| Index: webrtc/p2p/base/basicpacketsocketfactory.cc
|
| diff --git a/webrtc/p2p/base/basicpacketsocketfactory.cc b/webrtc/p2p/base/basicpacketsocketfactory.cc
|
| index a05f9df8dfc352cb53d6157893e946c02ff865e9..51e9b07fc0c660c0b7e743ea0b8ab75ba5d514a5 100644
|
| --- a/webrtc/p2p/base/basicpacketsocketfactory.cc
|
| +++ b/webrtc/p2p/base/basicpacketsocketfactory.cc
|
| @@ -10,6 +10,8 @@
|
|
|
| #include "webrtc/p2p/base/basicpacketsocketfactory.h"
|
|
|
| +#include <string>
|
| +
|
| #include "webrtc/p2p/base/asyncstuntcpsocket.h"
|
| #include "webrtc/p2p/base/stun.h"
|
| #include "webrtc/base/asynctcpsocket.h"
|
| @@ -47,9 +49,8 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
|
| uint16_t min_port,
|
| uint16_t max_port) {
|
| // UDP sockets are simple.
|
| - rtc::AsyncSocket* socket =
|
| - socket_factory()->CreateAsyncSocket(
|
| - address.family(), SOCK_DGRAM);
|
| + AsyncSocket* socket =
|
| + socket_factory()->CreateAsyncSocket(address.family(), SOCK_DGRAM);
|
| if (!socket) {
|
| return NULL;
|
| }
|
| @@ -59,7 +60,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
|
| delete socket;
|
| return NULL;
|
| }
|
| - return new rtc::AsyncUDPSocket(socket);
|
| + return new AsyncUDPSocket(socket);
|
| }
|
|
|
| AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
|
| @@ -73,9 +74,8 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
|
| return NULL;
|
| }
|
|
|
| - rtc::AsyncSocket* socket =
|
| - socket_factory()->CreateAsyncSocket(local_address.family(),
|
| - SOCK_STREAM);
|
| + AsyncSocket* socket =
|
| + socket_factory()->CreateAsyncSocket(local_address.family(), SOCK_STREAM);
|
| if (!socket) {
|
| return NULL;
|
| }
|
| @@ -90,24 +90,23 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateServerTcpSocket(
|
| // If using SSLTCP, wrap the TCP socket in a pseudo-SSL socket.
|
| if (opts & PacketSocketFactory::OPT_SSLTCP) {
|
| ASSERT(!(opts & PacketSocketFactory::OPT_TLS));
|
| - socket = new rtc::AsyncSSLSocket(socket);
|
| + socket = new AsyncSSLSocket(socket);
|
| }
|
|
|
| // Set TCP_NODELAY (via OPT_NODELAY) for improved performance.
|
| // See http://go/gtalktcpnodelayexperiment
|
| - socket->SetOption(rtc::Socket::OPT_NODELAY, 1);
|
| + socket->SetOption(Socket::OPT_NODELAY, 1);
|
|
|
| if (opts & PacketSocketFactory::OPT_STUN)
|
| return new cricket::AsyncStunTCPSocket(socket, true);
|
|
|
| - return new rtc::AsyncTCPSocket(socket, true);
|
| + return new AsyncTCPSocket(socket, true);
|
| }
|
|
|
| AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
|
| const SocketAddress& local_address, const SocketAddress& remote_address,
|
| const ProxyInfo& proxy_info, const std::string& user_agent, int opts) {
|
| -
|
| - rtc::AsyncSocket* socket =
|
| + AsyncSocket* socket =
|
| socket_factory()->CreateAsyncSocket(local_address.family(), SOCK_STREAM);
|
| if (!socket) {
|
| return NULL;
|
| @@ -121,20 +120,20 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
|
| }
|
|
|
| // If using a proxy, wrap the socket in a proxy socket.
|
| - if (proxy_info.type == rtc::PROXY_SOCKS5) {
|
| - socket = new rtc::AsyncSocksProxySocket(
|
| + if (proxy_info.type == PROXY_SOCKS5) {
|
| + socket = new AsyncSocksProxySocket(
|
| socket, proxy_info.address, proxy_info.username, proxy_info.password);
|
| - } else if (proxy_info.type == rtc::PROXY_HTTPS) {
|
| - socket = new rtc::AsyncHttpsProxySocket(
|
| - socket, user_agent, proxy_info.address,
|
| - proxy_info.username, proxy_info.password);
|
| + } else if (proxy_info.type == PROXY_HTTPS) {
|
| + socket =
|
| + new AsyncHttpsProxySocket(socket, user_agent, proxy_info.address,
|
| + proxy_info.username, proxy_info.password);
|
| }
|
|
|
| // If using TLS, wrap the socket in an SSL adapter.
|
| if (opts & PacketSocketFactory::OPT_TLS) {
|
| ASSERT(!(opts & PacketSocketFactory::OPT_SSLTCP));
|
|
|
| - rtc::SSLAdapter* ssl_adapter = rtc::SSLAdapter::Create(socket);
|
| + SSLAdapter* ssl_adapter = SSLAdapter::Create(socket);
|
| if (!ssl_adapter) {
|
| return NULL;
|
| }
|
| @@ -149,7 +148,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
|
| // If using SSLTCP, wrap the TCP socket in a pseudo-SSL socket.
|
| } else if (opts & PacketSocketFactory::OPT_SSLTCP) {
|
| ASSERT(!(opts & PacketSocketFactory::OPT_TLS));
|
| - socket = new rtc::AsyncSSLSocket(socket);
|
| + socket = new AsyncSSLSocket(socket);
|
| }
|
|
|
| if (socket->Connect(remote_address) < 0) {
|
| @@ -164,18 +163,18 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
|
| if (opts & PacketSocketFactory::OPT_STUN) {
|
| tcp_socket = new cricket::AsyncStunTCPSocket(socket, false);
|
| } else {
|
| - tcp_socket = new rtc::AsyncTCPSocket(socket, false);
|
| + tcp_socket = new AsyncTCPSocket(socket, false);
|
| }
|
|
|
| // Set TCP_NODELAY (via OPT_NODELAY) for improved performance.
|
| // See http://go/gtalktcpnodelayexperiment
|
| - tcp_socket->SetOption(rtc::Socket::OPT_NODELAY, 1);
|
| + tcp_socket->SetOption(Socket::OPT_NODELAY, 1);
|
|
|
| return tcp_socket;
|
| }
|
|
|
| AsyncResolverInterface* BasicPacketSocketFactory::CreateAsyncResolver() {
|
| - return new rtc::AsyncResolver();
|
| + return new AsyncResolver();
|
| }
|
|
|
| int BasicPacketSocketFactory::BindSocket(AsyncSocket* socket,
|
| @@ -189,8 +188,7 @@ int BasicPacketSocketFactory::BindSocket(AsyncSocket* socket,
|
| } else {
|
| // Otherwise, try to find a port in the provided range.
|
| for (int port = min_port; ret < 0 && port <= max_port; ++port) {
|
| - ret = socket->Bind(rtc::SocketAddress(local_address.ipaddr(),
|
| - port));
|
| + ret = socket->Bind(SocketAddress(local_address.ipaddr(), port));
|
| }
|
| }
|
| return ret;
|
|
|