Change WebRTC SslCipher to be exposed as number only.

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 29 matching lines @@
   SSL_PROTOCOL_TLS_10,
   SSL_PROTOCOL_TLS_11,
   SSL_PROTOCOL_TLS_12,
   SSL_PROTOCOL_DTLS_10 = SSL_PROTOCOL_TLS_11,
   SSL_PROTOCOL_DTLS_12 = SSL_PROTOCOL_TLS_12,
 };
 
 // Errors for Read -- in the high range so no conflict with OpenSSL.
 enum { SSE_MSG_TRUNC = 0xff0001 };
 
+// Helper struct to retrieve both IANA number and the RFC name.
+struct SslCipher {
+  uint16_t ssl_id;

[juberti, 2015/09/24 13:41:15]

prefer plain |int| to a sized type.

[guoweis_webrtc, 2015/09/24 18:27:13]

Done.

+  std::string rfc_name;

[juberti, 2015/09/24 13:41:15]

I would just call this |name|

[guoweis_webrtc, 2015/09/24 18:27:13]

Done.

+  SslCipher() : ssl_id(0) {}
+  SslCipher(uint16_t ssl_id, const std::string& rfc_name)
+      : ssl_id(ssl_id), rfc_name(rfc_name) {}
+};
+
 class SSLStreamAdapter : public StreamAdapterInterface {
  public:
   // Instantiate an SSLStreamAdapter wrapping the given stream,
   // (using the selected implementation for the platform).
   // Caller is responsible for freeing the returned object.
   static SSLStreamAdapter* Create(StreamInterface* stream);
 
   explicit SSLStreamAdapter(StreamInterface* stream)
       : StreamAdapterInterface(stream), ignore_bad_cert_(false),
         client_auth_enabled_(true) { }
@@ skipping 68 matching lines @@
                                         const unsigned char* digest_val,
                                         size_t digest_len) = 0;
 
   // Retrieves the peer's X.509 certificate, if a connection has been
   // established. It returns the transmitted over SSL, including the entire
   // chain. The returned certificate is owned by the caller.
   virtual bool GetPeerCertificate(SSLCertificate** cert) const = 0;
 
   // Retrieves the name of the cipher suite used for the connection
   // (e.g. "TLS_RSA_WITH_AES_128_CBC_SHA").
-  virtual bool GetSslCipher(std::string* cipher);
+  virtual bool GetSslCipher(SslCipher* cipher);
 
   // Key Exporter interface from RFC 5705
   // Arguments are:
   // label               -- the exporter label.
   //                        part of the RFC defining each exporter
   //                        usage (IN)
   // context/context_len -- a context to bind to for this connection;
   //                        optional, can be NULL, 0 (IN)
   // use_context         -- whether to use the context value
   //                        (needed to distinguish no context from
@@ skipping 12 matching lines @@
   virtual bool GetDtlsSrtpCipher(std::string* cipher);
 
   // 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(torbjorng@webrtc.org): Fix callers to avoid default parameter.
-  static std::string GetDefaultSslCipher(SSLProtocolVersion version,
-                                         KeyType key_type = KT_DEFAULT);
+  static const SslCipher& GetDefaultSslCipher(SSLProtocolVersion version,
+                                              KeyType key_type = KT_DEFAULT);
 
  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_