GN: Change rtc_source_set targets --> rtc_static_library
This changes most non-test related rtc_source_set targets to be
rtc_static_library instead. Targets without any .cc files are excluded.
This should bring back the build behavior we used to have with GYP
(i.e. same symbols exported in the libjingle_peerconnection.a file, which
are used by some downstream projects).
After doing an Android build with these changes:
$ nm --defined-only -g -C out/Release/lib.unstripped/libjingle_peerconnection_so.so | grep -i createpeerconnectionf
00077c51 T Java_org_webrtc_PeerConnectionFactory_nativeCreatePeerConnectionFactory
$ nm --defined-only -g -C out/Release/obj/webrtc/api/libjingle_peerconnection.a | grep -i createpeerconnectionf
00000001 T webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, webrtc::AudioDeviceModule*, cricket::WebRtcVideoEncoderFactory*, cricket::WebRtcVideoDecoderFactory*)
00000001 T webrtc::CreatePeerConnectionFactory()
See https://chromium.googlesource.com/chromium/src/+/master/tools/gn/docs/cookbook.md#Note-on-static-libraries
for more details on this.
NOTICE: This should be further cleaned up in the future, to reduce
binary bloat and unnecessary linking time. Right now it's more
important to restore the desired build output though.
BUG=webrtc:6410, chromium:630755
Review-Url: https://codereview.webrtc.org/2361623004
Cr-Commit-Position: refs/heads/master@{#14364}
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 47b757a..d4308e1 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -287,7 +287,7 @@
}
}
-rtc_source_set("webrtc_common") {
+rtc_static_library("webrtc_common") {
sources = [
"common_types.cc",
"common_types.h",
@@ -313,7 +313,7 @@
}
}
-rtc_source_set("rtc_event_log") {
+rtc_static_library("rtc_event_log") {
sources = [
"call/rtc_event_log.cc",
"call/rtc_event_log.h",
@@ -340,7 +340,7 @@
}
if (rtc_enable_protobuf) {
- rtc_source_set("rtc_event_log_parser") {
+ rtc_static_library("rtc_event_log_parser") {
sources = [
"call/rtc_event_log_parser.cc",
"call/rtc_event_log_parser.h",
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index d58371c..6fe43a7 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -43,7 +43,7 @@
}
}
-rtc_source_set("libjingle_peerconnection") {
+rtc_static_library("libjingle_peerconnection") {
cflags = []
sources = [
"audiotrack.cc",
@@ -160,7 +160,7 @@
}
}
- rtc_source_set("libjingle_peerconnection_jni") {
+ rtc_static_library("libjingle_peerconnection_jni") {
sources = [
"android/jni/androidmediacodeccommon.h",
"android/jni/androidmediadecoder_jni.cc",
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index d29ca8a..14607cd 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -8,7 +8,7 @@
import("../build/webrtc.gni")
-rtc_source_set("audio") {
+rtc_static_library("audio") {
sources = [
"audio_receive_stream.cc",
"audio_receive_stream.h",
diff --git a/webrtc/call/BUILD.gn b/webrtc/call/BUILD.gn
index 05a3192..32bcba3 100644
--- a/webrtc/call/BUILD.gn
+++ b/webrtc/call/BUILD.gn
@@ -8,7 +8,7 @@
import("../build/webrtc.gni")
-rtc_source_set("call") {
+rtc_static_library("call") {
sources = [
"bitrate_allocator.cc",
"call.cc",
diff --git a/webrtc/common_audio/BUILD.gn b/webrtc/common_audio/BUILD.gn
index 08a97b2..6258fb4 100644
--- a/webrtc/common_audio/BUILD.gn
