This is needed because the target is defined in webrtc/common.gyp
and its current location crosses package boundaries when generating
projects for some build systems.
NOTRY=True
Review URL: https://codereview.webrtc.org/1665603003
Cr-Commit-Position: refs/heads/master@{#11496}
Reason for revert:
Reverting due to problem with roll:
/b/build/slave/linux/build/src/buildtools/linux64/gn gen //out/Release '--args=ffmpeg_branding="Chrome" proprietary_codecs=true is_debug=false is_component_build=false use_goma=true goma_dir="/b/build/goma" symbol_level=1 dcheck_always_on=true' --check --runtime-deps-list-file=/b/build/slave/linux/build/src/out/Release/runtime_deps
-> returned 1
ERROR at //third_party/webrtc/BUILD.gn:245:18: Item not found
configs -= [ "//build/config/clang:find_bad_constructs" ]
^-----------------------------------------
You were trying to remove "//build/config/clang:find_bad_constructs"
from the list but it wasn't there.
GN gen failed: 1
step returned non-zero exit code: 1
@@@STEP_FAILURE@@@
Original issue's description:
> Use an explicit identifier in Config
>
> This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
>
> Committed: https://crrev.com/25249d92d3cf105bcc7b684c8924ccdbc9afcb93
> Cr-Commit-Position: refs/heads/master@{#11231}
TBR=henrik.lundin@webrtc.org,stefan@webrtc.org,tommi@chromium.org,aluebs@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1586563003
Cr-Commit-Position: refs/heads/master@{#11239}
This let's us use them to configure them when using WebRTC as an external library. One use case where this is necessary is in the Android OS.
Review URL: https://codereview.webrtc.org/1538643004
Cr-Commit-Position: refs/heads/master@{#11231}
Also removes listing of targets in webrtc_fuzzers which is very prone to
not being up to date. They're not required for ClusterFuzz integration
or building locally. This also means that adding fuzzers won't require
approval outside the fuzzers directory.
BUG=webrtc:4771
R=kjellander@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1518973003 .
Cr-Commit-Position: refs/heads/master@{#11067}
This change adds fuzzer tests for iLBC, iSAC fix and float, and
Opus. The fuzzer function takes a random input vector and splits it
into a number of payloads. The lengths of the payloads is also
determined by the random vector. The payloads are decoded with the
decoders.
BUG=webrtc:5306
R=kjellander@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1499093002 .
Cr-Commit-Position: refs/heads/master@{#10932}
This reverts commit f054819e257a4f9cbb7fa82ba51dc2335f4359ec,
2d3747de9b7c3014e106d3766dc07cf5da3e1881 and
7ef0553c85c5b373535d7f6161e9a6d3b5b9a826.
It seems harder than expected to get a GN build for rtc_sound
and we lack sufficient trybot support for the case where
WebRTC is built as part of Chromium.
The Debug builds failed like this:
[6939/7454] SOLINK ./libcontent.so
FAILED: ../../third_party/llvm-build/Release+Asserts/bin/clang++ -shared -Wl,--fatal-warnings -fPIC -Wl,-z,noexecstack -Wl,-z,now -Wl,-z,relro -Wl,-z,defs -B../../third_party/binutils/Linux_x64/Release/bin -fuse-ld=gold -Wl,--icf=all -pthread -m64 -Wl,--export-dynamic -o ./libcontent.so -Wl,-soname=libcontent.so @./libcontent.so.rsp && { readelf -d ./libcontent.so | grep SONAME ; nm -gD -f p ./libcontent.so | cut -f1-2 -d' '; } > ./libcontent.so.tmp && if ! cmp -s ./libcontent.so.tmp ./libcontent.so.TOC; then mv ./libcontent.so.tmp ./libcontent.so.TOC; fi
../../third_party/webrtc/sound/alsasoundsystem.cc:453: error: undefined reference to 'rtc::LateBindingSymbolTable::Load()'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.h.def:62: error: undefined reference to 'rtc::LateBindingSymbolTable::IsLoaded() const'
../../third_party/webrtc/base/latebindingsymboltable.cc.def:63: error: undefined reference to 'rtc::LateBindingSymbolTable::LateBindingSymbolTable(rtc::LateBindingSymbolTable::TableInfo const*, void**)'
../../third_party/webrtc/base/latebindingsymboltable.cc.def:65: error: undefined reference to 'rtc::LateBindingSymbolTable::~LateBindingSymbolTable()'
clang: error: linker command failed with exit code 1 (use -v to see invocation)
ninja: build stopped: subcommand failed.
