Background:
In Chrome mirroring which uses 500ms buffering mode,
audio video mismatch happens in the begining because of the lack of the api.
BUG=b/10538425
TEST=pass 'git try' except tests which is aleady broken in the bot. pass 'build/android/test_runner.py gtest -s modules_tests --verbose --release -f *InitialPlayoutDelayTest*'
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2144004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4671 4adac7df-926f-26a2-2b94-8c16560cd09d
Breaks out config part of FakeEncoder from VideoSendStream tests to
FakeEncoder. Also sets FakeEncoder as encoder for VideoSendStream tests.
Anticipated speedup didn't happen as VP8 is still initialized by default
when creating channels in the old API. This will be sped up when moving
off the old API as VP8 won't be enabled by default.
BUG=2312
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2155004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4659 4adac7df-926f-26a2-2b94-8c16560cd09d
On Windows, when tests are run under memory tools like
DrMemory and TSan, slashes occur in the path as directory
separators. This makes the fileutils.cc fail to figure out
the project root, making paths to resource files invalid.
By replacing slashes in the path with backslashes for Windows
this can be fixed.
BUG=2318
TEST=running the same command as the bot locally, succeeding after this patch is applied.
R=phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2137004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4640 4adac7df-926f-26a2-2b94-8c16560cd09d
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
The metrics_unittests tried to write an empty file to the
current working directory, which caused a crash on Android. By
using the output folder instead, this is avoided.
This makes it possible to re-enable the tests disabled in r4474
and also remove the empty dummy test added in r4627.
BUG=1882,
TEST=Setup a Chromium+WebRTC checkout as described in issue 1882
. build/android/envsetup.sh
export GYP_DEFINES="$GYP_DEFINES include_tests=1 enable_tracing=1"
gclient runhooks
ninja -C out/Release metrics_unittests_apk
build/android/test_runner.py gtest -s metrics_unittests --verbose --release
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2127006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4634 4adac7df-926f-26a2-2b94-8c16560cd09d
The ACM tests needed re-writing, because all tests were not individual gtests, and the result was difficult to interpret.
While doing the re-write, I discovered a bug related to 48 kHz CNG. We can't have the 48 kHz CNG active at the moment. The bug is fixed in this CL.
I also needed to rewrite parts of the VAD/DTX implementation, so that the status of VAD and DTX (enabled or not) is propagated back from the function SetVAD().
BUG=issue2173
R=minyue@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1961004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4625 4adac7df-926f-26a2-2b94-8c16560cd09d
I broke captured image flipping when refactoring this code while it was
still in chromium. Previously we had CaptureData that was returned from
capturers with correctly inverted stride, but frames were still stored
with positive stride. CaptureData was removed and so the returned frames
always had positive stride, which is not correct. Now ScreenCapturerMac
uses frames with inverted stride when capturing using OpenGL.
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/2105004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4621 4adac7df-926f-26a2-2b94-8c16560cd09d
Window enumeration is based on the code used by hangouts plugin
(see libjingle/talk/base/linuxwindowpicker.cc). XServerPixelBuffer
is used to capture windows. It had to be refactored to support window
capturing (previously it worked only for the whole screen).
R=wez@chromium.org
Review URL: https://webrtc-codereview.appspot.com/1741004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4605 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
