rtc_tools/data_channel_benchmark/data_channel_benchmark.cc - src - Git at Google

 /*
  *  Copyright (c) 2021 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.
  *
  *  Data Channel Benchmarking tool.
  *
  *  Create a server using: ./data_channel_benchmark --server --port 12345
  *  Start the flow of data from the server to a client using:
  *  ./data_channel_benchmark --port 12345 --transfer_size 100 --packet_size 8196
  *  The throughput is reported on the server console.
  *
  *  The negotiation does not require a 3rd party server and is done over a gRPC
  *  transport. No TURN server is configured, so both peers need to be reachable
  *  using STUN only.
  */
 #include <inttypes.h>

 #include <charconv>

 #include "absl/cleanup/cleanup.h"
 #include "absl/flags/flag.h"
 #include "absl/flags/parse.h"
 #include "rtc_base/event.h"
 #include "rtc_base/ssl_adapter.h"
 #include "rtc_base/thread.h"
 #include "rtc_tools/data_channel_benchmark/grpc_signaling.h"
 #include "rtc_tools/data_channel_benchmark/peer_connection_client.h"
 #include "system_wrappers/include/field_trial.h"

 ABSL_FLAG(int, verbose, 0, "verbosity level (0-5)");
 ABSL_FLAG(bool, server, false, "Server mode");
 ABSL_FLAG(bool, oneshot, true, "Terminate after serving a client");
 ABSL_FLAG(std::string, address, "localhost", "Connect to server address");
 ABSL_FLAG(uint16_t, port, 0, "Connect to port (0 for random)");
 ABSL_FLAG(uint64_t, transfer_size, 2, "Transfer size (MiB)");
 ABSL_FLAG(uint64_t, packet_size, 256 * 1024, "Packet size");
 ABSL_FLAG(std::string,
           force_fieldtrials,
           "",
           "Field trials control experimental feature code which can be forced. "
           "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
           " will assign the group Enable to field trial WebRTC-FooFeature.");

 struct SetupMessage {
   size_t packet_size;
   size_t transfer_size;

   std::string ToString() {
     char buffer[64];
     rtc::SimpleStringBuilder sb(buffer);
     sb << packet_size << "," << transfer_size;

     return sb.str();
   }

   static SetupMessage FromString(absl::string_view sv) {
     SetupMessage result;
     auto parameters = rtc::split(sv, ',');
     std::from_chars(parameters[0].data(),
                     parameters[0].data() + parameters[0].size(),
                     result.packet_size, 10);
     std::from_chars(parameters[1].data(),
                     parameters[1].data() + parameters[1].size(),
                     result.transfer_size, 10);
     return result;
   }
 };

 class DataChannelObserverImpl : public webrtc::DataChannelObserver {
  public:
   explicit DataChannelObserverImpl(webrtc::DataChannelInterface* dc)
       : dc_(dc), bytes_received_(0) {}
   void OnStateChange() override {
     RTC_LOG(LS_INFO) << "State changed to " << dc_->state();
     switch (dc_->state()) {
       case webrtc::DataChannelInterface::DataState::kOpen:
         open_event_.Set();
         break;
       case webrtc::DataChannelInterface::DataState::kClosed:
         closed_event_.Set();
         break;
       default:
         break;
     }
   }
   void OnMessage(const webrtc::DataBuffer& buffer) override {
     bytes_received_ += buffer.data.size();
     if (bytes_received_threshold_ &&
         bytes_received_ >= bytes_received_threshold_) {
       bytes_received_event_.Set();
     }

     if (setup_message_.empty() && !buffer.binary) {
       setup_message_.assign(buffer.data.cdata<char>(), buffer.data.size());
       setup_message_event_.Set();
     }
   }
   void OnBufferedAmountChange(uint64_t sent_data_size) override {
     if (dc_->buffered_amount() <
         webrtc::DataChannelInterface::MaxSendQueueSize() / 2)
       low_buffered_threshold_event_.Set();
     else
       low_buffered_threshold_event_.Reset();
   }

   bool WaitForOpenState() {
     return dc_->state() == webrtc::DataChannelInterface::DataState::kOpen ||
            open_event_.Wait(rtc::Event::kForever);
   }
   bool WaitForClosedState() {
     return dc_->state() == webrtc::DataChannelInterface::DataState::kClosed ||
            closed_event_.Wait(rtc::Event::kForever);
   }

