Commit Graph

101 Commits

Author SHA1 Message Date
931e6583b2 Remove unnecessary dependencies for voe when building with include_internal_audio_device==0.
In particular and practical terms, this avoids pulling in AudioDeviceModuleImpl and associated classes, in Chrome.

BUG=
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49999004

Cr-Commit-Position: refs/heads/master@{#9229}
2015-05-20 07:44:23 +00:00
cb7f8ce2df Clear ARM NEON flag
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}
2015-05-20 05:20:04 +00:00
b26198972c Adding support for OpenSL ES output in native WebRTC
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}
2015-05-18 14:49:04 +00:00
e6cefb60f8 GYP variables for building expat, icu, libsrtp, usrsctp
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}
2015-04-27 12:38:37 +00:00
d417c93c10 Remove android_webview_build conditions.
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}
2015-04-09 15:36:13 +00:00
3b71efc56e Revert "Explicitly set target_subarch for iOS."
This reverts commit f6b7265c6bd254b1558b8e04a2c6d2c1769ef4a6.

Reason: Did not have the desired effect of resolving
https://webrtc-codereview.appspot.com/50569004/ due to incorrect
local testing. I suspect these lines need to be at another
variables-level.

TBR=tkchin@webrtc.org
BUG=4503

Review URL: https://webrtc-codereview.appspot.com/47989004

Cr-Commit-Position: refs/heads/master@{#8939}
2015-04-07 14:23:36 +00:00
f6b7265c6b Explicitly set target_subarch for iOS.
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}
2015-04-07 14:06:18 +00:00
7c112f3e5a Adding build_opus as a switch in GYP.
This is to allow not building Opus. On non-chromium non-gyp chases, one can let WebRTC depend on other Opus builds.

BUG=
R=kjellander@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43739004

Cr-Commit-Position: refs/heads/master@{#8754}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8754 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-17 14:05:18 +00:00
d2c09dd339 Make building openmax_dl conditional in gyp.
Intentionally not modifying the GN build.

R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48479004

Cr-Commit-Position: refs/heads/master@{#8688}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8688 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-11 22:07:18 +00:00
a32f064e97 Fix build configuration bug with debug builds.
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
2015-03-08 07:39:22 +00:00
6dab6d700d Let Chromium declare the mips_dsp_rev build variable.
In https://codereview.chromium.org/883253003, the mips_dsp_rev
build variable is added to Chromium's GYP and GN build files. Remove
the declarations of mips_dsp_rev from WebRTC's GYP and GN build files.

Replace mips_fpu with mips_float_abi and remove the compiler flags that
are already set by Chromium.

The main review of this was done in https://webrtc-codereview.appspot.com/39779004
but since that CL wasn't created with the right base URL, I made
this in order to be able to run WebRTC trybots properly.

BUG=446234
TBR=wtc@chromium.org

Review URL: https://webrtc-codereview.appspot.com/44549004

Cr-Commit-Position: refs/heads/master@{#8590}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8590 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-04 09:51:17 +00:00
e8f50df6b9 Remove avi recorder and corresponding enable_video flags.
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42099004

Cr-Commit-Position: refs/heads/master@{#8554}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8554 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-02 13:07:44 +00:00
4b3618c7f3 Remove TraceImpl logging thread.
Simplifies TraceImpl significantly. Chromium uses trace callbacks and
logs directly through the trace callback interface. Added slowdowns when
logging to file are expected to only affect test targets.

BUG=
R=andresp@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34329004

Cr-Commit-Position: refs/heads/master@{#8529}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8529 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-27 15:05:50 +00:00
2b69eab077 Restructure GYP for vp9, opus and direct trace
This is needed to make the build more flexible for some use cases.

BUG=4185
R=andresp@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34099004

Cr-Commit-Position: refs/heads/master@{#8290}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8290 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-09 10:01:40 +00:00
f2ec814e0f Move use of DEPTH into build_with_chromium==1.
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
2015-01-30 14:55:20 +00:00
7d2b6a9346 Enable Clang warning implicit-fallthrough and annotate the code.
BUG=4242
R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34899004

Cr-Commit-Position: refs/heads/master@{#8187}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8187 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-28 18:38:13 +00:00
45db7eefa2 Use Java based audio as default for WebRTC.
The work landed in 4034 (use of HW AEC in AppRTC) is currently not
active by default since we build for Open SL. I missed that when I
did my initial change (since I always disabled OpenSL by GYP_DEFINES).

This CL ensures that Java based audio is used as default in WebRTC.
It would be great if we could shift over to Open SL (to cut latency)
but that would (today) mean that we can't support the HW AEC.
Hence, we are not ready to do so yet.

BUG=4034
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36699004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8040 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 14:27:23 +00:00
34ac956706 Do not use openmax_dl for MIPS64 platform.
This fix is intended for MIPS64 Chromium Android builds, which has no openmax_dl
support at this moment.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31339004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7983 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-30 18:19:56 +00:00
1090a6eccf Remove obsolete target_arch == armv7.
Also, use arm_version >= 7 so things will continue to work when building
for ARMv8 and higher targets.

