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}
This commit is contained in:
Harald Alvestrand
WebRTC LUCI CQ
parent
1b083a998b
commit
e47493b3c0
9
BUILD.gn
9
BUILD.gn
@ -12,6 +12,15 @@
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# you add a new build file, there must be some path of dependencies from this
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# file to your new one or GN won't know about it.
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# Use of visibility = clauses:
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# The default visibility for all rtc_ targets is equivalent to "//*", or
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# "all targets in webrtc can depend on this, nothing outside can".
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#
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# When overriding, the choices are:
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# - visibility = [ "*" ] - public. Stuff outside webrtc can use this.
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# - visibility = [ ":*" ] - directory private.
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# As a general guideline, only targets in api/ should have public visibility.
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import("//build/config/linux/pkg_config.gni")
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import("//build/config/sanitizers/sanitizers.gni")
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import("//third_party/google_benchmark/buildconfig.gni")
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85
pc/BUILD.gn
85
pc/BUILD.gn
@ -6,6 +6,30 @@
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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# Visibility considerations:
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#
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# Most targets in this file should have visibility ":*", as they are only
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# used internally.
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# Some functions are cleared for wider webrtc usage; these have default
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# visibility (set to "//*", not the gn default of "*").
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# These are:
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# - rtc_pc_base
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# - rtc_pc
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# - session_description
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# - simulcast_description
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# - peerconnection
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# - sdp_utils
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# - media_stream_observer
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# - video_track_source
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# - libjingle_peerconnection
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#
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# Some targets are depended on by external users for historical reasons,
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# and are therefore marked with visibility "*". This is in the process
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# of being removed.
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#
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# Some targets are only publicly visible in Chrome builds.
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# These are marked up as such.
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import("../webrtc.gni")
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if (is_android) {
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import("//build/config/android/config.gni")
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@ -24,6 +48,7 @@ config("rtc_pc_config") {
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}
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rtc_library("proxy") {
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visibility = [ ":*" ]
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sources = [
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"proxy.cc",
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"proxy.h",
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@ -38,7 +63,8 @@ rtc_library("proxy") {
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}
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rtc_library("rtc_pc_base") {
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visibility = [ "*" ]
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Used by Chromium and others
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defines = []
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sources = [
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"channel.cc",
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@ -176,9 +202,9 @@ rtc_library("rtc_pc_base") {
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}
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rtc_source_set("session_description") {
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visibility = [ "*" ]
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Used by Chrome and others
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sources = [
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"session_description.cc",
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"session_description.h",
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@ -202,9 +228,6 @@ rtc_source_set("session_description") {
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}
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rtc_source_set("simulcast_description") {
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visibility = [ "*" ]
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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sources = [
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"simulcast_description.cc",
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"simulcast_description.h",
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@ -218,7 +241,9 @@ rtc_source_set("simulcast_description") {
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}
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rtc_source_set("rtc_pc") {
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if (build_with_chromium) {
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visibility = [ "*" ]
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}
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allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
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deps = [
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":rtc_pc_base",
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@ -227,6 +252,7 @@ rtc_source_set("rtc_pc") {
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}
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rtc_library("media_protocol_names") {
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visibility = [ ":*" ]
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sources = [
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"media_protocol_names.cc",
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"media_protocol_names.h",
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@ -235,7 +261,8 @@ rtc_library("media_protocol_names") {
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}
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rtc_source_set("peerconnection") {
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visibility = [ "*" ]
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Used by Chromium and others
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cflags = []
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sources = []
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@ -374,6 +401,7 @@ rtc_source_set("peerconnection") {
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}
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rtc_library("sctp_data_channel") {
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visibility = [ ":*" ]
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sources = [
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"sctp_data_channel.cc",
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"sctp_data_channel.h",
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@ -406,9 +434,9 @@ rtc_library("sctp_data_channel") {
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}
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rtc_library("data_channel_utils") {
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Known to be used externally
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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sources = [
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"data_channel_utils.cc",
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"data_channel_utils.h",
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@ -421,6 +449,7 @@ rtc_library("data_channel_utils") {
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}
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rtc_library("connection_context") {
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visibility = [ ":*" ]
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sources = [
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"connection_context.cc",
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"connection_context.h",
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@ -447,10 +476,6 @@ rtc_library("connection_context") {
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]
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}
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# These empty targets are the future homes of files moved out of
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# the peerconnection target. They are broken out so that other
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# projects and modules can add dependencies as needed.
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# TODO(bugs.webrtc.org/13634): Fill these targets with files.
