3c089d751e
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. * DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN * DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN * DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS Related CL: https://codereview.webrtc.org/1335923002/ BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1345433002 Cr-Commit-Position: refs/heads/master@{#9953}
62 lines
1.7 KiB
C++
62 lines
1.7 KiB
C++
/*
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* Copyright 2005 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_BASE_SOCKETSTREAM_H_
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#define WEBRTC_BASE_SOCKETSTREAM_H_
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#include "webrtc/base/asyncsocket.h"
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#include "webrtc/base/common.h"
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#include "webrtc/base/stream.h"
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namespace rtc {
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///////////////////////////////////////////////////////////////////////////////
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class SocketStream : public StreamInterface, public sigslot::has_slots<> {
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public:
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explicit SocketStream(AsyncSocket* socket);
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~SocketStream() override;
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void Attach(AsyncSocket* socket);
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AsyncSocket* Detach();
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AsyncSocket* GetSocket() { return socket_; }
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StreamState GetState() const override;
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StreamResult Read(void* buffer,
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size_t buffer_len,
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size_t* read,
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int* error) override;
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StreamResult Write(const void* data,
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size_t data_len,
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size_t* written,
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int* error) override;
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void Close() override;
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private:
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void OnConnectEvent(AsyncSocket* socket);
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void OnReadEvent(AsyncSocket* socket);
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void OnWriteEvent(AsyncSocket* socket);
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void OnCloseEvent(AsyncSocket* socket, int err);
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AsyncSocket* socket_;
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RTC_DISALLOW_COPY_AND_ASSIGN(SocketStream);
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};
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///////////////////////////////////////////////////////////////////////////////
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} // namespace rtc
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#endif // WEBRTC_BASE_SOCKETSTREAM_H_
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