+++ b/webrtc/common_audio/BUILD.gn
@@ -17,7 +17,7 @@
]
}
-rtc_source_set("common_audio") {
+rtc_static_library("common_audio") {
sources = [
"audio_converter.cc",
"audio_converter.h",
@@ -186,7 +186,7 @@
}
if (current_cpu == "x86" || current_cpu == "x64") {
- rtc_source_set("common_audio_sse2") {
+ rtc_static_library("common_audio_sse2") {
sources = [
"fir_filter_sse.cc",
"resampler/sinc_resampler_sse.cc",
@@ -205,7 +205,7 @@
}
if (rtc_build_with_neon) {
- rtc_source_set("common_audio_neon") {
+ rtc_static_library("common_audio_neon") {
sources = [
"fir_filter_neon.cc",
"resampler/sinc_resampler_neon.cc",
diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn
index 8b553b2..845a5eb 100644
--- a/webrtc/common_video/BUILD.gn
+++ b/webrtc/common_video/BUILD.gn
@@ -15,7 +15,7 @@
]
}
-rtc_source_set("common_video") {
+rtc_static_library("common_video") {
sources = [
"bitrate_adjuster.cc",
"h264/h264_common.cc",
diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn
index ce99522..da50bfd 100644
--- a/webrtc/examples/BUILD.gn
+++ b/webrtc/examples/BUILD.gn
@@ -121,7 +121,7 @@
include_dirs = [ "objc/AppRTCDemo/common" ]
}
- rtc_source_set("apprtc_common") {
+ rtc_static_library("apprtc_common") {
sources = [
"objc/AppRTCDemo/common/ARDUtilities.h",
"objc/AppRTCDemo/common/ARDUtilities.m",
@@ -154,7 +154,7 @@
}
}
- rtc_source_set("apprtc_signaling") {
+ rtc_static_library("apprtc_signaling") {
sources = [
"objc/AppRTCDemo/ARDAppClient+Internal.h",
"objc/AppRTCDemo/ARDAppClient.h",
@@ -270,7 +270,7 @@
}
if (is_mac) {
- rtc_source_set("AppRTCDemo_app") {
+ rtc_static_library("AppRTCDemo_app") {
sources = [
"objc/AppRTCDemo/mac/APPRTCAppDelegate.h",
"objc/AppRTCDemo/mac/APPRTCAppDelegate.m",
@@ -338,7 +338,7 @@
}
}
- rtc_source_set("socketrocket") {
+ rtc_static_library("socketrocket") {
sources = [
"objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.h",
"objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.m",
diff --git a/webrtc/libjingle/xmllite/BUILD.gn b/webrtc/libjingle/xmllite/BUILD.gn
index 13211ac..496d452 100644
--- a/webrtc/libjingle/xmllite/BUILD.gn
+++ b/webrtc/libjingle/xmllite/BUILD.gn
@@ -14,7 +14,7 @@
]
}
-rtc_source_set("rtc_xmllite") {
+rtc_static_library("rtc_xmllite") {
sources = [
"qname.cc",
"qname.h",
diff --git a/webrtc/libjingle/xmpp/BUILD.gn b/webrtc/libjingle/xmpp/BUILD.gn
index 43a39aa..879d672 100644
--- a/webrtc/libjingle/xmpp/BUILD.gn
+++ b/webrtc/libjingle/xmpp/BUILD.gn
@@ -30,7 +30,7 @@
]
}
-rtc_source_set("rtc_xmpp") {
+rtc_static_library("rtc_xmpp") {
cflags = []
sources = [
"asyncsocket.h",
diff --git a/webrtc/media/BUILD.gn b/webrtc/media/BUILD.gn
index e31d7e5..8a4e808 100644
--- a/webrtc/media/BUILD.gn
+++ b/webrtc/media/BUILD.gn
@@ -42,7 +42,7 @@
}
}
-rtc_source_set("rtc_media") {
+rtc_static_library("rtc_media") {
defines = []
libs = []
deps = []
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index a69206e..fceff9b0 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -39,7 +39,7 @@
"../../system_wrappers",
]
-rtc_source_set("audio_decoder_factory_interface") {
+rtc_static_library("audio_decoder_factory_interface") {
sources = [
"codecs/audio_decoder_factory.h",
"codecs/audio_format.cc",
@@ -50,7 +50,7 @@
]
}
-rtc_source_set("builtin_audio_decoder_factory") {
+rtc_static_library("builtin_audio_decoder_factory") {
