OpenSSL/BoringSSL compatibility fixes.

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.
  */
 
 #ifndef WEBRTC_BASE_SSLSTREAMADAPTER_H_
 #define WEBRTC_BASE_SSLSTREAMADAPTER_H_
 
 #include <string>
 #include <vector>
 
 #include "webrtc/base/stream.h"
 #include "webrtc/base/sslidentity.h"
 
 namespace rtc {
 
 // Constants for SSL profile.
 const int TLS_NULL_WITH_NULL_NULL = 0;
 
 // Constants for SRTP profiles.
 const int SRTP_INVALID_CRYPTO_SUITE = 0;
+#ifndef SRTP_AES128_CM_SHA1_80
 const int SRTP_AES128_CM_SHA1_80 = 0x0001;
+#endif
+#ifndef SRTP_AES128_CM_SHA1_32
 const int SRTP_AES128_CM_SHA1_32 = 0x0002;
+#endif
 
 // 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[];
 
 // Given the DTLS-SRTP protection profile ID, as defined in
@@ skipping 170 matching lines @@
 
   // 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_