From 602f41e2edcd0b555a492b074e1e3f7a01a89ede Mon Sep 17 00:00:00 2001 From: kjellander Date: Mon, 4 Apr 2016 23:39:45 -0700 Subject: [PATCH] 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: https://chromium.googlesource.com/external/webrtc/+/9266cc066845c5d5be6551c583dcc831a2a47b67 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} --- webrtc/build/common.gypi | 64 +--------------------------------------- webrtc/media/media.gyp | 1 + 2 files changed, 2 insertions(+), 63 deletions(-) diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi index 1afb8f695b..6ec0f1c942 100644 --- a/webrtc/build/common.gypi +++ b/webrtc/build/common.gypi @@ -263,31 +263,8 @@ ['build_with_chromium==1', { 'defines': [ # Changes settings for Chromium build. - # TODO(kjellander): Cleanup unused ones and move defines closer to the - # source when webrtc:4256 is completed. - 'ENABLE_EXTERNAL_AUTH', - 'EXPAT_RELATIVE_PATH', - 'FEATURE_ENABLE_SSL', - 'GTEST_RELATIVE_PATH', - 'HAVE_OPENSSL_SSL_H', - 'HAVE_SCTP', - 'HAVE_SRTP', - 'HAVE_WEBRTC_VIDEO', - 'HAVE_WEBRTC_VOICE', - 'LOGGING_INSIDE_WEBRTC', - 'NO_MAIN_THREAD_WRAPPING', - 'NO_SOUND_SYSTEM', - 'SRTP_RELATIVE_PATH', - 'SSL_USE_OPENSSL', - 'USE_WEBRTC_DEV_BRANCH', 'WEBRTC_CHROMIUM_BUILD', - ], - 'conditions': [ - ['OS=="win" and target_arch=="ia32"', { - 'defines': [ - '_USE_32BIT_TIME_T', - ], - }], + 'LOGGING_INSIDE_WEBRTC', ], 'include_dirs': [ # Include the top-level directory when building in Chrome, so we can @@ -462,21 +439,6 @@ }], ], }], - ['chromeos==1', { - 'defines': [ - 'CHROMEOS', - ], - }], - ['os_bsd==1', { - 'defines': [ - 'BSD', - ], - }], - ['OS=="openbsd"', { - 'defines': [ - 'OPENBSD', - ], - }], ['include_internal_audio_device==1', { 'defines': [ 'WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE', @@ -494,14 +456,6 @@ ['build_with_chromium==1', { 'defines': [ # Changes settings for Chromium build. - # TODO(kjellander): Cleanup unused ones and move defines closer to - # the source when webrtc:4256 is completed. - 'FEATURE_ENABLE_SSL', - 'FEATURE_ENABLE_VOICEMAIL', - 'EXPAT_RELATIVE_PATH', - 'GTEST_RELATIVE_PATH', - 'NO_MAIN_THREAD_WRAPPING', - 'NO_SOUND_SYSTEM', 'WEBRTC_CHROMIUM_BUILD', ], 'include_dirs': [ @@ -529,7 +483,6 @@ ['OS=="win"', { 'defines': [ 'WEBRTC_WIN', - '_CRT_SECURE_NO_WARNINGS', # Suppres warnings about _vsnprinf ], }], ['OS=="linux"', { @@ -550,21 +503,6 @@ 'WEBRTC_POSIX', ], }], - ['chromeos==1', { - 'defines': [ - 'CHROMEOS', - ], - }], - ['os_bsd==1', { - 'defines': [ - 'BSD', - ], - }], - ['OS=="openbsd"', { - 'defines': [ - 'OPENBSD', - ], - }], ], }, }, # target_defaults diff --git a/webrtc/media/media.gyp b/webrtc/media/media.gyp index 9994cfc2b3..621af832e8 100644 --- a/webrtc/media/media.gyp +++ b/webrtc/media/media.gyp @@ -81,6 +81,7 @@ 'engine/webrtccommon.h', 'engine/webrtcmediaengine.cc', 'engine/webrtcmediaengine.h', + 'engine/webrtcmediaengine.cc', 'engine/webrtcvideocapturer.cc', 'engine/webrtcvideocapturer.h', 'engine/webrtcvideocapturerfactory.h',