test/scenario/call_client.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 "test/scenario/call_client.h"

 #include <utility>

 #include "absl/memory/memory.h"
 #include "api/rtc_event_log/rtc_event_log.h"
 #include "api/rtc_event_log/rtc_event_log_factory.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"

 namespace webrtc {
 namespace test {
 namespace {
 static constexpr size_t kNumSsrcs = 6;
 const uint32_t kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE, 0xBADCAFF,
                                            0xBADCB00, 0xBADCB01, 0xBADCB02};
 const uint32_t kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF,
                                              0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
 const uint32_t kVideoRecvLocalSsrcs[kNumSsrcs] = {0xDAB001, 0xDAB002, 0xDAB003,
                                                   0xDAB004, 0xDAB005, 0xDAB006};
 const uint32_t kAudioSendSsrc = 0xDEADBEEF;
 const uint32_t kReceiverLocalAudioSsrc = 0x1234567;

 const char* kPriorityStreamId = "priority-track";

 constexpr int kEventLogOutputIntervalMs = 5000;

 CallClientFakeAudio InitAudio(TimeController* time_controller) {
   CallClientFakeAudio setup;
   auto capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
   auto renderer = TestAudioDeviceModule::CreateDiscardRenderer(48000);
   setup.fake_audio_device = TestAudioDeviceModule::Create(
       time_controller->GetTaskQueueFactory(), std::move(capturer),
       std::move(renderer), 1.f);
   setup.apm = AudioProcessingBuilder().Create();
   setup.fake_audio_device->Init();
   AudioState::Config audio_state_config;
   audio_state_config.audio_mixer = AudioMixerImpl::Create();
   audio_state_config.audio_processing = setup.apm;
   audio_state_config.audio_device_module = setup.fake_audio_device;
   setup.audio_state = AudioState::Create(audio_state_config);
   setup.fake_audio_device->RegisterAudioCallback(
       setup.audio_state->audio_transport());
   return setup;
 }

 Call* CreateCall(TimeController* time_controller,
                  RtcEventLog* event_log,
                  CallClientConfig config,
                  LoggingNetworkControllerFactory* network_controller_factory,
                  rtc::scoped_refptr<AudioState> audio_state) {
   CallConfig call_config(event_log);
   call_config.bitrate_config.max_bitrate_bps =
       config.transport.rates.max_rate.bps_or(-1);
   call_config.bitrate_config.min_bitrate_bps =
       config.transport.rates.min_rate.bps();
   call_config.bitrate_config.start_bitrate_bps =
       config.transport.rates.start_rate.bps();
   call_config.task_queue_factory = time_controller->GetTaskQueueFactory();
   call_config.network_controller_factory = network_controller_factory;
   call_config.audio_state = audio_state;
   return Call::Create(call_config, time_controller->GetClock(),
                       time_controller->CreateProcessThread("CallModules"),
                       time_controller->CreateProcessThread("Pacer"));
 }

 std::unique_ptr<RtcEventLog> CreateEventLog(
     TaskQueueFactory* task_queue_factory,
     LogWriterFactoryInterface* log_writer_factory) {
   if (!log_writer_factory) {
     return absl::make_unique<RtcEventLogNull>();
   }
   auto event_log = RtcEventLogFactory(task_queue_factory)
                        .CreateRtcEventLog(RtcEventLog::EncodingType::NewFormat);
   bool success = event_log->StartLogging(log_writer_factory->Create(".rtc.dat"),
                                          kEventLogOutputIntervalMs);
   RTC_CHECK(success);
   return event_log;
 }
 }

 LoggingNetworkControllerFactory::LoggingNetworkControllerFactory(
     LogWriterFactoryInterface* log_writer_factory,
     TransportControllerConfig config) {
   if (config.cc_factory) {
     cc_factory_ = config.cc_factory;
     if (log_writer_factory)
       RTC_LOG(LS_WARNING)
           << "Can't log controller state for injected network controllers";
   } else {
     if (log_writer_factory) {
       goog_cc_factory_.AttachWriter(
           log_writer_factory->Create(".cc_state.txt"));
       print_cc_state_ = true;
     }
     cc_factory_ = &goog_cc_factory_;
   }
 }

