Break out ice_server_parsing from peerconnection target
Drive-by: Add empty dummy targets for all the things left in
the peerconnection target. They should all move out.
Bug: webrtc:13634
Change-Id: I93b193804668decf5feee2a8847403466330e128
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250123
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35870}
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index b5c1553..a399913 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -238,8 +238,6 @@
"data_channel_controller.h",
"data_channel_utils.cc",
"data_channel_utils.h",
- "ice_server_parsing.cc",
- "ice_server_parsing.h",
"jsep_ice_candidate.cc",
"jsep_session_description.cc",
"local_audio_source.cc",
@@ -281,9 +279,13 @@
":dtmf_sender",
":ice_server_parsing",
":jitter_buffer_delay",
+ ":jsep_ice_candidate",
+ ":jsep_session_description",
+ ":local_audio_source",
":media_protocol_names",
":media_stream",
":media_stream_observer",
+ ":peer_connection",
":peer_connection_factory",
":peer_connection_internal",
":peer_connection_message_handler",
@@ -291,6 +293,7 @@
":remote_audio_source",
":rtc_pc_base",
":rtc_stats_collector",
+ ":rtc_stats_traversal",
":rtp_parameters_conversion",
":rtp_receiver",
":rtp_sender",
@@ -298,15 +301,22 @@
":rtp_transmission_manager",
":sctp_data_channel",
":sdp_offer_answer",
+ ":sdp_serializer",
":sdp_state_provider",
+ ":sdp_utils",
":session_description",
":simulcast_description",
+ ":stats_collector",
":stats_collector_interface",
+ ":stream_collection",
+ ":track_media_info_map",
":transceiver_list",
":usage_pattern",
":video_rtp_receiver",
":video_track",
":video_track_source",
+ ":webrtc_sdp",
+ ":webrtc_session_description_factory",
"../api:array_view",
"../api:async_dns_resolver",
"../api:audio_options_api",
@@ -434,10 +444,76 @@
rtc_source_set("sctp_data_channel") {
visibility = [ ":*" ]
}
-rtc_source_set("ice_server_parsing") {
+rtc_source_set("peer_connection_factory") {
visibility = [ "*" ] # Known to be used externally
}
-rtc_source_set("media_stream_observer") {
+rtc_source_set("peer_connection_internal") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("rtc_stats_collector") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("sdp_offer_answer") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("jsep_ice_candidate") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("jsep_session_description") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("local_audio_source") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("peer_connection") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("rtc_stats_traversal") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("sdp_serializer") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("sdp_utils") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("stats_collector") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("stream_collection") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("track_media_info_map") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("webrtc_sdp") {
+ visibility = [ ":*" ]
+}
+rtc_source_set("webrtc_session_description_factory") {
+ visibility = [ ":*" ]
+}
+
+rtc_library("ice_server_parsing") {
+ visibility = [ "*" ] # Known to be used externally
+ sources = [
+ "ice_server_parsing.cc",
+ "ice_server_parsing.h",
+ ]
+ deps = [
+ "../api:libjingle_peerconnection_api",
+ "../api:rtc_error",
+ "../p2p:rtc_p2p",
+ "../rtc_base:checks",
+ "../rtc_base:ip_address",
+ "../rtc_base:logging",
+ "../rtc_base:macromagic",
+ "../rtc_base:socket_address",
+ "../rtc_base:stringutils",
+ "../rtc_base/system:rtc_export",
+ ]
+}
+
+rtc_library("media_stream_observer") {
visibility = [
":*",
"../sdk/android:*",
@@ -452,18 +528,6 @@
]
absl_deps = [ "//third_party/abseil-cpp/absl/algorithm:container" ]
}
-rtc_source_set("peer_connection_factory") {
- visibility = [ "*" ] # Known to be used externally
-}
-rtc_source_set("peer_connection_internal") {
- visibility = [ ":*" ]
-}
-rtc_source_set("rtc_stats_collector") {
- visibility = [ ":*" ]
-}
-rtc_source_set("sdp_offer_answer") {
- visibility = [ "*" ] # Known to be used externally
-}
rtc_library("peer_connection_message_handler") {
sources = [
@@ -1124,6 +1188,7 @@
":audio_rtp_receiver",
":audio_track",
":dtmf_sender",
+ ":ice_server_parsing",
":integration_test_helpers",
":jitter_buffer_delay",
":media_stream",