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}
diff --git a/media/BUILD.gn b/media/BUILD.gn
index af59b59..eedf96f 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -392,59 +392,73 @@
]
}
-rtc_library("rtc_data") {
- defines = [
- # "SCTP_DEBUG" # Uncomment for SCTP debugging.
- ]
+rtc_source_set("rtc_data_sctp_transport_internal") {
+ sources = [ "sctp/sctp_transport_internal.h" ]
deps = [
- ":rtc_media_base",
- "../api:call_api",
- "../api:sequence_checker",
- "../api:transport_api",
+ "../media:rtc_media_base",
"../p2p:rtc_p2p",
- "../rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:threading",
- "../rtc_base/synchronization:mutex",
- "../rtc_base/task_utils:pending_task_safety_flag",
- "../rtc_base/task_utils:to_queued_task",
"../rtc_base/third_party/sigslot",
- "../system_wrappers",
]
- 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) {
+if (rtc_build_usrsctp) {
+ rtc_library("rtc_data_usrsctp_transport") {
+ defines = [
+ # "SCTP_DEBUG" # Uncomment for SCTP debugging.
+ ]
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 {
- # libtool on mac does not like empty targets.
- sources = [ "sctp/noop.cc" ]
- }
-
- if (rtc_enable_sctp && rtc_build_usrsctp) {
- deps += [
- "../api/transport:sctp_transport_factory_interface",
+ deps = [
+ ":rtc_data_sctp_transport_internal",
+ "../media:rtc_media_base",
+ "../p2p:rtc_p2p",
+ "../rtc_base",
+ "../rtc_base:rtc_base_approved",
+ "../rtc_base:threading",
+ "../rtc_base/synchronization:mutex",
+ "../rtc_base/task_utils:pending_task_safety_flag",
+ "../rtc_base/task_utils:to_queued_task",
+ "../rtc_base/third_party/sigslot:sigslot",
"//third_party/usrsctp",
]
+ absl_deps = [
+ "//third_party/abseil-cpp/absl/algorithm:container",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
+ }
+}
+
+rtc_library("rtc_data_sctp_transport_factory") {
+ defines = []
+ sources = [
+ "sctp/sctp_transport_factory.cc",
+ "sctp/sctp_transport_factory.h",
+ ]
+ deps = [
+ ":rtc_data_sctp_transport_internal",
+ "../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_build_usrsctp) {
+ defines += [ "WEBRTC_HAVE_USRSCTP" ]
+ deps += [ ":rtc_data_usrsctp_transport" ]
}
}
rtc_source_set("rtc_media") {
visibility = [ "*" ]
allow_poison = [ "audio_codecs" ] # TODO(bugs.webrtc.org/8396): Remove.
- deps = [
- ":rtc_audio_video",
- ":rtc_data",
- ]
+ deps = [ ":rtc_audio_video" ]
}
if (rtc_include_tests) {
@@ -537,7 +551,6 @@
defines = []
deps = [
":rtc_audio_video",
- ":rtc_data",
":rtc_encoder_simulcast_proxy",
":rtc_internal_video_codecs",
":rtc_media",
@@ -641,15 +654,18 @@
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 (is_ios) {
deps += [ ":rtc_media_unittests_bundle_data" ]
}
-
- if (rtc_enable_sctp && rtc_build_usrsctp) {
- deps += [ "//third_party/usrsctp" ]
- }
}
}
}
diff --git a/media/sctp/noop.cc b/media/sctp/noop.cc
deleted file mode 100644
index a3523b1..0000000
--- a/media/sctp/noop.cc
+++ /dev/null
@@ -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.
diff --git a/media/sctp/sctp_transport_factory.cc b/media/sctp/sctp_transport_factory.cc
index dd53d04..40480e7 100644
--- a/media/sctp/sctp_transport_factory.cc
+++ b/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>(
- new UsrsctpTransport(network_thread_, transport));
+ 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
diff --git a/media/sctp/sctp_transport_factory.h b/media/sctp/sctp_transport_factory.h
index 7fe4de0..4fff214 100644
--- a/media/sctp/sctp_transport_factory.h
+++ b/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 {
diff --git a/media/sctp/usrsctp_transport.h b/media/sctp/usrsctp_transport.h
index 0241c2c..de018b9 100644
--- a/media/sctp/usrsctp_transport.h
+++ b/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"
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index a61e04d..2b70f59 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -108,7 +108,7 @@
"../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 @@
"../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 @@
"../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 @@
"../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 @@
"webrtc_sdp_unittest.cc",
]
- if (rtc_enable_sctp) {
- defines = [ "WEBRTC_HAVE_SCTP" ]
- }
-
deps = [
":audio_rtp_receiver",
":audio_track",
@@ -1065,6 +1061,7 @@
"../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 @@
"../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 @@
"../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",
diff --git a/pc/connection_context.h b/pc/connection_context.h
index 29ae99a..0c69c17 100644
--- a/pc/connection_context.h
+++ b/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"
diff --git a/pc/peer_connection_factory.h b/pc/peer_connection_factory.h
index 71d7830..d2bac7a 100644
--- a/pc/peer_connection_factory.h
+++ b/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"
diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
index 054cd36..b276870 100644
--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
@@ -777,7 +777,6 @@
"../../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",
diff --git a/test/pc/sctp/BUILD.gn b/test/pc/sctp/BUILD.gn
index 93ae1bf..b47cff2 100644
--- a/test/pc/sctp/BUILD.gn
+++ b/test/pc/sctp/BUILD.gn
@@ -11,5 +11,5 @@
rtc_source_set("fake_sctp_transport") {
visibility = [ "*" ]
sources = [ "fake_sctp_transport.h" ]
- deps = [ "../../../media:rtc_data" ]
+ deps = [ "../../../media:rtc_data_sctp_transport_internal" ]
}
diff --git a/webrtc.gni b/webrtc.gni
index 6c80d72..574df1f 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -233,7 +233,6 @@
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 @@
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"
