Config CreateMatchingAudioConfig(
       const AudioSendStream::Config& send_config,
       rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
       Transport* transport,
       std::string sync_group);
   void CreateMatchingFecConfig(
       Transport* transport,
       const VideoSendStream::Config& video_send_config);
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);

   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
                                              float speed,
                                              int framerate,
                                              int width,
                                              int height);
   void CreateFrameGeneratorCapturer(int framerate, int width, int height);
   void CreateFakeAudioDevices(
       std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
       std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);

   void CreateVideoStreams();
   void CreateVideoSendStreams();
   void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
   void CreateAudioStreams();
   void CreateFlexfecStreams();

   void ConnectVideoSourcesToStreams();

   void AssociateFlexfecStreamsWithVideoStreams();
   void DissociateFlexfecStreamsFromVideoStreams();

   void Start();
   void StartVideoStreams();
   void Stop();
   void StopVideoStreams();
   void DestroyStreams();
   void DestroyVideoSendStreams();
   void SetFakeVideoCaptureRotation(VideoRotation rotation);

   void SetVideoDegradation(DegradationPreference preference);

   VideoSendStream::Config* GetVideoSendConfig();
   void SetVideoSendConfig(const VideoSendStream::Config& config);
   VideoEncoderConfig* GetVideoEncoderConfig();
   void SetVideoEncoderConfig(const VideoEncoderConfig& config);
   VideoSendStream* GetVideoSendStream();
   FlexfecReceiveStream::Config* GetFlexFecConfig();

   Clock* const clock_;

   std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
   std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
   std::unique_ptr<Call> sender_call_;
   RtpTransportControllerSend* sender_call_transport_controller_;
   std::unique_ptr<PacketTransport> send_transport_;
   std::vector<VideoSendStream::Config> video_send_configs_;
   std::vector<VideoEncoderConfig> video_encoder_configs_;
   std::vector<VideoSendStream*> video_send_streams_;
   AudioSendStream::Config audio_send_config_;
   AudioSendStream* audio_send_stream_;

   std::unique_ptr<Call> receiver_call_;
   std::unique_ptr<PacketTransport> receive_transport_;
   std::vector<VideoReceiveStream::Config> video_receive_configs_;
   std::vector<VideoReceiveStream*> video_receive_streams_;
   std::vector<AudioReceiveStream::Config> audio_receive_configs_;
   std::vector<AudioReceiveStream*> audio_receive_streams_;
   std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
   std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;

   test::FrameGeneratorCapturer* frame_generator_capturer_;
   std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
       video_sources_;
   DegradationPreference degradation_preference_ =
       DegradationPreference::MAINTAIN_FRAMERATE;

   std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;

   test::FunctionVideoEncoderFactory fake_encoder_factory_;
   int fake_encoder_max_bitrate_ = -1;
   test::FunctionVideoDecoderFactory fake_decoder_factory_;
   std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
   // Number of simulcast substreams.
   size_t num_video_streams_;
   size_t num_audio_streams_;
   size_t num_flexfec_streams_;
   rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
   rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
   test::FakeVideoRenderer fake_renderer_;

   SingleThreadedTaskQueueForTesting task_queue_;

  private:
   rtc::scoped_refptr<AudioProcessing> apm_send_;
   rtc::scoped_refptr<AudioProcessing> apm_recv_;
   rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
   rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
 };

 class BaseTest : public RtpRtcpObserver {
  public:
   BaseTest();
   explicit BaseTest(int timeout_ms);
   virtual ~BaseTest();

   virtual void PerformTest() = 0;
   virtual bool ShouldCreateReceivers() const = 0;

   virtual size_t GetNumVideoStreams() const;
   virtual size_t GetNumAudioStreams() const;
   virtual size_t GetNumFlexfecStreams() const;

   virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
   virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
   virtual void OnFakeAudioDevicesCreated(
       TestAudioDeviceModule* send_audio_device,
       TestAudioDeviceModule* recv_audio_device);

   virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
   virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);

   virtual void OnRtpTransportControllerSendCreated(
       RtpTransportControllerSend* controller);
   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);

