Remove webrtc_all target
Move all its configuration to all.gyp instead, which is not processed by Chromium builds (webrtc.gyp is). BUG=webrtc:5949 NOTRY=True Review-Url: https://codereview.webrtc.org/2060873002 Cr-Commit-Position: refs/heads/master@{#13124}
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kjellander
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@ -4,8 +4,9 @@ niklas.enbom@webrtc.org
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tina.legrand@webrtc.org
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tommi@webrtc.org
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per-file .gitignore=*
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per-file AUTHORS=*
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per-file .gn=kjellander@webrtc.org
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per-file *.gyp=*
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per-file AUTHORS=*
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per-file BUILD.gn=kjellander@webrtc.org
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per-file DEPS=*
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per-file PRESUBMIT.py=kjellander@webrtc.org
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34
all.gyp
34
all.gyp
@ -11,19 +11,53 @@
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'include_examples%': 1,
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'include_tests%': 1,
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},
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'includes': [
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'webrtc/build/common.gypi',
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],
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'targets': [
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{
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'target_name': 'All',
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'type': 'none',
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'dependencies': [
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'webrtc/api/api.gyp:*',
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'webrtc/base/base.gyp:*',
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'webrtc/common.gyp:*',
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'webrtc/common_audio/common_audio.gyp:*',
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'webrtc/common_video/common_video.gyp:*',
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'webrtc/media/media.gyp:*',
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'webrtc/modules/modules.gyp:*',
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'webrtc/p2p/p2p.gyp:*',
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'webrtc/pc/pc.gyp:*',
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'webrtc/system_wrappers/system_wrappers.gyp:*',
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'webrtc/tools/tools.gyp:*',
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'webrtc/voice_engine/voice_engine.gyp:*',
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'webrtc/webrtc.gyp:*',
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'<(webrtc_vp8_dir)/vp8.gyp:*',
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'<(webrtc_vp9_dir)/vp9.gyp:*',
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],
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'conditions': [
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['include_tests==1', {
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'includes': [
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'webrtc/webrtc_tests.gypi',
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],
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'dependencies': [
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'webrtc/api/api_tests.gyp:*',
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'webrtc/common_video/common_video_unittests.gyp:*',
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'webrtc/system_wrappers/system_wrappers_tests.gyp:*',
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'webrtc/test/metrics.gyp:*',
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'webrtc/test/test.gyp:*',
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],
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}],
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['include_examples==1', {
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'dependencies': [
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'webrtc/webrtc_examples.gyp:*',
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],
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}],
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['(OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7"))', {
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'dependencies': [
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'webrtc/sdk/sdk.gyp:*',
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],
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}],
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['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
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'dependencies': [
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'talk/app/webrtc/legacy_objc_api.gyp:*',
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@ -6,101 +6,6 @@
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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{
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'variables': {
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'webrtc_all_dependencies': [
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'base/base.gyp:*',
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'common.gyp:*',
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'common_audio/common_audio.gyp:*',
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'common_video/common_video.gyp:*',
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'media/media.gyp:*',
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'modules/modules.gyp:*',
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'p2p/p2p.gyp:*',
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'pc/pc.gyp:*',
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'system_wrappers/system_wrappers.gyp:*',
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'tools/tools.gyp:*',
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'voice_engine/voice_engine.gyp:*',
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'<(webrtc_vp8_dir)/vp8.gyp:*',
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'<(webrtc_vp9_dir)/vp9.gyp:*',
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],
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},
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'conditions': [
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['build_with_chromium==0', {
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# TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/
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# has been moved to webrtc/. It can't be processed by Chromium since the
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# reference to buid/java.gypi is using an absolute path (and includes
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# entries cannot contain variables).
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'variables': {
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'webrtc_all_dependencies': [
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'api/api.gyp:*',
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],
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},
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}],
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['build_with_chromium==0 and'
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'(OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7"))', {
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# TODO(kjellander): Move this to webrtc_all_dependencies once all of talk/
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# has been moved to webrtc/. It can't be processed by Chromium since the
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# reference to buid/java.gypi is using an absolute path (and includes
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# entries cannot contain variables).
