Repo created

TMessagesProj/jni/voip/webrtc/rtc_base/test_utils.h

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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_