 LoggingNetworkControllerFactory::~LoggingNetworkControllerFactory() {
 }

 void LoggingNetworkControllerFactory::LogCongestionControllerStats(
     Timestamp at_time) {
   if (print_cc_state_)
     goog_cc_factory_.PrintState(at_time);
 }

 std::unique_ptr<NetworkControllerInterface>
 LoggingNetworkControllerFactory::Create(NetworkControllerConfig config) {
   return cc_factory_->Create(config);
 }

 TimeDelta LoggingNetworkControllerFactory::GetProcessInterval() const {
   return cc_factory_->GetProcessInterval();
 }

 CallClient::CallClient(
     TimeController* time_controller,
     std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
     CallClientConfig config)
     : time_controller_(time_controller),
       clock_(time_controller->GetClock()),
       log_writer_factory_(std::move(log_writer_factory)),
       network_controller_factory_(log_writer_factory_.get(), config.transport),
       header_parser_(RtpHeaderParser::Create()),
       task_queue_(time_controller->GetTaskQueueFactory()->CreateTaskQueue(
           "CallClient",
           TaskQueueFactory::Priority::NORMAL)) {
   SendTask([this, config] {
     event_log_ = CreateEventLog(time_controller_->GetTaskQueueFactory(),
                                 log_writer_factory_.get());
     fake_audio_setup_ = InitAudio(time_controller_);
     call_.reset(CreateCall(time_controller_, event_log_.get(), config,
                            &network_controller_factory_,
                            fake_audio_setup_.audio_state));
     transport_ = absl::make_unique<NetworkNodeTransport>(clock_, call_.get());
   });
 }

 CallClient::~CallClient() {
   SendTask([&] {
     call_.reset();
     fake_audio_setup_ = {};
     rtc::Event done;
     event_log_->StopLogging([&done] { done.Set(); });
     done.Wait(rtc::Event::kForever);
     event_log_.reset();
   });
 }

 ColumnPrinter CallClient::StatsPrinter() {
   return ColumnPrinter::Lambda(
       "pacer_delay call_send_bw",
       [this](rtc::SimpleStringBuilder& sb) {
         Call::Stats call_stats = call_->GetStats();
         sb.AppendFormat("%.3lf %.0lf", call_stats.pacer_delay_ms / 1000.0,
                         call_stats.send_bandwidth_bps / 8.0);
       },
       64);
 }

 Call::Stats CallClient::GetStats() {
   return call_->GetStats();
 }

 void CallClient::OnPacketReceived(EmulatedIpPacket packet) {
   // Removes added overhead before delivering packet to sender.
   size_t size =
       packet.data.size() - route_overhead_.at(packet.to.ipaddr()).bytes();
   RTC_DCHECK_GE(size, 0);
   packet.data.SetSize(size);

   MediaType media_type = MediaType::ANY;
   if (!RtpHeaderParser::IsRtcp(packet.cdata(), packet.data.size())) {
     auto ssrc = RtpHeaderParser::GetSsrc(packet.cdata(), packet.data.size());
     RTC_CHECK(ssrc.has_value());
     media_type = ssrc_media_types_[*ssrc];
   }
   struct Closure {
     void operator()() {
       call->Receiver()->DeliverPacket(media_type, packet.data,
                                       packet.arrival_time.us());
     }
     Call* call;
     MediaType media_type;
     EmulatedIpPacket packet;
   };
   task_queue_.PostTask(Closure{call_.get(), media_type, std::move(packet)});
 }