BUG=3906
R=kjellander@webrtc.org, tkchin@webrtc.org, zhongwei.yao@arm.com

Review URL: https://webrtc-codereview.appspot.com/38379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7957 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-18 21:36:18 +00:00
9f79fe684a Merge remote bitrate estimator changes.
R=stefan@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/33489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7811 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-04 15:34:06 +00:00
d1fac61e8f Remove need for assembly offset generation in aecm and ns module.
All *neon.S files in aecm and ns modules have been removed. We need no
assembly offset generation now.

Pass byte to byte conformance test for aecm and ns test in audioproc
between new NEON (written in intrinsics) version and C version on both
ARMv7 and ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Change-Id: I05d43d0c04d00bead65ca8c8fda25f0a42394b2b

Review URL: https://webrtc-codereview.appspot.com/32229004

Patch from Zhongwei Yai <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7800 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-03 17:54:38 +00:00
a56a2c57cf Enabling building with NEON on ARM64
This patch enables NEON on ARM64 platform. Passed building both on
Android ARMv7 and Android ARM64.

BUG=3580
R=andrew@webrtc.org, jridges@masque.com

Review URL: https://webrtc-codereview.appspot.com/25069004

Patch from Zhongwei Yao <zhongwei.yao@arm.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7751 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-26 17:01:40 +00:00
1153322cf8 Build fix for MIPS Android Webview build.
Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-21 16:28:32 +00:00
5b88317820 Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.

This is the same patch as https://code.google.com/p/webrtc/source/detail?r=7422, which was reverted when rolled into chrome (due to bss size increase). Relanding this again as we now have the clear to get this in:
see https://code.google.com/p/webrtc/issues/detail?id=3932

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7588 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-01 06:10:48 +00:00
508c91683c Build fix for MIPS32R6.
Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25989004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 16:26:17 +00:00
14146e40aa arm64 iOS build.
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
2014-10-31 00:14:39 +00:00
b1dac33cac Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."
BUG=3932
R=marpan@google.com

Review URL: https://webrtc-codereview.appspot.com/27779004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7470 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 18:54:46 +00:00
573c78e31c Add VP9 codec to VCM and vie_auto_test.
Include VP9 tests in videoprocessor_integrationtests.
Include end-to-end send/receiveVP9 test.
Passes trybots.

R=kjellander@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7422 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-10 16:44:47 +00:00
0001adcfef Use openmax_dl on all ARM (v7 or higher) platforms.
openmax_dl now works on non-Android ARM, but it still requires
arm_version >= 7, and doesn't work on iOS at all.

TEST=Chromium build for a ChromeOS ARM device passes.
BUG=chromium:415393
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7402 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-09 04:13:02 +00:00
4165f7aa22 Add a variable for deciding when to use openmax_dl.
Modifies the previous condition to additionally not use openmax_dl on
iOS. Remove the All target's direct dependency on it, as it is now
pulled in by the targets that need it.

Add gn support since an openmax_dl gn target is available.

BUG=chromium:415393, webrtc:3906
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7397 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-08 18:01:27 +00:00
2fe5893748 Mac: adds missing _DEBUG flag to mac debug builds.
BUG=3836
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25769004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7377 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-06 22:04:11 +00:00
d05756f0a2 Changed mips_arch_variant variable value corresponding to Chromium code changes.
Chromium commit URL: https://crrev.com/c8a5da7455b57b2399e4a69e8100c098d9870052

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23809004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7343 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-30 15:53:24 +00:00
e34a2e7475 Revert "Set minimum SDK level to 10.7 for Mac and iOS" (r7175)
Reverting this since it didn't fix the build failures.
We ended up passing mac_sdk=10.9 in GYP_DEFINES on the bots
to to make the build pass again
(https://codereview.chromium.org/573673002).

BUG=3120
R=mcasas@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7275 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 12:43:14 +00:00
ab071daab8 Split video_render_module implementation into default and internal implementation.
Targets must now link with implementation of their choice instead of at "gyp"-time.

Targets linking with libjingle_media:
- internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default render implementation:
- video_engine_tests
- video_loopback
- video_replay
- anything dependent on webrtc_test_common

Targets linking with internal render implementation:
- vie_auto_test
- video_render_tests
- libwebrtcdemo-jni
- video_engine_core_unittests

GN changes:
- Not many since there is almost no test definitions.

Work-around for chromium:
- Until chromium has updated libpeerconnection to link with video_capture_impl and video_render_impl, webrtc target automatically depends on it. This should fix the FYI bots and not require a webrtc roll to fix.

Re-enable android tests by reverting 7026 (some tests left disabled).

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.
BUG=3770
R=kjellander@webrtc.org, pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-18 08:58:15 +00:00
a74eda1b6f Split video_capture_module specific implementation (external vs internal capture)
into its own targets. Dependencies must link directly with the desired one.

Targets linking with libjingle_media:
 - internal implementation when build_with_chromium=0, default otherwise.

