Change WebRTC SslCipher to be exposed as number only.

talk/session/media/srtpfilter.h

 /*
  * libjingle
  * Copyright 2009 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 21 matching lines @@
 #include <map>
 #include <string>
 #include <vector>
 
 #include "talk/media/base/cryptoparams.h"
 #include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/sigslotrepeater.h"
+#include "webrtc/base/sslstreamadapter.h"
 
 // Forward declaration to avoid pulling in libsrtp headers here
 struct srtp_event_data_t;
 struct srtp_ctx_t;
 struct srtp_policy_t;
 
 namespace cricket {
 
-// 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[];
 // Key is 128 bits and salt is 112 bits == 30 bytes. B64 bloat => 40 bytes.
 extern const int SRTP_MASTER_KEY_BASE64_LEN;
 
 // Needed for DTLS-SRTP
 extern const int SRTP_MASTER_KEY_KEY_LEN;
 extern const int SRTP_MASTER_KEY_SALT_LEN;
 
 class SrtpSession;
 class SrtpStat;
 
@@ skipping 252 matching lines @@
   std::map<FailureKey, FailureStat> failures_;
   // Threshold in ms to silent the signaling errors.
   uint32 signal_silent_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(SrtpStat);
 };
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_MEDIA_SRTPFILTER_H_