 std::unique_ptr<RtcEventLogOutput> CallClient::GetLogWriter(std::string name) {
   if (!log_writer_factory_ || name.empty())
     return nullptr;
   return log_writer_factory_->Create(name);
 }

 uint32_t CallClient::GetNextVideoSsrc() {
   RTC_CHECK_LT(next_video_ssrc_index_, kNumSsrcs);
   return kVideoSendSsrcs[next_video_ssrc_index_++];
 }

 uint32_t CallClient::GetNextVideoLocalSsrc() {
   RTC_CHECK_LT(next_video_local_ssrc_index_, kNumSsrcs);
   return kVideoRecvLocalSsrcs[next_video_local_ssrc_index_++];
 }

 uint32_t CallClient::GetNextAudioSsrc() {
   RTC_CHECK_LT(next_audio_ssrc_index_, 1);
   next_audio_ssrc_index_++;
   return kAudioSendSsrc;
 }

 uint32_t CallClient::GetNextAudioLocalSsrc() {
   RTC_CHECK_LT(next_audio_local_ssrc_index_, 1);
   next_audio_local_ssrc_index_++;
   return kReceiverLocalAudioSsrc;
 }

 uint32_t CallClient::GetNextRtxSsrc() {
   RTC_CHECK_LT(next_rtx_ssrc_index_, kNumSsrcs);
   return kSendRtxSsrcs[next_rtx_ssrc_index_++];
 }

 std::string CallClient::GetNextPriorityId() {
   RTC_CHECK_LT(next_priority_index_++, 1);
   return kPriorityStreamId;
 }

 void CallClient::AddExtensions(std::vector<RtpExtension> extensions) {
   for (const auto& extension : extensions)
     header_parser_->RegisterRtpHeaderExtension(extension);
 }

 void CallClient::SendTask(std::function<void()> task) {
   time_controller_->InvokeWithControlledYield(
       [&] { task_queue_.SendTask(std::move(task)); });
 }

 CallClientPair::~CallClientPair() = default;

 }  // namespace 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 "test/scenario/call_client.h"

	#include <utility>

	#include "absl/memory/memory.h"
	#include "api/rtc_event_log/rtc_event_log.h"
	#include "api/rtc_event_log/rtc_event_log_factory.h"
	#include "modules/audio_mixer/audio_mixer_impl.h"

	namespace webrtc {
	namespace test {
	namespace {
	static constexpr size_t kNumSsrcs = 6;
	const uint32_t kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE, 0xBADCAFF,
	0xBADCB00, 0xBADCB01, 0xBADCB02};
	const uint32_t kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, 0xC0FFEF,
	0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
	const uint32_t kVideoRecvLocalSsrcs[kNumSsrcs] = {0xDAB001, 0xDAB002, 0xDAB003,
	0xDAB004, 0xDAB005, 0xDAB006};
	const uint32_t kAudioSendSsrc = 0xDEADBEEF;
	const uint32_t kReceiverLocalAudioSsrc = 0x1234567;

	const char* kPriorityStreamId = "priority-track";

	constexpr int kEventLogOutputIntervalMs = 5000;

	CallClientFakeAudio InitAudio(TimeController* time_controller) {
	CallClientFakeAudio setup;
	auto capturer = TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
	auto renderer = TestAudioDeviceModule::CreateDiscardRenderer(48000);
	setup.fake_audio_device = TestAudioDeviceModule::Create(
	time_controller->GetTaskQueueFactory(), std::move(capturer),
	std::move(renderer), 1.f);
	setup.apm = AudioProcessingBuilder().Create();
	setup.fake_audio_device->Init();
	AudioState::Config audio_state_config;
	audio_state_config.audio_mixer = AudioMixerImpl::Create();
	audio_state_config.audio_processing = setup.apm;
	audio_state_config.audio_device_module = setup.fake_audio_device;
	setup.audio_state = AudioState::Create(audio_state_config);
	setup.fake_audio_device->RegisterAudioCallback(
	setup.audio_state->audio_transport());
	return setup;
	}

