120 Commits

Author SHA1 Message Date
henrike@webrtc.org
e6e139159f Android: cleanup gtest_target_type conditions.
Ever since crrev.com/133053 OS==android implies:
gtest_target_type=shared_library

Similar to Chromium's crrev.com/271222 where base.gyp's conditions are changed
(which the affected conditions in this cl comes from).

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

Review URL: https://webrtc-codereview.appspot.com/13539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6332 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-04 20:46:50 +00:00
wu@webrtc.org
7a9a3b70b3 * Revert clock.cc changes made in 6178, but keep the changes to the test.
* Use the new appoach proposed by jib in https://review.webrtc.org/10439004/ to fix the windows clock issue.

BUg=3325

R=niklas.enbom@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15569005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6273 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-29 19:40:28 +00:00
henrik.lundin@webrtc.org
59336e85fb Adding R/W lock to SimulatedClock
This change makes the SimulatedClock class thread safe.

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

Review URL: https://webrtc-codereview.appspot.com/15569004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6251 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-27 09:34:58 +00:00
wu@webrtc.org
75718cf80a * Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly.
* Updated to ClockTest.NtpTime to verify the returned NTP is at least larger than kNtpJan1970.

Current implementation uses timeGetTime, which returns the time since windows is started, which can't be converted to NTP time.

BUG=3325
R=pwestin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6178 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-15 23:54:14 +00:00
andresp@webrtc.org
a36ad6929d Add webrtc field trials API.
From now on it is expected that code linking system_wrappers.gyp:system_wrappers
provides an implementation for field_trial API or links with the default one in
system_wrappers.gyp:field_trial_default.

Note: Since there is no use of webrtc::field_trial API inside webrtc this CL on
itself does not forces the clients to actually define it. It however lays the
API and updates the gyp rules to link with so that it is ready to use.

Tested: Introduced a use of field trial in system wrappers and make sure all
bots were building successfully.

BUG=crbug/367114
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6147 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-14 12:24:04 +00:00
henrike@webrtc.org
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
andresp@webrtc.org
93ec9c557b Revert "FieldTrial implementation for webrtc." (rev 6089)
New wiring plans require it to be landed first in chrome for a cleaner roll of webrtc.

BUG=crbug/367114
R=tommi@webrtc.org
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6122 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 14:09:40 +00:00
perkj@webrtc.org
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
henrike@webrtc.org
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
andresp@webrtc.org
6a8a6723d3 FieldTrial implementation for webrtc.
BUG=crbug/367114
R=asvitkine@chromium.org, mflodman@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6089 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-09 07:14:34 +00:00
wu@webrtc.org
66773a032a Move timestamp_extrapolator and rtp_to_ntp to system wrapper so that it can be shared by both audio and video engine.
BUG=3111
TEST=try bots
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6074 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-07 17:09:44 +00:00
andrew@webrtc.org
0daa8be9d6 Add Chromium's ScopedVector.
Trivial changes from the original excepting scoped_vector_unittest.cc,
diff here: https://paste.googleplex.com/6664017300946944

This is a prerequisite for:
http://review.webrtc.org/9919004/

TBR=henrike@webrtc.org
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5938 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-18 21:20:54 +00:00
kjellander@webrtc.org
7de47bce12 Remove use of tmpnam.
This solves compilation with the Mac SDK 10.9.

BUG=3120, 3151
TEST=git try -t modules_tests:VideoProcessorIntegrationTest*
R=fischman@webrtc.org, henrike@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10739005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5917 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-16 08:04:26 +00:00
fischman@webrtc.org
2c89b5cb27 Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.
This CL brought to you by:
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do echo -e "\n# These are for the common case of adding or renaming files. If you're doing\n# structural changes, please get a review from a reviewer in this file.\nper-file *.gyp=*\nper-file *.gypi=*" >> $d/OWNERS; done
$ for d in $(for f in $(git ls-files '*gyp' '*gypi'); do dirname $f; done|sort|uniq|grep -v '^\.$'); do git add $d/OWNERS; done

(and then removed the talk/ impact)

R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5903 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-14 20:08:03 +00:00
asapersson@webrtc.org
2a770828d8 Remove usage of webrtc trace in video processing modules.
BUG=3153
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5880 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-10 11:30:49 +00:00
andrew@webrtc.org
d617a44a4f Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
- Transition scoped_ptr_mallocs to scoped_ptr.
- AlignedFreeDeleter matches Chromium's version.

