Adding placeholder for low bandwidth audio test

webrtc/audio/BUILD.gn

Index: webrtc/audio/BUILD.gn
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index acd8dfa7e887b5dc5d7620b61cc27f4eca255d83..a6a22e652cce5cad47874baebe00acf9c1649cc2 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -7,6 +7,10 @@
 # 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 = [
@@ -79,4 +83,20 @@ if (rtc_include_tests) {
       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" ]
+      }
+    }
+  }
 }