Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.

webrtc/webrtc.gni

 # Copyright (c) 2014 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.
 
 import("//build/config/arm.gni")
 import("//build/config/features.gni")
@@ skipping 30 matching lines @@
 
   # Selects fixed-point code where possible.
   rtc_prefer_fixed_point = false
 
   # Enables the use of protocol buffers for debug recordings.
   rtc_enable_protobuf = true
 
   # Disable the code for the intelligibility enhancer by default.
   rtc_enable_intelligibility_enhancer = false
 
-  # Enable when an external authentication mechanism is used for performing
-  # packet authentication for RTP packets instead of libsrtp.
-  rtc_enable_external_auth = build_with_chromium
-
   # Selects whether debug dumps for the audio processing module
   # should be generated.
   apm_debug_dump = false
 
   # Set this to true to enable BWE test logging.
   rtc_enable_bwe_test_logging = false
 
   # Set this to disable building with support for SCTP data channels.
   rtc_enable_sctp = true
 
@@ skipping 285 matching lines @@
       configs += invoker.configs
       configs -= rtc_remove_configs
       configs -= invoker.suppressed_configs
       public_configs = [ rtc_common_inherited_config ]
       if (defined(invoker.public_configs)) {
         public_configs += invoker.public_configs
       }
     }
   }
 }