sctp: Reorganize build targets

Bug: webrtc:12614
Change-Id: I2d276139746bb8cafdd5c50fe4595e60a6b1c7fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215234
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33745}

media/BUILD.gn

@@ -392,15 +392,29 @@ rtc_library("rtc_media_engine_defaults") {
   ]
 }
 
-rtc_library("rtc_data") {
+rtc_source_set("rtc_data_sctp_transport_internal") {
+  sources = [ "sctp/sctp_transport_internal.h" ]
+  deps = [
+    "../media:rtc_media_base",
+    "../p2p:rtc_p2p",
+    "../rtc_base:rtc_base_approved",
+    "../rtc_base:threading",
+    "../rtc_base/third_party/sigslot",
+  ]
+}
+
+if (rtc_build_usrsctp) {
+  rtc_library("rtc_data_usrsctp_transport") {
     defines = [
       # "SCTP_DEBUG" # Uncomment for SCTP debugging.
     ]
+    sources = [
+      "sctp/usrsctp_transport.cc",
+      "sctp/usrsctp_transport.h",
+    ]
     deps = [
-    ":rtc_media_base",
+      ":rtc_data_sctp_transport_internal",
-    "../api:call_api",
+      "../media:rtc_media_base",
-    "../api:sequence_checker",
-    "../api:transport_api",
       "../p2p:rtc_p2p",
       "../rtc_base",
       "../rtc_base:rtc_base_approved",
@@ -408,43 +422,43 @@ rtc_library("rtc_data") {
       "../rtc_base/synchronization:mutex",
       "../rtc_base/task_utils:pending_task_safety_flag",
       "../rtc_base/task_utils:to_queued_task",
-    "../rtc_base/third_party/sigslot",
+      "../rtc_base/third_party/sigslot:sigslot",
-    "../system_wrappers",
+      "//third_party/usrsctp",
     ]
     absl_deps = [
       "//third_party/abseil-cpp/absl/algorithm:container",
-    "//third_party/abseil-cpp/absl/base:core_headers",
       "//third_party/abseil-cpp/absl/types:optional",
     ]
+  }
+}
 
-  if (rtc_enable_sctp) {
+rtc_library("rtc_data_sctp_transport_factory") {
+  defines = []
   sources = [
     "sctp/sctp_transport_factory.cc",
     "sctp/sctp_transport_factory.h",
-      "sctp/sctp_transport_internal.h",
-      "sctp/usrsctp_transport.cc",
-      "sctp/usrsctp_transport.h",
   ]
-  } else {
+  deps = [
-    # libtool on mac does not like empty targets.
+    ":rtc_data_sctp_transport_internal",
-    sources = [ "sctp/noop.cc" ]
+    "../api/transport:sctp_transport_factory_interface",
+    "../rtc_base:threading",
+    "../rtc_base/system:unused",
+  ]
+
+  if (rtc_enable_sctp) {
+    assert(rtc_build_usrsctp, "An SCTP backend is required to enable SCTP")
   }
 
-  if (rtc_enable_sctp && rtc_build_usrsctp) {
+  if (rtc_build_usrsctp) {
-    deps += [
+    defines += [ "WEBRTC_HAVE_USRSCTP" ]
-      "../api/transport:sctp_transport_factory_interface",
+    deps += [ ":rtc_data_usrsctp_transport" ]
-      "//third_party/usrsctp",
-    ]
   }
 }
 
 rtc_source_set("rtc_media") {
   visibility = [ "*" ]
   allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
-  deps = [
+  deps = [ ":rtc_audio_video" ]
-    ":rtc_audio_video",
-    ":rtc_data",
-  ]
 }
 
 if (rtc_include_tests) {
@@ -537,7 +551,6 @@ if (rtc_include_tests) {
       defines = []
       deps = [
         ":rtc_audio_video",
-        ":rtc_data",
         ":rtc_encoder_simulcast_proxy",
         ":rtc_internal_video_codecs",
         ":rtc_media",
@@ -641,15 +654,18 @@ if (rtc_include_tests) {
         sources += [ "engine/webrtc_voice_engine_unittest.cc" ]
       }
 
