Repo created

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

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+/*
+ *  Copyright 2005 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_SOCKET_STREAM_H_
+#define RTC_BASE_SOCKET_STREAM_H_
+
+#include <stddef.h>
+
+#include "rtc_base/socket.h"
+#include "rtc_base/stream.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+
+namespace rtc {
+
+///////////////////////////////////////////////////////////////////////////////
+
+class SocketStream : public StreamInterface, public sigslot::has_slots<> {
+ public:
+  explicit SocketStream(Socket* socket);
+  ~SocketStream() override;
+
+  SocketStream(const SocketStream&) = delete;
+  SocketStream& operator=(const SocketStream&) = delete;
+
+  void Attach(Socket* socket);
+  Socket* Detach();
+
+  Socket* GetSocket() { return socket_; }
+
+  StreamState GetState() const override;
+
+  StreamResult Read(rtc::ArrayView<uint8_t> buffer,
+                    size_t& read,
+                    int& error) override;
+
+  StreamResult Write(rtc::ArrayView<const uint8_t> data,
+                     size_t& written,
+                     int& error) override;
+
+  void Close() override;
+
+ private:
+  void OnConnectEvent(Socket* socket);
+  void OnReadEvent(Socket* socket);
+  void OnWriteEvent(Socket* socket);
+  void OnCloseEvent(Socket* socket, int err);
+
+  Socket* socket_;
+};
+
+///////////////////////////////////////////////////////////////////////////////
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_SOCKET_STREAM_H_