TESTED=try bots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8969005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-20 21:08:36 +00:00
michaelbai@google.com
f928f5c87c Use system's cpu_features library
Remove the copied cpu_featrues.c/h
Use the NDK's cpu_features.a or the one build from android source.
This issue blocked libvpx roll.

BUG=334447
R=andrew@webrtc.org, fischman@webrtc.org, henrike@webrtc.org, wjia@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5492 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-02-05 18:43:46 +00:00
henrike@webrtc.org
3907c2e7e5 Removes the remaining uses of the list wrapper class and the list wrapper class.
BUG=2164
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/7019007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5378 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-01-13 22:41:34 +00:00
andresp@webrtc.org
7fb75ecbd4 Add thread_annotations for clang targets.
TESTED: As expected clang bots catched a few issues which are fixed with this CL, other bots ignore the annotations and compile fine.

R=niklas.enbom@webrtc.org, phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/6209004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5328 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-20 20:20:50 +00:00
pbos@webrtc.org
5ab756703e Revert r5294 to re-roll r5293.
To fix races in test each stream now owns its own encoder/decoder.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5297 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-16 12:24:44 +00:00
turaj@webrtc.org
41e2615e02 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..."
> Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
> 
> BUG=
> R=mflodman@webrtc.org, stefan@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/5409004

TBR=solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5294 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-15 18:42:32 +00:00
solenberg@webrtc.org
341e91441a Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5293 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-13 23:57:54 +00:00
kjellander@webrtc.org
f9bdbe3619 Roll chromium_revision 232627:238260
This brings us the updated swarming_client
that has moved out from Chromium into a standalone
project.
Because of this, all .isolate files needed to be
updated as well, similar to the changes in
https://codereview.chromium.org/29993003

TEST=trybots passing
BUG=none
R=andrew@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4859004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5260 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-11 13:37:12 +00:00
henrikg@webrtc.org
863b536100 Allow opening an AEC dump from an existing file handle.
This is necessary for Chromium to be able enable the dump with the sanbox enabled. It will open the file in the browser process and pass the handle to the render process.

This changes FileWrapper to deal with the case were the file handle is not managed by the wrapper.

BUG=2567
R=andrew@webrtc.org, henrika@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5239 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-06 16:05:17 +00:00
pbos@webrtc.org
c49d5b7df8 Move implementation files out of the webrtc/ root.
Leaves the root for public headers. Also fixes the issue of requiring
root OWNERS approval for changes in the Call implementation and adding
end-to-end tests.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5049005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5223 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-12-05 12:11:47 +00:00
andrew@webrtc.org
5973f3a24a Remove unneeded includes from trace_posix.cc.
TESTED=trybots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5103 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-08 17:30:07 +00:00
andrew@webrtc.org
f1a48174d4 Replace disabled logging with a restricted logging mode.
This will enable some low-level webrtc logging in a Chromium build,
while limiting the binary size impact.

For a Mac Release build, it results in an increase to Chrome.app of 37k
and libpeerconnection.so of 25k. For comparison, enabling full logs
costs 230k and 218k respectively.

BUG=b/11470432
TESTED=voe_cmd_test produces logs of the appropriate severity.
R=fischman@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5097 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-07 23:47:26 +00:00
andrew@webrtc.org
b3731da68f Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_.
Will fix a redefinition error in Chromium against webrtc head.

TESTED=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5029 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-24 15:16:53 +00:00
andrew@webrtc.org
31628aae7e Upgrade scoped_ptr to Chromium's latest version.
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.

- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.

TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-22 12:50:00 +00:00
kjellander@webrtc.org
e7009f3b0d Revert "Disable tests for TSan v2"
This reverts r4787 since r4966 got us a newer Clang
version with these issues fixed.

