* Clarified that the test only supports specifying a single test (multiple didn't work before, so better not claim to support it).
* No longer needs test executable arguments to use ++ instead of --
* Only appends the build_dir path to the test executable if not already
present.
* Simplified suppression path handling.
* Fixed crash when -v was used (import logging was missing)
* Style fixes.
* Thorougly tested with all the supported flags.
I noted that the --gtest_filter flag does not work as expected (it's
only for 'additional gtest_filter arguments', which seems to mean
additional arguments to the gtest filter text files that are used by
Chrome. I left it in here anyway. If --gtest_filter is given after the
test executable it will work, since those arguments are added straight
to the test executable
the test
BUG=none
TEST=I ran the following commands and verified that the suppressions and flags were handled correct:
tools/valgrind-webrtc/webrtc_tests.sh -v --gtest_repeat=2 --keep_logs --tool_flags=--trace-children=yes -t out/Debug/test_support_unittests --foo --bar
tools/valgrind-webrtc/webrtc_tests.sh -v --gtest_repeat=2 --keep_logs --tool_flags=--trace-children=yes -b out/Debug -t test_support_unittests --foo --bar
tools/valgrind-webrtc/webrtc_tests.sh -v --gtest_repeat=2 --keep_logs --tool_flags=--trace-children=yes -b out/Debug -t out/Debug/test_support_unittests --foo --bar
tools/valgrind-webrtc/webrtc_tests.sh -v --tool=tsan --gtest_repeat=2 --keep_logs --tool_flags=--trace-children=yes -t out/Debug/test_support_unittests --foo --bar
tools/valgrind-webrtc/webrtc_tests.sh -v --tool=tsan --gtest_repeat=2 --keep_logs --tool_flags=--trace-children=yes -b out/Debug -t test_support_unittests --foo --bar
tools/valgrind-webrtc/webrtc_tests.sh -v --tool=tsan --gtest_repeat=2 --keep_logs --tool_flags=--trace-children=yes -b out/Debug -t out/Debug/test_support_unittests --foo --bar
Review URL: https://webrtc-codereview.appspot.com/1029005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3354 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I
though that is more risky, so I'll do that in a separate patch later (perhaps
we could purge the types from the whole module in one go?)
BUG=
TEST=Trybots, vie_ & voe_auto_test --automated
Review URL: https://webrtc-codereview.appspot.com/998007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
* Improved sort checker so we sort includes more.
* Fixed vars in brackets and varsLikeTHIS.
* Added automatic x++ to ++x conversion in for loops.
TEST=Ran on various source files and verified manually.
Review URL: https://webrtc-codereview.appspot.com/1017004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3325 4adac7df-926f-26a2-2b94-8c16560cd09d
The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.
I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.
Next step is to add an API to choose application mode.
BUG=issue1239
Review URL: https://webrtc-codereview.appspot.com/1007006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes it possible to compile on the bots without hardcoding paths
to Ant, Java and ffmpeg deep into the Python scripts (hardcoded paths exists only in the buildbot configuration).
For bots, the ANT_HOME, JAVA_HOME and FFMPEG_HOME environment variables must be set to the install locations for each of these dependencies, for Windows.
This CL also improves the return code handling to make failures easier to detect when things break.
TEST=running build_zxing.py without Ant or Java in the PATH, but with
ANT_HOME, JAVA_HOME and FFMPEG_HOME set. Running Chromium's src/chrome/test/functional/webrtc_video_quality.py.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1002005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3311 4adac7df-926f-26a2-2b94-8c16560cd09d
The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches.
In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on.
BUG=
TEST=vie/voe_auto_test, trybots
Review URL: https://webrtc-codereview.appspot.com/1001006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes the following files be written into the output dir instead of
the current working dir:
* out.pcm
* vad_out.dat
* ns_prob.dat
TEST=out/Debug/audioproc -aecm -ns -agc --fixed_digital --perf -pb
resources/audioproc.aecdump
All trybots passing.
BUG=none
Review URL: https://webrtc-codereview.appspot.com/1003005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3302 4adac7df-926f-26a2-2b94-8c16560cd09d
