Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
Reland the base->rtc_base without adding stub headers (will be
done in follow-up CL). This preserves git blame history of all files.
BUG=webrtc:7634
NOTRY=True
TBR=kwiberg@webrtc.org
Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012
Reviewed-on: https://chromium-review.googlesource.com/554611
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18821}diff --git a/webrtc/rtc_base/socketstream.h b/webrtc/rtc_base/socketstream.h
new file mode 100644
index 0000000..7991c61
--- /dev/null
+++ b/webrtc/rtc_base/socketstream.h
@@ -0,0 +1,61 @@
+/*
+ * 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 WEBRTC_RTC_BASE_SOCKETSTREAM_H_
+#define WEBRTC_RTC_BASE_SOCKETSTREAM_H_
+
+#include "webrtc/base/asyncsocket.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/stream.h"
+
+namespace rtc {
+
+///////////////////////////////////////////////////////////////////////////////
+
+class SocketStream : public StreamInterface, public sigslot::has_slots<> {
+ public:
+ explicit SocketStream(AsyncSocket* socket);
+ ~SocketStream() override;
+
+ void Attach(AsyncSocket* socket);
+ AsyncSocket* Detach();
+
+ AsyncSocket* GetSocket() { return socket_; }
+
+ StreamState GetState() const override;
+
+ StreamResult Read(void* buffer,
+ size_t buffer_len,
+ size_t* read,
+ int* error) override;
+
+ StreamResult Write(const void* data,
+ size_t data_len,
+ size_t* written,
+ int* error) override;
+
+ void Close() override;
+
+ private:
+ void OnConnectEvent(AsyncSocket* socket);
+ void OnReadEvent(AsyncSocket* socket);
+ void OnWriteEvent(AsyncSocket* socket);
+ void OnCloseEvent(AsyncSocket* socket, int err);
+
+ AsyncSocket* socket_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(SocketStream);
+};
+
+///////////////////////////////////////////////////////////////////////////////
+
+} // namespace rtc
+
+#endif // WEBRTC_RTC_BASE_SOCKETSTREAM_H_