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/asyncresolverinterface.h b/webrtc/rtc_base/asyncresolverinterface.h
new file mode 100644
index 0000000..1784019
--- /dev/null
+++ b/webrtc/rtc_base/asyncresolverinterface.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright 2013 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_ASYNCRESOLVERINTERFACE_H_
+#define WEBRTC_RTC_BASE_ASYNCRESOLVERINTERFACE_H_
+
+#include "webrtc/base/sigslot.h"
+#include "webrtc/base/socketaddress.h"
+
+namespace rtc {
+
+// This interface defines the methods to resolve the address asynchronously.
+class AsyncResolverInterface {
+ public:
+ AsyncResolverInterface();
+ virtual ~AsyncResolverInterface();
+
+ // Start address resolve process.
+ virtual void Start(const SocketAddress& addr) = 0;
+ // Returns top most resolved address of |family|
+ virtual bool GetResolvedAddress(int family, SocketAddress* addr) const = 0;
+ // Returns error from resolver.
+ virtual int GetError() const = 0;
+ // Delete the resolver.
+ virtual void Destroy(bool wait) = 0;
+ // Returns top most resolved IPv4 address if address is resolved successfully.
+ // Otherwise returns address set in SetAddress.
+ SocketAddress address() const {
+ SocketAddress addr;
+ GetResolvedAddress(AF_INET, &addr);
+ return addr;
+ }
+
+ // This signal is fired when address resolve process is completed.
+ sigslot::signal1<AsyncResolverInterface*> SignalDone;
+};
+
+} // namespace rtc
+
+#endif