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Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.

Browse Source 
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrike@webrtc.org
2014-05-12 18:03:09 +00:00
parent f3a5e6afc4
commit 2c7d1b39b9
386 changed files with 84973 additions and 6 deletions
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33
webrtc/base/sslconfig.h Normal file
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/*
* Copyright 2012 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.
*/
#ifndef WEBRTC_BASE_SSLCONFIG_H_
#define WEBRTC_BASE_SSLCONFIG_H_
// If no preference has been indicated, default to SChannel on Windows and
// OpenSSL everywhere else, if it is available.
#if !defined(SSL_USE_SCHANNEL) && !defined(SSL_USE_OPENSSL) && \
!defined(SSL_USE_NSS)
#if defined(WEBRTC_WIN)
#define SSL_USE_SCHANNEL 1
#else // defined(WEBRTC_WIN)
#if defined(HAVE_OPENSSL_SSL_H)
#define SSL_USE_OPENSSL 1
#elif defined(HAVE_NSS_SSL_H)
#define SSL_USE_NSS 1
#endif
#endif // !defined(WEBRTC_WIN)
#endif
#endif // WEBRTC_BASE_SSLCONFIG_H_