The find_depot_tools.py is needed to workaround the import
error we get from gyp_chromium when importing it in
webrtc/build/gyp_webrtc (to avoid code duplication).
gyp_chromium introduced a dependency on it in
http://crrev.com/245412 but as we cannot sync all of Chrome's
src/tools (it's quite big), we'll work around this by
adding an empty find_depot_tools module.
The removal of the Cygwin relates to
http://crrev.com/248802 which is a step on the way to remove
Cygwin in Chromium. We seem to already be able to remove it
entirely for WebRTC though.
Changes in the isolate framework required us to update our
copies of the isolate.gypi files.
BUG=none
TEST=trybots passing on all platforms
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8099004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5512 4adac7df-926f-26a2-2b94-8c16560cd09d
The reason for this is that http://crrev.com/245412
introduces a dependency of Chrome's src/build/gyp_chromium
to src/tools/find_depot_tools.py, which we don't have
synced in WebRTC (src/tools is very big).
Offline discussions shows that we cannot rely on syncing
individual subdirectories from Chrome in the future, but
maintaining our own gyp_webrtc file will at least buy us
some time for now, so we can roll past that chromium_revision
in WebRTC DEPS.
Overview of differences between gyp_webrtc and gyp_chromium
(and how we previously used gyp_chromium):
* No .gyp file needs to be passed (defaults to all.gyp)
* CHROMIUM_GYP_FILE is ignored (i.e. cannot be used to
specify an alternate .gyp file to process)
* Ninja is used by default on all platforms unless GYP_GENERATORS
is set.
* Gyp syntax check is always on
* Gyp circular dependency check is always on
* No support for automatic toolchain detection on Windows.
* --depth argument is no longer needed since calculated by
the script.
* Support for a webrtc.gyp_env file sitting next to the
.gclient file in the top dir of checkout, which can be
used to override Gyp variables similar to chromium.gyp_env.
* SKIP_WEBRTC_GYP_ENV can be set to skip reading webrtc.gyp_env.
BUG=2863
TEST=Ran and verified behavior on Linux with:
gclient runhooks
webrtc/build/gyp_webrtc
webrtc/build/gyp_webrtc -Dextra_gyp_flag=0
. build/android/envsetup.sh && gclient runhooks
SKIP_WEBRTC_GYP_ENV=1 webrtc/build/gyp_webrtc
GYP_GENERATORS=make webrtc/build/gyp_webrtc
The patch also passes runhooks and compile step on all trybots.
R=andrew@webrtc.org, fischman@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7759004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5467 4adac7df-926f-26a2-2b94-8c16560cd09d
It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.
BUG=2253
R=fischman@webrtc.org, juberti@google.com, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4679005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
There are uninitializion problem with normal_asyn_test.cc. This is fairly easy to solve and therefore is included in this CL.
The following is a memo on the selection of the version to roll. It may be a reference for similar missions.
How was this version picked?
1. The whole purpose of this work is to update to Clang to be able to compile Opus 1.1. In Chromium, Clang got updated to 198389 at r244540.
2. From r245412, gyp_chromium requires "tools\find_depot_tools.py". However, WebRTC does not sync up the root of folder "tools". An issue has been created to Chromium on this.
... So the version must be a good version between r244540 and r245411 (inclusive)
BUG=
TEST=passes all trybots
R=kjellander@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/7569005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5436 4adac7df-926f-26a2-2b94-8c16560cd09d
With help from hinoka@, we're now using a more efficient approach
to download only the files that have changed from Google Storge.
When uploading new resource files, use
upload_to_google_storage.py --bucket chromium-webrtc-resources ./filename
which of course requires gsutil authentication setup.
NOTICE: Before deploying this, svn:ignore should be removed for
the resources folder, or the bots will run into problems with a
non-versioned file being found in the checkout during sync (as
this CL adds resources to version control).
All developers will also need to be informed to wipe their local
resources dir to avoid getting an error during checkout due to the
already existing non-versioned resources directory.
BUG=2294
TEST=locally running gclient runhooks
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2095004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5076 4adac7df-926f-26a2-2b94-8c16560cd09d
This will speed up our buildbots and trybots since it also enables
incremental builds again (currently out/ was wiped for each build).
The Chromium toolchain somehow has changed behavior, since the out/
dir didn't use to be wiped. This change should get faster builds
back again.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4641 4adac7df-926f-26a2-2b94-8c16560cd09d
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
