Add low_bandwidth_audio_test to default build

webrtc/audio/BUILD.gn

 # Copyright (c) 2015 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("../webrtc.gni")
 if (is_android) {
@@ skipping 60 matching lines @@
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/congestion_controller:congestion_controller",
       "../modules/pacing:pacing",
       "../test:test_common",
       "../test:test_support",
       "utility:utility_tests",
       "//testing/gmock",
       "//testing/gtest",
     ]
+    if (rtc_enable_protobuf) {
+      deps += [ ":low_bandwidth_audio_test" ]
+    }
+
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 
   if (rtc_enable_protobuf) {
     rtc_executable("low_bandwidth_audio_test") {
       testonly = true
 
       sources = [
         "test/low_bandwidth_audio_test.cc",
       ]
 
       deps = []
 
       if (is_android) {
         deps += [ "//testing/android/native_test:native_test_support" ]
       }
     }
   }
 }