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GN: Change rtc_source_set targets --> rtc_static_library

Browse Source 
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}
This commit is contained in:
kjellander 2016-09-23 00:38:52 -07:00 committed by Commit bot
parent 25f6a39181
commit b62dbbe985
36 changed files with 91 additions and 91 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/BUILD.gn
View File

@ -287,7 +287,7 @@ if (!is_ios || !build_with_chromium) {
}
}
rtc_source_set("webrtc_common") {
rtc_static_library("webrtc_common") {
sources = [
"common_types.cc",
"common_types.h",
@ -313,7 +313,7 @@ if (rtc_enable_protobuf) {
}
}
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 @@ rtc_source_set("rtc_event_log") {
}
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",

4
webrtc/api/BUILD.gn
View File

@ -43,7 +43,7 @@ config("libjingle_peerconnection_warnings_config") {
}
}
rtc_source_set("libjingle_peerconnection") {
rtc_static_library("libjingle_peerconnection") {
cflags = []
sources = [
"audiotrack.cc",
@ -160,7 +160,7 @@ if (is_android && !build_with_chromium) {
}
}
rtc_source_set("libjingle_peerconnection_jni") {
rtc_static_library("libjingle_peerconnection_jni") {
sources = [
"android/jni/androidmediacodeccommon.h",
"android/jni/androidmediadecoder_jni.cc",

2
webrtc/audio/BUILD.gn
View File

@ -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",

2
webrtc/call/BUILD.gn
View File

@ -8,7 +8,7 @@
import("../build/webrtc.gni")
rtc_source_set("call") {
rtc_static_library("call") {
sources = [
"bitrate_allocator.cc",
"call.cc",

6
webrtc/common_audio/BUILD.gn
View File

@ -17,7 +17,7 @@ config("common_audio_config") {
]
}
rtc_source_set("common_audio") {
rtc_static_library("common_audio") {
sources = [
"audio_converter.cc",
"audio_converter.h",
@ -186,7 +186,7 @@ rtc_source_set("common_audio") {
}
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 (current_cpu == "x86" || current_cpu == "x64") {
}
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",

2
webrtc/common_video/BUILD.gn
View File

@ -15,7 +15,7 @@ config("common_video_config") {
]
}
rtc_source_set("common_video") {
rtc_static_library("common_video") {
sources = [
"bitrate_adjuster.cc",
"h264/h264_common.cc",

8
webrtc/examples/BUILD.gn
View File

@ -121,7 +121,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
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 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
}
}
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_ios || (is_mac && target_cpu != "x86")) {
}
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 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
}
}
rtc_source_set("socketrocket") {
rtc_static_library("socketrocket") {
sources = [
"objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.h",
"objc/AppRTCDemo/third_party/SocketRocket/SRWebSocket.m",

2
webrtc/libjingle/xmllite/BUILD.gn
View File

@ -14,7 +14,7 @@ group("xmllite") {
]
}
rtc_source_set("rtc_xmllite") {
rtc_static_library("rtc_xmllite") {
sources = [
"qname.cc",
"qname.h",

2
webrtc/libjingle/xmpp/BUILD.gn
View File

@ -30,7 +30,7 @@ config("xmpp_inherited_config") {
]
}
rtc_source_set("rtc_xmpp") {
rtc_static_library("rtc_xmpp") {
cflags = []
sources = [
"asyncsocket.h",

2
webrtc/media/BUILD.gn
View File

@ -42,7 +42,7 @@ if (is_linux && rtc_use_gtk) {
}
}
rtc_source_set("rtc_media") {
rtc_static_library("rtc_media") {
defines = []
libs = []
deps = []