	Call* CreateCall(TimeController* time_controller,
	RtcEventLog* event_log,
	CallClientConfig config,
	LoggingNetworkControllerFactory* network_controller_factory,
	rtc::scoped_refptr<AudioState> audio_state) {
	CallConfig call_config(event_log);
	call_config.bitrate_config.max_bitrate_bps =
	config.transport.rates.max_rate.bps_or(-1);
	call_config.bitrate_config.min_bitrate_bps =
	config.transport.rates.min_rate.bps();
	call_config.bitrate_config.start_bitrate_bps =
	config.transport.rates.start_rate.bps();
	call_config.task_queue_factory = time_controller->GetTaskQueueFactory();
	call_config.network_controller_factory = network_controller_factory;
	call_config.audio_state = audio_state;
	return Call::Create(call_config, time_controller->GetClock(),
	time_controller->CreateProcessThread("CallModules"),
	time_controller->CreateProcessThread("Pacer"));
	}

	std::unique_ptr<RtcEventLog> CreateEventLog(
	TaskQueueFactory* task_queue_factory,
	LogWriterFactoryInterface* log_writer_factory) {
	if (!log_writer_factory) {
	return absl::make_unique<RtcEventLogNull>();
	}
	auto event_log = RtcEventLogFactory(task_queue_factory)
	.CreateRtcEventLog(RtcEventLog::EncodingType::NewFormat);
	bool success = event_log->StartLogging(log_writer_factory->Create(".rtc.dat"),
	kEventLogOutputIntervalMs);
	RTC_CHECK(success);
	return event_log;
	}
	}

	LoggingNetworkControllerFactory::LoggingNetworkControllerFactory(
	LogWriterFactoryInterface* log_writer_factory,
	TransportControllerConfig config) {
	if (config.cc_factory) {
	cc_factory_ = config.cc_factory;
	if (log_writer_factory)
	RTC_LOG(LS_WARNING)
	<< "Can't log controller state for injected network controllers";
	} else {
	if (log_writer_factory) {
	goog_cc_factory_.AttachWriter(
	log_writer_factory->Create(".cc_state.txt"));
	print_cc_state_ = true;
	}
	cc_factory_ = &goog_cc_factory_;
	}
	}

	LoggingNetworkControllerFactory::~LoggingNetworkControllerFactory() {
	}

	void LoggingNetworkControllerFactory::LogCongestionControllerStats(
	Timestamp at_time) {
	if (print_cc_state_)
	goog_cc_factory_.PrintState(at_time);
	}

	std::unique_ptr<NetworkControllerInterface>
	LoggingNetworkControllerFactory::Create(NetworkControllerConfig config) {
	return cc_factory_->Create(config);
	}

	TimeDelta LoggingNetworkControllerFactory::GetProcessInterval() const {
	return cc_factory_->GetProcessInterval();
	}

	CallClient::CallClient(
	TimeController* time_controller,
	std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
	CallClientConfig config)
	: time_controller_(time_controller),
	clock_(time_controller->GetClock()),
	log_writer_factory_(std::move(log_writer_factory)),
	network_controller_factory_(log_writer_factory_.get(), config.transport),
	header_parser_(RtpHeaderParser::Create()),
	task_queue_(time_controller->GetTaskQueueFactory()->CreateTaskQueue(
	"CallClient",
	TaskQueueFactory::Priority::NORMAL)) {
	SendTask([this, config] {
	event_log_ = CreateEventLog(time_controller_->GetTaskQueueFactory(),
	log_writer_factory_.get());
	fake_audio_setup_ = InitAudio(time_controller_);
	call_.reset(CreateCall(time_controller_, event_log_.get(), config,
	&network_controller_factory_,
	fake_audio_setup_.audio_state));
	transport_ = absl::make_unique<NetworkNodeTransport>(clock_, call_.get());
	});
	}

