chromium/third_party/webrtc/test/run_loop.cc

/*
 *  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() { … }

RunLoop::~RunLoop() { … }

TaskQueueBase* RunLoop::task_queue() { … }

void RunLoop::Run() { … }

void RunLoop::Quit() { … }

void RunLoop::Flush() { … }

RunLoop::FakeSocketServer::FakeSocketServer() = default;
RunLoop::FakeSocketServer::~FakeSocketServer() = default;

void RunLoop::FakeSocketServer::FailNextWait() { … }

bool RunLoop::FakeSocketServer::Wait(webrtc::TimeDelta max_wait_duration,
                                     bool process_io) { … }

void RunLoop::FakeSocketServer::WakeUp() { … }

rtc::Socket* RunLoop::FakeSocketServer::CreateSocket(int family, int type) { … }

RunLoop::WorkerThread::WorkerThread(rtc::SocketServer* ss)
    : … { … }

}  // namespace test
}  // namespace webrtc