Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"""

This reverts commit c73e1f437889d882cbf2987f7fb3a029a6150613.

Reason for revert: 
The problem with failed deps in chrome content/renderer had already been fixed in https://webrtc-review.googlesource.com/c/src/+/38660

Original change's description:
> Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""
> 
> This reverts commit 588c548657b3ddf76e7b3f241263eef7f5799f16.
> 
> Reason for revert: 
> 
> Breaks Chrome FYI:
> 
> /b/c/b/Linux_Builder/src/buildtools/linux64/gn gen //out/Release --check
>   -> returned 1
> ERROR at //build/split_static_library.gni:12:5: Dependency not allowed.
>     static_library(target_name) {
>     ^----------------------------
> The item //content/renderer:renderer
> can not depend on //third_party/webrtc/media:rtc_internal_video_codecs
> because it is not in //third_party/webrtc/media:rtc_internal_video_codecs's visibility list: [
>   //third_party/webrtc/*
>   //third_party/webrtc_overrides/*
> ]
> 
>  https://logs.chromium.org/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F23560%2F%2B%2Frecipes%2Fsteps%2Fgenerate_build_files%2F0%2Fstdout
> 
> Original change's description:
> > GN rtc_* templates: Set default visibility to webrtc_root + "/*"
> > 
> > This means that by default, targets are visible to everything under
> > the WebRTC root, but not visible to anything else.
> > 
> > API targets are manually tagged with visibility "*", so that targets
> > outside the WebRTC tree can see them.
> > 
> > BUG=webrtc:8254
> > 
> > Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
> > Reviewed-on: https://webrtc-review.googlesource.com/24140
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21548}
> 
> TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
> 
> Change-Id: I06620ce3d6f67482935c22efa231dd6cab91625a
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8254
> Reviewed-on: https://webrtc-review.googlesource.com/38760
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21555}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org,perkj@webrtc.org

Change-Id: I6f720078ce21bd172e0a6471bae8c4c011e4a657
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8254
Reviewed-on: https://webrtc-review.googlesource.com/38860
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21558}
diff --git a/modules/BUILD.gn b/modules/BUILD.gn
index 1f93e99..b5de323 100644
--- a/modules/BUILD.gn
+++ b/modules/BUILD.gn
@@ -43,6 +43,7 @@
 }
 
 rtc_source_set("module_api") {
+  visibility = [ "*" ]
   sources = [
     "include/module.h",
     "include/module_common_types.h",
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index 1d6a70c8..f9eea0c 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -40,7 +40,7 @@
                     ]
 
 rtc_static_library("audio_format_conversion") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "codecs/audio_format_conversion.cc",
     "codecs/audio_format_conversion.h",
@@ -57,7 +57,7 @@
 }
 
 rtc_static_library("rent_a_codec") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "acm2/acm_codec_database.cc",
     "acm2/acm_codec_database.h",
@@ -297,7 +297,7 @@
 }
 
 rtc_static_library("ilbc") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "codecs/ilbc/audio_decoder_ilbc.cc",
     "codecs/ilbc/audio_decoder_ilbc.h",
@@ -823,7 +823,7 @@
 }
 
 rtc_static_library("webrtc_opus") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "codecs/opus/audio_decoder_opus.cc",
     "codecs/opus/audio_decoder_opus.h",
@@ -884,7 +884,7 @@
 
 if (rtc_enable_protobuf) {
   proto_library("ana_debug_dump_proto") {
-    visibility += [ "*" ]
+    visibility += webrtc_default_visibility
     sources = [
       "audio_network_adaptor/debug_dump.proto",
     ]
@@ -902,7 +902,7 @@
 }
 
 rtc_static_library("audio_network_adaptor_config") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "audio_network_adaptor/audio_network_adaptor_config.cc",
     "audio_network_adaptor/include/audio_network_adaptor_config.h",
@@ -913,7 +913,7 @@
 }
 
 rtc_static_library("audio_network_adaptor") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "audio_network_adaptor/audio_network_adaptor_impl.cc",
     "audio_network_adaptor/audio_network_adaptor_impl.h",
@@ -984,7 +984,7 @@
 }
 
 rtc_static_library("neteq") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "neteq/accelerate.cc",
     "neteq/accelerate.h",
@@ -1082,7 +1082,7 @@
 # that ultimately are built and run as a part of the Chromium ecosystem, which
 # does not set the rtc_include_tests flag.
 rtc_source_set("neteq_tools_minimal") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "neteq/tools/audio_sink.cc",
     "neteq/tools/audio_sink.h",
@@ -1119,7 +1119,7 @@
 }
 
