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Enabling 'gn check' on webrtc/sdk

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BUG=webrtc:7499

Review-Url: https://codereview.webrtc.org/2818433003
Cr-Commit-Position: refs/heads/master@{#17805}
This commit is contained in:
mbonadei
2017-04-21 00:02:02 -07:00
committed by Commit bot
parent bf34b23ab5
commit 1e060c6b0c
8 changed files with 74 additions and 31 deletions
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1
.gn
View File

@ -33,6 +33,7 @@ check_targets = [
"//webrtc/media/*",
"//webrtc/modules/*",
"//webrtc/ortc/*",
"//webrtc/sdk/*",
"//webrtc/stats/*",
"//webrtc/tools/*",
"//webrtc/test/*",

1
webrtc/pc/BUILD.gn
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@ -81,7 +81,6 @@ config("libjingle_peerconnection_warnings_config") {
}
rtc_static_library("libjingle_peerconnection") {
check_includes = false # TODO(kjellander): Remove (bugs.webrtc.org/6828)
cflags = []
sources = [
"audiotrack.cc",

71
webrtc/sdk/BUILD.gn
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@ -28,14 +28,6 @@ if (is_ios || is_mac) {
}
rtc_static_library("rtc_sdk_common_objc") {
deps = [
"../base:rtc_base",
]
configs += [
"..:common_objc",
"//build/config/compiler:enable_arc",
]
public_configs = [ ":rtc_sdk_common_objc_config" ]
sources = [
"objc/Framework/Classes/NSString+StdString.h",
"objc/Framework/Classes/NSString+StdString.mm",
@ -58,6 +50,14 @@ if (is_ios || is_mac) {
"objc/Framework/Headers/WebRTC/RTCSSLAdapter.h",
"objc/Framework/Headers/WebRTC/RTCTracing.h",
]
configs += [
"..:common_objc",
"//build/config/compiler:enable_arc",
]
public_configs = [ ":rtc_sdk_common_objc_config" ]
deps = [
"../base:rtc_base",
]
if (is_ios) {
sources += [
"objc/Framework/Classes/RTCCameraPreviewView.m",
@ -73,6 +73,7 @@ if (is_ios || is_mac) {
"objc/Framework/Classes/Metal/RTCMTLNV12Renderer.mm",
"objc/Framework/Classes/Metal/RTCMTLVideoView.m",
]
deps += [ "../api:video_frame_api" ]
}
libs = [ "AVFoundation.framework" ]
@ -194,8 +195,22 @@ if (is_ios || is_mac) {
"objc/Framework/Headers/WebRTC/RTCVideoTrack.h",
]
configs += [
"..:common_objc",
"//build/config/compiler:enable_arc",
]
public_configs = [ ":rtc_sdk_common_objc_config" ]
deps = [
":webrtc_h264_video_toolbox",
"../api:video_frame_api",
"../base:rtc_base",
"../base:rtc_base_approved",
"../common_video",
"../media:rtc_media",
"../media:rtc_media_base",
"../system_wrappers",
]
if (is_ios) {
@ -230,13 +245,6 @@ if (is_ios || is_mac) {
]
}
configs += [
"..:common_objc",
"//build/config/compiler:enable_arc",
]
public_configs = [ ":rtc_sdk_common_objc_config" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@ -276,11 +284,6 @@ if (is_ios || is_mac) {
sources += [ "objc/Framework/Classes/Metal/RTCMTLVideoView.m" ]
}
}
deps = [
":rtc_sdk_peerconnection_objc",
"//third_party/ocmock",
"//webrtc/system_wrappers:system_wrappers_default",
]
# TODO(tkchin): Cleanup this warning.
cflags = [ "-Wno-objc-property-no-attribute" ]
@ -291,6 +294,13 @@ if (is_ios || is_mac) {
ldflags = [ "-ObjC" ]
defines = [ "GTEST_RELATIVE_PATH" ]
deps = [
":rtc_sdk_peerconnection_objc",
"../base:rtc_base_tests_utils",
"//third_party/ocmock",
"//webrtc/system_wrappers:system_wrappers_default",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@ -358,6 +368,13 @@ if (is_ios || is_mac) {
"@rpath/$output_name.framework/$output_name",
]
configs += [
"..:common_objc",
"//build/config/compiler:enable_arc",
]
public_configs = [ ":rtc_sdk_common_objc_config" ]
deps = [
":rtc_sdk_peerconnection_objc",
"../system_wrappers:field_trial_default",
@ -373,13 +390,6 @@ if (is_ios || is_mac) {
"VideoToolbox.framework",
]
configs += [
"..:common_objc",
"//build/config/compiler:enable_arc",
]
public_configs = [ ":rtc_sdk_common_objc_config" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
configs -= [ "//build/config/clang:find_bad_constructs" ]
@ -413,6 +423,13 @@ if (is_ios || is_mac) {
deps = [
":rtc_sdk_common_objc",
"../api:video_frame_api",
"../base:rtc_base_approved",
"../common_video",
"../media:rtc_media_base",
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_h264",
"../system_wrappers",
]
libs = [

