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-Original-Commit-Position: refs/heads/master@{#18821}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: c03627683fc2181981bd5207c2fdba6b5bd7a67c
diff --git a/rtc_base/asyncudpsocket.h b/rtc_base/asyncudpsocket.h
new file mode 100644
index 0000000..bff70f1
--- /dev/null
+++ b/rtc_base/asyncudpsocket.h
@@ -0,0 +1,67 @@
+/*
+ * 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 WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_
+#define WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_
+
+#include <memory>
+
+#include "webrtc/base/asyncpacketsocket.h"
+#include "webrtc/base/socketfactory.h"
+
+namespace rtc {
+
+// Provides the ability to receive packets asynchronously. Sends are not
+// buffered since it is acceptable to drop packets under high load.
+class AsyncUDPSocket : public AsyncPacketSocket {
+ public:
+ // Binds |socket| and creates AsyncUDPSocket for it. Takes ownership
+ // of |socket|. Returns null if bind() fails (|socket| is destroyed
+ // in that case).
+ static AsyncUDPSocket* Create(AsyncSocket* socket,
+ const SocketAddress& bind_address);
+ // Creates a new socket for sending asynchronous UDP packets using an
+ // asynchronous socket from the given factory.
+ static AsyncUDPSocket* Create(SocketFactory* factory,
+ const SocketAddress& bind_address);
+ explicit AsyncUDPSocket(AsyncSocket* socket);
+ ~AsyncUDPSocket() override;
+
+ SocketAddress GetLocalAddress() const override;
+ SocketAddress GetRemoteAddress() const override;
+ int Send(const void* pv,
+ size_t cb,
+ const rtc::PacketOptions& options) override;
+ int SendTo(const void* pv,
+ size_t cb,
+ const SocketAddress& addr,
+ const rtc::PacketOptions& options) override;
+ int Close() override;
+
+ State GetState() const override;
+ int GetOption(Socket::Option opt, int* value) override;
+ int SetOption(Socket::Option opt, int value) override;
+ int GetError() const override;
+ void SetError(int error) override;
+
+ private:
+ // Called when the underlying socket is ready to be read from.
+ void OnReadEvent(AsyncSocket* socket);
+ // Called when the underlying socket is ready to send.
+ void OnWriteEvent(AsyncSocket* socket);
+
+ std::unique_ptr<AsyncSocket> socket_;
+ char* buf_;
+ size_t size_;
+};
+
+} // namespace rtc
+
+#endif // WEBRTC_RTC_BASE_ASYNCUDPSOCKET_H_