sources = [
"codecs/builtin_audio_decoder_factory.cc",
"codecs/builtin_audio_decoder_factory.h",
@@ -62,7 +62,7 @@
defines = audio_codec_defines
}
-rtc_source_set("rent_a_codec") {
+rtc_static_library("rent_a_codec") {
sources = [
"acm2/acm_codec_database.cc",
"acm2/acm_codec_database.h",
@@ -80,7 +80,7 @@
]
}
-rtc_source_set("audio_coding") {
+rtc_static_library("audio_coding") {
sources = [
"acm2/acm_common_defs.h",
"acm2/acm_receiver.cc",
@@ -124,7 +124,7 @@
defines = audio_coding_defines
}
-rtc_source_set("audio_decoder_interface") {
+rtc_static_library("audio_decoder_interface") {
sources = [
"codecs/audio_decoder.cc",
"codecs/audio_decoder.h",
@@ -137,7 +137,7 @@
]
}
-rtc_source_set("audio_encoder_interface") {
+rtc_static_library("audio_encoder_interface") {
sources = [
"codecs/audio_encoder.cc",
"codecs/audio_encoder.h",
@@ -155,7 +155,7 @@
]
}
-rtc_source_set("cng") {
+rtc_static_library("cng") {
sources = [
"codecs/cng/audio_encoder_cng.cc",
"codecs/cng/audio_encoder_cng.h",
@@ -175,7 +175,7 @@
include_dirs = [ "codecs/red" ]
}
-rtc_source_set("red") {
+rtc_static_library("red") {
sources = [
"codecs/red/audio_encoder_copy_red.cc",
"codecs/red/audio_encoder_copy_red.h",
@@ -196,7 +196,7 @@
]
}
-rtc_source_set("g711") {
+rtc_static_library("g711") {
sources = [
"codecs/g711/audio_decoder_pcm.cc",
"codecs/g711/audio_decoder_pcm.h",
@@ -223,7 +223,7 @@
]
}
-rtc_source_set("g722") {
+rtc_static_library("g722") {
sources = [
"codecs/g722/audio_decoder_g722.cc",
"codecs/g722/audio_decoder_g722.h",
@@ -251,7 +251,7 @@
]
}
-rtc_source_set("ilbc") {
+rtc_static_library("ilbc") {
sources = [
"codecs/ilbc/abs_quant.c",
"codecs/ilbc/abs_quant.h",
@@ -408,7 +408,7 @@
]
}
-rtc_source_set("isac_common") {
+rtc_static_library("isac_common") {
sources = [
"codecs/isac/audio_encoder_isac_t.h",
"codecs/isac/audio_encoder_isac_t_impl.h",
@@ -424,7 +424,7 @@
]
}
-rtc_source_set("isac") {
+rtc_static_library("isac") {
sources = [
"codecs/isac/main/include/audio_decoder_isac.h",
"codecs/isac/main/include/audio_encoder_isac.h",
@@ -505,7 +505,7 @@
]
}
-rtc_source_set("isac_fix") {
+rtc_static_library("isac_fix") {
sources = [
"codecs/isac/fix/include/audio_decoder_isacfix.h",
"codecs/isac/fix/include/audio_encoder_isacfix.h",
@@ -608,7 +608,7 @@
}
if (rtc_build_with_neon) {
- rtc_source_set("isac_neon") {
+ rtc_static_library("isac_neon") {
sources = [
"codecs/isac/fix/source/entropy_coding_neon.c",
"codecs/isac/fix/source/filterbanks_neon.c",
@@ -647,7 +647,7 @@
]
}
-rtc_source_set("pcm16b") {
+rtc_static_library("pcm16b") {
sources = [
"codecs/pcm16b/audio_decoder_pcm16b.cc",
"codecs/pcm16b/audio_decoder_pcm16b.h",
@@ -670,7 +670,7 @@
include_dirs = [ "../../.." ]
}
-rtc_source_set("webrtc_opus") {
+rtc_static_library("webrtc_opus") {
sources = [
"codecs/opus/audio_decoder_opus.cc",
"codecs/opus/audio_decoder_opus.h",
@@ -749,7 +749,7 @@
]
}
-rtc_source_set("neteq") {
+rtc_static_library("neteq") {
sources = [
"neteq/accelerate.cc",
"neteq/accelerate.h",
@@ -1002,7 +1002,7 @@
proto_out_dir = "webrtc/audio_coding/neteq"
}
- rtc_source_set("rtc_event_log_source") {
+ rtc_static_library("rtc_event_log_source") {
testonly = true
sources = [
"neteq/tools/rtc_event_log_source.cc",
diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn
