Add rtc_include_alsa GN variable

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 154 matching lines @@
 
 # A second declare_args block, so that declarations within it can
 # depend on the possibly overridden variables in the first
 # declare_args block.
 declare_args() {
   # Include the iLBC audio codec?
   rtc_include_ilbc = !(build_with_chromium || build_with_mozilla)
 
   rtc_restrict_logging = build_with_chromium
 
+  # Set to false to disable Alsa.
+  rtc_include_alsa = true
+
   # Excluded in Chromium since its prerequisites don't require Pulse Audio.
   rtc_include_pulse_audio = !build_with_chromium
 
   # Chromium uses its own IO handling, so the internal ADM is only built for
   # standalone WebRTC.
   rtc_include_internal_audio_device = !build_with_chromium
 
   # Include tests in standalone checkout.
   rtc_include_tests = !build_with_chromium
 }
@@ skipping 187 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
       }
     }
   }
 }