From eda3a9b58295f436665ac39a9f66864d01f7ab96 Mon Sep 17 00:00:00 2001 From: kjellander Date: Thu, 2 Mar 2017 22:27:29 -0800 Subject: [PATCH] Revert of Enable GN check for webrtc/examples (patchset #7 id:120001 of https://codereview.webrtc.org/2714343002/ ) Reason for revert: I wasn't able to resolve it with that CL so I'll have to revert this by now. Will have another look at this when time permits. Original issue's description: > Enable GN check for webrtc/examples > > BUG=webrtc:6828 > NOTRY=True > > Review-Url: https://codereview.webrtc.org/2714343002 > Cr-Commit-Position: refs/heads/master@{#16987} > Committed: https://chromium.googlesource.com/external/webrtc/+/81db74a3841b42c0a84c2f35b91eab66fd3f4e79 TBR=perkj@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:6828 Review-Url: https://codereview.webrtc.org/2727253003 Cr-Commit-Position: refs/heads/master@{#16991} --- .gn | 1 - webrtc/examples/BUILD.gn | 60 ++++++++-------------------------------- 2 files changed, 11 insertions(+), 50 deletions(-) diff --git a/.gn b/.gn index ef7acf153a..2070fa86d3 100644 --- a/.gn +++ b/.gn @@ -27,7 +27,6 @@ check_targets = [ "//webrtc/call/*", "//webrtc/common_video/*", "//webrtc/common_audio/*", - "//webrtc/examples/*", "//webrtc/modules/audio_coding/*", "//webrtc/modules/audio_conference_mixer/*", "//webrtc/modules/audio_device/*", diff --git a/webrtc/examples/BUILD.gn b/webrtc/examples/BUILD.gn index 74d340fa1e..d575c65b09 100644 --- a/webrtc/examples/BUILD.gn +++ b/webrtc/examples/BUILD.gn @@ -50,7 +50,6 @@ group("examples") { if (is_android) { android_apk("AppRTCMobile") { - testonly = true apk_name = "AppRTCMobile" android_manifest = "androidapp/AndroidManifest.xml" @@ -65,7 +64,6 @@ if (is_android) { } android_library("AppRTCMobile_javalib") { - testonly = true android_manifest = "androidapp/AndroidManifest.xml" java_files = [ @@ -103,7 +101,6 @@ if (is_android) { } android_resources("AppRTCMobile_resources") { - testonly = true resource_dirs = [ "androidapp/res" ] custom_package = "org.appspot.apprtc" } @@ -172,7 +169,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) { } rtc_static_library("apprtc_common") { - testonly = true sources = [ "objc/AppRTCMobile/common/ARDUtilities.h", "objc/AppRTCMobile/common/ARDUtilities.m", @@ -184,9 +180,9 @@ if (is_ios || (is_mac && target_cpu != "x86")) { public_configs = [ ":apprtc_common_config" ] deps = [ - "//webrtc/sdk:rtc_sdk_common_objc", - "//webrtc/system_wrappers:field_trial_default", - "//webrtc/system_wrappers:metrics_default", + "../sdk:rtc_sdk_common_objc", + "../system_wrappers:field_trial_default", + "../system_wrappers:metrics_default", ] } @@ -203,7 +199,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) { } rtc_static_library("apprtc_signaling") { - testonly = true sources = [ "objc/AppRTCMobile/ARDAppClient+Internal.h", "objc/AppRTCMobile/ARDAppClient.h", @@ -250,14 +245,13 @@ if (is_ios || (is_mac && target_cpu != "x86")) { ":socketrocket", ] public_deps = [ - "//webrtc/sdk:rtc_sdk_peerconnection_objc", + "../sdk:rtc_sdk_peerconnection_objc", ] libs = [ "QuartzCore.framework" ] } if (is_ios) { rtc_static_library("AppRTCMobile_lib") { - testonly = true sources = [ "objc/AppRTCMobile/ios/ARDAppDelegate.m", "objc/AppRTCMobile/ios/ARDMainView.h", @@ -289,12 +283,10 @@ if (is_ios || (is_mac && target_cpu != "x86")) { deps = [ ":apprtc_common", ":apprtc_signaling", - "//webrtc/modules/audio_device", ] } ios_app_bundle("AppRTCMobile") { - testonly = true sources = [ "objc/AppRTCMobile/ios/main.m", ] @@ -354,7 +346,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) { if (is_mac) { rtc_static_library("AppRTCMobile_lib") { - testonly = true sources = [ "objc/AppRTCMobile/mac/APPRTCAppDelegate.h", "objc/AppRTCMobile/mac/APPRTCAppDelegate.m", @@ -362,7 +353,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) { "objc/AppRTCMobile/mac/APPRTCViewController.m", ] configs += [ - "//webrtc:common_objc", + "..:common_objc", "//build/config/compiler:enable_arc", ] deps = [ @@ -372,7 +363,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) { } mac_app_bundle("AppRTCMobile") { - testonly = true output_name = "AppRTCMobile" sources = [ @@ -413,7 +403,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) { } rtc_static_library("socketrocket") { - testonly = true sources = [ "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h", "objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m", @@ -434,7 +423,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) { # TODO(kthelgason): compile xctests on mac when chromium supports it. if (is_ios) { rtc_source_set("apprtcmobile_test_sources") { - testonly = true include_dirs = [ "objc/AppRTCMobile", "objc/AppRTCMobile/ios", @@ -445,9 +433,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) { "objc/AppRTCMobile/tests/ARDSDPUtils_xctest.mm", "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm", ] - deps = [ - "//webrtc/base:rtc_base", - ] public_deps = [ ":AppRTCMobile_ios_frameworks", ":AppRTCMobile_lib", @@ -502,7 +487,6 @@ if (is_linux || is_win) { } rtc_executable("peerconnection_client") { - testonly = true sources = [ "peerconnection/client/conductor.cc", "peerconnection/client/conductor.h", @@ -516,7 +500,6 @@ if (is_linux || is_win) { # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] } - deps = [] if (is_win) { sources += [ "peerconnection/client/flagdefs.h", @@ -526,8 +509,13 @@ if (is_linux || is_win) { ] cflags = [ "/wd4245" ] configs += [ "//build/config/win:windowed" ] - deps += [ "//webrtc/media:rtc_media_base" ] } + deps = [ + "//third_party/libyuv", + "//webrtc/pc:libjingle_peerconnection", + "//webrtc/system_wrappers:field_trial_default", + "//webrtc/system_wrappers:metrics_default", + ] if (is_linux) { sources += [ "peerconnection/client/linux/main.cc", @@ -543,26 +531,12 @@ if (is_linux || is_win) { deps += [ "//build/config/linux/gtk" ] } configs += [ ":peerconnection_client_warnings_config" ] - - deps += [ - "//third_party/libyuv", - "//webrtc/api:libjingle_peerconnection_test_api", - "//webrtc/api:video_frame_api", - "//webrtc/base:rtc_base", - "//webrtc/base:rtc_base_approved", - "//webrtc/media:rtc_media", - "//webrtc/modules/video_capture:video_capture_module", - "//webrtc/pc:libjingle_peerconnection", - "//webrtc/system_wrappers:field_trial_default", - "//webrtc/system_wrappers:metrics_default", - ] if (rtc_build_json) { deps += [ "//third_party/jsoncpp" ] } } rtc_executable("peerconnection_server") { - testonly = true sources = [ "peerconnection/server/data_socket.cc", "peerconnection/server/data_socket.h", @@ -583,14 +557,11 @@ if (is_linux || is_win) { } } rtc_executable("relayserver") { - testonly = true sources = [ "relayserver/relayserver_main.cc", ] deps = [ - "../base:rtc_base", "//webrtc/base:rtc_base_approved", - "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", @@ -601,14 +572,11 @@ if (is_linux || is_win) { } } rtc_executable("turnserver") { - testonly = true sources = [ "turnserver/turnserver_main.cc", ] deps = [ - "../base:rtc_base", "//webrtc/base:rtc_base_approved", - "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", @@ -619,14 +587,11 @@ if (is_linux || is_win) { } } rtc_executable("stunserver") { - testonly = true sources = [ "stunserver/stunserver_main.cc", ] deps = [ - "../base:rtc_base", "//webrtc/base:rtc_base_approved", - "//webrtc/p2p:rtc_p2p", "//webrtc/pc:rtc_pc", "//webrtc/system_wrappers:field_trial_default", "//webrtc/system_wrappers:metrics_default", @@ -641,7 +606,6 @@ if (is_linux || is_win) { if (!build_with_chromium) { # Doesn't build within Chrome on Win. rtc_executable("stun_prober") { - testonly = true sources = [ "stunprober/main.cc", ] @@ -653,8 +617,6 @@ if (!build_with_chromium) { } deps = [ - "../base:rtc_base", - "../base:rtc_base_approved", "../p2p:libstunprober", "../p2p:rtc_p2p", "../system_wrappers:field_trial_default",