Only use BoringSSL time callback in unit tests.

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 226 matching lines @@
   // 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.
   static std::string SslCipherSuiteToName(int cipher_suite);
 
+  // Use our timeutils.h source of timing in BoringSSL, allowing us to test
+  // using a fake clock.
+  static void enable_time_callback_for_testing();
+
   sigslot::signal1<SSLHandshakeError> SignalSSLHandshakeError;
 
  private:
   // If true, the server certificate need not match the configured
   // server_name, and in fact missing certificate authority and other
   // verification errors are ignored.
   bool ignore_bad_cert_;
 
   // 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_