38
webrtc/modules/audio_coding/BUILD.gn
View File

@ -39,7 +39,7 @@ audio_coding_deps = audio_codec_deps + [
"../../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("audio_decoder_factory_interface") {
]
}
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 @@ rtc_source_set("builtin_audio_decoder_factory") {
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 @@ config("audio_coding_config") {
]
}
rtc_source_set("audio_coding") {
rtc_static_library("audio_coding") {
sources = [
"acm2/acm_common_defs.h",
"acm2/acm_receiver.cc",
@ -124,7 +124,7 @@ rtc_source_set("audio_coding") {
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_decoder_interface") {
]
}
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 @@ config("cng_config") {
]
}
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 @@ config("red_config") {
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 @@ config("g711_config") {
]
}
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 @@ config("g722_config") {
]
}
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 @@ config("ilbc_config") {
]
}
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("ilbc") {
]
}
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 @@ config("isac_config") {
]
}
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 @@ config("isac_fix_config") {
]
}
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 @@ rtc_source_set("isac_fix") {
}
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 @@ config("pcm16b_config") {
]
}
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 @@ config("opus_config") {
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 @@ config("neteq_config") {
]
}
rtc_source_set("neteq") {
rtc_static_library("neteq") {
sources = [
"neteq/accelerate.cc",
"neteq/accelerate.h",
@ -1002,7 +1002,7 @@ if (rtc_include_tests) {
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",

2
webrtc/modules/audio_conference_mixer/BUILD.gn
View File

@ -16,7 +16,7 @@ config("audio_conference_mixer_config") {
]
}
rtc_source_set("audio_conference_mixer") {
rtc_static_library("audio_conference_mixer") {
sources = [
"include/audio_conference_mixer.h",
"include/audio_conference_mixer_defines.h",

2
webrtc/modules/audio_device/BUILD.gn
View File

@ -48,7 +48,7 @@ config("audio_device_warnings_config") {
}
}
rtc_source_set("audio_device") {
rtc_static_library("audio_device") {
public_configs = [ ":audio_device_config" ]
deps = [

2
webrtc/modules/audio_mixer/BUILD.gn
View File

@ -12,7 +12,7 @@ config("audio_conference_mixer_config") {
include_dirs = [ "../../modules/include" ]
}
rtc_source_set("audio_mixer") {
rtc_static_library("audio_mixer") {
sources = [
"audio_frame_manipulator.cc",
"audio_frame_manipulator.h",

8
webrtc/modules/audio_processing/BUILD.gn
View File

@ -17,7 +17,7 @@ declare_args() {
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 (rtc_enable_protobuf) {
}
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 (current_cpu == "x86" || current_cpu == "x64") {
}
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 @@ if (rtc_include_tests) {
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",

2
webrtc/modules/bitrate_controller/BUILD.gn
View File

@ -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",

2
webrtc/modules/congestion_controller/BUILD.gn
View File

@ -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",

6
webrtc/modules/desktop_capture/BUILD.gn
View File

@ -12,7 +12,7 @@ import("../../build/webrtc.gni")
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("primitives") {
]
}
rtc_source_set("desktop_capture") {
rtc_static_library("desktop_capture") {
sources = [
"cropped_desktop_frame.cc",
"cropped_desktop_frame.h",
@ -168,7 +168,7 @@ rtc_source_set("desktop_capture") {
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",

2
webrtc/modules/media_file/BUILD.gn
View File

@ -12,7 +12,7 @@ config("media_file_config") {
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",

2
webrtc/modules/pacing/BUILD.gn
View File

@ -8,7 +8,7 @@
import("../../build/webrtc.gni")
rtc_source_set("pacing") {
rtc_static_library("pacing") {
sources = [
"bitrate_prober.cc",
"bitrate_prober.h",

4
webrtc/modules/remote_bitrate_estimator/BUILD.gn
View File

@ -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 @@ rtc_source_set("remote_bitrate_estimator") {
}
if (rtc_include_tests) {
rtc_source_set("bwe_simulator") {
rtc_static_library("bwe_simulator") {
testonly = true
sources = [
"bwe_simulations.cc",

2
webrtc/modules/rtp_rtcp/BUILD.gn
View File

@ -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",

2
webrtc/modules/utility/BUILD.gn
View File

@ -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",

6
webrtc/modules/video_capture/BUILD.gn
View File

@ -12,7 +12,7 @@ import("../../build/webrtc.gni")
# 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_module") {
}
}
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 @@ if (!build_with_chromium) {
}
}
rtc_source_set("video_capture_internal_impl") {
rtc_static_library("video_capture_internal_impl") {
configs += [ ":video_capture_internal_impl_warnings_config" ]
deps = [