   // Set how many received bytes are required until
   // WaitForBytesReceivedThreshold return true.
   void SetBytesReceivedThreshold(uint64_t bytes_received_threshold) {
     bytes_received_threshold_ = bytes_received_threshold;
     if (bytes_received_ >= bytes_received_threshold_)
       bytes_received_event_.Set();
   }
   // Wait until the received byte count reaches the desired value.
   bool WaitForBytesReceivedThreshold() {
     return (bytes_received_threshold_ &&
             bytes_received_ >= bytes_received_threshold_) ||
            bytes_received_event_.Wait(rtc::Event::kForever);
   }

   bool WaitForLowbufferedThreshold() {
     return low_buffered_threshold_event_.Wait(rtc::Event::kForever);
   }
   std::string SetupMessage() { return setup_message_; }
   bool WaitForSetupMessage() {
     return setup_message_event_.Wait(rtc::Event::kForever);
   }

  private:
   webrtc::DataChannelInterface* dc_;
   rtc::Event open_event_;
   rtc::Event closed_event_;
   rtc::Event bytes_received_event_;
   absl::optional<uint64_t> bytes_received_threshold_;
   uint64_t bytes_received_;
   rtc::Event low_buffered_threshold_event_;
   std::string setup_message_;
   rtc::Event setup_message_event_;
 };

 int RunServer() {
   bool oneshot = absl::GetFlag(FLAGS_oneshot);
   uint16_t port = absl::GetFlag(FLAGS_port);

   auto signaling_thread = rtc::Thread::Create();
   signaling_thread->Start();
   {
     auto factory = webrtc::PeerConnectionClient::CreateDefaultFactory(
         signaling_thread.get());

     auto grpc_server = webrtc::GrpcSignalingServerInterface::Create(
         [factory = rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>(
              factory)](webrtc::SignalingInterface* signaling) {
           webrtc::PeerConnectionClient client(factory.get(), signaling);
           client.StartPeerConnection();
           auto peer_connection = client.peerConnection();

           // Set up the data channel
           auto dc_or_error =
               peer_connection->CreateDataChannelOrError("benchmark", nullptr);
           auto data_channel = dc_or_error.MoveValue();
           auto data_channel_observer =
               std::make_unique<DataChannelObserverImpl>(data_channel.get());
           data_channel->RegisterObserver(data_channel_observer.get());
           absl::Cleanup unregister_observer(
               [data_channel] { data_channel->UnregisterObserver(); });

           // Wait for a first message from the remote peer.
           // It configures how much data should be sent and how big the packets
           // should be.
           // First message is "packet_size,transfer_size".
           data_channel_observer->WaitForSetupMessage();
           auto parameters =
               SetupMessage::FromString(data_channel_observer->SetupMessage());

           // Wait for the sender and receiver peers to stabilize (send all ACKs)
           // This makes it easier to isolate the sending part when profiling.
           absl::SleepFor(absl::Seconds(1));

           std::string data(parameters.packet_size, '0');
           size_t remaining_data = parameters.transfer_size;

           auto begin_time = webrtc::Clock::GetRealTimeClock()->CurrentTime();

           while (remaining_data) {
             if (remaining_data < data.size())
               data.resize(remaining_data);

             rtc::CopyOnWriteBuffer buffer(data);
             webrtc::DataBuffer data_buffer(buffer, true);
             if (!data_channel->Send(data_buffer)) {
               // If the send() call failed, the buffers are full.
               // We wait until there's more room.
               data_channel_observer->WaitForLowbufferedThreshold();
               continue;
             }
             remaining_data -= buffer.size();
             fprintf(stderr, "Progress: %zu / %zu (%zu%%)\n",
                     (parameters.transfer_size - remaining_data),
                     parameters.transfer_size,
                     (100 - remaining_data * 100 / parameters.transfer_size));
           }