index 73446ba..129ff5b 100644
--- a/webrtc/modules/audio_conference_mixer/BUILD.gn
+++ b/webrtc/modules/audio_conference_mixer/BUILD.gn
@@ -16,7 +16,7 @@
]
}
-rtc_source_set("audio_conference_mixer") {
+rtc_static_library("audio_conference_mixer") {
sources = [
"include/audio_conference_mixer.h",
"include/audio_conference_mixer_defines.h",
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index 2771876..31a02d1 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -48,7 +48,7 @@
}
}
-rtc_source_set("audio_device") {
+rtc_static_library("audio_device") {
public_configs = [ ":audio_device_config" ]
deps = [
diff --git a/webrtc/modules/audio_mixer/BUILD.gn b/webrtc/modules/audio_mixer/BUILD.gn
index c0e5bb5..40b9f3b 100644
--- a/webrtc/modules/audio_mixer/BUILD.gn
+++ b/webrtc/modules/audio_mixer/BUILD.gn
@@ -12,7 +12,7 @@
include_dirs = [ "../../modules/include" ]
}
-rtc_source_set("audio_mixer") {
+rtc_static_library("audio_mixer") {
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 2387c2e..54c33e3 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -17,7 +17,7 @@
aec_untrusted_delay_for_testing = false
}
-rtc_source_set("audio_processing") {
+rtc_static_library("audio_processing") {
sources = [
"aec/aec_core.cc",
"aec/aec_core.h",
@@ -253,7 +253,7 @@
}
if (current_cpu == "x86" || current_cpu == "x64") {
- rtc_source_set("audio_processing_sse2") {
+ rtc_static_library("audio_processing_sse2") {
sources = [
"aec/aec_core_sse2.cc",
"aec/aec_rdft_sse2.cc",
@@ -272,7 +272,7 @@
}
if (rtc_build_with_neon) {
- rtc_source_set("audio_processing_neon") {
+ rtc_static_library("audio_processing_neon") {
sources = [
"aec/aec_core_neon.cc",
"aec/aec_rdft_neon.cc",
@@ -453,7 +453,7 @@
proto_out_dir = "webrtc/modules/audio_processing"
}
- rtc_source_set("audioproc_protobuf_utils") {
+ rtc_static_library("audioproc_protobuf_utils") {
sources = [
"test/protobuf_utils.cc",
"test/protobuf_utils.h",
diff --git a/webrtc/modules/bitrate_controller/BUILD.gn b/webrtc/modules/bitrate_controller/BUILD.gn
index 7f10c5d..7bf28e4 100644
--- a/webrtc/modules/bitrate_controller/BUILD.gn
+++ b/webrtc/modules/bitrate_controller/BUILD.gn
@@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
-rtc_source_set("bitrate_controller") {
+rtc_static_library("bitrate_controller") {
sources = [
"bitrate_controller_impl.cc",
"bitrate_controller_impl.h",
diff --git a/webrtc/modules/congestion_controller/BUILD.gn b/webrtc/modules/congestion_controller/BUILD.gn
index a07acc5..758fba5 100644
--- a/webrtc/modules/congestion_controller/BUILD.gn
+++ b/webrtc/modules/congestion_controller/BUILD.gn
@@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
-rtc_source_set("congestion_controller") {
+rtc_static_library("congestion_controller") {
sources = [
"congestion_controller.cc",
"delay_based_bwe.cc",
diff --git a/webrtc/modules/desktop_capture/BUILD.gn b/webrtc/modules/desktop_capture/BUILD.gn
index 8887ba9..46a768a 100644
--- a/webrtc/modules/desktop_capture/BUILD.gn
+++ b/webrtc/modules/desktop_capture/BUILD.gn
@@ -12,7 +12,7 @@
use_desktop_capture_differ_sse2 =
!is_ios && (current_cpu == "x86" || current_cpu == "x64")
-rtc_source_set("primitives") {
+rtc_static_library("primitives") {
sources = [
"desktop_capture_types.h",
"desktop_frame.cc",
@@ -24,7 +24,7 @@
]
