Revert of Removing #defines previously used for building without BoringSSL/OpenSSL.

webrtc/base/sslstreamadapter.h

 /*
  *  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.
  */
 
@@ skipping 210 matching lines @@
   // DTLS-SRTP interface
   virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& crypto_suites);
   virtual bool GetDtlsSrtpCryptoSuite(int* crypto_suite);
 
   // Returns true if a TLS connection has been established.
   // The only difference between this and "GetState() == SE_OPEN" is that if
   // the peer certificate digest hasn't been verified, the state will still be
   // SS_OPENING but IsTlsConnected should return true.
   virtual bool IsTlsConnected() = 0;
 
-  // Capabilities testing.
-  // Used to have "DTLS supported", "DTLS-SRTP supported" etc. methods, but now
-  // that's assumed.
+  // Capabilities testing
+  static bool HaveDtls();
+  static bool HaveDtlsSrtp();
+  static bool HaveExporter();
   static bool IsBoringSsl();
 
   // Returns true iff the supplied cipher is deemed to be strong.
   // TODO(torbjorng): Consider removing the KeyType argument.
   static bool IsAcceptableCipher(int cipher, KeyType key_type);
   static bool IsAcceptableCipher(const std::string& cipher, KeyType key_type);
 
   // TODO(guoweis): Move this away from a static class method. Currently this is
   // introduced such that any caller could depend on sslstreamadapter.h without
   // depending on specific SSL implementation.
@@ skipping 13 matching lines @@
 
   // If true (default), the client is required to provide a certificate during
   // handshake. If no certificate is given, handshake fails. This applies to
   // server mode only.
   bool client_auth_enabled_;
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_SSLSTREAMADAPTER_H_