Break out pc/session_description in its own build target (part 1)

As a side effect, break out pc/simulcast_description.

Step 1: Don't move the {h,cc} files; just declare the targets
so that downstream projects can add dependencies on it.

Bug: webtc:11967
Change-Id: Iad3d77513af418b664c1bef46070177ed24027fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221603
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34254}

pc/BUILD.gn

@@ -81,8 +81,14 @@ rtc_library("rtc_pc_base") {
     "sctp_transport.h",
     "sctp_utils.cc",
     "sctp_utils.h",
+
+    # Will move to "session_description" target
+    # when downstream projects depend on that.
     "session_description.cc",
     "session_description.h",
+
+    # Will move to "simulcast_description" target
+    # when downstream projects depend on that.
     "simulcast_description.cc",
     "simulcast_description.h",
     "srtp_filter.cc",
@@ -101,6 +107,8 @@ rtc_library("rtc_pc_base") {
   deps = [
     ":media_protocol_names",
     ":proxy",
+    ":session_description",
+    ":simulcast_description",
     "../api:array_view",
     "../api:async_dns_resolver",
     "../api:audio_options_api",
@@ -176,6 +184,43 @@ rtc_library("rtc_pc_base") {
   public_configs = [ ":rtc_pc_config" ]
 }
 
+rtc_source_set("session_description") {
+  sources = [
+    # Future home of
+    #"session_description.cc",
+    #"session_description.h",
+  ]
+  deps = [
+    ":media_protocol_names",
+    ":simulcast_description",
+    "../api:libjingle_peerconnection_api",
+    "../api:rtp_parameters",
+    "../api:rtp_transceiver_direction",
+    "../media:rtc_media_base",
+    "../p2p:rtc_p2p",
+    "../rtc_base:checks",
+    "../rtc_base:socket_address",
+    "../rtc_base/system:rtc_export",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/algorithm:container",
+    "//third_party/abseil-cpp/absl/memory:memory",
+  ]
+}
+
+rtc_source_set("simulcast_description") {
+  sources = [
+    # Future home of
+    # "simulcast_description.cc",
+    # "simulcast_description.h",
+  ]
+  deps = [
+    "../rtc_base:checks",
+    "../rtc_base:socket_address",
+    "../rtc_base/system:rtc_export",
+  ]
+}
+
 rtc_source_set("rtc_pc") {
   visibility = [ "*" ]
   allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
@@ -254,6 +299,8 @@ rtc_library("peerconnection") {
     ":rtp_transceiver",
     ":rtp_transmission_manager",
     ":sdp_state_provider",
+    ":session_description",
+    ":simulcast_description",
     ":stats_collector_interface",
     ":transceiver_list",
     ":usage_pattern",
@@ -414,6 +461,7 @@ rtc_library("rtp_transceiver") {
     ":rtp_parameters_conversion",
     ":rtp_receiver",
     ":rtp_sender",
+    ":session_description",
     "../api:array_view",
     "../api:libjingle_peerconnection_api",
     "../api:rtc_error",
@@ -735,6 +783,7 @@ rtc_library("rtp_parameters_conversion") {
   ]
   deps = [
     ":rtc_pc_base",
+    ":session_description",
     "../api:array_view",
     "../api:libjingle_peerconnection_api",
     "../api:rtc_error",
@@ -865,6 +914,7 @@ if (rtc_include_tests && !build_with_chromium) {
       ":peerconnection",
       ":rtc_pc",
       ":rtc_pc_base",
+      ":session_description",
       ":video_rtp_receiver",
       "../api:array_view",
       "../api:audio_options_api",
@@ -1036,6 +1086,7 @@ if (rtc_include_tests && !build_with_chromium) {
       ":rtp_receiver",
       ":rtp_sender",
       ":rtp_transceiver",
+      ":session_description",
       ":usage_pattern",
       ":video_rtp_receiver",
       ":video_rtp_track_source",
@@ -1195,6 +1246,7 @@ if (rtc_include_tests && !build_with_chromium) {
       ":rtp_receiver",
       ":rtp_sender",
       ":rtp_transceiver",
+      ":session_description",
       ":usage_pattern",
       ":video_rtp_receiver",
       ":video_rtp_track_source",

test/pc/e2e/BUILD.gn

@@ -289,6 +289,7 @@ if (!build_with_chromium) {
         "../../../api:peer_connection_quality_test_fixture_api",
         "../../../api/video:video_frame",
         "../../../pc:peerconnection",
+        "../../../pc:session_description",
         "../../../pc:video_track_source",
       ]
       absl_deps = [ "//third_party/abseil-cpp/absl/types:variant" ]
@@ -737,6 +738,8 @@ if (!build_with_chromium) {
       "../../../p2p:rtc_p2p",
       "../../../pc:peerconnection",
       "../../../pc:rtc_pc_base",
+      "../../../pc:session_description",
+      "../../../pc:simulcast_description",
       "../../../rtc_base:stringutils",
     ]
     absl_deps = [

test/peer_scenario/BUILD.gn

@@ -47,6 +47,7 @@ if (rtc_include_tests) {
       "../../p2p:rtc_p2p",
       "../../pc:pc_test_utils",
       "../../pc:rtc_pc_base",
+      "../../pc:session_description",
       "../../rtc_base",
       "../../rtc_base:null_socket_server",
       "../../rtc_base:stringutils",

test/peer_scenario/tests/BUILD.gn

@@ -25,6 +25,7 @@ if (rtc_include_tests) {
       "../../../modules/rtp_rtcp:rtp_rtcp",
       "../../../modules/rtp_rtcp:rtp_rtcp_format",
       "../../../pc:rtc_pc_base",
+      "../../../pc:session_description",
     ]
   }
 }