Add restrictive visibility to all targets in //pc

This CL sets all visibility to ":*" (this buildfile) where no users
outside this directory are known, and marks up publicly exported
targets and Chrome dependencies explicitly.

Bug: webrtc:13661
Change-Id: I9b2c304ea222f401d71a1ec86eb7a052051f0be3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251690
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36004}

BUILD.gn

@@ -12,6 +12,15 @@
 # you add a new build file, there must be some path of dependencies from this
 # file to your new one or GN won't know about it.
 
+# Use of visibility = clauses:
+# The default visibility for all rtc_ targets is equivalent to "//*", or
+# "all targets in webrtc can depend on this, nothing outside can".
+#
+# When overriding, the choices are:
+# - visibility = [ "*" ] - public. Stuff outside webrtc can use this.
+# - visibility = [ ":*" ] - directory private.
+# As a general guideline, only targets in api/ should have public visibility.
+
 import("//build/config/linux/pkg_config.gni")
 import("//build/config/sanitizers/sanitizers.gni")
 import("//third_party/google_benchmark/buildconfig.gni")

pc/BUILD.gn

@@ -6,6 +6,30 @@
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
+# Visibility considerations:
+#
+# Most targets in this file should have visibility ":*", as they are only
+# used internally.
+# Some functions are cleared for wider webrtc usage; these have default
+# visibility (set to "//*", not the gn default of "*").
+# These are:
+# - rtc_pc_base
+# - rtc_pc
+# - session_description
+# - simulcast_description
+# - peerconnection
+# - sdp_utils
+# - media_stream_observer
+# - video_track_source
+# - libjingle_peerconnection
+#
+# Some targets are depended on by external users for historical reasons,
+# and are therefore marked with visibility "*". This is in the process
+# of being removed.
+#
+# Some targets are only publicly visible in Chrome builds.
+# These are marked up as such.
+
 import("../webrtc.gni")
 if (is_android) {
   import("//build/config/android/config.gni")
@@ -24,6 +48,7 @@ config("rtc_pc_config") {
 }
 
 rtc_library("proxy") {
+  visibility = [ ":*" ]
   sources = [
     "proxy.cc",
     "proxy.h",
@@ -38,7 +63,8 @@ rtc_library("proxy") {
 }
 
 rtc_library("rtc_pc_base") {
-  visibility = [ "*" ]
+  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
+  visibility = [ "*" ]  # Used by Chromium and others
   defines = []
   sources = [
     "channel.cc",
@@ -176,9 +202,9 @@ rtc_library("rtc_pc_base") {
 }
 
 rtc_source_set("session_description") {
-  visibility = [ "*" ]
-
   # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
+  visibility = [ "*" ]  # Used by Chrome and others
+
   sources = [
     "session_description.cc",
     "session_description.h",
@@ -202,9 +228,6 @@ rtc_source_set("session_description") {
 }
 
 rtc_source_set("simulcast_description") {
-  visibility = [ "*" ]
-
-  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   sources = [
     "simulcast_description.cc",
     "simulcast_description.h",
@@ -218,7 +241,9 @@ rtc_source_set("simulcast_description") {
 }
 
 rtc_source_set("rtc_pc") {
-  visibility = [ "*" ]
+  if (build_with_chromium) {
+    visibility = [ "*" ]
+  }
   allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
   deps = [
     ":rtc_pc_base",
@@ -227,6 +252,7 @@ rtc_source_set("rtc_pc") {
 }
 
 rtc_library("media_protocol_names") {
+  visibility = [ ":*" ]
   sources = [
     "media_protocol_names.cc",
     "media_protocol_names.h",
@@ -235,7 +261,8 @@ rtc_library("media_protocol_names") {
 }
 
 rtc_source_set("peerconnection") {
-  visibility = [ "*" ]
+  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
+  visibility = [ "*" ]  # Used by Chromium and others
   cflags = []
   sources = []
 
