The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
if not on Android/iOS.
This is a re-land of https://codereview.webrtc.org/1674103002/.
The reason Chromium FYI turned red was due to deps not
being relative. See kjellander's CL:
https://codereview.webrtc.org/1681493002/.
This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.
This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).
Third time's the charm?
TBR=kjellander@webrtc.org
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1675143003
Cr-Commit-Position: refs/heads/master@{#11523}
Reason for revert:
Chromium FYI turns red.
Original issue's description:
> Default build flag |rtc_use_h264| to |proprietary_codecs|
> if not on Android/iOS.
>
> This means proprietary_codecs=1 && ffmpeg_branding=Chrome
> can be used to enable this H.264 enc/dec implementation
> instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
> This is used by both Chromium trybots (but not default
> Chromium build) and offical Chrome build, meaning we will
> be able to test and enable H.264 in chromium.
>
> This change would otherwise be enough to launch this
> feature in Chrome, but because we do not want to do that
> before we have chromium browser tests and are ready to flip
> the switch, this CL prevents chromium from using H.264 just
> yet: https://codereview.chromium.org/1641163002/ (landing
> this after that CL).
>
> Note: This is a re-land of
> https://codereview.webrtc.org/1660403004/. Reverting it
> was not necessary.
>
> TBR=kjellander@webrtc.org
> BUG=chromium:500605, chromium:468365
>
> Committed: https://crrev.com/10b9dd7ab1a8c3f80b2d2924be658e43131a4fbe
> Cr-Commit-Position: refs/heads/master@{#11517}
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1675113002
Cr-Commit-Position: refs/heads/master@{#11518}
if not on Android/iOS.
This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.
This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).
Note: This is a re-land of
https://codereview.webrtc.org/1660403004/. Reverting it
was not necessary.
TBR=kjellander@webrtc.org
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1674103002
Cr-Commit-Position: refs/heads/master@{#11517}
The files that are built when use_gtk==1 are not included in the Chromium build
(webrtc/media/devices/gtkvideorenderer.cc and webrtc/media/devices/gtkvideorenderer.h)
so to preserve previous behavior in Chromium before/after
https://codereview.webrtc.org/1587193006, this is the right thing to do.
The reason this was discovered was that a Chrome OS build started failing, since
it was lacking the gtk+2.0 package.
NOTRY=True
BUG=chromium:584722
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1677693002
Cr-Commit-Position: refs/heads/master@{#11510}
Reason for revert:
Trybots red? Don't have time to intvestigate
Original issue's description:
> Default build flag |rtc_use_h264| to |proprietary_codecs|
> if not on Android/iOS.
>
> This means proprietary_codecs=1 && ffmpeg_branding=Chrome
> can be used to enable this H.264 enc/dec implementation
> instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
> This is used by both Chromium trybots (but not default
> Chromium build) and offical Chrome build, meaning we will
> be able to test and enable H.264 in chromium.
>
> This change would otherwise be enough to launch this
> feature in Chrome, but because we do not want to do that
> before we have chromium browser tests and are ready to flip
> the switch, this CL prevents chromium from using H.264 just
> yet: https://codereview.chromium.org/1641163002/ (landing
> this after that CL).
>
> BUG=chromium:500605, chromium:468365
>
> Committed: https://crrev.com/7cd94f66ebfe5bf808d7dcb8da069d35f4a2b31a
> Cr-Commit-Position: refs/heads/master@{#11506}
TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1677623002
Cr-Commit-Position: refs/heads/master@{#11508}
if not on Android/iOS.
This means proprietary_codecs=1 && ffmpeg_branding=Chrome
can be used to enable this H.264 enc/dec implementation
instead of rtc_use_h264=1 && ffmpeg_branding=Chrome.
This is used by both Chromium trybots (but not default
Chromium build) and offical Chrome build, meaning we will
be able to test and enable H.264 in chromium.
This change would otherwise be enough to launch this
feature in Chrome, but because we do not want to do that
before we have chromium browser tests and are ready to flip
the switch, this CL prevents chromium from using H.264 just
yet: https://codereview.chromium.org/1641163002/ (landing
this after that CL).
BUG=chromium:500605, chromium:468365
Review URL: https://codereview.webrtc.org/1660403004
Cr-Commit-Position: refs/heads/master@{#11506}
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
New flag: rtc_initialize_ffmpeg, default value = !build_with_chromium.