}
-rtc_source_set("desktop_capture") {
+rtc_static_library("desktop_capture") {
sources = [
"cropped_desktop_frame.cc",
"cropped_desktop_frame.h",
@@ -168,7 +168,7 @@
if (use_desktop_capture_differ_sse2) {
# Have to be compiled as a separate target because it needs to be compiled
# with SSE2 enabled.
- rtc_source_set("desktop_capture_differ_sse2") {
+ rtc_static_library("desktop_capture_differ_sse2") {
visibility = [ ":*" ]
sources = [
"differ_vector_sse2.cc",
diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn
index 385052c..f73c44b 100644
--- a/webrtc/modules/media_file/BUILD.gn
+++ b/webrtc/modules/media_file/BUILD.gn
@@ -12,7 +12,7 @@
visibility = [ ":*" ] # Only targets in this file can depend on this.
}
-rtc_source_set("media_file") {
+rtc_static_library("media_file") {
sources = [
"media_file.h",
"media_file_defines.h",
diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn
index 26ad2b1..7a16297 100644
--- a/webrtc/modules/pacing/BUILD.gn
+++ b/webrtc/modules/pacing/BUILD.gn
@@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
-rtc_source_set("pacing") {
+rtc_static_library("pacing") {
sources = [
"bitrate_prober.cc",
"bitrate_prober.h",
diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
index 1e42ce9..ea0e214 100644
--- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn
+++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
-rtc_source_set("remote_bitrate_estimator") {
+rtc_static_library("remote_bitrate_estimator") {
sources = [
"aimd_rate_control.cc",
"aimd_rate_control.h",
@@ -56,7 +56,7 @@
}
if (rtc_include_tests) {
- rtc_source_set("bwe_simulator") {
+ rtc_static_library("bwe_simulator") {
testonly = true
sources = [
"bwe_simulations.cc",
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index e448af0..8b9d4b3 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
-rtc_source_set("rtp_rtcp") {
+rtc_static_library("rtp_rtcp") {
sources = [
"include/fec_receiver.h",
"include/receive_statistics.h",
diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn
index bb51b13..4d17776 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
-rtc_source_set("utility") {
+rtc_static_library("utility") {
sources = [
"include/audio_frame_operations.h",
"include/file_player.h",
diff --git a/webrtc/modules/video_capture/BUILD.gn b/webrtc/modules/video_capture/BUILD.gn
index d6e7dce..60573ba 100644
--- a/webrtc/modules/video_capture/BUILD.gn
+++ b/webrtc/modules/video_capture/BUILD.gn
@@ -12,7 +12,7 @@
# Targets must link with either 'video_capture' or
# 'video_capture_internal_impl' depending on whether they want to
# use the internal capturer.
-rtc_source_set("video_capture_module") {
+rtc_static_library("video_capture_module") {
sources = [
"device_info_impl.cc",
"device_info_impl.h",
@@ -40,7 +40,7 @@
}
}
-rtc_source_set("video_capture") {
+rtc_static_library("video_capture") {
sources = [
"external/device_info_external.cc",
"external/video_capture_external.cc",
@@ -87,7 +87,7 @@
}
}
- rtc_source_set("video_capture_internal_impl") {
+ rtc_static_library("video_capture_internal_impl") {
configs += [ ":video_capture_internal_impl_warnings_config" ]
deps = [
diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn
index 3db9b02..d9b392d 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
-rtc_source_set("video_coding") {
+rtc_static_library("video_coding") {
sources = [
"codec_database.cc",
"codec_database.h",
@@ -95,7 +95,7 @@
]
}
-rtc_source_set("video_coding_utility") {
+rtc_static_library("video_coding_utility") {
sources = [
"utility/frame_dropper.cc",
"utility/frame_dropper.h",
@@ -127,7 +127,7 @@
]
}
