Adding placeholder for low bandwidth audio test

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) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
 
 rtc_static_library("audio") {
   sources = [
     "audio_receive_stream.cc",
     "audio_receive_stream.h",
     "audio_send_stream.cc",
     "audio_send_stream.h",
     "audio_state.cc",
     "audio_state.h",
     "audio_transport_proxy.cc",
@@ skipping 52 matching lines @@
       "../test:test_support",
       "utility:utility_tests",
       "//testing/gmock",
       "//testing/gtest",
     ]
     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" ]
+      }
+    }
+  }
 }