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48 lines
1.4 KiB
48 lines
1.4 KiB
7 months ago
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/*
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* Copyright 2004 The WebRTC Project Authors. All rights reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "rtc_base/async_tcp_socket.h"
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#include <memory>
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#include <string>
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#include "rtc_base/gunit.h"
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#include "rtc_base/virtual_socket_server.h"
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namespace rtc {
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class AsyncTCPSocketTest : public ::testing::Test, public sigslot::has_slots<> {
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public:
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AsyncTCPSocketTest()
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: vss_(new rtc::VirtualSocketServer()),
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socket_(vss_->CreateAsyncSocket(SOCK_STREAM)),
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tcp_socket_(new AsyncTCPSocket(socket_, true)),
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ready_to_send_(false) {
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tcp_socket_->SignalReadyToSend.connect(this,
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&AsyncTCPSocketTest::OnReadyToSend);
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}
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void OnReadyToSend(rtc::AsyncPacketSocket* socket) { ready_to_send_ = true; }
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protected:
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std::unique_ptr<VirtualSocketServer> vss_;
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AsyncSocket* socket_;
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std::unique_ptr<AsyncTCPSocket> tcp_socket_;
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bool ready_to_send_;
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};
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TEST_F(AsyncTCPSocketTest, OnWriteEvent) {
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EXPECT_FALSE(ready_to_send_);
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socket_->SignalWriteEvent(socket_);
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EXPECT_TRUE(ready_to_send_);
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}
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} // namespace rtc
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