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'variables': {
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'webrtc_all_dependencies': [
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'sdk/sdk.gyp:*',
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],
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},
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}],
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['include_tests==1', {
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'includes': [
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'webrtc_tests.gypi',
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],
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}],
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['enable_protobuf==1', {
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'targets': [
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{
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# This target should only be built if enable_protobuf is defined
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'target_name': 'rtc_event_log_proto',
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'type': 'static_library',
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'sources': ['call/rtc_event_log.proto',],
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'variables': {
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'proto_in_dir': 'call',
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'proto_out_dir': 'webrtc/call',
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},
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'includes': ['build/protoc.gypi'],
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},
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],
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}],
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['enable_protobuf==1', {
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'targets': [
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{
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'target_name': 'rtc_event_log_parser',
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'type': 'static_library',
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'sources': [
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'call/rtc_event_log_parser.cc',
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'call/rtc_event_log_parser.h',
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],
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'dependencies': [
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'rtc_event_log_proto',
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],
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'export_dependent_settings': [
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'rtc_event_log_proto',
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],
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},
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],
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}],
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['include_tests==1 and enable_protobuf==1', {
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'targets': [
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{
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'target_name': 'rtc_event_log2rtp_dump',
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'type': 'executable',
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'sources': ['call/rtc_event_log2rtp_dump.cc',],
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'dependencies': [
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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'rtc_event_log_parser',
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'rtc_event_log_proto',
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'test/test.gyp:rtp_test_utils'
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],
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},
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],
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}],
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],
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'includes': [
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'build/common.gypi',
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'audio/webrtc_audio.gypi',
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@ -108,27 +13,6 @@
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'video/webrtc_video.gypi',
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],
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'targets': [
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{
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'target_name': 'webrtc_all',
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'type': 'none',
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'dependencies': [
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'<@(webrtc_all_dependencies)',
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'webrtc',
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],
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'conditions': [
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['include_tests==1', {
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'dependencies': [
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'api/api_tests.gyp:*',
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'common_video/common_video_unittests.gyp:*',
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'rtc_unittests',
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'system_wrappers/system_wrappers_tests.gyp:*',
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'test/metrics.gyp:*',
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'test/test.gyp:*',
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'webrtc_tests',
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],
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}],
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],
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},
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{
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'target_name': 'webrtc',
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'type': 'static_library',
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@ -185,6 +69,56 @@
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}],
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],
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},
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],
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], # targets
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'conditions': [
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['include_tests==1', {
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'includes': [
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'webrtc_tests.gypi',
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],
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}],
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['enable_protobuf==1', {
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'targets': [
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{
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# This target should only be built if enable_protobuf is defined
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'target_name': 'rtc_event_log_proto',
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'type': 'static_library',
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'sources': ['call/rtc_event_log.proto',],
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'variables': {
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'proto_in_dir': 'call',
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'proto_out_dir': 'webrtc/call',
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},
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'includes': ['build/protoc.gypi'],
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},
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{
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'target_name': 'rtc_event_log_parser',
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'type': 'static_library',
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'sources': [
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'call/rtc_event_log_parser.cc',
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'call/rtc_event_log_parser.h',
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],
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'dependencies': [
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'rtc_event_log_proto',
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],
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'export_dependent_settings': [
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'rtc_event_log_proto',
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],
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},
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],
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}],
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['include_tests==1 and enable_protobuf==1', {
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'targets': [
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{
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'target_name': 'rtc_event_log2rtp_dump',
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'type': 'executable',
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'sources': ['call/rtc_event_log2rtp_dump.cc',],
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'dependencies': [
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'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
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'rtc_event_log_parser',
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'rtc_event_log_proto',
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'test/test.gyp:rtp_test_utils'
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],
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},
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],
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}],
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], # conditions
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}
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