   virtual test::PacketTransport* CreateSendTransport(
       SingleThreadedTaskQueueForTesting* task_queue,
       Call* sender_call);
   virtual test::PacketTransport* CreateReceiveTransport(
       SingleThreadedTaskQueueForTesting* task_queue);

   virtual void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config);
   virtual void ModifyVideoCaptureStartResolution(int* width,
                                                  int* heigt,
                                                  int* frame_rate);
   virtual void ModifyVideoDegradationPreference(
       DegradationPreference* degradation_preference);

   virtual void OnVideoStreamsCreated(
       VideoSendStream* send_stream,
       const std::vector<VideoReceiveStream*>& receive_streams);

   virtual void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs);
   virtual void OnAudioStreamsCreated(
       AudioSendStream* send_stream,
       const std::vector<AudioReceiveStream*>& receive_streams);

   virtual void ModifyFlexfecConfigs(
       std::vector<FlexfecReceiveStream::Config>* receive_configs);
   virtual void OnFlexfecStreamsCreated(
       const std::vector<FlexfecReceiveStream*>& receive_streams);

   virtual void OnFrameGeneratorCapturerCreated(
       FrameGeneratorCapturer* frame_generator_capturer);

   virtual void OnStreamsStopped();
 };

 class SendTest : public BaseTest {
  public:
   explicit SendTest(int timeout_ms);

   bool ShouldCreateReceivers() const override;
 };

 class EndToEndTest : public BaseTest {
  public:
   EndToEndTest();
   explicit EndToEndTest(int timeout_ms);

   bool ShouldCreateReceivers() const override;
 };

 }  // namespace test
 }  // namespace webrtc

 #endif  // TEST_CALL_TEST_H_
	/*
	* Copyright (c) 2014 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.
	*/
	#ifndef TEST_CALL_TEST_H_
	#define TEST_CALL_TEST_H_

	#include <memory>
	#include <vector>

	#include "api/test/video/function_video_decoder_factory.h"
	#include "api/test/video/function_video_encoder_factory.h"
	#include "api/video/video_bitrate_allocator_factory.h"
	#include "call/call.h"
	#include "call/rtp_transport_controller_send.h"
	#include "logging/rtc_event_log/rtc_event_log.h"
	#include "modules/audio_device/include/test_audio_device.h"
	#include "test/encoder_settings.h"
	#include "test/fake_decoder.h"
	#include "test/fake_videorenderer.h"
	#include "test/fake_vp8_encoder.h"
	#include "test/frame_generator_capturer.h"
	#include "test/rtp_rtcp_observer.h"
	#include "test/single_threaded_task_queue.h"

	namespace webrtc {
	namespace test {

	class BaseTest;

	class CallTest : public ::testing::Test {
	public:
	CallTest();
	virtual ~CallTest();

	static constexpr size_t kNumSsrcs = 6;
	static const int kNumSimulcastStreams = 3;
	static const int kDefaultWidth = 320;
	static const int kDefaultHeight = 180;
	static const int kDefaultFramerate = 30;
	static const int kDefaultTimeoutMs;
	static const int kLongTimeoutMs;
	enum classPayloadTypes : uint8_t {
	kSendRtxPayloadType = 98,
	kRtxRedPayloadType = 99,
	kVideoSendPayloadType = 100,
	kAudioSendPayloadType = 103,
	kRedPayloadType = 118,
	kUlpfecPayloadType = 119,
	kFlexfecPayloadType = 120,
	kPayloadTypeH264 = 122,
	kPayloadTypeVP8 = 123,
	kPayloadTypeVP9 = 124,
	kFakeVideoSendPayloadType = 125,
	};
	static const uint32_t kSendRtxSsrcs[kNumSsrcs];
	static const uint32_t kVideoSendSsrcs[kNumSsrcs];
	static const uint32_t kAudioSendSsrc;
	static const uint32_t kFlexfecSendSsrc;
	static const uint32_t kReceiverLocalVideoSsrc;
	static const uint32_t kReceiverLocalAudioSsrc;
	static const int kNackRtpHistoryMs;
	static const uint8_t kDefaultKeepalivePayloadType;
	static const std::map<uint8_t, MediaType> payload_type_map_;