           // Receiver signals the data channel close event when it has received
           // all the data it requested.
           data_channel_observer->WaitForClosedState();

           auto end_time = webrtc::Clock::GetRealTimeClock()->CurrentTime();
           auto duration_ms = (end_time - begin_time).ms<size_t>();
           double throughput = (parameters.transfer_size / 1024. / 1024.) /
                               (duration_ms / 1000.);
           printf("Elapsed time: %zums %gMiB/s\n", duration_ms, throughput);
         },
         port, oneshot);
     grpc_server->Start();

     printf("Server listening on port %d\n", grpc_server->SelectedPort());
     grpc_server->Wait();
   }

   signaling_thread->Quit();
   return 0;
 }

 int RunClient() {
   uint16_t port = absl::GetFlag(FLAGS_port);
   std::string server_address = absl::GetFlag(FLAGS_address);
   size_t transfer_size = absl::GetFlag(FLAGS_transfer_size) * 1024 * 1024;
   size_t packet_size = absl::GetFlag(FLAGS_packet_size);

   auto signaling_thread = rtc::Thread::Create();
   signaling_thread->Start();
   {
     auto factory = webrtc::PeerConnectionClient::CreateDefaultFactory(
         signaling_thread.get());
     auto grpc_client = webrtc::GrpcSignalingClientInterface::Create(
         server_address + ":" + std::to_string(port));
     webrtc::PeerConnectionClient client(factory.get(),
                                         grpc_client->signaling_client());

     // Set up the callback to receive the data channel from the sender.
     rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel;
     rtc::Event got_data_channel;
     client.SetOnDataChannel(
         [&data_channel, &got_data_channel](
             rtc::scoped_refptr<webrtc::DataChannelInterface> channel) {
           data_channel = channel;
           got_data_channel.Set();
         });

     // Connect to the server.
     if (!grpc_client->Start()) {
       fprintf(stderr, "Failed to connect to server\n");
       return 1;
     }

     // Wait for the data channel to be received
     got_data_channel.Wait(rtc::Event::kForever);

     // DataChannel needs an observer to start draining the read queue
     DataChannelObserverImpl observer(data_channel.get());
     observer.SetBytesReceivedThreshold(transfer_size);
     data_channel->RegisterObserver(&observer);
     absl::Cleanup unregister_observer(
         [data_channel] { data_channel->UnregisterObserver(); });

     // Send a configuration string to the server to tell it to send
     // 'packet_size' bytes packets and send a total of 'transfer_size' MB.
     observer.WaitForOpenState();
     SetupMessage setup_message = {
         .packet_size = packet_size,
         .transfer_size = transfer_size,
     };
     if (!data_channel->Send(webrtc::DataBuffer(setup_message.ToString()))) {
       fprintf(stderr, "Failed to send parameter string\n");
       return 1;
     }

     // Wait until we have received all the data
     observer.WaitForBytesReceivedThreshold();

     // Close the data channel, signaling to the server we have received
     // all the requested data.
     data_channel->Close();
   }

   signaling_thread->Quit();

   return 0;
 }

 int main(int argc, char** argv) {
   rtc::InitializeSSL();
   absl::ParseCommandLine(argc, argv);