Targets linking with default/external capture implementation:
 - anything dependent on webrtc_test_common
 - anything dependent on video_engine_core

Targets linking with internal capture implementation:
 - vie_auto_test
 - anything dependent on webrtc_test_renderer

GN changes:
 - Not many since there is almost no test definitions.

TESTED: passes all the bots. If this inadvertently breaks a target please fix the linking rules so the target has the desired implementation linked in.

BUG=3768
R=glaznev@webrtc.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7209 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-17 11:50:19 +00:00
192ab710ce Set minimum SDK level to 10.7 for Mac and iOS.
This is needed since r7174 introduced a dependency
on AVFoundation, which is not present in the 10.6 SDK which is
still the default for Chromium.

BUG=
TESTED=Passing compile on trybots.
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7175 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 08:02:43 +00:00
67eabc0938 Add schannel webrtc_base build using a new use_schannel gyp variable.
R=henrike@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/28409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7141 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-10 18:06:47 +00:00
0372b93118 Partial revert of r7014 (Android APK refactor)
This reverts selected parts of r7014 to enable
rolling WebRTC in Chromium DEPS.

This works around the problem with GYP includes
being processed in the first pass (i.e. variables
cannot be used for paths). Using a dependency with
a path using a variable that is conditioned for
build_with_chromium being 0 or 1 solves the Chromium
build.

These changes will be restored once I've finished
a major GYP refactoring that will break out all
test related code (at least the parts that includes
the Android APK targets) into a separate chain
of GYP targets that are not processed when generating
projects for Chromium (which is why r7014 is breaking
the Chromium build).

BUG=3741
TESTED=Passing compilation of standalone using:
GYP_DEFINES="OS=android component=static_library fastbuild=1 target_arch=arm" webrtc/build/gyp_webrtc
ninja -C out/Debug
Then verified the *_apk  targets are generated and compiled.

Passing compilation from a Chromium checkout with third_party/webrtc
directory removed and a new empty third_party/webrtc mapped to the
standalone checkout using:
sudo mount --bind /path/to/trunk/webrtc third_party/webrtc
Then running build/gyp_chromium
I also verified WebRTC GYP targets exist and are able to compile.

R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7040 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-03 14:34:46 +00:00
d72a7599d4 Create a copy of talk/xmllite under webrtc/xmllite.
BUG=3379
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7027 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-02 15:41:12 +00:00
3bd4156d75 Android APK tests built from a normal WebRTC checkout.
Restructure how the Android APK tests are compiled now
that we have a Chromium checkout available (since r6938).

This removes the need of several hacks that were needed when
building these targets from inside a Chromium checkout.
By creating a symlink to Chromium's base we can compile the required
targets. This also removes the need of the previously precompiled
binaries we keep in /deps/tools/android at Google code.

All the user needs to do is to add the target_os = ["android"]
entry to his .gclient as described at
https://code.google.com/p/chromium/wiki/AndroidBuildInstructions

Before committing this CL, the Android APK buildbots will need
to be updated.
This also solves http://crbug.com/402594 since the apply_svn_patch.py
usage will be similar to the other standalone bots.
It also solves http://crbug.com/399297

BUG=chromium:399297, chromium:402594
TESTED=Locally compiled all APK targets by running:
GYP_DEFINES="OS=android include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release

checkdeps

R=henrike@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7014 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-01 11:06:37 +00:00
af7fdfcde8 Add LTO support for Android Chromium.
This is to add support for a Link-Time Optimizations experiment in Android Chromium. As it is disabled by default, it won't change anything for most configurations.
BUG=chromium:407544
R=andrew@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7009 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-29 17:41:13 +00:00
a685c9df62 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6467 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-17 14:48:44 +00:00
9f36c087f1 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries.
BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6455 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-16 21:35:20 +00:00
c6db88b0cf Make it possible to build webrtc for arm64.
- Bump revision of protobuf lib
- Remove -Wextra for arm64 gcc targets (warnings in stlport)
- Add MemoryBarrier implementation in single_rw_fifo.cc.
- [pending 15619004]: Bump revision of /deps/tools/android to get md5sum_bin for arm64.

BUG=chromium:354405,chromium:354539
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6330 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 17:15:42 +00:00
f048872e91 Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.

BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 18:00:26 +00:00
e9a604accd Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.

http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457


> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
> 
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12199004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:15:48 +00:00
2c7d1b39b9 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 18:03:09 +00:00
413d001132 Removed the disabling of include_tests from r2729.
BUG=N/A
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11259004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5856 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-07 15:52:31 +00:00
0aa04f9f24 Restore support for code coverage in WebRTC
In https://codereview.chromium.org/68193002
Chromium dropped the support for the coverage=1 flag.
This restores it for WebRTC purposes for the Linux platform.

TEST=Manually ran the coverage steps on my machine, verified
that .gcno files are generated.

BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5806 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-28 13:14:00 +00:00
cc08e3f9b1 Moves WEBRTC_POSIX define from header file to gyp-settings.
Makes WEBRTC_POSIX defined in the same place as the other OSs also this is needed to prevent excessive changes to talk/base files when migrating them to webrtc/base

BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5659 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-07 15:30:21 +00:00