BUG=2259,2334
TEST=test_support_unittests and system_wrappers_unittests passed execution under TSan v2 for 100 iterations:
GYP_DEFINES='tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only"' gclient runhooks
ninja -C out/Release test_support_unittests system_wrappers_unittests

TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/system_wrappers_unittests --gtest_repeat=100 --gtest_break_on_failure

TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/test_support_unittests --gtest_repeat=100 --gtest_break_on_failure
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2407004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4974 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 12:26:48 +00:00
kjellander@webrtc.org
5ed4f46af1 Remove TSan v2 disabled test in condition_variable_unittest.cc
When we rolled our chromium_revision 226126:228675 in r4966
we picked up Clang r191856, which fixes the problem we've
seen earlier in condition_variable_unittest.cc.
Because of this, I'm now re-enabling this test.

TEST=trybots passing
BUG=2259
R=hta@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2404004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4971 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-16 10:50:41 +00:00
kjellander@webrtc.org
3f9288f987 Add APK and isolate target for video_engine_tests
Add .isolate file and _run target for video_engine_tests.

Move tools/swarm_client to be untracked in all .isolate file,
so refactorings in swarm_client doesn't require us updating
all our .isolate files (similar to the changes for the
Chromium tests done in:
https://src.chromium.org/viewvc/chrome?view=rev&revision=218844)

Update modules_unittests.isolate with new NetEq4 reference files
needed.

TEST=trybots passing
I also setup a Chromium workspace where I patched third_party/webrtc
with the changes in this CL, followed by compiling with the settings
described in
https://code.google.com/p/webrtc/issues/detail?id=1882#c11
I then verified that the video_engine_tests_apk dir was created
in the output folder.
BUG=1916,2462
R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2344007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4925 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-04 18:20:38 +00:00
fischman@webrtc.org
4e65e07e41 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android.
Besides being ~40% the size of the previous implementation, this makes it so
that VideoCaptureAndroid can stop and restart capture, which is necessary to
support onPause/onResume reasonably on Android.

BUG=1407
R=henrike@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2334004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4915 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-03 18:23:13 +00:00
kjellander@webrtc.org
2a97317953 Fix include of isolate.gypi
Recent changes in GYP seem to have broken our previous
"hack" for getting the GYP rule for .isolate files
imported from the Chromium build/isolate.gypi.

The best solution for now is to remove the hack
and check in a copy of Chromium's src/build/isolate.gypi
in WebRTC's build/ dir instead. A similar approach is
used for our build/protoc.gypi file.

TEST=On Linux, I successfully ran:
gclient runhooks
ninja -C out/Release
and verified a bunch of .isolated files were created in
out/Release (which didn't happen before this patch).

I also renamed the build/isolate.gypi from Chromium to
ensure that our own is used and not that one (in case any
paths would be incorrect).

I also ran build/gyp_chromium in a Chromium checkout
with WebRTC in third_party/webrtc having this patch applied
to ensure GYP processing was still working.

Finally, I verified that the same project generation and
compilation from a Chromium checkout worked the way we build
our Android native tests, using:
. build/android/envsetup.sh
GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1" gclient runhooks
ninja -C out/Release android_builder_webrtc

BUG=1916
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2338004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4907 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 19:31:16 +00:00
pbos@webrtc.org
c0167702d3 Stop timer in ~EventWindows().
Running out of handles seems to have been an issue when adding another
test target, this should solve it.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4897 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-02 13:11:15 +00:00
fischman@webrtc.org
7e4d0df8ee PeerConnection(Android): enable tracing to logcat.
BUG=1295
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2258007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4888 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-01 02:40:43 +00:00
kjellander@webrtc.org
d8a9b86671 Disable tests for TSan v2
These are tests that fail and that I haven't been able to suppress.
I assume they're caused by the same bug in TSan v2 as described in
webrtc:2259.

