Index: webrtc/base/sslstreamadapter.h |
diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h |
index f6f0befa05a8d320da31255c06bf86a579be1584..9a9446267d34f9b1f3564d1c6c48a510fc1f745d 100644 |
--- a/webrtc/base/sslstreamadapter.h |
+++ b/webrtc/base/sslstreamadapter.h |
@@ -24,8 +24,12 @@ 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. |