BUG=webrtc:4160
TBR=tfarina@chromium.org
Review URL: https://codereview.webrtc.org/1407893005 .
Cr-Commit-Position: refs/heads/master@{#10411}
Tested on Linux with the following command lines:
$ gn gen out-gn/Release --args='is_debug=false target_cpu="x64"
build_with_chromium=false'
$ ninja -C out-gn/Release rtc_sound
BUG=webrtc:4160
R=kjellander@webrtc.org
Review URL: https://codereview.webrtc.org/1425583002
Cr-Commit-Position: refs/heads/master@{#10405}
Reason for revert:
Breaks FYI bots.
ninja: error: '../../third_party/webrtc_overrides/webrtc/base/logging.cc', needed by 'obj/third_party/webrtc_overrides/webrtc/base/rtc_base.logging.o', missing and no known rule to make it
Original issue's description:
> Update build files to use webrtc_overrides in Chromium instead of overrides.
>
> This is a part of moving the overrides to Chromium. See bug comment #65 for all steps.
>
> BUG=chromium:468375
>
> Committed: https://crrev.com/baae0a8a6c873ddf812a5687b84638359b2e7e5b
> Cr-Commit-Position: refs/heads/master@{#9996}
TBR=kjellander@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:468375
Review URL: https://codereview.webrtc.org/1352423002
Cr-Commit-Position: refs/heads/master@{#9998}
This is a part of moving the overrides to Chromium. See bug comment #65 for all steps.
BUG=chromium:468375
Review URL: https://codereview.webrtc.org/1354933002
Cr-Commit-Position: refs/heads/master@{#9996}
The disabling of the sin,cos,sinf,cosf functions had the wrong
condition for GN. This fixes that and also makes the condition
in common.gypi a bit more readable.
BUG=
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1307633008 .
Cr-Commit-Position: refs/heads/master@{#9871}
Re-lands "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module."
This reverts commit b933667a7f97697d6390d1eee5f378cedd9ca208.
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1259683003 .
Cr-Commit-Position: refs/heads/master@{#9661}
Placed the protobuf structures in the namespace webrtc::rtclog. Removed the message field from the DebugEvent structure, since it was not used.
Added an interface to set config information for VideoReceiveStream and VideoSendStream in the event log.
Added function to log full RTCP packets and changed RTP-logging to only log headers.
Significantly extended the unit tests for RtcEventLog.
R=ivoc@webrtc.org, minyue@webrtc.org, pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1230973005 .
Cr-Commit-Position: refs/heads/master@{#9656}
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
Merge WEBRTC_ARCH_ARM64_NEON and WEBRTC_ARCH_ARM_NEON into one
WEBRTC_HAS_NEON.
Replace WEBRTC_DETECT_ARM_NEON by WEBRTC_DETECT_NEON.
Replace WEBRTC_ARCH_ARM by WEBRTC_ARCH_ARM64 for arm64 cpu.
BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org
Change-Id: I870a4d0682b80633b671c9aab733153f6d95a980
Review URL: https://webrtc-codereview.appspot.com/49309004
Cr-Commit-Position: refs/heads/master@{#9228}
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
Mostly, it's about moving constructors and descructors to the .cc
files, so that they won't be inlined everywhere.
The reason this CL is so big is that a lot of code was using
common_types.h without declaring a dependency on webrtc_common, which
broke the build once common_types.h started to depend on
common_types.cc.
BUG=163
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26089004
Cr-Commit-Position: refs/heads/master@{#8516}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8516 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.
Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)
GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture
video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render
BUG=456815
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35099004
Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
Allows successful build of arm64 libraries using
GYP_DEFINES="OS=ios target_arch=arm64 target_subarch=arm64".
Note that not all libraries will be NEON optimized (eg common_audio),
however most importantly libvpx will be. WEBRTC_ARCH_ARM needs to be
defined so that libvpx doesn't post-process, which is significantly
detrimental to performance.
BUG=3898
R=kjellander@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7573 4adac7df-926f-26a2-2b94-8c16560cd09d
Libvpx now supports GN and this CL turns on compiling it.
I also introduced an executable target 'webrtc_tests'
that depends on all in WeBRTC + tests in order to get a full
linking step executed (since we've seen link problems for GN
when rolling WebRTC into Chromium).
I also converted a few test targets and made a GN file for
third_party/gflags.
BUG=3441
TESTED=Trybots + full Chromium build with a symlinked src/third_party/webrtc
dir to a workspace wit this CL applied.
