webrtc/test/run_loop.cc
Niels Möller d0b8879770 Delete AsyncSocket class, merge into Socket class
Bug: webrtc:13065
Change-Id: I13afee2386ea9c4de0e4fa95133f0c4d3ec826e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227031
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34787}
2021-08-17 15:39:25 +00:00

68 lines
1.6 KiB
C++

/*
* 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/task_utils/to_queued_task.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(
ToQueuedTask([this]() { socket_server_.FailNextWait(); }));
worker_thread_.ProcessMessages(1000);
}
RunLoop::FakeSocketServer::FakeSocketServer() = default;
RunLoop::FakeSocketServer::~FakeSocketServer() = default;
void RunLoop::FakeSocketServer::FailNextWait() {
fail_next_wait_ = true;
}
bool RunLoop::FakeSocketServer::Wait(int cms, 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