Reason for revert:
I suspect that this CL breaks Chromium WebRTC FYI bots. (Thanks kjellander@ for spotting.) The added dep in BUILD.gn would be the problem.
Example:
https://luci-logdog.appspot.com/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F15058%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
FAILED: newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o
/b/c/goma_client/gomacc ../../native_client/toolchain/linux_x86/pnacl_newlib/bin/pnacl-clang++ -MMD -MF newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o.d -DNACL_TC_REV=5dfe030a71ca66e72c5719ef5034c2ed24706c43 -DV8_DEPRECATION_WARNINGS -DUSE_OPENSSL_CERTS=1 -DNO_TCMALLOC -DFULL_SAFE_BROWSING -DSAFE_BROWSING_CSD -DSAFE_BROWSING_DB_LOCAL -DCHROMIUM_BUILD -DENABLE_MEDIA_ROUTER=1 -DFIELDTRIAL_TESTING_ENABLED -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -D_FORTIFY_SOURCE=2 -DNDEBUG -DNVALGRIND -DWEBRTC_RESTRICT_LOGGING -DEXPAT_RELATIVE_PATH -DHAVE_SCTP -DENABLE_EXTERNAL_AUTH -DHAVE_WEBRTC_VIDEO -DHAVE_WEBRTC_VOICE -DLOGGING_INSIDE_WEBRTC -DUSE_WEBRTC_DEV_BRANCH -DFEATURE_ENABLE_VOICEMAIL -DEXPAT_RELATIVE_PATH -DGTEST_RELATIVE_PATH -DNO_MAIN_THREAD_WRAPPING -DNO_SOUND_SYSTEM -DWEBRTC_CHROMIUM_BUILD -DWEBRTC_POSIX -I../.. -Inewlib_pnacl/gen -I../../third_party/webrtc_overrides -I../../third_party -fno-strict-aliasing -Wno-builtin-macro-redefined -D__DATE__= -D__TIME__= -D__TIMESTAMP__= -fcolor-diagnostics -Wall -Werror -Wextra -Wno-missing-field-initializers -Wno-unused-parameter -Wno-c++11-narrowing -Wno-covered-switch-default -Wno-deprecated-register -Wno-unneeded-internal-declaration -Wno-inconsistent-missing-override -O2 -fno-ident -fdata-sections -ffunction-sections -g0 -fvisibility=hidden -fvisibility-inlines-hidden -std=gnu++11 -fno-rtti -fno-exceptions -c ../../third_party/webrtc/api/mediaconstraintsinterface.cc -o newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o
In file included from ../../third_party/webrtc/api/mediaconstraintsinterface.cc:11:
In file included from ../../third_party/webrtc/api/mediaconstraintsinterface.h:27:
In file included from ../../third_party/webrtc/api/peerconnectioninterface.h:77:
In file included from ../../third_party/webrtc/api/dtmfsenderinterface.h:16:
In file included from ../../third_party/webrtc/api/mediastreaminterface.h:33:
In file included from ../../third_party/webrtc/media/base/mediachannel.h:28:
../../third_party/webrtc/base/socket.h:18:10: fatal error: 'sys/socket.h' file not found
#include <sys/socket.h>
^
1 error generated.
Original issue's description:
> Add DesktopCapturerId and attach it to DesktopFrame
>
> This change adds a DesktopCapturerId namespace, and attaches an int to each
> DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
> this field to differentiate themselves.
>
> BUG=679523, 650926
>
> Review-Url: https://codereview.webrtc.org/2759493002
> Cr-Commit-Position: refs/heads/master@{#17329}
> Committed: 41e3d9ff3b
TBR=sergeyu@chromium.org,zijiehe@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2767003002
Cr-Commit-Position: refs/heads/master@{#17336}
Revert of Add DesktopCapturerId and attach it to DesktopFrame (patchset #4 id:100002 of https://codereview.webrtc.org/2759493002/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Revert of PyLint fixes for tools-webrtc and webrtc/tools (patchset #3 id:40001 of https://codereview.webrtc.org/2736233003/ )
…
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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