| /* |
| * 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 "test/run_loop.h" |
| |
| #include "rtc_base/time_utils.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| RunLoop::RunLoop() { |
| worker_thread_.WrapCurrent(); |
| } |
| |
| RunLoop::~RunLoop() { |
| worker_thread_.UnwrapCurrent(); |
| } |
| |
| TaskQueueBase* RunLoop::task_queue() { |
| return &worker_thread_; |
| } |
| |
| void RunLoop::Run() { |
| worker_thread_.ProcessMessages(WorkerThread::kForever); |
| } |
| |
| void RunLoop::Quit() { |
| socket_server_.FailNextWait(); |
| } |
| |
| void RunLoop::Flush() { |
| worker_thread_.PostTask([this]() { socket_server_.FailNextWait(); }); |
| // If a test clock is used, like with GlobalSimulatedTimeController then the |
| // thread will loop forever since time never increases. Since the clock is |
| // simulated, 0ms can be used as the loop delay, which will process all |
| // messages ready for execution. |
| int cms = rtc::GetClockForTesting() ? 0 : 1000; |
| worker_thread_.ProcessMessages(cms); |
| } |
| |
| RunLoop::FakeSocketServer::FakeSocketServer() = default; |
| RunLoop::FakeSocketServer::~FakeSocketServer() = default; |
| |
| void RunLoop::FakeSocketServer::FailNextWait() { |
| fail_next_wait_ = true; |
| } |
| |
| bool RunLoop::FakeSocketServer::Wait(webrtc::TimeDelta max_wait_duration, |
| bool process_io) { |
| if (fail_next_wait_) { |
| fail_next_wait_ = false; |
| return false; |
| } |
| return true; |
| } |
| |
| void RunLoop::FakeSocketServer::WakeUp() {} |
| |
| rtc::Socket* RunLoop::FakeSocketServer::CreateSocket(int family, int type) { |
| return nullptr; |
| } |
| |
| RunLoop::WorkerThread::WorkerThread(rtc::SocketServer* ss) |
| : rtc::Thread(ss), tq_setter_(this) {} |
| |
| } // namespace test |
| } // namespace webrtc |