* audio_coding_module_test: enabling on Windows.
* audio_conference_mixer_unittests: enabling on Windows.
* audio_device_test_api: disabling on Mac, since this test is failing but not reporting failure. See issue 257 for more details.
* media_file_unittests: enabling on Windows.
* rtp_rtcp_unittests: enabling on Windows.
* test_bwe: enabling on Windows.
* test_fec: enabling on all platforms. See CL 369008 and 379010.
* test_support_unittests: enabling on all platforms.
* udp_transport_unittests: enabling on Windows.
* video_codecs_test_framework_unittests: adding disabled test on all platforms.
* video_codecs_test_framework_integrationtests: enabling on all platforms.
* video_processing_unittests: enabling on Windows, since issue 247 is fixed.
BUG=
TEST=Tried out on the master during after-office hours.
Review URL: http://webrtc-codereview.appspot.com/379011
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1621 4adac7df-926f-26a2-2b94-8c16560cd09d
The files are shorter (7 s) with one set provided for each sample rate.
Will be accompanied by the following set of files in the resource bundle:
far8_stereo.pcm
far16_stereo.pcm
far32_stereo.pcm
near8_stereo.pcm
near16_stereo.pcm
near32_stereo.pcm
BUG=114
TEST=audioproc_unittest
Review URL: https://webrtc-codereview.appspot.com/380003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1617 4adac7df-926f-26a2-2b94-8c16560cd09d
Focused responsibility of supported tests in master.cfg instead of being in utils.py (hard to overview and maintain).
Enabled the following empty tests on all platforms:
- audio_conference_mixer_unittests
- cng_unittests
- g711_unittests
- g722_unittests
- pcm16b_unittests
- media_file_unittests
- udp_transport_unittests
- webrtc_utility_unittests
Removed "headless tests" concept since everything is now compiled in the make all step (no need for compile only, no execution tests).
Removed audio_device_test_func test since not a proper test (dev tool) that was configured as headless.
BUG=
TEST=Ran local master and successfully built and executed all tests with Mac build slave.
Review URL: http://webrtc-codereview.appspot.com/384002
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1603 4adac7df-926f-26a2-2b94-8c16560cd09d
Now the unit test is included in the vie_auto_test target and executed when the automated flag is used.
TBR=mflodman
BUG=
TEST=vie_auto_test --automated --gtest_filter=FrameDropPrimitivesTest.FixOutputFileForComparison
Review URL: https://webrtc-codereview.appspot.com/381003
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1598 4adac7df-926f-26a2-2b94-8c16560cd09d
Refactoring of FrameDropHandler: It now also tracks when frames are leaving the encoder and is being sent to external transport.
Previous 'Sent' state is now renamed to 'Created'.
NOTICE: The test seems to be a little flaky on Linux so it's not ready for buildbots yet. Since this might be caused by unstable production code further investigation should be performed to clear out the flakiness. I will file an issue for this when this CL is submitted (since I don't have any code to refer to before that). Usually the flakiness is caused by a decoded/rendered callback that is left out for the last frame, but I have seen other flaky failures too, which means it's not as simple as ignoring the last frame.
These errors occur even if 400kbps bit rate and 0% PL and 0 delay is configured.
BUG=
TEST=vie_auto_test --automated --gtest_filter="ViEVideoVerificationTest.RunsFullStackWithoutErrors" in Debug+Release on Linux, Mac and Windows.
Review URL: http://webrtc-codereview.appspot.com/339005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1597 4adac7df-926f-26a2-2b94-8c16560cd09d
and touched VoEBaseImpl::NeedMorePlayData and AudioCodingModuleImpl::PlayoutData10Ms(), for
performance reasons in Android platforms.
The two functions used about 6% of VoE originally. After the change, the percentage reduced
to about 0.2%.
Review URL: https://webrtc-codereview.appspot.com/379001
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1589 4adac7df-926f-26a2-2b94-8c16560cd09d
Note that all files were moved to a new directory. The diffs won't be 100% friendly because of this.
Extracted common handling for OAuth on both sides of the connection in order to add a new build status
data handler. This data handler will be used to report build status data. Don't look too closely at the
details of what data is transferred as this will change in the next patch. We will also extract data from
a different page in a slightly different way, but there won't be huge differences.
In particular, we won't look at the /one_box_per_builder page on the master but rather at the transposed
grid (/tgrid) on the build master since we also need the revision number. The regular expressions to
extract the data will be slightly more complex.
BUG=
TEST=
Review URL: https://webrtc-codereview.appspot.com/367023
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1586 4adac7df-926f-26a2-2b94-8c16560cd09d
Investigation with corrupt payloads revealed a few places we could
be using stronger checks. These are not foolproof by any means, but
I figure the earlier we catch this the better.
BUG=242
TEST=loopback call with a hacked ViE to insert corrupt payloads, and vie_auto_test without the hacks.
Review URL: https://webrtc-codereview.appspot.com/369015
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1585 4adac7df-926f-26a2-2b94-8c16560cd09d
Style and return value changes. No impact externally, since audio_processing, audio_conference_mixer and audio_coding either already assumes 'int' as return value, assumes nothing or doesn't take care of the return value.
TESTS=vad_unittests, audioproc_unittest
Review URL: https://webrtc-codereview.appspot.com/374006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1581 4adac7df-926f-26a2-2b94-8c16560cd09d
