GN: Tighten up test target visibility + refactorings

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 36 matching lines @@
     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
     "../modules/rtp_rtcp:rtp_rtcp",
     "../system_wrappers",
     "../voice_engine",
   ]
 }
 if (rtc_include_tests) {
   rtc_source_set("audio_tests") {
     testonly = true
 
+    # Skip restricting visibility on mobile platforms since the tests on those
+    # gets additional generated targets which would require many lines here to
+    # cover (which would be confusing to read and hard to maintain).
+    if (!is_android && !is_ios) {
+      visibility = [ "//webrtc:video_engine_tests" ]
+    }
+
     # TODO(kjellander): Remove (bugs.webrtc.org/6828)
     # This needs remote_bitrate_estimator to be moved to webrtc/api first.
     check_includes = false
 
     sources = [
       "audio_receive_stream_unittest.cc",
       "audio_send_stream_unittest.cc",
       "audio_state_unittest.cc",
     ]
     deps = [
@@ skipping 44 matching lines @@
         "//resources/voice_engine/audio_tiny48.wav",
       ]
 
       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" ]
       }
     }
   }
 }