This refactoring takes a careful approach to avoid rushing the change: * stub headers are left in all the old locations of webrtc/base * existing GN targets are kept and now just forward to the moved ones using public_deps. The only exception to the above is the base_java target and its .java files, which were moved to webrtc/rtc_base right away since it's not possible to use public_deps for android_library. To avoid breaking builds, a temporary Dummy.java file was added to the new intermediate target in webrtc/rtc_base:base_java as well to avoid hitting a GN assert in the android_library template. The above approach should make the transition smooth without breaking downstream. A helper script was created (https://codereview.webrtc.org/2879203002/) and was run like this: stub-headers.py -s webrtc/base -d webrtc/rtc_base -i 7634 stub-headers.py -s webrtc/base/numerics -d webrtc/rtc_base/numerics -i 7634 Fixed invalid header guards in the following files: webrtc/base/base64.h webrtc/base/cryptstring.h webrtc/base/event.h webrtc/base/flags.h webrtc/base/httpbase.h webrtc/base/httpcommon-inl.h webrtc/base/httpcommon.h webrtc/base/httpserver.h webrtc/base/logsinks.h webrtc/base/macutils.h webrtc/base/nattypes.h webrtc/base/openssladapter.h webrtc/base/opensslstreamadapter.h webrtc/base/pathutils.h webrtc/base/physicalsocketserver.h webrtc/base/proxyinfo.h webrtc/base/sigslot.h webrtc/base/sigslotrepeater.h webrtc/base/socket.h webrtc/base/socketaddresspair.h webrtc/base/socketfactory.h webrtc/base/stringutils.h webrtc/base/testbase64.h webrtc/base/testutils.h webrtc/base/transformadapter.h webrtc/base/win32filesystem.h Added new header guards to: sslroots.h testbase64.h BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True R=kwiberg@webrtc.org Review-Url: https://codereview.webrtc.org/2877023002 . Cr-Commit-Position: refs/heads/master@{#18816}
172 lines
5.0 KiB
C++
172 lines
5.0 KiB
C++
/*
|
|
* 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 "webrtc/base/testclient.h"
|
|
|
|
#include "webrtc/base/gunit.h"
|
|
#include "webrtc/base/ptr_util.h"
|
|
#include "webrtc/base/thread.h"
|
|
#include "webrtc/base/timeutils.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,
|
|
FakeClock* fake_clock)
|
|
: fake_clock_(fake_clock),
|
|
socket_(std::move(socket)),
|
|
prev_packet_timestamp_(-1) {
|
|
socket_->SignalReadPacket.connect(this, &TestClient::OnPacket);
|
|
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) {
|
|
{
|
|
CritScope cs(&crit_);
|
|
if (packets_.size() != 0) {
|
|
break;
|
|
}
|
|
}
|
|
AdvanceTime(1);
|
|
}
|
|
|
|
// Return the first packet placed in the queue.
|
|
std::unique_ptr<Packet> packet;
|
|
CritScope cs(&crit_);
|
|
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->size == size && memcmp(packet->buf, buf, size) == 0 &&
|
|
CheckTimestamp(packet->packet_time.timestamp));
|
|
if (addr)
|
|
*addr = packet->addr;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
bool TestClient::CheckTimestamp(int64_t packet_timestamp) {
|
|
bool res = true;
|
|
if (packet_timestamp == -1) {
|
|
res = false;
|
|
}
|
|
if (prev_packet_timestamp_ != -1) {
|
|
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 char* buf,
|
|
size_t size, const SocketAddress& remote_addr,
|
|
const PacketTime& packet_time) {
|
|
CritScope cs(&crit_);
|
|
packets_.push_back(MakeUnique<Packet>(remote_addr, buf, size, packet_time));
|
|
}
|
|
|
|
void TestClient::OnReadyToSend(AsyncPacketSocket* socket) {
|
|
++ready_to_send_count_;
|
|
}
|
|
|
|
TestClient::Packet::Packet(const SocketAddress& a,
|
|
const char* b,
|
|
size_t s,
|
|
const PacketTime& packet_time)
|
|
: addr(a), buf(0), size(s), packet_time(packet_time) {
|
|
buf = new char[size];
|
|
memcpy(buf, b, size);
|
|
}
|
|
|
|
TestClient::Packet::Packet(const Packet& p)
|
|
: addr(p.addr), buf(0), size(p.size), packet_time(p.packet_time) {
|
|
buf = new char[size];
|
|
memcpy(buf, p.buf, size);
|
|
}
|
|
|
|
TestClient::Packet::~Packet() {
|
|
delete[] buf;
|
|
}
|
|
|
|
} // namespace rtc
|