R=brettw@chromium.orgTBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25569004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7344 4adac7df-926f-26a2-2b94-8c16560cd09d
LOGGING_INSIDE_WEBRTC was being set in the inherited config, whereas in the GYP build this define is not inherited. This caused duplicate logging macros to be defined in Chrome files dependening on WebRTC targets.
Move LOGGING_INSIDE_WEBRTC to the common config (non-inherited).
TBR=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/25449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7122 4adac7df-926f-26a2-2b94-8c16560cd09d
Fixes the following failure for mips:
"ERROR at //third_party/webrtc/BUILD.gn:136:7: Undefined variable for +=.
cflags += [ "-mhard-float" ]
^-----
I don't have something with this name in scope now."
BUG=3441
TEST=In Chromium. Passing compile locally on Linux using:
gn gen out-gn/mips --args="is_debug=false os=\"android\" cpu_arch=\"mipsel\"" --verbose && ninja -C out-gn/mips all
gn gen out-gn/arm --args="is_debug=false os=\"android\" cpu_arch=\"arm\"" --verbose && ninja -C out-gn/arm all
gn gen out-gn/x86-linux --args="is_debug=false os=\"linux\"" --verbose && ninja -C out-gn/x86-linux webrtc
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/15349004
Patch from Gordana Cmiljanovic <Gordana.Cmiljanovic@imgtec.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7063 4adac7df-926f-26a2-2b94-8c16560cd09d
Now that WebRTC has rolled the chromium_revision past
http://crrev.com/284372 in r6784, clang has become the
default compiler. Since WebRTC standalone code doesn't
yet compile the Chromium Clang plugins enabled, this CL
disables them for the parts of the code that doesn't yet pass
compilation with them enabled.
The buildbots are using Goma which is not yet switched
over to Clang by default. That's why they're not red yet.
BUG=163
TEST=Passing compile locally on Linux using:
gn gen out/Debug --args="build_with_chromium=false is_debug=true" && ninja
-C out/Debug
gn gen out/Release --args="build_with_chromium=false is_debug=false" && ninja
-C out/Release
gn gen out/Default --args="build_with_chromium=false os=\"android\" cpu_arch=\"arm\" arm_version=7" && ninja -C out/Default
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/16279004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6966 4adac7df-926f-26a2-2b94-8c16560cd09d
Most WebRTC source files are using full paths for includes which
requires the root to be in the include path.
This is currently handled in the common_inherited_config config in
webrtc/BUILD.gn: the .. include_dir.
However, when built from Chromium, the include
paths are not inherited in the same way when building the all target.
Building the 'webrtc' target of Chrome works without the changes
in this CL, but the default target fails.
BUG=3441
TEST=Built the default target from a Chromium checkout with
https://codereview.chromium.org/321313006/ applied and
src/third_party/webrtc linked to the webrtc folder of the WebRTC
workspace.
R=brettw@chromium.org
Review URL: https://webrtc-codereview.appspot.com/15989004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6670 4adac7df-926f-26a2-2b94-8c16560cd09d
Refactor webrtc/base/BUILD.gn to not have any subtracted
source entries.
Also fix an error in webrtc/BUILD.gn that occurs when running
on Chormium trybots as a part of enabling WebRTC for GN in
https://codereview.chromium.org/321313006/
The error is that pkg-config for dbus-glib fails. Workaround
this by putting the pkg-config entry within the proper condition.
BUG=webrtc:3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default
I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc
R=brettw@chromium.org, henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/16759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6560 4adac7df-926f-26a2-2b94-8c16560cd09d
This should work as a foundation for all the work that is
left to do to make the parts of WebRTC that Chromium uses
to build with GN.
I implemented some the smaller modules myself in this CL.
The remaining work (TODO's in the .gn files) will be distributed
to various team members.
I'm adding myself to OWNERS files for BUILD.gn files in all the
directories where I'm adding a BUILD.gn file.
BUG=3441
TEST=
Successful compilation of WebRTC as standalone:
gn gen out/Default --args="build_with_chromium=false" && ninja -C out/Default
gn gen out/Default --args="build_with_chromium=false is_clang=true clang_use_chrome_plugins=false" && ninja -C out/Default
I built successfully from a Chromium checkout (with
https://codereview.chromium.org/321313006/ applied) using:
gn gen out/Default && ninja -C out/Default webrtc
R=brettw@chromium.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13749004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6523 4adac7df-926f-26a2-2b94-8c16560cd09d
