GN: New conventions, default target and refactorings

webrtc/modules/BUILD.gn

Index: webrtc/modules/BUILD.gn
diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn
index 8f90a00f16e986c369170632ec5b07c4d6e69abf..a7e388ea8fab76c14c1e6db82fc558996a485cfa 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -19,13 +19,17 @@ group("modules") {
     "audio_coding",
     "audio_conference_mixer",
     "audio_device",
-    "audio_mixer:audio_mixer_impl",
+    "audio_mixer",
     "audio_processing",
     "bitrate_controller",
+    "congestion_controller",
     "desktop_capture",
     "media_file",
+    "pacing",
+    "remote_bitrate_estimator",
     "rtp_rtcp",
     "utility",
+    "video_capture",
     "video_coding",
     "video_processing",
   ]
@@ -625,7 +629,7 @@ if (rtc_include_tests) {
       "../base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
       "../common_audio",
       "../common_video",
-      "../system_wrappers:system_wrappers",
+      "../system_wrappers",
       "../test:rtp_test_utils",
       "../test:test_common",
       "../test:test_support",
@@ -645,8 +649,7 @@ if (rtc_include_tests) {
       "audio_coding:webrtc_opus",
       "audio_conference_mixer",
       "audio_device",
-      "audio_mixer:audio_frame_manipulator",
-      "audio_mixer:audio_mixer_impl",
+      "audio_mixer",
       "audio_processing",
       "audio_processing:audioproc_test_utils",
       "bitrate_controller",