-rtc_source_set("webrtc_h264") {
+rtc_static_library("webrtc_h264") {
sources = [
"codecs/h264/h264.cc",
"codecs/h264/include/h264.h",
@@ -178,7 +178,7 @@
}
}
- rtc_source_set("webrtc_h264_video_toolbox") {
+ rtc_static_library("webrtc_h264_video_toolbox") {
sources = [
"codecs/h264/h264_video_toolbox_decoder.cc",
"codecs/h264/h264_video_toolbox_decoder.h",
@@ -223,7 +223,7 @@
}
}
-rtc_source_set("webrtc_i420") {
+rtc_static_library("webrtc_i420") {
sources = [
"codecs/i420/i420.cc",
"codecs/i420/include/i420.h",
@@ -240,7 +240,7 @@
]
}
-rtc_source_set("webrtc_vp8") {
+rtc_static_library("webrtc_vp8") {
sources = [
"codecs/vp8/default_temporal_layers.cc",
"codecs/vp8/default_temporal_layers.h",
@@ -278,7 +278,7 @@
}
}
-rtc_source_set("webrtc_vp9") {
+rtc_static_library("webrtc_vp9") {
sources = [
"codecs/vp9/include/vp9.h",
"codecs/vp9/screenshare_layers.cc",
diff --git a/webrtc/modules/video_processing/BUILD.gn b/webrtc/modules/video_processing/BUILD.gn
index 2cd6068..baf2e85 100644
--- a/webrtc/modules/video_processing/BUILD.gn
+++ b/webrtc/modules/video_processing/BUILD.gn
@@ -11,7 +11,7 @@
build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64"
-rtc_source_set("video_processing") {
+rtc_static_library("video_processing") {
sources = [
"frame_preprocessor.cc",
"frame_preprocessor.h",
@@ -56,7 +56,7 @@
}
if (build_video_processing_sse2) {
- rtc_source_set("video_processing_sse2") {
+ rtc_static_library("video_processing_sse2") {
sources = [
"util/denoiser_filter_sse2.cc",
"util/denoiser_filter_sse2.h",
@@ -75,7 +75,7 @@
}
if (rtc_build_with_neon) {
- rtc_source_set("video_processing_neon") {
+ rtc_static_library("video_processing_neon") {
sources = [
"util/denoiser_filter_neon.cc",
"util/denoiser_filter_neon.h",
diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn
index 8e1f37b..526155c 100644
--- a/webrtc/p2p/BUILD.gn
+++ b/webrtc/p2p/BUILD.gn
@@ -18,7 +18,7 @@
defines = [ "FEATURE_ENABLE_VOICEMAIL" ]
}
-rtc_source_set("rtc_p2p") {
+rtc_static_library("rtc_p2p") {
sources = [
"base/asyncstuntcpsocket.cc",
"base/asyncstuntcpsocket.h",
@@ -142,7 +142,7 @@
}
}
-rtc_source_set("libstunprober") {
+rtc_static_library("libstunprober") {
sources = [
"stunprober/stunprober.cc",
]
diff --git a/webrtc/pc/BUILD.gn b/webrtc/pc/BUILD.gn
index 94b0d94..42a933f 100644
--- a/webrtc/pc/BUILD.gn
+++ b/webrtc/pc/BUILD.gn
@@ -22,7 +22,7 @@
]
}
-rtc_source_set("rtc_pc") {
+rtc_static_library("rtc_pc") {
defines = []
sources = [
"audiomonitor.cc",
diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn
index 70954ee..96d8ef7 100644
--- a/webrtc/sdk/BUILD.gn
+++ b/webrtc/sdk/BUILD.gn
@@ -23,7 +23,7 @@
]
}
- rtc_source_set("rtc_sdk_common_objc") {
+ rtc_static_library("rtc_sdk_common_objc") {
deps = [
"../base:rtc_base",
]
@@ -77,7 +77,7 @@
}
}
- rtc_source_set("rtc_sdk_peerconnection_objc") {
+ rtc_static_library("rtc_sdk_peerconnection_objc") {
sources = [
"objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h",
"objc/Framework/Classes/RTCAVFoundationVideoSource.mm",
diff --git a/webrtc/stats/BUILD.gn b/webrtc/stats/BUILD.gn
index a9c9fb8..8966744 100644
--- a/webrtc/stats/BUILD.gn
+++ b/webrtc/stats/BUILD.gn
@@ -15,7 +15,7 @@
}
# GYP version: webrtc/stats/stats.gyp:rtc_stats
-rtc_source_set("rtc_stats") {
+rtc_static_library("rtc_stats") {
cflags = []
sources = [
"rtcstats.cc",
diff --git a/webrtc/system_wrappers/BUILD.gn b/webrtc/system_wrappers/BUILD.gn
index 8087f69..4dd85c2 100644
--- a/webrtc/system_wrappers/BUILD.gn
+++ b/webrtc/system_wrappers/BUILD.gn
@@ -12,7 +12,7 @@
}
import("../build/webrtc.gni")
-rtc_source_set("system_wrappers") {
