Reland "Move webrtc/{base => rtc_base}" (https://codereview.webrtc.org/2877023002)
Reland the base->rtc_base without adding stub headers (will be done in follow-up CL). This preserves git blame history of all files. BUG=webrtc:7634 NOTRY=True TBR=kwiberg@webrtc.org Change-Id: Iea3bb6f3f67b8374c96337b63e8f5aa3e6181012 Reviewed-on: https://chromium-review.googlesource.com/554611 Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#18821}
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webrtc/rtc_base/testclient.cc
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webrtc/rtc_base/testclient.cc
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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 "webrtc/base/testclient.h"
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#include "webrtc/base/gunit.h"
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#include "webrtc/base/ptr_util.h"
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#include "webrtc/base/thread.h"
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#include "webrtc/base/timeutils.h"
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namespace rtc {
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// DESIGN: Each packet received is put it into a list of packets.
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// Callers can retrieve received packets from any thread by calling
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// NextPacket.
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TestClient::TestClient(std::unique_ptr<AsyncPacketSocket> socket)
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: TestClient(std::move(socket), nullptr) {}
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TestClient::TestClient(std::unique_ptr<AsyncPacketSocket> socket,
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FakeClock* fake_clock)
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: fake_clock_(fake_clock),
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socket_(std::move(socket)),
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prev_packet_timestamp_(-1) {
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socket_->SignalReadPacket.connect(this, &TestClient::OnPacket);
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socket_->SignalReadyToSend.connect(this, &TestClient::OnReadyToSend);
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}
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TestClient::~TestClient() {}
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bool TestClient::CheckConnState(AsyncPacketSocket::State state) {
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// Wait for our timeout value until the socket reaches the desired state.
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int64_t end = TimeAfter(kTimeoutMs);
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while (socket_->GetState() != state && TimeUntil(end) > 0) {
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AdvanceTime(1);
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}
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return (socket_->GetState() == state);
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}
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int TestClient::Send(const char* buf, size_t size) {
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rtc::PacketOptions options;
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return socket_->Send(buf, size, options);
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}
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int TestClient::SendTo(const char* buf, size_t size,
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const SocketAddress& dest) {
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rtc::PacketOptions options;
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return socket_->SendTo(buf, size, dest, options);
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}
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std::unique_ptr<TestClient::Packet> TestClient::NextPacket(int timeout_ms) {
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// If no packets are currently available, we go into a get/dispatch loop for
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// at most timeout_ms. If, during the loop, a packet arrives, then we can
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// stop early and return it.
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// Note that the case where no packet arrives is important. We often want to
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// test that a packet does not arrive.
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// Note also that we only try to pump our current thread's message queue.
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// Pumping another thread's queue could lead to messages being dispatched from
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// the wrong thread to non-thread-safe objects.
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int64_t end = TimeAfter(timeout_ms);
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while (TimeUntil(end) > 0) {
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{
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CritScope cs(&crit_);
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if (packets_.size() != 0) {
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break;
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}
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}
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AdvanceTime(1);
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}
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// Return the first packet placed in the queue.
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std::unique_ptr<Packet> packet;
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CritScope cs(&crit_);
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if (packets_.size() > 0) {
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packet = std::move(packets_.front());
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packets_.erase(packets_.begin());
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}
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return packet;
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}
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bool TestClient::CheckNextPacket(const char* buf, size_t size,
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SocketAddress* addr) {
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bool res = false;
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std::unique_ptr<Packet> packet = NextPacket(kTimeoutMs);
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if (packet) {
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res = (packet->size == size && memcmp(packet->buf, buf, size) == 0 &&
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CheckTimestamp(packet->packet_time.timestamp));
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if (addr)
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*addr = packet->addr;
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}
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return res;
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}
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bool TestClient::CheckTimestamp(int64_t packet_timestamp) {
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bool res = true;
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if (packet_timestamp == -1) {
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res = false;
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}
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if (prev_packet_timestamp_ != -1) {
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if (packet_timestamp < prev_packet_timestamp_) {
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res = false;
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}
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}
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prev_packet_timestamp_ = packet_timestamp;
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return res;
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}
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void TestClient::AdvanceTime(int ms) {
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// If the test is using a fake clock, we must advance the fake clock to
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// advance time. Otherwise, ProcessMessages will work.
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if (fake_clock_) {
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SIMULATED_WAIT(false, ms, *fake_clock_);
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} else {
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Thread::Current()->ProcessMessages(1);
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}
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}
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bool TestClient::CheckNoPacket() {
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return NextPacket(kNoPacketTimeoutMs) == nullptr;
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}
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int TestClient::GetError() {
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return socket_->GetError();
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}
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int TestClient::SetOption(Socket::Option opt, int value) {
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return socket_->SetOption(opt, value);
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}
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void TestClient::OnPacket(AsyncPacketSocket* socket, const char* buf,
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size_t size, const SocketAddress& remote_addr,
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const PacketTime& packet_time) {
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CritScope cs(&crit_);
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packets_.push_back(MakeUnique<Packet>(remote_addr, buf, size, packet_time));
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}
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void TestClient::OnReadyToSend(AsyncPacketSocket* socket) {
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++ready_to_send_count_;
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}
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TestClient::Packet::Packet(const SocketAddress& a,
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const char* b,
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size_t s,
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const PacketTime& packet_time)
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: addr(a), buf(0), size(s), packet_time(packet_time) {
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buf = new char[size];
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memcpy(buf, b, size);
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}
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TestClient::Packet::Packet(const Packet& p)
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: addr(p.addr), buf(0), size(p.size), packet_time(p.packet_time) {
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buf = new char[size];
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memcpy(buf, p.buf, size);
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}
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TestClient::Packet::~Packet() {
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delete[] buf;
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}
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} // namespace rtc
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