	protected:
	// RunBaseTest overwrites the audio_state of the send and receive Call configs
	// to simplify test code.
	void RunBaseTest(BaseTest* test);

	void CreateCalls();
	void CreateCalls(const Call::Config& sender_config,
	const Call::Config& receiver_config);
	void CreateSenderCall();
	void CreateSenderCall(const Call::Config& config);
	void CreateReceiverCall(const Call::Config& config);
	void DestroyCalls();

	void CreateVideoSendConfig(VideoSendStream::Config* video_config,
	size_t num_video_streams,
	size_t num_used_ssrcs,
	Transport* send_transport);
	void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
	size_t num_flexfec_streams,
	Transport* send_transport);
	void SetAudioConfig(const AudioSendStream::Config& config);

	void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
	void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
	void SetReceiveUlpFecConfig(VideoReceiveStream::Config* receive_config);
	void CreateSendConfig(size_t num_video_streams,
	size_t num_audio_streams,
	size_t num_flexfec_streams,
	Transport* send_transport);

	void CreateMatchingVideoReceiveConfigs(
	const VideoSendStream::Config& video_send_config,
	Transport* rtcp_send_transport);
	void CreateMatchingVideoReceiveConfigs(
	const VideoSendStream::Config& video_send_config,
	Transport* rtcp_send_transport,
	bool send_side_bwe,
	VideoDecoderFactory* decoder_factory,
	absl::optional<size_t> decode_sub_stream,
	bool receiver_reference_time_report,
	int rtp_history_ms);
	void AddMatchingVideoReceiveConfigs(
	std::vector<VideoReceiveStream::Config>* receive_configs,
	const VideoSendStream::Config& video_send_config,
	Transport* rtcp_send_transport,
	bool send_side_bwe,
	VideoDecoderFactory* decoder_factory,
	absl::optional<size_t> decode_sub_stream,
	bool receiver_reference_time_report,
	int rtp_history_ms);

	void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
	void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
	static AudioReceiveStream::Config CreateMatchingAudioConfig(
	const AudioSendStream::Config& send_config,
	rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
	Transport* transport,
	std::string sync_group);
	void CreateMatchingFecConfig(
	Transport* transport,
	const VideoSendStream::Config& video_send_config);
	void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);

	void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
	float speed,
	int framerate,
	int width,
	int height);
	void CreateFrameGeneratorCapturer(int framerate, int width, int height);
	void CreateFakeAudioDevices(
	std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
	std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);

	void CreateVideoStreams();
	void CreateVideoSendStreams();
	void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
	void CreateAudioStreams();
	void CreateFlexfecStreams();

	void ConnectVideoSourcesToStreams();

	void AssociateFlexfecStreamsWithVideoStreams();
	void DissociateFlexfecStreamsFromVideoStreams();

	void Start();
	void StartVideoStreams();
	void Stop();
	void StopVideoStreams();
	void DestroyStreams();
	void DestroyVideoSendStreams();
	void SetFakeVideoCaptureRotation(VideoRotation rotation);

	void SetVideoDegradation(DegradationPreference preference);

	VideoSendStream::Config* GetVideoSendConfig();
	void SetVideoSendConfig(const VideoSendStream::Config& config);
	VideoEncoderConfig* GetVideoEncoderConfig();
	void SetVideoEncoderConfig(const VideoEncoderConfig& config);
	VideoSendStream* GetVideoSendStream();
	FlexfecReceiveStream::Config* GetFlexFecConfig();

	Clock* const clock_;

	std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
	std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
	std::unique_ptr<Call> sender_call_;
	RtpTransportControllerSend* sender_call_transport_controller_;
	std::unique_ptr<PacketTransport> send_transport_;
	std::vector<VideoSendStream::Config> video_send_configs_;
	std::vector<VideoEncoderConfig> video_encoder_configs_;
	std::vector<VideoSendStream*> video_send_streams_;
	AudioSendStream::Config audio_send_config_;
	AudioSendStream* audio_send_stream_;