In WebRTC standalone we initialize FFmpeg by default, in Chromium we don't by default.
Chromium is an external project that also use FFmpeg. If both projects do FFmpeg initialization code things will break. The flag makes it possible for other external projects than chromium to decide whether or not WebRTC should initialize FFmpeg.
BUG=chromium:500605, chromium:468365, webrtc:5427
Review URL: https://codereview.webrtc.org/1639273002
Cr-Commit-Position: refs/heads/master@{#11456}
It works on all platforms except Android and iOS (FFmpeg limitation).
Implemented behind compile time flags, off by default.
The plan is to have it enabled in Chrome (see bug), but not in Chromium/webrtc by default.
Flags to turn it on:
- rtc_use_h264 = true
- ffmpeg_branding = "Chrome" (or other brand that includes H.264 decoder)
Tests using H264:
- video_loopback --codec=H264
- screenshare_loopback --codec=H264
- video_engine_tests (EndToEndTest.SendsAndReceivesH264)
NOTRY=True
BUG=500605, 468365
BUG=https://bugs.chromium.org/p/webrtc/issues/detail?id=5424
Review URL: https://codereview.webrtc.org/1306813009
Cr-Commit-Position: refs/heads/master@{#11390}
That these declarations were missing was a bug, which apparently
didn't actually cause build problems in either Chromium or WebRTC
standalone. (Presumably, because rent_a_codec was always linked
together with other build targets that did declare such dependencies.)
BUG=webrtc:5435
Review URL: https://codereview.webrtc.org/1607463002
Cr-Commit-Position: refs/heads/master@{#11303}
This makes it possible to use protobuffers with
an external protobuf library instead of the one that
comes with the WebRTC code.
NOTRY=True
Review URL: https://codereview.webrtc.org/1589433002
Cr-Commit-Position: refs/heads/master@{#11236}
Defining use_third_party_h264 directly, and indirectly defining use_openh264 (from third_party/openh264) and ffmpeg_branding (from third_party/ffmpeg).
These will be used in a follow-up CL that adds an encoder and decoder implementation.
The flags are added in this CL so that they can be used by trybots/waterfall bots in GN without "Build argument had no effect" errors. Equivalent GYP changes are also added.
BUG=468365
Review URL: https://codereview.webrtc.org/1575913003
Cr-Commit-Position: refs/heads/master@{#11204}
This will make it possible to remove the build_with_libjingle=1 and key=''
GYP_DEFINES the bots are using (https://codereview.chromium.org/1450313002/).
It will also pave the road for enabling more WebRTC native tests on iOS.
BUG=webrtc:4755,webrtc:3185,webrtc:5165
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
Local compilation with:
GYP_DEFINES='OS=ios target_arch=arm' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm chromium_ios_signing=0' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=ia32' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator
GYP_DEFINES='OS=ios target_arch=x64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1457053003 .
Cr-Commit-Position: refs/heads/master@{#10711}
JNI already has jstring<->UTF8 string conversion, so using that should
save ~1mb off android binaries (ICU is *large*), probably around
300-400k after compression.
BUG=
Review URL: https://codereview.webrtc.org/1430023005
Cr-Commit-Position: refs/heads/master@{#10545}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means,
likely just the linker omitting compilation units with no incoming
references.
(This was previously landed as revisions 10046 and 10060, and got
reverted because it broke several of the Chromium FYI bots.)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368843003
Cr-Commit-Position: refs/heads/master@{#10127}
Our perf test suite webrtc_perf_tests timed out, which caused most
of the delay landing this (https://crbug.comn/535973 and
https://codereview.chromium.org/1370133004).
Other problems with executing Android tests also needed to be
resolved in order to land this (http://crbug.com/534849).
Libvpx has moved from third_party/libvpx to third_party/libvpx_new
as of https://codereview.chromium.org/1323333002/
Android GN was blocking this roll due to a problem that ended up
being caused by a bug (http://crbug.com/534849).
