Add IsAcceptableCipher, use instead of GetDefaultCipher.

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 176 matching lines @@
   // Capabilities testing
   static bool HaveDtls();
   static bool HaveDtlsSrtp();
   static bool HaveExporter();
 
   // Returns the default Ssl cipher used between streams of this class
   // for the given protocol version. This is used by the unit tests.
   // TODO(guoweis): Move this away from a static class method.
   static int GetDefaultSslCipherForTest(SSLProtocolVersion version,
                                         KeyType key_type);
+  static bool IsAcceptableCipher(int cipher,
+                                 SSLProtocolVersion version,
+                                 KeyType key_type);
+  static bool IsAcceptableCipher(std::string cipher,
+                                 SSLProtocolVersion version,
+                                 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);
 
  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_