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/*
* Copyright 2004 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.
*/
#ifndef RTC_BASE_TEST_UTILS_H_
#define RTC_BASE_TEST_UTILS_H_
// Utilities for testing rtc infrastructure in unittests
#include <map>
#include <utility>
#include "rtc_base/socket.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
namespace webrtc {
namespace testing {
///////////////////////////////////////////////////////////////////////////////
// StreamSink - Monitor asynchronously signalled events from Socket.
///////////////////////////////////////////////////////////////////////////////
// Note: Any event that is an error is treated as SSE_ERROR instead of that
// event.
enum StreamSinkEvent {
SSE_OPEN = 1,
SSE_READ = 2,
SSE_WRITE = 4,
SSE_CLOSE = 8,
SSE_ERROR = 16
};
class StreamSink : public sigslot::has_slots<> {
public:
StreamSink();
~StreamSink() override;
void Monitor(rtc::Socket* socket) {
socket->SignalConnectEvent.connect(this, &StreamSink::OnConnectEvent);
socket->SignalReadEvent.connect(this, &StreamSink::OnReadEvent);
socket->SignalWriteEvent.connect(this, &StreamSink::OnWriteEvent);
socket->SignalCloseEvent.connect(this, &StreamSink::OnCloseEvent);
// In case you forgot to unmonitor a previous object with this address
events_.erase(socket);
}
void Unmonitor(rtc::Socket* socket) {
socket->SignalConnectEvent.disconnect(this);
socket->SignalReadEvent.disconnect(this);
socket->SignalWriteEvent.disconnect(this);
socket->SignalCloseEvent.disconnect(this);
events_.erase(socket);
}
bool Check(rtc::Socket* socket, StreamSinkEvent event, bool reset = true) {
return DoCheck(socket, event, reset);
}
private:
typedef std::map<rtc::Socket*, int> EventMap;
void OnConnectEvent(rtc::Socket* socket) { AddEvents(socket, SSE_OPEN); }
void OnReadEvent(rtc::Socket* socket) { AddEvents(socket, SSE_READ); }
void OnWriteEvent(rtc::Socket* socket) { AddEvents(socket, SSE_WRITE); }
void OnCloseEvent(rtc::Socket* socket, int error) {
AddEvents(socket, (0 == error) ? SSE_CLOSE : SSE_ERROR);
}
void AddEvents(rtc::Socket* obj, int events) {
EventMap::iterator it = events_.find(obj);
if (events_.end() == it) {
events_.insert(EventMap::value_type(obj, events));
} else {
it->second |= events;
}
}
bool DoCheck(rtc::Socket* obj, StreamSinkEvent event, bool reset) {
EventMap::iterator it = events_.find(obj);
if ((events_.end() == it) || (0 == (it->second & event))) {
return false;
}
if (reset) {
it->second &= ~event;
}
return true;
}
EventMap events_;
};
} // namespace testing
} // namespace webrtc
#endif // RTC_BASE_TEST_UTILS_H_