+rtc_static_library("system_wrappers") {
sources = [
"include/aligned_array.h",
"include/aligned_malloc.h",
@@ -136,21 +136,21 @@
deps += [ "../base:rtc_base_approved" ]
}
-rtc_source_set("field_trial_default") {
+rtc_static_library("field_trial_default") {
sources = [
"include/field_trial_default.h",
"source/field_trial_default.cc",
]
}
-rtc_source_set("metrics_default") {
+rtc_static_library("metrics_default") {
sources = [
"include/metrics_default.h",
"source/metrics_default.cc",
]
}
-rtc_source_set("system_wrappers_default") {
+group("system_wrappers_default") {
deps = [
":field_trial_default",
":metrics_default",
@@ -159,7 +159,7 @@
}
if (is_android) {
- rtc_source_set("cpu_features_android") {
+ rtc_static_library("cpu_features_android") {
sources = [
"source/cpu_features_android.c",
]
@@ -171,7 +171,7 @@
}
if (is_linux) {
- rtc_source_set("cpu_features_linux") {
+ rtc_static_library("cpu_features_linux") {
sources = [
"source/cpu_features_linux.c",
]
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index 5ac393d..249c78c 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -12,7 +12,7 @@
import("//build/config/android/rules.gni")
}
-rtc_source_set("test") {
+group("test") {
testonly = true
deps = [
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index 0bff7bd..4e5d671 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -117,7 +117,7 @@
]
}
-rtc_source_set("audio_decoder_fuzzer") {
+rtc_static_library("audio_decoder_fuzzer") {
sources = [
"audio_decoder_fuzzer.cc",
"audio_decoder_fuzzer.h",
diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn
index dc4fce2..7d9cd40 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -9,7 +9,7 @@
import("//third_party/protobuf/proto_library.gni")
import("../build/webrtc.gni")
-rtc_source_set("tools") {
+group("tools") {
deps = [
":command_line_parser",
]
@@ -23,7 +23,7 @@
}
}
-rtc_source_set("command_line_parser") {
+rtc_static_library("command_line_parser") {
sources = [
"simple_command_line_parser.cc",
"simple_command_line_parser.h",
@@ -33,7 +33,7 @@
]
}
-rtc_source_set("video_quality_analysis") {
+rtc_static_library("video_quality_analysis") {
sources = [
"frame_analyzer/video_quality_analysis.cc",
"frame_analyzer/video_quality_analysis.h",
@@ -85,7 +85,7 @@
]
}
-rtc_source_set("frame_editing_lib") {
+rtc_static_library("frame_editing_lib") {
sources = [
"frame_editing/frame_editing_lib.cc",
"frame_editing/frame_editing_lib.h",
@@ -146,7 +146,7 @@
proto_out_dir = "webrtc/tools/event_log_visualizer"
}
- rtc_source_set("event_log_visualizer_utils") {
+ rtc_static_library("event_log_visualizer_utils") {
sources = [
"event_log_visualizer/analyzer.cc",
"event_log_visualizer/analyzer.h",
diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn
index b877a24..14266a6 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -8,7 +8,7 @@
import("../build/webrtc.gni")
-rtc_source_set("video") {
+rtc_static_library("video") {
sources = [
"call_stats.cc",
"call_stats.h",
diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn
index 8a8050b..8ea5328 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -8,7 +8,7 @@
import("../build/webrtc.gni")
-rtc_source_set("voice_engine") {
+rtc_static_library("voice_engine") {
sources = [
"channel.cc",
"channel.h",
@@ -106,7 +106,7 @@
]
}
-rtc_source_set("level_indicator") {
+rtc_static_library("level_indicator") {
sources = [
"level_indicator.cc",
"level_indicator.h",
@@ -142,7 +142,7 @@
}
}
- rtc_source_set("channel_transport") {
+ rtc_static_library("channel_transport") {
testonly = true
sources = [
"test/channel_transport/channel_transport.cc",
