Only use BoringSSL time callback in unit tests.

webrtc/base/opensslstreamadapter.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 100 matching lines @@
 
   // Capabilities interfaces
   static bool HaveDtls();
   static bool HaveDtlsSrtp();
   static bool HaveExporter();
   static bool IsBoringSsl();
 
   static bool IsAcceptableCipher(int cipher, KeyType key_type);
   static bool IsAcceptableCipher(const std::string& cipher, KeyType key_type);
 
+  // Use our timeutils.h source of timing in BoringSSL, allowing us to test
+  // using a fake clock.
+  static void enable_time_callback_for_testing();
+
  protected:
   void OnEvent(StreamInterface* stream, int events, int err) override;
 
  private:
   enum SSLState {
     // Before calling one of the StartSSL methods, data flows
     // in clear text.
     SSL_NONE,
     SSL_WAIT,  // waiting for the stream to open to start SSL negotiation
     SSL_CONNECTING,  // SSL negotiation in progress
@@ skipping 80 matching lines @@
 
   // Max. allowed protocol version
   SSLProtocolVersion ssl_max_version_;
 };
 
 /////////////////////////////////////////////////////////////////////////////
 
 }  // namespace rtc
 
 #endif  // WEBRTC_BASE_OPENSSLSTREAMADAPTER_H__