Convert internal representation of Srtp cryptos from string to int.

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 13 matching lines @@
 const int SRTP_AES128_CM_SHA1_32 = 0x0002;
 
 // Cipher suite to use for SRTP. Typically a 80-bit HMAC will be used, except
 // in applications (voice) where the additional bandwidth may be significant.
 // A 80-bit HMAC is always used for SRTCP.
 // 128-bit AES with 80-bit SHA-1 HMAC.
 extern const char CS_AES_CM_128_HMAC_SHA1_80[];
 // 128-bit AES with 32-bit SHA-1 HMAC.
 extern const char CS_AES_CM_128_HMAC_SHA1_32[];
 
-// Returns the DTLS-SRTP protection profile ID, as defined in
-// https://tools.ietf.org/html/rfc5764#section-4.1.2, for the given SRTP
-// Crypto-suite, as defined in https://tools.ietf.org/html/rfc4568#section-6.2
-int GetSrtpCryptoSuiteFromName(const std::string& cipher_rfc_name);
+// Given the DTLS-SRTP protection profile ID, as defined in
+// https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile
+// name, as defined in https://tools.ietf.org/html/rfc5764#section-4.1.2.
+std::string SrtpCryptoSuiteToName(int srtp_crypto);
+
+// The reverse of above conversion.
+int SrtpCryptoSuiteNameToId(const std::string& crypto);
 
 // SSLStreamAdapter : A StreamInterfaceAdapter that does SSL/TLS.
 // After SSL has been started, the stream will only open on successful
 // SSL verification of certificates, and the communication is
 // encrypted of course.
 //
 // This class was written with SSLAdapter as a starting point. It
 // offers a similar interface, with two differences: there is no
 // support for a restartable SSL connection, and this class has a
 // peer-to-peer mode.
@@ skipping 119 matching lines @@
   // result              -- where to put the computed value
   // result_len          -- the length of the computed value
   virtual bool ExportKeyingMaterial(const std::string& label,
                                     const uint8_t* context,
                                     size_t context_len,
                                     bool use_context,
                                     uint8_t* result,
                                     size_t result_len);
 
   // DTLS-SRTP interface
-  virtual bool SetDtlsSrtpCiphers(const std::vector<std::string>& ciphers);
-  virtual bool GetDtlsSrtpCipher(std::string* cipher);
+  virtual bool SetDtlsSrtpCryptoSuites(const std::vector<int>& ciphers);
+  virtual bool GetDtlsSrtpCryptoSuite(int* cipher);

[pthatcher1, 2015/11/11 19:59:41]

More here.

I'm going to stop commenting on these.  Please find the rest and remain
consistent.

[guoweis_webrtc, 2015/11/17 01:21:16]

Done.

 
   // 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);
 
   // 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 GetSslCipherSuiteName(int cipher);
+  static std::string SslCipherSuiteToName(int cipher);
 
  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_