Revert of Set defines for Chromium build. (patchset #3 id:40001 of https://codereview.webrtc.org/1847013002/ )
Reason for revert:
This breaks remoting_unittests on Windows in Chromium:
[5116:2536:0404/012329:5457156:ERROR:webrtcsession.cc(1388)] ConnectDataChannel called when data_channel_ is NULL.
[5116:2536:0404/012329:5457187:ERROR:opensslidentity.cc(154)] Generating certificate: error:0c000071:ASN.1 encoding routines:OPENSSL_internal:ERROR_GETTING_TIME
[5116:2536:0404/012329:5457218:ERROR:opensslidentity.cc(154)] Generating certificate: error:0c000071:ASN.1 encoding routines:OPENSSL_internal:ERROR_GETTING_TIME
[5116:2536:0404/012329:5457218:WARNING:dtlsidentitystore.cc(221)] Failed to generate DTLS identity.
[
Original issue's description:
> Set defines for Chromium build.
>
> Copy the defines from the target_defaults section of Chromium's
> src/third_party/libjingle.gyp into our webrtc/build/common.gypi
> in order to ensure the same defines are used for the Chromium build
> when removing the source listings in src/third_party/libjingle.gyp.
> With this CL landed, it should be possible to replace them with
> dependencies on:
> * webrtc/api/api.gyp:libjingle_peerconnections
> * webrtc/media/media.gyp:rtc_media
> * webrtc/pc/pc.gyp:rtc_pc
> * webrtc/pp2/p2p.gyp:rtc_p2p
> * webrtc/libjingle/xmpp/xmpp.gyp:rtc_xmpp
>
> Not ported (Windows specific):
> * Precompiled headers (build/win_precompile.gypi):
> since it only seems to offer a compile speedup. Will be landed
> for all of WebRTC in separate CL.
>
> BUG=webrtc:4256
> NOTRY=True
> R=perkj@webrtc.org, tommi@webrtc.org
>
> Committed: 9266cc0668
TBR=perkj@webrtc.org,tommi@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4256
Review URL: https://codereview.webrtc.org/1861603002
Cr-Commit-Position: refs/heads/master@{#12229}
This commit is contained in:
parent
de81ea8524
commit
602f41e2ed
@ -263,31 +263,8 @@
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['build_with_chromium==1', {
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'defines': [
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# Changes settings for Chromium build.
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# TODO(kjellander): Cleanup unused ones and move defines closer to the
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# source when webrtc:4256 is completed.
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'ENABLE_EXTERNAL_AUTH',
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'EXPAT_RELATIVE_PATH',
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'FEATURE_ENABLE_SSL',
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'GTEST_RELATIVE_PATH',
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'HAVE_OPENSSL_SSL_H',
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'HAVE_SCTP',
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'HAVE_SRTP',
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'HAVE_WEBRTC_VIDEO',
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'HAVE_WEBRTC_VOICE',
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'LOGGING_INSIDE_WEBRTC',
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'NO_MAIN_THREAD_WRAPPING',
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'NO_SOUND_SYSTEM',
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'SRTP_RELATIVE_PATH',
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'SSL_USE_OPENSSL',
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'USE_WEBRTC_DEV_BRANCH',
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'WEBRTC_CHROMIUM_BUILD',
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],
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'conditions': [
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['OS=="win" and target_arch=="ia32"', {
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'defines': [
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'_USE_32BIT_TIME_T',
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],
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}],
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'LOGGING_INSIDE_WEBRTC',
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],
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'include_dirs': [
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# Include the top-level directory when building in Chrome, so we can
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@ -462,21 +439,6 @@
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}],
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],
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}],
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['chromeos==1', {
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'defines': [
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'CHROMEOS',
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],
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}],
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['os_bsd==1', {
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'defines': [
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'BSD',
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],
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}],
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['OS=="openbsd"', {
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'defines': [
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'OPENBSD',
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],
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}],
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['include_internal_audio_device==1', {
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'defines': [
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'WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE',
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@ -494,14 +456,6 @@
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['build_with_chromium==1', {
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'defines': [
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# Changes settings for Chromium build.
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# TODO(kjellander): Cleanup unused ones and move defines closer to
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# the source when webrtc:4256 is completed.
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'FEATURE_ENABLE_SSL',
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'FEATURE_ENABLE_VOICEMAIL',
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'EXPAT_RELATIVE_PATH',
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'GTEST_RELATIVE_PATH',
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'NO_MAIN_THREAD_WRAPPING',
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'NO_SOUND_SYSTEM',
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'WEBRTC_CHROMIUM_BUILD',
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],
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'include_dirs': [
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@ -529,7 +483,6 @@
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['OS=="win"', {
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'defines': [
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'WEBRTC_WIN',
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'_CRT_SECURE_NO_WARNINGS', # Suppres warnings about _vsnprinf
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],
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}],
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['OS=="linux"', {
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@ -550,21 +503,6 @@
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'WEBRTC_POSIX',
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],
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}],
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['chromeos==1', {
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'defines': [
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'CHROMEOS',
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],
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}],
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['os_bsd==1', {
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'defines': [
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'BSD',
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],
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}],
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['OS=="openbsd"', {
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'defines': [
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'OPENBSD',
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],
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}],
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],
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},
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}, # target_defaults
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@ -81,6 +81,7 @@
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'engine/webrtccommon.h',
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'engine/webrtcmediaengine.cc',
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'engine/webrtcmediaengine.h',
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'engine/webrtcmediaengine.cc',
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'engine/webrtcvideocapturer.cc',
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'engine/webrtcvideocapturer.h',
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'engine/webrtcvideocapturerfactory.h',
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