@@ -374,6 +401,7 @@ rtc_source_set("peerconnection") {
 }
 
 rtc_library("sctp_data_channel") {
+  visibility = [ ":*" ]
   sources = [
     "sctp_data_channel.cc",
     "sctp_data_channel.h",
@@ -406,9 +434,9 @@ rtc_library("sctp_data_channel") {
 }
 
 rtc_library("data_channel_utils") {
+  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   visibility = [ "*" ]  # Known to be used externally
 
-  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   sources = [
     "data_channel_utils.cc",
     "data_channel_utils.h",
@@ -421,6 +449,7 @@ rtc_library("data_channel_utils") {
 }
 
 rtc_library("connection_context") {
+  visibility = [ ":*" ]
   sources = [
     "connection_context.cc",
     "connection_context.h",
@@ -447,10 +476,6 @@ rtc_library("connection_context") {
   ]
 }
 
-# These empty targets are the future homes of files moved out of
-# the peerconnection target. They are broken out so that other
-# projects and modules can add dependencies as needed.
-# TODO(bugs.webrtc.org/13634): Fill these targets with files.
 rtc_source_set("data_channel_controller") {
   visibility = [ ":*" ]
   sources = [
@@ -745,10 +770,6 @@ rtc_source_set("sdp_serializer") {
   ]
 }
 rtc_source_set("sdp_utils") {
-  visibility = [
-    ":*",
-    "../test/pc/e2e:*",
-  ]
   sources = [
     "sdp_utils.cc",
     "sdp_utils.h",
@@ -824,9 +845,9 @@ rtc_source_set("track_media_info_map") {
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 rtc_source_set("webrtc_sdp") {
+  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   visibility = [ "*" ]  # Used by Chrome and more
 
-  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   sources = [
     "jsep_ice_candidate.cc",
     "jsep_session_description.cc",
@@ -891,9 +912,9 @@ rtc_source_set("webrtc_session_description_factory") {
 }
 
 rtc_library("ice_server_parsing") {
+  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   visibility = [ "*" ]  # Known to be used externally
 
-  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   sources = [
     "ice_server_parsing.cc",
     "ice_server_parsing.h",
@@ -913,10 +934,6 @@ rtc_library("ice_server_parsing") {
 }
 
 rtc_library("media_stream_observer") {
-  visibility = [
-    ":*",
-    "../sdk/android:*",
-  ]
   sources = [
     "media_stream_observer.cc",
     "media_stream_observer.h",
@@ -928,6 +945,7 @@ rtc_library("media_stream_observer") {
   absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
 }
 rtc_source_set("peer_connection_factory") {
+  # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
   visibility = [ "*" ]  # Known to be used externally
   sources = [
     "peer_connection_factory.cc",
@@ -982,6 +1000,7 @@ rtc_source_set("peer_connection_factory") {
 }
 
 rtc_library("peer_connection_message_handler") {
+  visibility = [ ":*" ]
   sources = [
     "peer_connection_message_handler.cc",
     "peer_connection_message_handler.h",
@@ -1000,6 +1019,7 @@ rtc_library("peer_connection_message_handler") {
 }
 
 rtc_library("usage_pattern") {
+  visibility = [ ":*" ]
   sources = [
     "usage_pattern.cc",
     "usage_pattern.h",
@@ -1012,6 +1032,7 @@ rtc_library("usage_pattern") {
 }
 
 rtc_library("rtp_transceiver") {
+  visibility = [ ":*" ]
   sources = [
     "rtp_transceiver.cc",
     "rtp_transceiver.h",
@@ -1049,6 +1070,7 @@ rtc_library("rtp_transceiver") {
 }
 
 rtc_library("rtp_transmission_manager") {
+  visibility = [ ":*" ]
   sources = [
     "rtp_transmission_manager.cc",
     "rtp_transmission_manager.h",
@@ -1085,6 +1107,7 @@ rtc_library("rtp_transmission_manager") {
 }
 
 rtc_library("transceiver_list") {
+  visibility = [ ":*" ]
   sources = [
     "transceiver_list.cc",
     "transceiver_list.h",
@@ -1104,6 +1127,7 @@ rtc_library("transceiver_list") {
 }
 