BUG=2259,2334
TEST=Ran the tests similar to the instructions in the bugs, passed 100 iterations.
R=andrew@webrtc.org, hta@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2250004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4787 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-19 07:15:36 +00:00
henrike@webrtc.org
82f014aa0b OpenSL (not default): Enables low latency audio on Android.
BUG=1669
R=andrew@webrtc.org, fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2032004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4719 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-10 18:24:07 +00:00
andrew@webrtc.org
9080518a39 Restore severity precondition to logging.h.
I mistakenly ommitted the checks when logging.h was ported from
libjingle to webrtc. This caused a significant CPU cost for logs which
were later filtered out anyway.

Verified with LS_VERBOSE logging in neteq4, running:
$ out/Release/modules_unittests \
--gtest_filter=NetEqDecodingTest.TestBitExactness \
--gtest_repeat=50 > time.txt
$ grep "case ran" time.txt | grep "[0-9]* ms" -o | sort

Results on a MacBook Retina, averaged over 5 runs:
Verbose logs disabled:                          666 ms
Exisiting implementation, verbose logs enabled: 944 ms (1.42x)
New implementation, verbose logs enabled:       673 ms (1.01x)

BUG=2314
R=henrik.lundin@webrtc.org, henrike@webrtc.org, kjellander@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2160005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4682 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-05 16:40:43 +00:00
kjellander@webrtc.org
f8c16b8bf4 Disabling CondVarTest for TSan v2 (take 2)
This test goes into an eternal loop sometimes when run under
TSan v2, therefore I'm disabling it until this is resolved.
Doing that, the TSan v2 bot in our new WebRTC FYI waterfall at
http://build.chromium.org/p/client.webrtc.fyi/waterfall
will hopefully go green.

The define used only exists for TSan v2, so v1 should be
unaffected.

The suppression added in http://review.webrtc.org/2126005/
turns out actually has no effect, so I'm removing it again.
It's probably useless since TSan v2 obviously alters the code's
behavior, causing these errors. Since the errors are not TSan v2
failures, the suppression has no effect.

BUG=2259
TEST=test passing locally when run as described in 2259 (since
no tests are run).

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2115005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4639 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-29 09:47:18 +00:00
kjellander@webrtc.org
e141373b8a Add isolate configuration for Android for all tests.
In https://code.google.com/p/webrtc/source/detail?r=4407
henrike@ added the path to the WebRTC resources and
data directories for Android that are required in order to
use isolate for test execution on Android.

This CL adds similar entries to the rest of the .isolate
files added in
https://code.google.com/p/webrtc/source/detail?r=4590.

It also removes three accidentally added .isolate files that originated
from old test names:
* audio_device_test_api
* video_capture_module_test
* video_render_module_test

BUG=1882,1916
TEST=trybots passing.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2107004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4627 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-27 12:10:09 +00:00
kjellander@webrtc.org
3365422c41 Isolate GYP target and .isolate files for tests
This is a re-land attempt of http://review.webrtc.org/1673004/
It now includes a build/isolate.gypi in WebRTC that includes the same
file as the one that would be included when WebRTC is used in a Chromium
checkout. It is needed since it is not possible to use variables in GYP's
includes sections.

Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing

Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
   actually executing it:
   tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
   tools/swarm_client/googletest/fix_test_cases.py --isolated out/Release/testname.isolated

All tests that run on the bots for WebRTC has got _run target
and .isolate file created.

"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_tests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests

Tests that requires bare-metal and audio/video devices:
* audio_device_tests
* video_capture_tests

I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_tests
* vie_auto_test
* voe_auto_test

TEST=running isolate.py as described above. WebRTC trybots passing. Created a Chromium checkout with third_party/webrtc ToT and this patch applied, passing the runhooks step.
BUG=1916
R=henrike@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2056004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4590 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-22 07:57:00 +00:00
pbos@webrtc.org
e6dc38ea9b Lock resources in event_posix.cc.
Fixes errors reported by Helgrind from event_posix.cc when running video_engine_tests.