14
webrtc/modules/video_coding/BUILD.gn
View File

@ -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") {
]
}
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("video_coding_utility") {
]
}
rtc_source_set("webrtc_h264") {
rtc_static_library("webrtc_h264") {
sources = [
"codecs/h264/h264.cc",
"codecs/h264/include/h264.h",
@ -178,7 +178,7 @@ if (is_ios) {
}
}
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 @@ if (is_ios) {
}
}
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_i420") {
]
}
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_vp8") {
}
}
rtc_source_set("webrtc_vp9") {
rtc_static_library("webrtc_vp9") {
sources = [
"codecs/vp9/include/vp9.h",
"codecs/vp9/screenshare_layers.cc",

6
webrtc/modules/video_processing/BUILD.gn
View File

@ -11,7 +11,7 @@ import("../../build/webrtc.gni")
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 @@ rtc_source_set("video_processing") {
}
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 (build_video_processing_sse2) {
}
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",

4
webrtc/p2p/BUILD.gn
View File

@ -18,7 +18,7 @@ config("rtc_p2p_inherited_config") {
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("rtc_p2p") {
}
}
rtc_source_set("libstunprober") {
rtc_static_library("libstunprober") {
sources = [
"stunprober/stunprober.cc",
]

2
webrtc/pc/BUILD.gn
View File

@ -22,7 +22,7 @@ config("rtc_pc_config") {
]
}
rtc_source_set("rtc_pc") {
rtc_static_library("rtc_pc") {
defines = []
sources = [
"audiomonitor.cc",

4
webrtc/sdk/BUILD.gn
View File

@ -23,7 +23,7 @@ if (is_ios || (is_mac && mac_deployment_target == "10.7")) {
]
}
rtc_source_set("rtc_sdk_common_objc") {
rtc_static_library("rtc_sdk_common_objc") {
deps = [
"../base:rtc_base",
]
@ -77,7 +77,7 @@ if (is_ios || (is_mac && mac_deployment_target == "10.7")) {
}
}
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",

2
webrtc/stats/BUILD.gn
View File

@ -15,7 +15,7 @@ group("stats") {
}
# GYP version: webrtc/stats/stats.gyp:rtc_stats
rtc_source_set("rtc_stats") {
rtc_static_library("rtc_stats") {
cflags = []
sources = [
"rtcstats.cc",

12
webrtc/system_wrappers/BUILD.gn
View File

@ -12,7 +12,7 @@ if (is_android) {
}
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 @@ rtc_source_set("system_wrappers") {
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 @@ rtc_source_set("system_wrappers_default") {
}
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_android) {
}
if (is_linux) {
rtc_source_set("cpu_features_linux") {
rtc_static_library("cpu_features_linux") {
sources = [
"source/cpu_features_linux.c",
]

2
webrtc/test/BUILD.gn
View File

@ -12,7 +12,7 @@ if (is_android) {
import("//build/config/android/rules.gni")
}
rtc_source_set("test") {
group("test") {
testonly = true
deps = [

2
webrtc/test/fuzzers/BUILD.gn
View File

@ -117,7 +117,7 @@ webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
]
}
rtc_source_set("audio_decoder_fuzzer") {
rtc_static_library("audio_decoder_fuzzer") {
sources = [
"audio_decoder_fuzzer.cc",
"audio_decoder_fuzzer.h",

10
webrtc/tools/BUILD.gn
View File

@ -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("tools") {
}
}
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("command_line_parser") {
]
}
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_executable("rgba_to_i420_converter") {
]
}
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 @@ if (rtc_enable_protobuf) {
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",

2
webrtc/video/BUILD.gn
View File

@ -8,7 +8,7 @@
import("../build/webrtc.gni")
rtc_source_set("video") {
rtc_static_library("video") {
sources = [
"call_stats.cc",
"call_stats.h",

6
webrtc/voice_engine/BUILD.gn
View File

@ -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("voice_engine") {
]
}
rtc_source_set("level_indicator") {
rtc_static_library("level_indicator") {
sources = [
"level_indicator.cc",
"level_indicator.h",
@ -142,7 +142,7 @@ if (rtc_include_tests) {
}
}
rtc_source_set("channel_transport") {
rtc_static_library("channel_transport") {
testonly = true
sources = [
"test/channel_transport/channel_transport.cc",