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Diffstat (limited to 'third_party/libwebrtc/rtc_tools/data_channel_benchmark/data_channel_benchmark.cc')
| -rw-r--r-- | third_party/libwebrtc/rtc_tools/data_channel_benchmark/data_channel_benchmark.cc | 322 |
1 files changed, 322 insertions, 0 deletions
diff --git a/third_party/libwebrtc/rtc_tools/data_channel_benchmark/data_channel_benchmark.cc b/third_party/libwebrtc/rtc_tools/data_channel_benchmark/data_channel_benchmark.cc new file mode 100644 index 0000000000..33776f37aa --- /dev/null +++ b/third_party/libwebrtc/rtc_tools/data_channel_benchmark/data_channel_benchmark.cc @@ -0,0 +1,322 @@ +/* + * 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(); +} |