Add Windows, Mac, Android support to low bandwidth audio test

webrtc/BUILD.gn

 # 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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 298 matching lines @@
       } else {
         deps += [ "modules/video_capture:video_capture_tests" ]
       }
       if (!is_ios) {
         deps += [
           "modules/audio_device:audio_device_tests",
           "voice_engine:voe_auto_test",
         ]
       }
       if (rtc_enable_protobuf) {
-        deps += [ "logging:rtc_event_log2rtp_dump" ]
+        deps += [
+          "audio:low_bandwidth_audio_test",
+          "logging:rtc_event_log2rtp_dump",
+        ]
       }
     }
   }
 }
 
 rtc_static_library("webrtc_common") {
   sources = [
     "common_types.cc",
     "common_types.h",
     "config.cc",
@@ skipping 171 matching lines @@
       ]
 
       deps = [
         "//base:base_java_test_support",
         "//webrtc/examples:AppRTCMobile_javalib",
         "//webrtc/sdk/android:libjingle_peerconnection_java",
       ]
     }
   }
 }