-      if (rtc_enable_sctp) {
+      if (rtc_build_usrsctp) {
         sources += [
           "sctp/usrsctp_transport_reliability_unittest.cc",
           "sctp/usrsctp_transport_unittest.cc",
         ]
         deps += [
+          ":rtc_data_sctp_transport_internal",
+          ":rtc_data_usrsctp_transport",
           "../rtc_base:rtc_event",
           "../rtc_base/task_utils:pending_task_safety_flag",
           "../rtc_base/task_utils:to_queued_task",
+          "//third_party/usrsctp",
         ]
       }
 
@@ -669,10 +685,6 @@ if (rtc_include_tests) {
       if (is_ios) {
         deps += [ ":rtc_media_unittests_bundle_data" ]
       }
-
-      if (rtc_enable_sctp && rtc_build_usrsctp) {
-        deps += [ "//third_party/usrsctp" ]
-      }
     }
   }
 }

media/sctp/noop.cc

@@ -1,13 +0,0 @@
-/*
- *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file is only needed to make ninja happy on some platforms.
-// On some platforms it is not possible to link an rtc_static_library
-// without any source file listed in the GN target.

media/sctp/sctp_transport_factory.cc

@@ -10,16 +10,30 @@
 
 #include "media/sctp/sctp_transport_factory.h"
 
+#include "rtc_base/system/unused.h"
+
+#ifdef WEBRTC_HAVE_USRSCTP
+#include "media/sctp/usrsctp_transport.h"  // nogncheck
+#endif
+
 namespace cricket {
 
 SctpTransportFactory::SctpTransportFactory(rtc::Thread* network_thread)
-    : network_thread_(network_thread) {}
+    : network_thread_(network_thread) {
+  RTC_UNUSED(network_thread_);
+}
 
 std::unique_ptr<SctpTransportInternal>
 SctpTransportFactory::CreateSctpTransport(
     rtc::PacketTransportInternal* transport) {
-  return std::unique_ptr<SctpTransportInternal>(
+  std::unique_ptr<SctpTransportInternal> result;
+#ifdef WEBRTC_HAVE_USRSCTP
+  if (!result) {
+    result = std::unique_ptr<SctpTransportInternal>(
         new UsrsctpTransport(network_thread_, transport));
   }
+#endif
+  return result;
+}
 
 }  // namespace cricket

media/sctp/sctp_transport_factory.h

@@ -14,7 +14,7 @@
 #include <memory>
 
 #include "api/transport/sctp_transport_factory_interface.h"
-#include "media/sctp/usrsctp_transport.h"
+#include "media/sctp/sctp_transport_internal.h"
 #include "rtc_base/thread.h"
 
 namespace cricket {

media/sctp/usrsctp_transport.h

@@ -21,7 +21,6 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "api/transport/sctp_transport_factory_interface.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
 #include "rtc_base/copy_on_write_buffer.h"

pc/BUILD.gn

@@ -108,7 +108,7 @@ rtc_library("rtc_pc_base") {
     "../common_video",
     "../common_video:common_video",
     "../logging:ice_log",
-    "../media:rtc_data",
+    "../media:rtc_data_sctp_transport_internal",
     "../media:rtc_h264_profile_id",
     "../media:rtc_media_base",
     "../media:rtc_media_config",
@@ -281,7 +281,7 @@ rtc_library("peerconnection") {
     "../call:call_interfaces",
     "../common_video",
     "../logging:ice_log",
-    "../media:rtc_data",
+    "../media:rtc_data_sctp_transport_internal",
     "../media:rtc_media_base",
     "../media:rtc_media_config",
     "../modules/audio_processing:audio_processing_statistics",
@@ -336,7 +336,7 @@ rtc_library("connection_context") {
     "../api/transport:field_trial_based_config",
     "../api/transport:sctp_transport_factory_interface",
     "../api/transport:webrtc_key_value_config",
-    "../media:rtc_data",
+    "../media:rtc_data_sctp_transport_factory",
     "../media:rtc_media_base",
     "../p2p:rtc_p2p",
     "../rtc_base",
@@ -869,7 +869,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/video/test:mock_recordable_encoded_frame",
       "../call:rtp_interfaces",
       "../call:rtp_receiver",
-      "../media:rtc_data",
+      "../media:rtc_data_sctp_transport_internal",
       "../media:rtc_media_base",
       "../media:rtc_media_tests_utils",
       "../modules/rtp_rtcp:rtp_rtcp_format",
@@ -1011,10 +1011,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "webrtc_sdp_unittest.cc",
     ]
 