Relevant changes:
* src/buildtools: f7310ee..8d89c1b
* src/third_party/boringssl/src: 1d128f3..4c60d35
* src/third_party/icu: 6b3ce81..423fc7e
* src/third_party/libjpeg_turbo: 631e2dd..e4e7503
* src/third_party/libvpx: ac1772e..70db223
* src/third_party/libyuv: fcacbfb..62c49dc
* src/tools/gyp: 5d01a8c..01528c7
* src/tools/swarming_client: 77f720b..6e5d2b2
Details: 310ea93..8cf53d6/DEPS
Clang version changed 245965:247874
Details: 310ea93..8cf53d6/tools/clang/scripts/update.sh
BUG=481034, 535973
TBR=marpan@webrtc.org
Review URL: https://codereview.webrtc.org/1355083002
Cr-Commit-Position: refs/heads/master@{#10101}
Reason for revert:
Breaking Chromium FYI bots.
Original issue's description:
> Don't link with audio codecs that we don't use
>
> We used to link with all audio codecs unconditionally (except Opus);
> this patch makes gyp and gn only link to the ones that are used.
>
> (This unfortunately fails to have a measurable impact on Chromium
> binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
> fix were already being excluded from Chromium by some other means
> (likely just the linker omitting compilation units with no incoming
> references).)
>
> BUG=webrtc:4557
>
> Committed: https://crrev.com/f66a9251424351ea6d631c54dd1feb64cc13d809
> Cr-Commit-Position: refs/heads/master@{#10046}
TBR=henrik.lundin@webrtc.org,tina.legrand@webrtc.org,kjellander@webrtc.org,kwiberg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1368933002
Cr-Commit-Position: refs/heads/master@{#10069}
We used to link with all audio codecs unconditionally (except Opus);
this patch makes gyp and gn only link to the ones that are used.
(This unfortunately fails to have a measurable impact on Chromium
binary size, at least on x86_64 Linux; it turns out that iLBC and iSAC
fix were already being excluded from Chromium by some other means
(likely just the linker omitting compilation units with no incoming
references).)
BUG=webrtc:4557
Review URL: https://codereview.webrtc.org/1349393003
Cr-Commit-Position: refs/heads/master@{#10046}
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}
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}
This makes the build more flexible when linking against
prebuilt external libraries.
Use existing build_* variables for libyuv and json in talk/
(already in use in webrtc/).
Also make it possible to avoid building the GTK parts of the Linux build.
BUG=4242
R=andrew@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44179005
Cr-Commit-Position: refs/heads/master@{#9087}
Now that android_webview_build is no longer supported, remove build
conditionals referencing it and also remove the extra level of
indirection used to reference the cpufeatures target.
BUG=chromium:440793
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44119005
Patch from Richard Coles <torne@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8963}
WebRTC doesn't use or support the target_subarch
GYP variable in the same way as Chromium does, and we
currently don't build fat binaries at our buildbots.
This is needed to unblock the chromium_revision roll in
https://webrtc-codereview.appspot.com/50569004/ and it
also makes it a bit easier to build for iOS since it
is no longer needed to set target_subarch=arm64 when
target_arch=arm64 (i.e. when you build for ARM64).
TBR=tkchin@webrtc.org
BUG=4503
Review URL: https://webrtc-codereview.appspot.com/48869004
Cr-Commit-Position: refs/heads/master@{#8937}
The problem we were running into on the Mac 10.9 debug bot in Chrome turned out to be good ol'fashion memory corruption. Part of webrtc was being compiled with _DEBUG, another half without it. This caused the definition of some symbols to be out of sync (notably pthread_mutex_t) and would cause code built from common.gypi, to overwrite memory allocated via common types from base/base.gypi derived code. Fun stuff to track down. This was a problem in particular with base/criticalsection.h since it's inlined into multiple object files but will have different definitions of what a mutex is.
TBR=pbos,kjellander
BUG=
Review URL: https://webrtc-codereview.appspot.com/43659004
Cr-Commit-Position: refs/heads/master@{#8646}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8646 4adac7df-926f-26a2-2b94-8c16560cd09d
For build_with_chromium==1 the includes will be the same.
For build_with_chromium==0 the <(DEPTH) variable is replaced by ../..
which should be the same in all common use cases.
This change makes the include paths for all GYP targets
more similar to the setup in the
direct_dependent_settings section further down.
BUG=4185
TESTED=Trybots + build in Chromium with third_party/webrtc patched with this CL.
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41739004
Cr-Commit-Position: refs/heads/master@{#8219}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8219 4adac7df-926f-26a2-2b94-8c16560cd09d
