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d07900c848
This is done instead of directly using AsyncPacketSocket::SignalReceived. Bug: webrtc:15368, webrtc:11943 Change-Id: I5671e66b270355188b1252138eced8e6c78ba7ad Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327521 Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41180}
114 lines
4 KiB
C++
114 lines
4 KiB
C++
/*
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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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#ifndef RTC_BASE_TEST_CLIENT_H_
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#define RTC_BASE_TEST_CLIENT_H_
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#include <memory>
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#include <vector>
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#include "api/units/timestamp.h"
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#include "rtc_base/async_udp_socket.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/fake_clock.h"
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#include "rtc_base/network/received_packet.h"
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#include "rtc_base/synchronization/mutex.h"
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namespace rtc {
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// A simple client that can send TCP or UDP data and check that it receives
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// what it expects to receive. Useful for testing server functionality.
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class TestClient : public sigslot::has_slots<> {
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public:
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// Records the contents of a packet that was received.
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struct Packet {
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Packet(const rtc::ReceivedPacket& received_packet);
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Packet(const Packet& p);
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SocketAddress addr;
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Buffer buf;
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absl::optional<webrtc::Timestamp> packet_time;
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};
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// Default timeout for NextPacket reads.
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static const int kTimeoutMs = 5000;
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// Creates a client that will send and receive with the given socket and
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// will post itself messages with the given thread.
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explicit TestClient(std::unique_ptr<AsyncPacketSocket> socket);
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// Create a test client that will use a fake clock. NextPacket needs to wait
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// for a packet to be received, and thus it needs to advance the fake clock
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// if the test is using one, rather than just sleeping.
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TestClient(std::unique_ptr<AsyncPacketSocket> socket,
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ThreadProcessingFakeClock* fake_clock);
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~TestClient() override;
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TestClient(const TestClient&) = delete;
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TestClient& operator=(const TestClient&) = delete;
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SocketAddress address() const { return socket_->GetLocalAddress(); }
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SocketAddress remote_address() const { return socket_->GetRemoteAddress(); }
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// Checks that the socket moves to the specified connect state.
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bool CheckConnState(AsyncPacketSocket::State state);
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// Checks that the socket is connected to the remote side.
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bool CheckConnected() {
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return CheckConnState(AsyncPacketSocket::STATE_CONNECTED);
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}
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// Sends using the clients socket.
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int Send(const char* buf, size_t size);
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// Sends using the clients socket to the given destination.
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int SendTo(const char* buf, size_t size, const SocketAddress& dest);
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// Returns the next packet received by the client or null if none is received
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// within the specified timeout.
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std::unique_ptr<Packet> NextPacket(int timeout_ms);
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// Checks that the next packet has the given contents. Returns the remote
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// address that the packet was sent from.
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bool CheckNextPacket(const char* buf, size_t len, SocketAddress* addr);
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// Checks that no packets have arrived or will arrive in the next second.
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bool CheckNoPacket();
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int GetError();
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int SetOption(Socket::Option opt, int value);
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bool ready_to_send() const { return ready_to_send_count() > 0; }
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// How many times SignalReadyToSend has been fired.
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int ready_to_send_count() const { return ready_to_send_count_; }
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private:
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// Timeout for reads when no packet is expected.
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static const int kNoPacketTimeoutMs = 1000;
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// Workaround for the fact that AsyncPacketSocket::GetConnState doesn't exist.
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Socket::ConnState GetState();
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void OnPacket(AsyncPacketSocket* socket,
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const rtc::ReceivedPacket& received_packet);
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void OnReadyToSend(AsyncPacketSocket* socket);
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bool CheckTimestamp(absl::optional<webrtc::Timestamp> packet_timestamp);
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void AdvanceTime(int ms);
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ThreadProcessingFakeClock* fake_clock_ = nullptr;
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webrtc::Mutex mutex_;
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std::unique_ptr<AsyncPacketSocket> socket_;
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std::vector<std::unique_ptr<Packet>> packets_;
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int ready_to_send_count_ = 0;
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absl::optional<webrtc::Timestamp> prev_packet_timestamp_;
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};
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} // namespace rtc
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#endif // RTC_BASE_TEST_CLIENT_H_
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