	CallClient::~CallClient() {
	SendTask([&] {
	call_.reset();
	fake_audio_setup_ = {};
	rtc::Event done;
	event_log_->StopLogging([&done] { done.Set(); });
	done.Wait(rtc::Event::kForever);
	event_log_.reset();
	});
	}

	ColumnPrinter CallClient::StatsPrinter() {
	return ColumnPrinter::Lambda(
	"pacer_delay call_send_bw",
	[this](rtc::SimpleStringBuilder& sb) {
	Call::Stats call_stats = call_->GetStats();
	sb.AppendFormat("%.3lf %.0lf", call_stats.pacer_delay_ms / 1000.0,
	call_stats.send_bandwidth_bps / 8.0);
	},
	64);
	}

	Call::Stats CallClient::GetStats() {
	return call_->GetStats();
	}

	void CallClient::OnPacketReceived(EmulatedIpPacket packet) {
	// Removes added overhead before delivering packet to sender.
	size_t size =
	packet.data.size() - route_overhead_.at(packet.to.ipaddr()).bytes();
	RTC_DCHECK_GE(size, 0);
	packet.data.SetSize(size);

	MediaType media_type = MediaType::ANY;
	if (!RtpHeaderParser::IsRtcp(packet.cdata(), packet.data.size())) {
	auto ssrc = RtpHeaderParser::GetSsrc(packet.cdata(), packet.data.size());
	RTC_CHECK(ssrc.has_value());
	media_type = ssrc_media_types_[*ssrc];
	}
	struct Closure {
	void operator()() {
	call->Receiver()->DeliverPacket(media_type, packet.data,
	packet.arrival_time.us());
	}
	Call* call;
	MediaType media_type;
	EmulatedIpPacket packet;
	};
	task_queue_.PostTask(Closure{call_.get(), media_type, std::move(packet)});
	}

	std::unique_ptr<RtcEventLogOutput> CallClient::GetLogWriter(std::string name) {
	if (!log_writer_factory_ \|\| name.empty())
	return nullptr;
	return log_writer_factory_->Create(name);
	}

	uint32_t CallClient::GetNextVideoSsrc() {
	RTC_CHECK_LT(next_video_ssrc_index_, kNumSsrcs);
	return kVideoSendSsrcs[next_video_ssrc_index_++];
	}

	uint32_t CallClient::GetNextVideoLocalSsrc() {
	RTC_CHECK_LT(next_video_local_ssrc_index_, kNumSsrcs);
	return kVideoRecvLocalSsrcs[next_video_local_ssrc_index_++];
	}

	uint32_t CallClient::GetNextAudioSsrc() {
	RTC_CHECK_LT(next_audio_ssrc_index_, 1);
	next_audio_ssrc_index_++;
	return kAudioSendSsrc;
	}

	uint32_t CallClient::GetNextAudioLocalSsrc() {
	RTC_CHECK_LT(next_audio_local_ssrc_index_, 1);
	next_audio_local_ssrc_index_++;
	return kReceiverLocalAudioSsrc;
	}

	uint32_t CallClient::GetNextRtxSsrc() {
	RTC_CHECK_LT(next_rtx_ssrc_index_, kNumSsrcs);
	return kSendRtxSsrcs[next_rtx_ssrc_index_++];
	}

	std::string CallClient::GetNextPriorityId() {
	RTC_CHECK_LT(next_priority_index_++, 1);
	return kPriorityStreamId;
	}

	void CallClient::AddExtensions(std::vector<RtpExtension> extensions) {
	for (const auto& extension : extensions)
	header_parser_->RegisterRtpHeaderExtension(extension);
	}

	void CallClient::SendTask(std::function<void()> task) {
	time_controller_->InvokeWithControlledYield(
	[&] { task_queue_.SendTask(std::move(task)); });
	}

	CallClientPair::~CallClientPair() = default;

	} // namespace test
	} // namespace webrtc