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rtc_source_set("data_channel_controller") {
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visibility = [ ":*" ]
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sources = [
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@ -745,10 +770,6 @@ rtc_source_set("sdp_serializer") {
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]
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}
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rtc_source_set("sdp_utils") {
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visibility = [
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":*",
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"../test/pc/e2e:*",
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]
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sources = [
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"sdp_utils.cc",
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"sdp_utils.h",
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@ -824,9 +845,9 @@ rtc_source_set("track_media_info_map") {
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absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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}
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rtc_source_set("webrtc_sdp") {
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Used by Chrome and more
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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sources = [
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"jsep_ice_candidate.cc",
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"jsep_session_description.cc",
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@ -891,9 +912,9 @@ rtc_source_set("webrtc_session_description_factory") {
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}
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rtc_library("ice_server_parsing") {
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Known to be used externally
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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sources = [
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"ice_server_parsing.cc",
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"ice_server_parsing.h",
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@ -913,10 +934,6 @@ rtc_library("ice_server_parsing") {
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}
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rtc_library("media_stream_observer") {
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visibility = [
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":*",
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"../sdk/android:*",
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]
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sources = [
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"media_stream_observer.cc",
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"media_stream_observer.h",
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@ -928,6 +945,7 @@ rtc_library("media_stream_observer") {
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absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
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}
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rtc_source_set("peer_connection_factory") {
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Known to be used externally
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sources = [
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"peer_connection_factory.cc",
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@ -982,6 +1000,7 @@ rtc_source_set("peer_connection_factory") {
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}
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rtc_library("peer_connection_message_handler") {
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visibility = [ ":*" ]
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sources = [
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"peer_connection_message_handler.cc",
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"peer_connection_message_handler.h",
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@ -1000,6 +1019,7 @@ rtc_library("peer_connection_message_handler") {
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}
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rtc_library("usage_pattern") {
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visibility = [ ":*" ]
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sources = [
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"usage_pattern.cc",
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"usage_pattern.h",
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@ -1012,6 +1032,7 @@ rtc_library("usage_pattern") {
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}
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rtc_library("rtp_transceiver") {
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visibility = [ ":*" ]
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sources = [
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"rtp_transceiver.cc",
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"rtp_transceiver.h",
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@ -1049,6 +1070,7 @@ rtc_library("rtp_transceiver") {
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}
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rtc_library("rtp_transmission_manager") {
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visibility = [ ":*" ]
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sources = [
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"rtp_transmission_manager.cc",
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"rtp_transmission_manager.h",
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@ -1085,6 +1107,7 @@ rtc_library("rtp_transmission_manager") {
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}
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rtc_library("transceiver_list") {
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visibility = [ ":*" ]
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sources = [
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"transceiver_list.cc",
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"transceiver_list.h",
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@ -1104,6 +1127,7 @@ rtc_library("transceiver_list") {
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}
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rtc_library("rtp_receiver") {
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visibility = [ ":*" ]
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sources = [
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"rtp_receiver.cc",
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"rtp_receiver.h",
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@ -1133,6 +1157,7 @@ rtc_library("rtp_receiver") {
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}
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rtc_library("audio_rtp_receiver") {
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visibility = [ ":*" ]
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sources = [
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"audio_rtp_receiver.cc",
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"audio_rtp_receiver.h",
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@ -1169,6 +1194,7 @@ rtc_library("audio_rtp_receiver") {
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}
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rtc_library("video_rtp_receiver") {
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visibility = [ ":*" ]
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sources = [
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"video_rtp_receiver.cc",
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"video_rtp_receiver.h",
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@ -1205,6 +1231,7 @@ rtc_library("video_rtp_receiver") {
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}
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rtc_library("video_rtp_track_source") {
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visibility = [ ":*" ]
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sources = [
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"video_rtp_track_source.cc",
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"video_rtp_track_source.h",
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@ -1223,6 +1250,7 @@ rtc_library("video_rtp_track_source") {
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}
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rtc_library("audio_track") {
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visibility = [ ":*" ]
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sources = [
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"audio_track.cc",
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"audio_track.h",
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@ -1237,6 +1265,7 @@ rtc_library("audio_track") {
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}
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rtc_library("video_track") {
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visibility = [ ":*" ]
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sources = [
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"video_track.cc",
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"video_track.h",
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@ -1256,6 +1285,7 @@ rtc_library("video_track") {
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}
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rtc_source_set("sdp_state_provider") {
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visibility = [ ":*" ]
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sources = [ "sdp_state_provider.h" ]
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deps = [
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":rtc_pc_base",
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@ -1264,6 +1294,7 @@ rtc_source_set("sdp_state_provider") {
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}
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rtc_library("jitter_buffer_delay") {
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visibility = [ ":*" ]
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sources = [
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"jitter_buffer_delay.cc",
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"jitter_buffer_delay.h",
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@ -1279,6 +1310,7 @@ rtc_library("jitter_buffer_delay") {
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}
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rtc_library("remote_audio_source") {
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visibility = [ ":*" ]
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sources = [
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"remote_audio_source.cc",
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"remote_audio_source.h",
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@ -1307,6 +1339,7 @@ rtc_library("remote_audio_source") {
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}
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rtc_library("rtp_sender") {
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visibility = [ ":*" ]
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sources = [
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"rtp_sender.cc",
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"rtp_sender.h",
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@ -1338,6 +1371,7 @@ rtc_library("rtp_sender") {
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}
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rtc_library("rtp_parameters_conversion") {
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visibility = [ ":*" ]
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sources = [
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"rtp_parameters_conversion.cc",
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"rtp_parameters_conversion.h",
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@ -1361,6 +1395,7 @@ rtc_library("rtp_parameters_conversion") {
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}
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rtc_library("dtmf_sender") {
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visibility = [ ":*" ]
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sources = [
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"dtmf_sender.cc",
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"dtmf_sender.h",
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@ -1384,6 +1419,7 @@ rtc_library("dtmf_sender") {
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}
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rtc_library("media_stream") {
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visibility = [ ":*" ]
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sources = [
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"media_stream.cc",
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"media_stream.h",
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@ -1423,6 +1459,7 @@ rtc_library("video_track_source") {
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}
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rtc_source_set("stats_collector_interface") {
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visibility = [ ":*" ]
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sources = [ "stats_collector_interface.h" ]
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deps = [
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"../api:libjingle_peerconnection_api",
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@ -1431,9 +1468,9 @@ rtc_source_set("stats_collector_interface") {
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}
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rtc_source_set("libjingle_peerconnection") {
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visibility = [ "*" ]
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# TODO(bugs.webrtc.org/13661): Reduce visibility if possible
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visibility = [ "*" ] # Used by Chrome and others
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deps = [
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":peerconnection",
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"../api:libjingle_peerconnection_api",
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