 rtc_library("rtp_receiver") {
+  visibility = [ ":*" ]
   sources = [
     "rtp_receiver.cc",
     "rtp_receiver.h",
@@ -1133,6 +1157,7 @@ rtc_library("rtp_receiver") {
 }
 
 rtc_library("audio_rtp_receiver") {
+  visibility = [ ":*" ]
   sources = [
     "audio_rtp_receiver.cc",
     "audio_rtp_receiver.h",
@@ -1169,6 +1194,7 @@ rtc_library("audio_rtp_receiver") {
 }
 
 rtc_library("video_rtp_receiver") {
+  visibility = [ ":*" ]
   sources = [
     "video_rtp_receiver.cc",
     "video_rtp_receiver.h",
@@ -1205,6 +1231,7 @@ rtc_library("video_rtp_receiver") {
 }
 
 rtc_library("video_rtp_track_source") {
+  visibility = [ ":*" ]
   sources = [
     "video_rtp_track_source.cc",
     "video_rtp_track_source.h",
@@ -1223,6 +1250,7 @@ rtc_library("video_rtp_track_source") {
 }
 
 rtc_library("audio_track") {
+  visibility = [ ":*" ]
   sources = [
     "audio_track.cc",
     "audio_track.h",
@@ -1237,6 +1265,7 @@ rtc_library("audio_track") {
 }
 
 rtc_library("video_track") {
+  visibility = [ ":*" ]
   sources = [
     "video_track.cc",
     "video_track.h",
@@ -1256,6 +1285,7 @@ rtc_library("video_track") {
 }
 
 rtc_source_set("sdp_state_provider") {
+  visibility = [ ":*" ]
   sources = [ "sdp_state_provider.h" ]
   deps = [
     ":rtc_pc_base",
@@ -1264,6 +1294,7 @@ rtc_source_set("sdp_state_provider") {
 }
 
 rtc_library("jitter_buffer_delay") {
+  visibility = [ ":*" ]
   sources = [
     "jitter_buffer_delay.cc",
     "jitter_buffer_delay.h",
@@ -1279,6 +1310,7 @@ rtc_library("jitter_buffer_delay") {
 }
 
 rtc_library("remote_audio_source") {
+  visibility = [ ":*" ]
   sources = [
     "remote_audio_source.cc",
     "remote_audio_source.h",
@@ -1307,6 +1339,7 @@ rtc_library("remote_audio_source") {
 }
 
 rtc_library("rtp_sender") {
+  visibility = [ ":*" ]
   sources = [
     "rtp_sender.cc",
     "rtp_sender.h",
@@ -1338,6 +1371,7 @@ rtc_library("rtp_sender") {
 }
 
 rtc_library("rtp_parameters_conversion") {
+  visibility = [ ":*" ]
   sources = [
     "rtp_parameters_conversion.cc",
     "rtp_parameters_conversion.h",
@@ -1361,6 +1395,7 @@ rtc_library("rtp_parameters_conversion") {
 }
 
 rtc_library("dtmf_sender") {
+  visibility = [ ":*" ]
   sources = [
     "dtmf_sender.cc",
     "dtmf_sender.h",
@@ -1384,6 +1419,7 @@ rtc_library("dtmf_sender") {
 }
 
 rtc_library("media_stream") {
+  visibility = [ ":*" ]
   sources = [
     "media_stream.cc",
     "media_stream.h",
@@ -1423,6 +1459,7 @@ rtc_library("video_track_source") {
 }
 
 rtc_source_set("stats_collector_interface") {
+  visibility = [ ":*" ]
   sources = [ "stats_collector_interface.h" ]
   deps = [
     "../api:libjingle_peerconnection_api",
@@ -1431,9 +1468,9 @@ rtc_source_set("stats_collector_interface") {
 }
 
 rtc_source_set("libjingle_peerconnection") {
-  visibility = [ "*" ]
-
   # TODO(bugs.webrtc.org/13661): Reduce visibility if possible
+  visibility = [ "*" ]  # Used by Chrome and others
+
   deps = [
     ":peerconnection",
     "../api:libjingle_peerconnection_api",