-    if (rtc_enable_sctp) {
-      defines = [ "WEBRTC_HAVE_SCTP" ]
-    }
-
     deps = [
       ":audio_rtp_receiver",
       ":audio_track",
@@ -1065,6 +1061,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/video:video_rtp_headers",
       "../call/adaptation:resource_adaptation_test_utilities",
       "../logging:fake_rtc_event_log",
+      "../media:rtc_data_sctp_transport_internal",
       "../media:rtc_media_config",
       "../media:rtc_media_engine_defaults",
       "../modules/audio_device:audio_device_api",
@@ -1118,8 +1115,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
       "../media:rtc_audio_video",
-      "../media:rtc_data",  # TODO(phoglund): AFAIK only used for one sctp
-                            # constant.
       "../media:rtc_media_base",
       "../media:rtc_media_tests_utils",
       "../modules/audio_processing",
@@ -1328,7 +1323,6 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/video_codecs:builtin_video_encoder_factory",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
-      "../media:rtc_data",
       "../media:rtc_media",
       "../media:rtc_media_base",
       "../media:rtc_media_tests_utils",

pc/connection_context.h

@@ -22,7 +22,6 @@
 #include "api/transport/sctp_transport_factory_interface.h"
 #include "api/transport/webrtc_key_value_config.h"
 #include "media/base/media_engine.h"
-#include "media/sctp/sctp_transport_internal.h"
 #include "p2p/base/basic_packet_socket_factory.h"
 #include "pc/channel_manager.h"
 #include "rtc_base/checks.h"

pc/peer_connection_factory.h

@@ -37,7 +37,6 @@
 #include "api/transport/sctp_transport_factory_interface.h"
 #include "api/transport/webrtc_key_value_config.h"
 #include "call/call.h"
-#include "media/sctp/sctp_transport_internal.h"
 #include "p2p/base/port_allocator.h"
 #include "pc/channel_manager.h"
 #include "pc/connection_context.h"

sdk/android/BUILD.gn

@@ -777,7 +777,6 @@ if (current_os == "linux" || is_android) {
       "../../api/video_codecs:video_codecs_api",
       "../../call:call_interfaces",
       "../../media:rtc_audio_video",
-      "../../media:rtc_data",
       "../../media:rtc_media_base",
       "../../modules/audio_device",
       "../../modules/audio_processing:api",

test/pc/sctp/BUILD.gn

@@ -11,5 +11,5 @@ import("../../../webrtc.gni")
 rtc_source_set("fake_sctp_transport") {
   visibility = [ "*" ]
   sources = [ "fake_sctp_transport.h" ]
-  deps = [ "../../../media:rtc_data" ]
+  deps = [ "../../../media:rtc_data_sctp_transport_internal" ]
 }

webrtc.gni

@@ -233,7 +233,6 @@ declare_args() {
   rtc_libvpx_build_vp9 = !build_with_mozilla
   rtc_build_opus = !build_with_mozilla
   rtc_build_ssl = !build_with_mozilla
-  rtc_build_usrsctp = !build_with_mozilla
 
   # Enable libevent task queues on platforms that support it.
   if (is_win || is_mac || is_ios || is_nacl || is_fuchsia ||
@@ -290,6 +289,11 @@ declare_args() {
   rtc_exclude_transient_suppressor = false
 }
 
+declare_args() {
+  # Enable the usrsctp backend for DataChannels and related unittests
+  rtc_build_usrsctp = !build_with_mozilla && rtc_enable_sctp
+}
+
 # Make it possible to provide custom locations for some libraries (move these
 # up into declare_args should we need to actually use them for the GN build).
 rtc_libvpx_dir = "//third_party/libvpx"