 rtc_source_set("neteq_test_tools") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   testonly = true
   sources = [
     "neteq/tools/audio_checksum.h",
@@ -1178,7 +1178,7 @@
 }
 
 rtc_source_set("neteq_tools") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "neteq/tools/fake_decode_from_file.cc",
     "neteq/tools/fake_decode_from_file.h",
@@ -1214,7 +1214,7 @@
 }
 
 rtc_source_set("neteq_input_audio_tools") {
-  visibility += [ "*" ]
+  visibility += webrtc_default_visibility
   sources = [
     "neteq/tools/input_audio_file.cc",
     "neteq/tools/input_audio_file.h",
@@ -1275,7 +1275,7 @@
   }
 
   group("audio_coding_tests") {
-    visibility += [ "*" ]
+    visibility += webrtc_default_visibility
     testonly = true
     public_deps = [
       ":acm_receive_test",
@@ -1310,7 +1310,7 @@
 
   rtc_source_set("audio_coding_modules_tests") {
     testonly = true
-    visibility += [ "*" ]
+    visibility += webrtc_default_visibility
 
     sources = [
       "test/ACMTest.h",
@@ -1374,7 +1374,7 @@
 
   rtc_source_set("audio_coding_perf_tests") {
     testonly = true
-    visibility += [ "*" ]
+    visibility += webrtc_default_visibility
 
     sources = [
       "codecs/opus/opus_complexity_unittest.cc",
@@ -2053,7 +2053,7 @@
 
   rtc_source_set("audio_coding_unittests") {
     testonly = true
-    visibility += [ "*" ]
+    visibility += webrtc_default_visibility
 
     sources = [
       "acm2/acm_receiver_unittest.cc",
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 3b70c07..eac2dc0 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -46,6 +46,7 @@
 }
 
 rtc_source_set("audio_device") {
+  visibility = [ "*" ]
   public_deps = [
     ":audio_device_generic",
   ]
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index c5724e5..4de77c7 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -26,6 +26,7 @@
 }
 
 rtc_static_library("audio_processing") {
+  visibility = [ "*" ]
   configs += [ ":apm_debug_dump" ]
   sources = [
     "aec/aec_resampler.cc",
@@ -290,6 +291,7 @@
 }
 
 rtc_source_set("audio_processing_statistics") {
+  visibility = [ "*" ]
   sources = [
     "include/audio_processing_statistics.cc",
     "include/audio_processing_statistics.h",
@@ -781,6 +783,7 @@
   }
 
   rtc_source_set("audioproc_test_utils") {
+    visibility = [ "*" ]
     testonly = true
     sources = [
       "test/audio_buffer_tools.cc",
diff --git a/modules/audio_processing/aec_dump/BUILD.gn b/modules/audio_processing/aec_dump/BUILD.gn
index 2e44509..aac4b0a 100644
--- a/modules/audio_processing/aec_dump/BUILD.gn
+++ b/modules/audio_processing/aec_dump/BUILD.gn
@@ -9,6 +9,7 @@
 import("../../../webrtc.gni")  # This contains def of 'rtc_enable_protobuf'
 
 rtc_source_set("aec_dump") {
+  visibility = [ "*" ]
   sources = [
     "aec_dump_factory.h",
   ]
diff --git a/modules/bitrate_controller/BUILD.gn b/modules/bitrate_controller/BUILD.gn
index a791c33..165db62 100644
--- a/modules/bitrate_controller/BUILD.gn
+++ b/modules/bitrate_controller/BUILD.gn
@@ -9,6 +9,7 @@
 import("../../webrtc.gni")
 
 rtc_static_library("bitrate_controller") {
+  visibility = [ "*" ]
   sources = [
     "bitrate_controller_impl.cc",
     "bitrate_controller_impl.h",
diff --git a/modules/congestion_controller/BUILD.gn b/modules/congestion_controller/BUILD.gn
index e2c79a0..46c3047 100644
--- a/modules/congestion_controller/BUILD.gn
+++ b/modules/congestion_controller/BUILD.gn
@@ -17,6 +17,7 @@
 }
 
 rtc_static_library("congestion_controller") {
+  visibility = [ "*" ]
   configs += [ ":bwe_test_logging" ]
   sources = [
     "include/receive_side_congestion_controller.h",
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index fa414b7..04fef21 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -12,6 +12,7 @@
 use_desktop_capture_differ_sse2 = current_cpu == "x86" || current_cpu == "x64"
 
