Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/base/asyncudpsocket_unittest.cc b/base/asyncudpsocket_unittest.cc
new file mode 100644
index 0000000..bd65940
--- /dev/null
+++ b/base/asyncudpsocket_unittest.cc
@@ -0,0 +1,53 @@
+/*
+ * 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.
+ */
+
+#include <string>
+
+#include "webrtc/base/asyncudpsocket.h"
+#include "webrtc/base/gunit.h"
+#include "webrtc/base/physicalsocketserver.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/virtualsocketserver.h"
+
+namespace rtc {
+
+class AsyncUdpSocketTest
+ : public testing::Test,
+ public sigslot::has_slots<> {
+ public:
+ AsyncUdpSocketTest()
+ : pss_(new rtc::PhysicalSocketServer),
+ vss_(new rtc::VirtualSocketServer(pss_.get())),
+ socket_(vss_->CreateAsyncSocket(SOCK_DGRAM)),
+ udp_socket_(new AsyncUDPSocket(socket_)),
+ ready_to_send_(false) {
+ udp_socket_->SignalReadyToSend.connect(this,
+ &AsyncUdpSocketTest::OnReadyToSend);
+ }
+
+ void OnReadyToSend(rtc::AsyncPacketSocket* socket) {
+ ready_to_send_ = true;
+ }
+
+ protected:
+ scoped_ptr<PhysicalSocketServer> pss_;
+ scoped_ptr<VirtualSocketServer> vss_;
+ AsyncSocket* socket_;
+ scoped_ptr<AsyncUDPSocket> udp_socket_;
+ bool ready_to_send_;
+};
+
+TEST_F(AsyncUdpSocketTest, OnWriteEvent) {
+ EXPECT_FALSE(ready_to_send_);
+ socket_->SignalWriteEvent(socket_);
+ EXPECT_TRUE(ready_to_send_);
+}
+
+} // namespace rtc