Move webrtc/{base => rtc_base}
This refactoring takes a careful approach to avoid rushing the change:
* stub headers are left in all the old locations of webrtc/base
* existing GN targets are kept and now just forward to the moved ones
using public_deps.
The only exception to the above is the base_java target and its .java files,
which were moved to webrtc/rtc_base right away since it's not possible
to use public_deps for android_library.
To avoid breaking builds, a temporary Dummy.java file was added to
the new intermediate target in webrtc/rtc_base:base_java as well to avoid
hitting a GN assert in the android_library template.
The above approach should make the transition smooth without breaking
downstream.
A helper script was created (https://codereview.webrtc.org/2879203002/)
and was run like this:
stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634
stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634
Fixed invalid header guards in the following files:
webrtc/base/base64.h
webrtc/base/cryptstring.h
webrtc/base/event.h
webrtc/base/flags.h
webrtc/base/httpbase.h
webrtc/base/httpcommon-inl.h
webrtc/base/httpcommon.h
webrtc/base/httpserver.h
webrtc/base/logsinks.h
webrtc/base/macutils.h
webrtc/base/nattypes.h
webrtc/base/openssladapter.h
webrtc/base/opensslstreamadapter.h
webrtc/base/pathutils.h
webrtc/base/physicalsocketserver.h
webrtc/base/proxyinfo.h
webrtc/base/sigslot.h
webrtc/base/sigslotrepeater.h
webrtc/base/socket.h
webrtc/base/socketaddresspair.h
webrtc/base/socketfactory.h
webrtc/base/stringutils.h
webrtc/base/testbase64.h
webrtc/base/testutils.h
webrtc/base/transformadapter.h
webrtc/base/win32filesystem.h
Added new header guards to:
sslroots.h
testbase64.h
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2877023002 .
Cr-Commit-Position: refs/heads/master@{#18816}
diff --git a/webrtc/rtc_base/asyncudpsocket.h b/webrtc/rtc_base/asyncudpsocket.h
new file mode 100644
index 0000000..bff70f1
--- /dev/null
+++ b/webrtc/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_