   // Make sure that higher severity number means more logs by reversing the
   // rtc::LoggingSeverity values.
   auto logging_severity =
       std::max(0, rtc::LS_NONE - absl::GetFlag(FLAGS_verbose));
   rtc::LogMessage::LogToDebug(
       static_cast<rtc::LoggingSeverity>(logging_severity));

   bool is_server = absl::GetFlag(FLAGS_server);
   std::string field_trials = absl::GetFlag(FLAGS_force_fieldtrials);

   webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str());

   return is_server ? RunServer() : RunClient();
 }
	/*
	* Copyright (c) 2021 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.
	*
	* Data Channel Benchmarking tool.
	*
	* Create a server using: ./data_channel_benchmark --server --port 12345
	* Start the flow of data from the server to a client using:
	* ./data_channel_benchmark --port 12345 --transfer_size 100 --packet_size 8196
	* The throughput is reported on the server console.
	*
	* The negotiation does not require a 3rd party server and is done over a gRPC
	* transport. No TURN server is configured, so both peers need to be reachable
	* using STUN only.
	*/
	#include <inttypes.h>

	#include <charconv>

	#include "absl/cleanup/cleanup.h"
	#include "absl/flags/flag.h"
	#include "absl/flags/parse.h"
	#include "rtc_base/event.h"
	#include "rtc_base/ssl_adapter.h"
	#include "rtc_base/thread.h"
	#include "rtc_tools/data_channel_benchmark/grpc_signaling.h"
	#include "rtc_tools/data_channel_benchmark/peer_connection_client.h"
	#include "system_wrappers/include/field_trial.h"

	ABSL_FLAG(int, verbose, 0, "verbosity level (0-5)");
	ABSL_FLAG(bool, server, false, "Server mode");
	ABSL_FLAG(bool, oneshot, true, "Terminate after serving a client");
	ABSL_FLAG(std::string, address, "localhost", "Connect to server address");
	ABSL_FLAG(uint16_t, port, 0, "Connect to port (0 for random)");
	ABSL_FLAG(uint64_t, transfer_size, 2, "Transfer size (MiB)");
	ABSL_FLAG(uint64_t, packet_size, 256 * 1024, "Packet size");
	ABSL_FLAG(std::string,
	force_fieldtrials,
	"",
	"Field trials control experimental feature code which can be forced. "
	"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
	" will assign the group Enable to field trial WebRTC-FooFeature.");

	struct SetupMessage {
	size_t packet_size;
	size_t transfer_size;

	std::string ToString() {
	char buffer[64];
	rtc::SimpleStringBuilder sb(buffer);
	sb << packet_size << "," << transfer_size;

	return sb.str();
	}

	static SetupMessage FromString(absl::string_view sv) {
	SetupMessage result;
	auto parameters = rtc::split(sv, ',');
	std::from_chars(parameters[0].data(),
	parameters[0].data() + parameters[0].size(),
	result.packet_size, 10);
	std::from_chars(parameters[1].data(),
	parameters[1].data() + parameters[1].size(),
	result.transfer_size, 10);
	return result;
	}
	};

	class DataChannelObserverImpl : public webrtc::DataChannelObserver {
	public:
	explicit DataChannelObserverImpl(webrtc::DataChannelInterface* dc)
	: dc_(dc), bytes_received_(0) {}
	void OnStateChange() override {
	RTC_LOG(LS_INFO) << "State changed to " << dc_->state();
	switch (dc_->state()) {
	case webrtc::DataChannelInterface::DataState::kOpen:
	open_event_.Set();
	break;
	case webrtc::DataChannelInterface::DataState::kClosed:
	closed_event_.Set();
	break;
	default:
	break;
	}
	}
	void OnMessage(const webrtc::DataBuffer& buffer) override {
	bytes_received_ += buffer.data.size();
	if (bytes_received_threshold_ &&
	bytes_received_ >= bytes_received_threshold_) {
	bytes_received_event_.Set();
	}

	if (setup_message_.empty() && !buffer.binary) {
	setup_message_.assign(buffer.data.cdata<char>(), buffer.data.size());
	setup_message_event_.Set();
	}
	}
	void OnBufferedAmountChange(uint64_t sent_data_size) override {
	if (dc_->buffered_amount() <
	webrtc::DataChannelInterface::MaxSendQueueSize() / 2)
	low_buffered_threshold_event_.Set();
	else
	low_buffered_threshold_event_.Reset();
	}

	bool WaitForOpenState() {
	return dc_->state() == webrtc::DataChannelInterface::DataState::kOpen \|\|
	open_event_.Wait(rtc::Event::kForever);
	}
	bool WaitForClosedState() {
	return dc_->state() == webrtc::DataChannelInterface::DataState::kClosed \|\|
	closed_event_.Wait(rtc::Event::kForever);
	}

