By changing the working directory for the executing script to the same
directory as the script is located in, it is possible to run the script
standing in a higher-level directory (otherwise the input file relative
paths become invalid).
This CL also changes the input file path for the audio_e2e_test test to
assume the file is located resources.
BUG=none
TEST=locally executed the tests standing in trunk/
Review URL: https://webrtc-codereview.appspot.com/1061009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3422 4adac7df-926f-26a2-2b94-8c16560cd09d
With http://review.webrtc.org/1064004 we got use of headers in testing/ for the first time in production code (which is what gtest_prod.h is meant for). This showed that when WebRTC is built inside Chrome, the include path doesn't include the top-level directory, so testing/gtest/ could not be found.
By adding <(DEPTH) to the WebRTC global include path list in common.gypi, this is resolved.
Having this directory in the global include path list will also make it possible for us to use full paths for common third party libraries, which should be something we aim for.
BUG=none
TEST=Successfully compiled the webrtc_test_tools target on Linux in a Chromium checkout with third_party/webrtc replaced by ToT trunk with this patch applied (with Python 2.6 installed, which is needed to get the pyautolib target generated).
Review URL: https://webrtc-codereview.appspot.com/1082004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3421 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL is one step in a larger change of the DelayEstimator where we will open up for multiple near-end signals.
This particular CL separates the low level far-end parts without affecting the usage externally. This is a first step towards separating the far-end and near-end parts giving the user the control.
BUG=None
TEST=audioproc_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1068005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3415 4adac7df-926f-26a2-2b94-8c16560cd09d
The path in test.gyp in r3411 was incorrect since it was based on the symlink that does not exist on Windows.
This CL changes it to reference the actual path in /tools/perf
TBR=phoglund
BUG=none
TEST=win try bot.
Review URL: https://webrtc-codereview.appspot.com/1074006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3413 4adac7df-926f-26a2-2b94-8c16560cd09d
This script is an attempt to move flags and argumetns to tests from the
buildbot configuration to the source tree.
This will make it easier for anyone to modify test execution behavior
and also has the benefit that it's easier to run the tests in a similar
fashion on a developer workstation.
NOTICE: The audio comparison tool will need to be moved to ~/bin when bots are going to switch over to using this script for execution.
TEST=local execution.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1021006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3411 4adac7df-926f-26a2-2b94-8c16560cd09d
The function WebRtcOpus_DurationEst returned the number of samples
per packet in the native 48 kHz sample rate, while the decoder
function returns data in 32 kHz sample rate. This creates a discrepancy
that makes NetEQ's lip-sync functionality add too little delay.
BUG=1334
TEST=try bots
Review URL: https://webrtc-codereview.appspot.com/1069006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3403 4adac7df-926f-26a2-2b94-8c16560cd09d
The test fails since it's assuming out/testfile.yuv exists when running the test. Just opening the file at a later time than the SetUp function seems to break the test so that's not a viable solution. This CL uses a simple workaround that simply truncates the file before opening it, which works.
BUG=none
TEST=tools_unittests in Debug+Release on Mac, Win and Linux + memcheck, tsan, asan.
Review URL: https://webrtc-codereview.appspot.com/1067004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3401 4adac7df-926f-26a2-2b94-8c16560cd09d
Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine.
BUG=
TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test
Review URL: https://webrtc-codereview.appspot.com/1022011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.
Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.
TEST=vie_auto_test, rtp_rtcp_unittests, trybots
Review URL: https://webrtc-codereview.appspot.com/1041004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