	std::unique_ptr<Call> receiver_call_;
	std::unique_ptr<PacketTransport> receive_transport_;
	std::vector<VideoReceiveStream::Config> video_receive_configs_;
	std::vector<VideoReceiveStream*> video_receive_streams_;
	std::vector<AudioReceiveStream::Config> audio_receive_configs_;
	std::vector<AudioReceiveStream*> audio_receive_streams_;
	std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
	std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;

	test::FrameGeneratorCapturer* frame_generator_capturer_;
	std::vector<std::unique_ptr<rtc::VideoSourceInterface<VideoFrame>>>
	video_sources_;
	DegradationPreference degradation_preference_ =
	DegradationPreference::MAINTAIN_FRAMERATE;

	std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;

	test::FunctionVideoEncoderFactory fake_encoder_factory_;
	int fake_encoder_max_bitrate_ = -1;
	test::FunctionVideoDecoderFactory fake_decoder_factory_;
	std::unique_ptr<VideoBitrateAllocatorFactory> bitrate_allocator_factory_;
	// Number of simulcast substreams.
	size_t num_video_streams_;
	size_t num_audio_streams_;
	size_t num_flexfec_streams_;
	rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
	rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
	test::FakeVideoRenderer fake_renderer_;

	SingleThreadedTaskQueueForTesting task_queue_;

	private:
	rtc::scoped_refptr<AudioProcessing> apm_send_;
	rtc::scoped_refptr<AudioProcessing> apm_recv_;
	rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
	rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
	};

	class BaseTest : public RtpRtcpObserver {
	public:
	BaseTest();
	explicit BaseTest(int timeout_ms);
	virtual ~BaseTest();

	virtual void PerformTest() = 0;
	virtual bool ShouldCreateReceivers() const = 0;

	virtual size_t GetNumVideoStreams() const;
	virtual size_t GetNumAudioStreams() const;
	virtual size_t GetNumFlexfecStreams() const;

	virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
	virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
	virtual void OnFakeAudioDevicesCreated(
	TestAudioDeviceModule* send_audio_device,
	TestAudioDeviceModule* recv_audio_device);

	virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
	virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);

	virtual void OnRtpTransportControllerSendCreated(
	RtpTransportControllerSend* controller);
	virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);

	virtual test::PacketTransport* CreateSendTransport(
	SingleThreadedTaskQueueForTesting* task_queue,
	Call* sender_call);
	virtual test::PacketTransport* CreateReceiveTransport(
	SingleThreadedTaskQueueForTesting* task_queue);

	virtual void ModifyVideoConfigs(
	VideoSendStream::Config* send_config,
	std::vector<VideoReceiveStream::Config>* receive_configs,
	VideoEncoderConfig* encoder_config);
	virtual void ModifyVideoCaptureStartResolution(int* width,
	int* heigt,
	int* frame_rate);
	virtual void ModifyVideoDegradationPreference(
	DegradationPreference* degradation_preference);

	virtual void OnVideoStreamsCreated(
	VideoSendStream* send_stream,
	const std::vector<VideoReceiveStream*>& receive_streams);

	virtual void ModifyAudioConfigs(
	AudioSendStream::Config* send_config,
	std::vector<AudioReceiveStream::Config>* receive_configs);
	virtual void OnAudioStreamsCreated(
	AudioSendStream* send_stream,
	const std::vector<AudioReceiveStream*>& receive_streams);

	virtual void ModifyFlexfecConfigs(
	std::vector<FlexfecReceiveStream::Config>* receive_configs);
	virtual void OnFlexfecStreamsCreated(
	const std::vector<FlexfecReceiveStream*>& receive_streams);

	virtual void OnFrameGeneratorCapturerCreated(
	FrameGeneratorCapturer* frame_generator_capturer);

	virtual void OnStreamsStopped();
	};

	class SendTest : public BaseTest {
	public:
	explicit SendTest(int timeout_ms);

	bool ShouldCreateReceivers() const override;
	};

	class EndToEndTest : public BaseTest {
	public:
	EndToEndTest();
	explicit EndToEndTest(int timeout_ms);

	bool ShouldCreateReceivers() const override;
	};

	} // namespace test
	} // namespace webrtc

	#endif // TEST_CALL_TEST_H_