	// Set how many received bytes are required until
	// WaitForBytesReceivedThreshold return true.
	void SetBytesReceivedThreshold(uint64_t bytes_received_threshold) {
	bytes_received_threshold_ = bytes_received_threshold;
	if (bytes_received_ >= bytes_received_threshold_)
	bytes_received_event_.Set();
	}
	// Wait until the received byte count reaches the desired value.
	bool WaitForBytesReceivedThreshold() {
	return (bytes_received_threshold_ &&
	bytes_received_ >= bytes_received_threshold_) \|\|
	bytes_received_event_.Wait(rtc::Event::kForever);
	}

	bool WaitForLowbufferedThreshold() {
	return low_buffered_threshold_event_.Wait(rtc::Event::kForever);
	}
	std::string SetupMessage() { return setup_message_; }
	bool WaitForSetupMessage() {
	return setup_message_event_.Wait(rtc::Event::kForever);
	}

	private:
	webrtc::DataChannelInterface* dc_;
	rtc::Event open_event_;
	rtc::Event closed_event_;
	rtc::Event bytes_received_event_;
	absl::optional<uint64_t> bytes_received_threshold_;
	uint64_t bytes_received_;
	rtc::Event low_buffered_threshold_event_;
	std::string setup_message_;
	rtc::Event setup_message_event_;
	};

	int RunServer() {
	bool oneshot = absl::GetFlag(FLAGS_oneshot);
	uint16_t port = absl::GetFlag(FLAGS_port);

	auto signaling_thread = rtc::Thread::Create();
	signaling_thread->Start();
	{
	auto factory = webrtc::PeerConnectionClient::CreateDefaultFactory(
	signaling_thread.get());

	auto grpc_server = webrtc::GrpcSignalingServerInterface::Create(
	[factory = rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>(
	factory)](webrtc::SignalingInterface* signaling) {
	webrtc::PeerConnectionClient client(factory.get(), signaling);
	client.StartPeerConnection();
	auto peer_connection = client.peerConnection();

	// Set up the data channel
	auto dc_or_error =
	peer_connection->CreateDataChannelOrError("benchmark", nullptr);
	auto data_channel = dc_or_error.MoveValue();
	auto data_channel_observer =
	std::make_unique<DataChannelObserverImpl>(data_channel.get());
	data_channel->RegisterObserver(data_channel_observer.get());
	absl::Cleanup unregister_observer(
	[data_channel] { data_channel->UnregisterObserver(); });

	// Wait for a first message from the remote peer.
	// It configures how much data should be sent and how big the packets
	// should be.
	// First message is "packet_size,transfer_size".
	data_channel_observer->WaitForSetupMessage();
	auto parameters =
	SetupMessage::FromString(data_channel_observer->SetupMessage());

	// Wait for the sender and receiver peers to stabilize (send all ACKs)
	// This makes it easier to isolate the sending part when profiling.
	absl::SleepFor(absl::Seconds(1));

	std::string data(parameters.packet_size, '0');
	size_t remaining_data = parameters.transfer_size;

	auto begin_time = webrtc::Clock::GetRealTimeClock()->CurrentTime();

	while (remaining_data) {
	if (remaining_data < data.size())
	data.resize(remaining_data);

	rtc::CopyOnWriteBuffer buffer(data);
	webrtc::DataBuffer data_buffer(buffer, true);
	if (!data_channel->Send(data_buffer)) {
	// If the send() call failed, the buffers are full.
	// We wait until there's more room.
	data_channel_observer->WaitForLowbufferedThreshold();
	continue;
	}
	remaining_data -= buffer.size();
	fprintf(stderr, "Progress: %zu / %zu (%zu%%)\n",
	(parameters.transfer_size - remaining_data),
	parameters.transfer_size,
	(100 - remaining_data * 100 / parameters.transfer_size));
	}

