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/*
* 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 "rtc_base/async_tcp_socket.h"
#include <memory>
#include <string>
#include "rtc_base/gunit.h"
#include "rtc_base/virtual_socket_server.h"
namespace rtc {
class AsyncTCPSocketTest : public ::testing::Test, public sigslot::has_slots<> {
public:
AsyncTCPSocketTest()
: vss_(new rtc::VirtualSocketServer()),
socket_(vss_->CreateSocket(SOCK_STREAM)),
tcp_socket_(new AsyncTCPSocket(socket_, true)),
ready_to_send_(false) {
tcp_socket_->SignalReadyToSend.connect(this,
&AsyncTCPSocketTest::OnReadyToSend);
}
void OnReadyToSend(rtc::AsyncPacketSocket* socket) { ready_to_send_ = true; }
protected:
std::unique_ptr<VirtualSocketServer> vss_;
Socket* socket_;
std::unique_ptr<AsyncTCPSocket> tcp_socket_;
bool ready_to_send_;
};
TEST_F(AsyncTCPSocketTest, OnWriteEvent) {
EXPECT_FALSE(ready_to_send_);
socket_->SignalWriteEvent(socket_);
EXPECT_TRUE(ready_to_send_);
}
} // namespace rtc
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