Adding the argv[0] path to the working directory to make it possible find the project root even when the test is executed from outside the project dir (like it is on some buildbots).
Removed tests that moves into directories since they no longer work with this approach. The new functionality have been verified by manual tests of the following cases, example with a checkout root dir called webrtc/:
Working dir: Command line:
webrtc trunk/out/Debug/test
webrtc/trunk out/Debug/test
webrtc/trunk/out Debug/test
webrtc/trunk/out ./Debug/test
webrtc/trunk/out/Debug ./test
webrtc/trunk/out/Debug/subdir ../test
webrtc/trunk/out/Debug/subdir ./../test
I also made another program with its own main method (only links with 'test_support', not 'test_support_main') and made sure that it was still possible to use as before (i.e. works within the project tree but not above it):
#include "testsupport/fileutils.h"
int main(int argc, char** argv) {
printf("Working dir: %s\n", webrtc::test::WorkingDir().c_str());
printf("Project root: %s\n", webrtc::test::ProjectRootPath().c_str());
printf("Output path: %s\n", webrtc::test::OutputPath().c_str());
}
BUG=Existing implementation cannot handle when the working directory is outside the project checkout.
TEST=test_support_unittests and manual tests with video_codecs_test_framework_integration_tests + passing all trybots + memcheck tool
Review URL: https://webrtc-codereview.appspot.com/858014
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2927 4adac7df-926f-26a2-2b94-8c16560cd09d
Problem: When building webrtc in ARM but not Android:
third_party/webrtc/modules/audio_processing/ns/nsx_core_neon.c: In function 'WebRtcNsx_NoiseEstimationNeon':
third_party/webrtc/modules/audio_processing/ns/nsx_core_neon.c:248:7: note: use -flax-vector-conversions to permit conversions between vectors with differing element types or numbers of subparts
third_party/webrtc/modules/audio_processing/ns/nsx_core_neon.c:248:17: error: incompatible types when assigning to type 'int16x8_t' from type 'uint16x8_t'
Review URL: https://webrtc-codereview.appspot.com/859013
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2917 4adac7df-926f-26a2-2b94-8c16560cd09d
-added internal spatial resize and frame dropper as codec config parameters.
-made one test dedicated to internal spatial resize.
-set the denoiser to be on for the temporal layer test.
-correct the setting for number of spatial resizes for temporal layer test.
Review URL: https://webrtc-codereview.appspot.com/866007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2912 4adac7df-926f-26a2-2b94-8c16560cd09d
./content/renderer/media/webrtc_audio_device_impl.h:19:81: error: third_party/webrtc/modules/audio_device/main/interface/audio_device.h: No such file or directory
We have to create a temporary files until the new webrtc is rolled the chrome.
Review URL: https://webrtc-codereview.appspot.com/867008
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2905 4adac7df-926f-26a2-2b94-8c16560cd09d
These changes make the execution abort earlier on an error (like a tool is not found) and makes it easier to figure out what's wrong.
Made build_zxing.py executable.
BUG=None
TEST=Local runs of the PyAuto test src/chrome/test/functional/webrtc_video_quality.py in Chromium.
Review URL: https://webrtc-codereview.appspot.com/840005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2899 4adac7df-926f-26a2-2b94-8c16560cd09d
Also Suppressed FakeMediaProcess errors (bug 898) and took out a test and suppressed general errors (bug 332). Lastly, fixed memory leak in misc test.
BUG=898, 332
TEST=Ran voe_auto_test with repeat=10 through the whole standard suite, under valgrind. Ran without valgrind. Tested that the extended and standard tests still start and are reachable from the menu.
Review URL: https://webrtc-codereview.appspot.com/855009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2898 4adac7df-926f-26a2-2b94-8c16560cd09d
Added new helper for getting regular input. Rewrote remaining input handling for custom calls. Added a --choose_defaults flag which makes it possible to default on everything (e.g. with the flag, choosing custom call will accept all defaults and go directly to the call).
The next patch will add support for overriding arbitrary choices using flags. That is the point I want to arrive at and this patch paves the way for that. Fortunately it gets rid of some repetitive and bug-prone code on the way.
BUG=
Review URL: https://webrtc-codereview.appspot.com/858005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2878 4adac7df-926f-26a2-2b94-8c16560cd09d
Make sure that playing dummy RTP files works, i.e., RTP dumps with only
headers, no actual payloads. It turs out that some RTP dumps did not
state how long the actual packet was (including the discarded payload),
which resulted in packets with zero length payload being inserted into
NetEQ. Now, the dummy packet will contain at least one byte of payload
(set to zero).
Also added capability to filter out a specific SSRC from the RTP
dump file.
Finally, added SSRC printout from RTPanalyze, and corresponding
change in RTPchange.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/857007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2876 4adac7df-926f-26a2-2b94-8c16560cd09d
