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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/test_client.h"
#include <string.h>
#include <memory>
#include <utility>
#include "api/units/timestamp.h"
#include "rtc_base/gunit.h"
#include "rtc_base/network/received_packet.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
namespace rtc {
// DESIGN: Each packet received is put it into a list of packets.
// Callers can retrieve received packets from any thread by calling
// NextPacket.
TestClient::TestClient(std::unique_ptr<AsyncPacketSocket> socket)
: TestClient(std::move(socket), nullptr) {}
TestClient::TestClient(std::unique_ptr<AsyncPacketSocket> socket,
ThreadProcessingFakeClock* fake_clock)
: fake_clock_(fake_clock), socket_(std::move(socket)) {
socket_->RegisterReceivedPacketCallback(
[&](rtc::AsyncPacketSocket* socket, const rtc::ReceivedPacket& packet) {
OnPacket(socket, packet);
});
socket_->SignalReadyToSend.connect(this, &TestClient::OnReadyToSend);
}
TestClient::~TestClient() {}
bool TestClient::CheckConnState(AsyncPacketSocket::State state) {
// Wait for our timeout value until the socket reaches the desired state.
int64_t end = TimeAfter(kTimeoutMs);
while (socket_->GetState() != state && TimeUntil(end) > 0) {
AdvanceTime(1);
}
return (socket_->GetState() == state);
}
int TestClient::Send(const char* buf, size_t size) {
rtc::PacketOptions options;
return socket_->Send(buf, size, options);
}
int TestClient::SendTo(const char* buf,
size_t size,
const SocketAddress& dest) {
rtc::PacketOptions options;
return socket_->SendTo(buf, size, dest, options);
}
std::unique_ptr<TestClient::Packet> TestClient::NextPacket(int timeout_ms) {
// If no packets are currently available, we go into a get/dispatch loop for
// at most timeout_ms. If, during the loop, a packet arrives, then we can
// stop early and return it.
// Note that the case where no packet arrives is important. We often want to
// test that a packet does not arrive.
// Note also that we only try to pump our current thread's message queue.
// Pumping another thread's queue could lead to messages being dispatched from
// the wrong thread to non-thread-safe objects.
int64_t end = TimeAfter(timeout_ms);
while (TimeUntil(end) > 0) {
{
webrtc::MutexLock lock(&mutex_);
if (packets_.size() != 0) {
break;
}
}
AdvanceTime(1);
}
// Return the first packet placed in the queue.
std::unique_ptr<Packet> packet;
webrtc::MutexLock lock(&mutex_);
if (packets_.size() > 0) {
packet = std::move(packets_.front());
packets_.erase(packets_.begin());
}
return packet;
}
bool TestClient::CheckNextPacket(const char* buf,
size_t size,
SocketAddress* addr) {
bool res = false;
std::unique_ptr<Packet> packet = NextPacket(kTimeoutMs);
if (packet) {
res = (packet->buf.size() == size &&
memcmp(packet->buf.data(), buf, size) == 0 &&
CheckTimestamp(packet->packet_time));
if (addr)
*addr = packet->addr;
}
return res;
}
bool TestClient::CheckTimestamp(
absl::optional<webrtc::Timestamp> packet_timestamp) {
bool res = true;
if (!packet_timestamp) {
res = false;
}
if (prev_packet_timestamp_) {
if (packet_timestamp < prev_packet_timestamp_) {
res = false;
}
}
prev_packet_timestamp_ = packet_timestamp;
return res;
}
void TestClient::AdvanceTime(int ms) {
// If the test is using a fake clock, we must advance the fake clock to
// advance time. Otherwise, ProcessMessages will work.
if (fake_clock_) {
SIMULATED_WAIT(false, ms, *fake_clock_);
} else {
Thread::Current()->ProcessMessages(1);
}
}
bool TestClient::CheckNoPacket() {
return NextPacket(kNoPacketTimeoutMs) == nullptr;
}
int TestClient::GetError() {
return socket_->GetError();
}
int TestClient::SetOption(Socket::Option opt, int value) {
return socket_->SetOption(opt, value);
}
void TestClient::OnPacket(AsyncPacketSocket* socket,
const rtc::ReceivedPacket& received_packet) {
webrtc::MutexLock lock(&mutex_);
packets_.push_back(std::make_unique<Packet>(received_packet));
}
void TestClient::OnReadyToSend(AsyncPacketSocket* socket) {
++ready_to_send_count_;
}
TestClient::Packet::Packet(const rtc::ReceivedPacket& received_packet)
: addr(received_packet.source_address()),
// Copy received_packet payload to a buffer owned by Packet.
buf(received_packet.payload().data(), received_packet.payload().size()),
packet_time(received_packet.arrival_time()) {}
TestClient::Packet::Packet(const Packet& p)
: addr(p.addr),
buf(p.buf.data(), p.buf.size()),
packet_time(p.packet_time) {}
} // namespace rtc
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