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Use source_sets in component builds and static_library in release builds.

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Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).

Source sets always pass all the object files to the linker.

On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.

See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set

Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
This commit is contained in:
Mirko Bonadei
2019-10-17 21:32:04 +02:00
committed by Commit Bot
parent e8b962bd3c
commit 86d053c2db
100 changed files with 756 additions and 654 deletions
Download Patch File Download Diff File Expand all files Collapse all files

28
logging/BUILD.gn
View File

@ -36,7 +36,7 @@ rtc_source_set("rtc_event_log_api") {
]
}
rtc_source_set("rtc_stream_config") {
rtc_library("rtc_stream_config") {
sources = [
"rtc_event_log/rtc_stream_config.cc",
"rtc_event_log/rtc_stream_config.h",
@ -48,7 +48,7 @@ rtc_source_set("rtc_stream_config") {
]
}
rtc_source_set("rtc_event_pacing") {
rtc_library("rtc_event_pacing") {
sources = [
"rtc_event_log/events/rtc_event_alr_state.cc",
"rtc_event_log/events/rtc_event_alr_state.h",
@ -61,7 +61,7 @@ rtc_source_set("rtc_event_pacing") {
]
}
rtc_source_set("rtc_event_audio") {
rtc_library("rtc_event_audio") {
sources = [
"rtc_event_log/events/rtc_event_audio_network_adaptation.cc",
"rtc_event_log/events/rtc_event_audio_network_adaptation.h",
@ -83,7 +83,7 @@ rtc_source_set("rtc_event_audio") {
]
}
rtc_source_set("rtc_event_bwe") {
rtc_library("rtc_event_bwe") {
sources = [
"rtc_event_log/events/rtc_event_bwe_update_delay_based.cc",
"rtc_event_log/events/rtc_event_bwe_update_delay_based.h",
@ -110,7 +110,7 @@ rtc_source_set("rtc_event_bwe") {
]
}
rtc_source_set("rtc_event_generic_packet_events") {
rtc_library("rtc_event_generic_packet_events") {
visibility = [ "*" ]
sources = [
"rtc_event_log/events/rtc_event_generic_ack_received.cc",
@ -128,7 +128,7 @@ rtc_source_set("rtc_event_generic_packet_events") {
]
}
rtc_source_set("rtc_event_rtp_rtcp") {
rtc_library("rtc_event_rtp_rtcp") {
sources = [
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc",
"rtc_event_log/events/rtc_event_rtcp_packet_incoming.h",
@ -151,7 +151,7 @@ rtc_source_set("rtc_event_rtp_rtcp") {
]
}
rtc_source_set("rtc_event_video") {
rtc_library("rtc_event_video") {
sources = [
"rtc_event_log/events/rtc_event_video_receive_stream_config.cc",
"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
@ -170,7 +170,7 @@ rtc_source_set("rtc_event_video") {
# TODO(eladalon): Break down into (1) encoder and (2) decoder; we don't need
# the decoder code in the WebRTC library, only in unit tests and tools.
rtc_static_library("rtc_event_log_impl_encoder") {
rtc_library("rtc_event_log_impl_encoder") {
sources = [
"rtc_event_log/encoder/blob_encoding.cc",
"rtc_event_log/encoder/blob_encoding.h",
@ -235,7 +235,7 @@ rtc_source_set("rtc_event_log_impl_output") {
}
if (rtc_enable_protobuf) {
rtc_source_set("rtc_event_log_impl") {
rtc_library("rtc_event_log_impl") {
visibility = [ "../api/rtc_event_log:rtc_event_log_factory" ]
sources = [
"rtc_event_log/rtc_event_log_impl.cc",
@ -258,7 +258,7 @@ if (rtc_enable_protobuf) {
}
}
rtc_source_set("fake_rtc_event_log") {
rtc_library("fake_rtc_event_log") {
testonly = true
sources = [
"rtc_event_log/fake_rtc_event_log.cc",
@ -292,7 +292,7 @@ if (rtc_enable_protobuf) {
proto_out_dir = "logging/rtc_event_log"
}
rtc_static_library("rtc_event_log_parser") {
rtc_library("rtc_event_log_parser") {
visibility = [ "*" ]
sources = [
"rtc_event_log/logged_events.cc",
@ -335,7 +335,7 @@ if (rtc_enable_protobuf) {
}
if (rtc_include_tests) {
rtc_source_set("rtc_event_log_tests") {
rtc_library("rtc_event_log_tests") {
testonly = true
assert(rtc_enable_protobuf)
defines = [ "ENABLE_RTC_EVENT_LOG" ]
@ -412,7 +412,7 @@ if (rtc_enable_protobuf) {
}
}
rtc_source_set("ice_log") {
rtc_library("ice_log") {
sources = [
"rtc_event_log/events/rtc_event_dtls_transport_state.cc",
"rtc_event_log/events/rtc_event_dtls_transport_state.h",
@ -436,7 +436,7 @@ rtc_source_set("ice_log") {
}
if (rtc_include_tests) {
rtc_source_set("mocks") {
rtc_library("mocks") {
testonly = true
sources = [
"rtc_event_log/mock/mock_rtc_event_log.cc",