This CL reduce the size of chrome in release build by 70KB.
With this patch and r1592 , sizes.py reports 92255640 bytes with webrtc, down from 92485792 bytes.
The size is 88839360 bytes without webrtc.
BR,
/SX
Review URL: https://webrtc-codereview.appspot.com/380007
git-svn-id: http://webrtc.googlecode.com/svn/trunk@1637 4adac7df-926f-26a2-2b94-8c16560cd09d
* 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