16
webrtc/sdk/android/BUILD.gn
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@ -69,6 +69,19 @@ rtc_static_library("libjingle_peerconnection_jni") {
}
deps = [
"../..:webrtc_common",
"../../api:video_frame_api",
"../../api/video_codecs:video_codecs_api",
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../base:rtc_task_queue",
"../../common_video:common_video",
"../../media:rtc_media",
"../../media:rtc_media_base",
"../../modules/utility:utility",
"../../modules/video_coding:video_coding_utility",
"../../system_wrappers:system_wrappers",
"../../voice_engine:voice_engine",
"//webrtc/pc:libjingle_peerconnection",
]
@ -91,6 +104,8 @@ rtc_static_library("libjingle_peerconnection_metrics_default_jni") {
configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
deps = [
":libjingle_peerconnection_jni",
"../../system_wrappers",
"//webrtc/pc:libjingle_peerconnection",
"//webrtc/system_wrappers:field_trial_default",
"//webrtc/system_wrappers:metrics_default",
@ -108,6 +123,7 @@ rtc_shared_library("libjingle_peerconnection_so") {
deps = [
":libjingle_peerconnection_jni",
":libjingle_peerconnection_metrics_default_jni",
"../../base:rtc_base",
"//webrtc/pc:libjingle_peerconnection",
]
output_extension = "so"

1
webrtc/sdk/android/src/jni/DEPS
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@ -5,4 +5,5 @@ include_rules = [
"+webrtc/modules/utility/include/jvm_android.h",
"+webrtc/modules/video_coding/utility/vp8_header_parser.h",
"+webrtc/pc",
"+webrtc/system_wrappers/include/field_trial_default.h",
]

5
webrtc/sdk/android/src/jni/peerconnection_jni.cc
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@ -74,7 +74,10 @@
#include "webrtc/sdk/android/src/jni/jni_helpers.h"
#include "webrtc/sdk/android/src/jni/native_handle_impl.h"
#include "webrtc/sdk/android/src/jni/rtcstatscollectorcallbackwrapper.h"
#include "webrtc/system_wrappers/include/field_trial_default.h"
// Adding 'nogncheck' to disable the gn include headers check.
// We don't want to depend on 'system_wrappers:field_trial_default' because
// clients should be able to provide their own implementation.
#include "webrtc/system_wrappers/include/field_trial_default.h" // nogncheck
#include "webrtc/system_wrappers/include/logcat_trace_context.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/include/voe_base.h"

5
webrtc/sdk/objc/Framework/Classes/RTCFieldTrials.mm
View File

@ -14,7 +14,10 @@
#import "WebRTC/RTCLogging.h"
#include "webrtc/system_wrappers/include/field_trial_default.h"
// Adding 'nogncheck' to disable the gn include headers check.
// We don't want to depend on 'system_wrappers:field_trial_default' because
// clients should be able to provide their own implementation.
#include "webrtc/system_wrappers/include/field_trial_default.h" // nogncheck
NSString * const kRTCFieldTrialAudioSendSideBweKey = @"WebRTC-Audio-SendSideBwe";
NSString * const kRTCFieldTrialSendSideBweWithOverheadKey = @"WebRTC-SendSideBwe-WithOverhead";

5
webrtc/sdk/objc/Framework/Classes/RTCMetricsSampleInfo+Private.h
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@ -10,7 +10,10 @@
#import "WebRTC/RTCMetricsSampleInfo.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
// Adding 'nogncheck' to disable the gn include headers check.
// We don't want to depend on 'system_wrappers:metrics_default' because
// clients should be able to provide their own implementation.
#include "webrtc/system_wrappers/include/metrics_default.h" // nogncheck
NS_ASSUME_NONNULL_BEGIN