Add support for GCM cipher suites from RFC 7714.

webrtc/base/sslstreamadapter.h

Index: webrtc/base/sslstreamadapter.h
diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h
index c57056b14ad1b2336b1527c00bd7c24059ff02f0..5af0fdd21175d4a76244ab7a03e3468cbe61e92e 100644
--- a/webrtc/base/sslstreamadapter.h
+++ b/webrtc/base/sslstreamadapter.h
@@ -26,6 +26,8 @@ const int TLS_NULL_WITH_NULL_NULL = 0;
 const int SRTP_INVALID_CRYPTO_SUITE = 0;
 const int SRTP_AES128_CM_SHA1_80 = 0x0001;
 const int SRTP_AES128_CM_SHA1_32 = 0x0002;
+const int SRTP_AEAD_AES_128_GCM = 0x0007;
+const int SRTP_AEAD_AES_256_GCM = 0x0008;
 
 // 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.
@@ -34,6 +36,10 @@ const int SRTP_AES128_CM_SHA1_32 = 0x0002;
 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[];
+// 128-bit AES GCM with 16 byte AEAD auth tag.
+extern const char CS_AEAD_AES_128_GCM[];
+// 256-bit AES GCM with 16 byte AEAD auth tag.
+extern const char CS_AEAD_AES_256_GCM[];
 
 // Given the DTLS-SRTP protection profile ID, as defined in
 // https://tools.ietf.org/html/rfc4568#section-6.2 , return the SRTP profile
@@ -43,6 +49,10 @@ std::string SrtpCryptoSuiteToName(int crypto_suite);
 // The reverse of above conversion.
 int SrtpCryptoSuiteFromName(const std::string& crypto_suite);
 
+// Get key length and salt length for given crypto suite. Returns true for
+// valid suites, otherwise false.
+bool SrtpCryptoSuiteParams(int crypto_suite, int *key_length, int *salt_length);
+
 // 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