Files
platform-external-webrtc/webrtc/rtc_base/socketstream.cc
Henrik Kjellander 6776518bea Move webrtc/{base => rtc_base}
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}
2017-06-28 18:58:10 +00:00

124 lines
3.2 KiB
C++

/*
* Copyright 2010 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/socketstream.h"
#include "webrtc/base/checks.h"
namespace rtc {
SocketStream::SocketStream(AsyncSocket* socket) : socket_(nullptr) {
Attach(socket);
}
SocketStream::~SocketStream() {
delete socket_;
}
void SocketStream::Attach(AsyncSocket* socket) {
if (socket_)
delete socket_;
socket_ = socket;
if (socket_) {
socket_->SignalConnectEvent.connect(this, &SocketStream::OnConnectEvent);
socket_->SignalReadEvent.connect(this, &SocketStream::OnReadEvent);
socket_->SignalWriteEvent.connect(this, &SocketStream::OnWriteEvent);
socket_->SignalCloseEvent.connect(this, &SocketStream::OnCloseEvent);
}
}
AsyncSocket* SocketStream::Detach() {
AsyncSocket* socket = socket_;
if (socket_) {
socket_->SignalConnectEvent.disconnect(this);
socket_->SignalReadEvent.disconnect(this);
socket_->SignalWriteEvent.disconnect(this);
socket_->SignalCloseEvent.disconnect(this);
socket_ = nullptr;
}
return socket;
}
StreamState SocketStream::GetState() const {
RTC_DCHECK(socket_ != nullptr);
switch (socket_->GetState()) {
case Socket::CS_CONNECTED:
return SS_OPEN;
case Socket::CS_CONNECTING:
return SS_OPENING;
case Socket::CS_CLOSED:
default:
return SS_CLOSED;
}
}
StreamResult SocketStream::Read(void* buffer, size_t buffer_len,
size_t* read, int* error) {
RTC_DCHECK(socket_ != nullptr);
int result = socket_->Recv(buffer, buffer_len, nullptr);
if (result < 0) {
if (socket_->IsBlocking())
return SR_BLOCK;
if (error)
*error = socket_->GetError();
return SR_ERROR;
}
if ((result > 0) || (buffer_len == 0)) {
if (read)
*read = result;
return SR_SUCCESS;
}
return SR_EOS;
}
StreamResult SocketStream::Write(const void* data, size_t data_len,
size_t* written, int* error) {
RTC_DCHECK(socket_ != nullptr);
int result = socket_->Send(data, data_len);
if (result < 0) {
if (socket_->IsBlocking())
return SR_BLOCK;
if (error)
*error = socket_->GetError();
return SR_ERROR;
}
if (written)
*written = result;
return SR_SUCCESS;
}
void SocketStream::Close() {
RTC_DCHECK(socket_ != nullptr);
socket_->Close();
}
void SocketStream::OnConnectEvent(AsyncSocket* socket) {
RTC_DCHECK(socket == socket_);
SignalEvent(this, SE_OPEN | SE_READ | SE_WRITE, 0);
}
void SocketStream::OnReadEvent(AsyncSocket* socket) {
RTC_DCHECK(socket == socket_);
SignalEvent(this, SE_READ, 0);
}
void SocketStream::OnWriteEvent(AsyncSocket* socket) {
RTC_DCHECK(socket == socket_);
SignalEvent(this, SE_WRITE, 0);
}
void SocketStream::OnCloseEvent(AsyncSocket* socket, int err) {
RTC_DCHECK(socket == socket_);
SignalEvent(this, SE_CLOSE, err);
}
} // namespace rtc