	// Receiver signals the data channel close event when it has received
	// all the data it requested.
	data_channel_observer->WaitForClosedState();

	auto end_time = webrtc::Clock::GetRealTimeClock()->CurrentTime();
	auto duration_ms = (end_time - begin_time).ms<size_t>();
	double throughput = (parameters.transfer_size / 1024. / 1024.) /
	(duration_ms / 1000.);
	printf("Elapsed time: %zums %gMiB/s\n", duration_ms, throughput);
	},
	port, oneshot);
	grpc_server->Start();

	printf("Server listening on port %d\n", grpc_server->SelectedPort());
	grpc_server->Wait();
	}

	signaling_thread->Quit();
	return 0;
	}

	int RunClient() {
	uint16_t port = absl::GetFlag(FLAGS_port);
	std::string server_address = absl::GetFlag(FLAGS_address);
	size_t transfer_size = absl::GetFlag(FLAGS_transfer_size) * 1024 * 1024;
	size_t packet_size = absl::GetFlag(FLAGS_packet_size);

	auto signaling_thread = rtc::Thread::Create();
	signaling_thread->Start();
	{
	auto factory = webrtc::PeerConnectionClient::CreateDefaultFactory(
	signaling_thread.get());
	auto grpc_client = webrtc::GrpcSignalingClientInterface::Create(
	server_address + ":" + std::to_string(port));
	webrtc::PeerConnectionClient client(factory.get(),
	grpc_client->signaling_client());

	// Set up the callback to receive the data channel from the sender.
	rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel;
	rtc::Event got_data_channel;
	client.SetOnDataChannel(
	[&data_channel, &got_data_channel](
	rtc::scoped_refptr<webrtc::DataChannelInterface> channel) {
	data_channel = channel;
	got_data_channel.Set();
	});

	// Connect to the server.
	if (!grpc_client->Start()) {
	fprintf(stderr, "Failed to connect to server\n");
	return 1;
	}

	// Wait for the data channel to be received
	got_data_channel.Wait(rtc::Event::kForever);

	// DataChannel needs an observer to start draining the read queue
	DataChannelObserverImpl observer(data_channel.get());
	observer.SetBytesReceivedThreshold(transfer_size);
	data_channel->RegisterObserver(&observer);
	absl::Cleanup unregister_observer(
	[data_channel] { data_channel->UnregisterObserver(); });

	// Send a configuration string to the server to tell it to send
	// 'packet_size' bytes packets and send a total of 'transfer_size' MB.
	observer.WaitForOpenState();
	SetupMessage setup_message = {
	.packet_size = packet_size,
	.transfer_size = transfer_size,
	};
	if (!data_channel->Send(webrtc::DataBuffer(setup_message.ToString()))) {
	fprintf(stderr, "Failed to send parameter string\n");
	return 1;
	}

	// Wait until we have received all the data
	observer.WaitForBytesReceivedThreshold();

	// Close the data channel, signaling to the server we have received
	// all the requested data.
	data_channel->Close();
	}

	signaling_thread->Quit();

	return 0;
	}

	int main(int argc, char** argv) {
	rtc::InitializeSSL();
	absl::ParseCommandLine(argc, argv);

	// Make sure that higher severity number means more logs by reversing the
	// rtc::LoggingSeverity values.
	auto logging_severity =
	std::max(0, rtc::LS_NONE - absl::GetFlag(FLAGS_verbose));
	rtc::LogMessage::LogToDebug(
	static_cast<rtc::LoggingSeverity>(logging_severity));

	bool is_server = absl::GetFlag(FLAGS_server);
	std::string field_trials = absl::GetFlag(FLAGS_force_fieldtrials);

	webrtc::field_trial::InitFieldTrialsFromString(field_trials.c_str());

	return is_server ? RunServer() : RunClient();
	}