chromium/third_party/webrtc/rtc_base/test_client.h

/*
 *  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 RTC_BASE_TEST_CLIENT_H_
#define RTC_BASE_TEST_CLIENT_H_

#include <memory>
#include <vector>

#include "api/units/timestamp.h"
#include "rtc_base/async_udp_socket.h"
#include "rtc_base/buffer.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/network/received_packet.h"
#include "rtc_base/synchronization/mutex.h"

namespace rtc {

// A simple client that can send TCP or UDP data and check that it receives
// what it expects to receive. Useful for testing server functionality.
class TestClient : public sigslot::has_slots<> { … };

}  // namespace rtc

#endif  // RTC_BASE_TEST_CLIENT_H_