 rtc_static_library("primitives") {
+  visibility = [ "*" ]
   sources = [
     "desktop_capture_types.h",
     "desktop_frame.cc",
@@ -159,6 +160,7 @@
 }
 
 rtc_source_set("desktop_capture") {
+  visibility = [ "*" ]
   public_deps = [
     ":desktop_capture_generic",
   ]
diff --git a/modules/media_file/BUILD.gn b/modules/media_file/BUILD.gn
index 589b281..4fff59d 100644
--- a/modules/media_file/BUILD.gn
+++ b/modules/media_file/BUILD.gn
@@ -13,6 +13,7 @@
 }
 
 rtc_static_library("media_file") {
+  visibility = [ "*" ]
   sources = [
     "media_file.h",
     "media_file_defines.h",
diff --git a/modules/remote_bitrate_estimator/BUILD.gn b/modules/remote_bitrate_estimator/BUILD.gn
index 0ca6b87..c57b498 100644
--- a/modules/remote_bitrate_estimator/BUILD.gn
+++ b/modules/remote_bitrate_estimator/BUILD.gn
@@ -9,6 +9,7 @@
 import("../../webrtc.gni")
 
 rtc_static_library("remote_bitrate_estimator") {
+  visibility = [ "*" ]
   sources = [
     "aimd_rate_control.cc",
     "aimd_rate_control.h",
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index f07ff1a..bf94013 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -97,6 +97,7 @@
 }
 
 rtc_static_library("rtp_rtcp") {
+  visibility = [ "*" ]
   sources = [
     "include/flexfec_receiver.h",
     "include/flexfec_sender.h",
@@ -230,6 +231,7 @@
 }
 
 rtc_source_set("rtcp_transceiver") {
+  visibility = [ "*" ]
   public = [
     "source/rtcp_transceiver.h",
     "source/rtcp_transceiver_config.h",
diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index e3de364..fbfa61e 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -9,6 +9,7 @@
 import("../../webrtc.gni")
 
 rtc_static_library("utility") {
+  visibility = [ "*" ]
   sources = [
     "include/audio_frame_operations.h",
     "include/helpers_android.h",
@@ -45,6 +46,7 @@
 
 rtc_source_set("mock_process_thread") {
   testonly = true
+  visibility = [ "*" ]
   sources = [
     "include/mock/mock_process_thread.h",
   ]
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index 8df5205..6378564 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -47,6 +47,7 @@
 }
 
 rtc_static_library("video_capture") {
+  visibility = [ "*" ]
   sources = [
     "external/device_info_external.cc",
     "external/video_capture_external.cc",
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 2ca7f14..efd0f25 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -9,6 +9,7 @@
 import("../../webrtc.gni")
 
 rtc_static_library("video_coding") {
+  visibility = [ "*" ]
   sources = [
     "codec_database.cc",
     "codec_database.h",
@@ -168,6 +169,7 @@
 }
 
 rtc_source_set("video_coding_utility") {
+  visibility = [ "*" ]
   sources = [
     "codecs/interface/video_codec_interface.h",
     "codecs/interface/video_error_codes.h",
@@ -212,6 +214,7 @@
 }
 
 rtc_static_library("webrtc_h264") {
+  visibility = [ "*" ]
   sources = [
     "codecs/h264/h264.cc",
     "codecs/h264/include/h264.h",
@@ -255,6 +258,7 @@
 }
 
 rtc_static_library("webrtc_i420") {
+  visibility = [ "*" ]
   sources = [
     "codecs/i420/i420.cc",
     "codecs/i420/include/i420.h",
@@ -332,6 +336,7 @@
 
 # This target includes the internal SW codec.
 rtc_static_library("webrtc_vp8") {
+  visibility = [ "*" ]
   sources = [
     "codecs/vp8/default_temporal_layers.cc",
     "codecs/vp8/default_temporal_layers.h",
@@ -376,6 +381,7 @@
 }
 
 rtc_static_library("webrtc_vp9") {
+  visibility = [ "*" ]
   if (rtc_libvpx_build_vp9) {
     sources = [
       "codecs/vp9/include/vp9.h",
diff --git a/modules/video_processing/BUILD.gn b/modules/video_processing/BUILD.gn
index 4fd5e88..351e579 100644
--- a/modules/video_processing/BUILD.gn
+++ b/modules/video_processing/BUILD.gn
@@ -12,6 +12,7 @@
 build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64"
 
 rtc_static_library("video_processing") {
+  visibility = [ "*" ]
   sources = [
     "util/denoiser_filter.cc",
     "util/denoiser_filter_c.cc",