Set use_gtk=0 as default for Chromium builds.

webrtc/build/common.gypi

 # Copyright (c) 2012 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.
 
 # This file contains common settings for building WebRTC components.
 
@@ skipping 103 matching lines @@
     # Make it possible to provide custom locations for some libraries.
     'libvpx_dir%': '<(DEPTH)/third_party/libvpx_new',
     'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
     'opus_dir%': '<(opus_dir)',
 
     # Use Java based audio layer as default for Android.
     # Change this setting to 1 to use Open SL audio instead.
     # TODO(henrika): add support for Open SL ES.
     'enable_android_opensl%': 0,
 
-    # Disable this to skip building source requiring GTK.
-    'use_gtk%': 1,
-
     # Link-Time Optimizations
     # Executes code generation at link-time instead of compile-time
     # https://gcc.gnu.org/wiki/LinkTimeOptimization
     'use_lto%': 0,
 
     # Defer ssl perference to that specified through sslconfig.h instead of
     # choosing openssl or nss directly.  In practice, this can be used to
     # enable schannel on windows.
     'use_legacy_ssl_defaults%': 0,
 
@@ skipping 20 matching lines @@
       # FFmpeg must only be initialized once. Projects that initialize FFmpeg
       # externally, such as Chromium, must turn this flag off so that WebRTC
       # does not also initialize.
       ['build_with_chromium==0', {
         'rtc_initialize_ffmpeg%': 1,
       }, {
         'rtc_initialize_ffmpeg%': 0,
       }],
 
       ['build_with_chromium==1', {
+        # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
+        # build environments, even if available for Chromium builds.
+        'use_gtk%': 0,
         # Exclude pulse audio on Chromium since its prerequisites don't require
         # pulse audio.
         'include_pulse_audio%': 0,
 
         # Exclude internal ADM since Chromium uses its own IO handling.
         'include_internal_audio_device%': 0,
 
         # Remove tests for Chromium to avoid slowing down GYP generation.
         'include_tests%': 0,
         'restrict_webrtc_logging%': 1,
       }, {  # Settings for the standalone (not-in-Chromium) build.
+        'use_gtk%': 1,
         # TODO(andrew): For now, disable the Chrome plugins, which causes a
         # flood of chromium-style warnings. Investigate enabling them:
         # http://code.google.com/p/webrtc/issues/detail?id=163
         'clang_use_chrome_plugins%': 0,
 
         'include_pulse_audio%': 1,
         'include_internal_audio_device%': 1,
         'include_tests%': 1,
         'restrict_webrtc_logging%': 0,
       }],
@@ skipping 290 matching lines @@
           # of a more specific macro.
           'defines': [
             'WEBRTC_POSIX',
           ],
         }],
       ],
     },
   }, # target_defaults
 }