BUG=
TEST=helgrind,trybots
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2060004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4572 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-20 09:49:19 +00:00
kjellander@webrtc.org
4298f73031 Revert 4547 "Isolate GYP target and .isolate files for tests"
As this breaks the FYI bots in 
http://build.chromium.org/p/chromium.webrtc.fyi/waterfall
due to different path to isolate.gypi (which cannot easily
be resolved due to limitations in GYP)

> Isolate GYP target and .isolate files for tests
> 
> Implemented according to the instructions at
> http://www.chromium.org/developers/testing/isolated-testing
> 
> Workflow has been like this:
> 1. create _run GYP target
> 2. create a stripped down .isolate file
> 3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
> 4. runhooks
> 5. compile
> 6. test if the test would run (i.e. find it's dependencies) without
>    actually executing it:
>    tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
> 7. If failing, run the fix_test_cases.py script like this:
>    tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated
> 
> All tests that run on the bots for WebRTC has got _run target
> and .isolate file created.
> 
> "Normal tests" that run fine on any machine:
> * audio_decoder_unittests
> * common_audio_unittests
> * common_video_unittests
> * metrics_unittests
> * modules_integrationtests
> * modules_unittests
> * neteq_unittests
> * system_wrappers_unittests
> * test_support_unittests
> * tools_unittests
> * video_engine_core_unittests
> * voice_engine_unittests
> 
> Tests that requires bare-metal and audio/video devices:
> * audio_device_integrationtests
> * video_capture_integrationtests
> 
> I also added the isolate boilerplate code for the following
> tests that are not yet pure gtest binaries (which means they
> cannot run isolated yet):
> * video_render_integrationtests
> * vie_auto_test
> * voe_auto_test
> 
> TEST=running isolate.py as described above.
> BUG=1916
> R=tommi@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/1673004

TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2040004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4548 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 11:29:58 +00:00
kjellander@webrtc.org
d7a4d235d2 Isolate GYP target and .isolate files for tests
Implemented according to the instructions at
http://www.chromium.org/developers/testing/isolated-testing

Workflow has been like this:
1. create _run GYP target
2. create a stripped down .isolate file
3. export GYP_DEFINES="$GYP_DEFINES test_isolation_mode=check"
4. runhooks
5. compile
6. test if the test would run (i.e. find it's dependencies) without
   actually executing it:
   tools/swarm_client/isolate.py run --isolated out/Release/testname.isolated
7. If failing, run the fix_test_cases.py script like this:
   tools/swarm_client/fix_test_cases.py --isolated out/Release/testname.isolated

All tests that run on the bots for WebRTC has got _run target
and .isolate file created.

"Normal tests" that run fine on any machine:
* audio_decoder_unittests
* common_audio_unittests
* common_video_unittests
* metrics_unittests
* modules_integrationtests
* modules_unittests
* neteq_unittests
* system_wrappers_unittests
* test_support_unittests
* tools_unittests
* video_engine_core_unittests
* voice_engine_unittests

Tests that requires bare-metal and audio/video devices:
* audio_device_integrationtests
* video_capture_integrationtests

I also added the isolate boilerplate code for the following
tests that are not yet pure gtest binaries (which means they
cannot run isolated yet):
* video_render_integrationtests
* vie_auto_test
* voe_auto_test

TEST=running isolate.py as described above.
BUG=1916
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1673004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4547 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-15 10:02:06 +00:00
pbos@webrtc.org
4ca7d3f9fe Replace MapWrapper with std::map<>.
MapWrapper was needed on some platforms where STL wasn't supported, we
now use std::map<> directly.

BUG=2164
TEST=trybots
R=henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4530 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-12 19:51:57 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents.
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-05 16:22:53 +00:00
henrike@webrtc.org
89c674053e Adds all unittests to android NDK-APK framework.
BUG=N/A
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1872004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4474 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-02 16:53:47 +00:00
pbos@webrtc.org
a2a2718a6c Fix some chromium-style warnings in webrtc/system_wrappers/
BUG=163
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1906004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4464 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-01 17:26:15 +00:00
henrike@webrtc.org
42581545eb Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle.
Libjingle sets the variables here: https://code.google.com/p/libjingle/source/browse/trunk/talk/supplement.gypi

BUG=N/A
R=andrew@webrtc.org, fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1787005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4354 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-07-16 16:37:22 +00:00