kjellander@webrtc.org
6c35e0b0f7
Reorganize test targets in WebRTC
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This CL will lower the number of test targets in WebRTC by:
Add common_audio_unittests and merge the following targets into it (copied from http://review.webrtc.org/1584006 ):
* resampler_unittests
* signal_processing_unittests
* vad_unittests
Merge into modules_unittests:
* bitrate_controller_unittests
* desktop_capture_unittests
* media_file_unittests
* remote_bitrate_estimator_unittests
* rtp_rtcp_unittests
* paced_sender_unittests
Merge into test_support_unittests:
* channel_transport_unittests
channel_transport.gyp was also removed in favor for test.gyp.
I had to remove a main method from rtcp_format_remb_unittest.cc
since it caused the fileutils.h code to not be able to find the
right project root path in ordrer to provide correct paths
to test files.
Buildbot configuration update will be synced with the commit of this CL.
TEST=trybots
BUG=1843
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1639004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4213 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-11 08:29:17 +00:00
andrew@webrtc.org
f791b1cebf
Rename voice_engine_core -> voice_engine and move targets to voice_engine.gyp.
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R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1574004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4126 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-29 00:38:02 +00:00
pwestin@webrtc.org
e30823911c
Move the VoE tests to use external transport instead of the built in udp transport
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Review URL: https://webrtc-codereview.appspot.com/1223006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3708 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-22 16:12:57 +00:00
pwestin@webrtc.org
684f0577fb
Revert r3667 and r3665
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Review URL: https://webrtc-codereview.appspot.com/1199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3668 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 23:20:57 +00:00
pwestin@webrtc.org
361bac7a4f
Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work.
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Review URL: https://webrtc-codereview.appspot.com/1029004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3665 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-13 17:52:42 +00:00
phoglund@webrtc.org
5d37139374
Fixed a ton of Python lint errors, enabled python lint checking.
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BUG=
Review URL: https://webrtc-codereview.appspot.com/1166004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3627 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-03-07 09:59:43 +00:00
phoglund@webrtc.org
acc54b41da
Added perf expectations and corrected existing tests to remove spaces from series names.
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The make_expectations.py script is copied from Chrome. It can look at a test's result history and automagically compute good regress/improve values.
This patch comes into play together with https://chromereviews.googleplex.com/6242014/ .
NOTE TO SELF: save history on master when deploying!
Review URL: https://webrtc-codereview.appspot.com/1019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3339 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-07 12:05:53 +00:00
phoglund@webrtc.org
3bb42ef0d6
Made e2e audio quality test write its results to perf.
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The https://chromereviews.googleplex.com/5573026/ patch will mark the test step as perf-printing - this cl will make the test actually print perf lines.
TEST=Ran locally.
Review URL: https://webrtc-codereview.appspot.com/933036
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3252 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-12-10 10:08:00 +00:00
andrew@webrtc.org
14b43beb7c
Move src/ -> webrtc/
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TBR=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/915006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
andrew@webrtc.org
6f8db36e04
Reorganize voice_engine/.
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The usual changes:
voice_engine/main/source -> voice_engine/
voice_engine/main/interface -> voice_engine/include
voice_engine/main/test -> voice_engine/test
Include path changes.
BUG=none
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/705004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2535 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-27 21:49:28 +00:00
andrew@webrtc.org
d41f59a23f
Fix Mac-gcc warnings.
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Resolves:
- warning: allocating zero-element array
- warning: suggest a space before ‘;’ or explicit braces around empty
body in ‘for’ statement
BUG=none
TEST=build on Mac-gcc, trybots
Review URL: https://webrtc-codereview.appspot.com/675006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2519 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-07-16 17:05:47 +00:00
andrew@webrtc.org
9f6577b304
Restore default source in e2e test.
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Also wait for VoE to start up.
TBR=kjellander@webrtc.org
BUG=issue502
TEST=audio_e2e_test
Review URL: https://webrtc-codereview.appspot.com/643006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2391 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-12 03:25:03 +00:00
andrew@webrtc.org
f4c6aa2e81
Improve the reliablity of the audio e2e test.
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- Use higher quality resampling.
- Add a longer delay before pacat recording.
TBR=kjellander@webrtc.org
BUG=issue502
TEST=manually
Review URL: https://webrtc-codereview.appspot.com/646005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2374 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-06-07 04:00:15 +00:00
andrew@webrtc.org
aa5f410db5
Add a README to e2e_quality.
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TBR=kjellander@webrtc.org
BUG=issue502
TEST=none
Review URL: https://webrtc-codereview.appspot.com/590004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2278 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-23 15:11:20 +00:00
andrew@webrtc.org
2b004655b5
Add comparison to e2e and clean up.
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BUG=issue502
TEST=manual
Review URL: https://webrtc-codereview.appspot.com/577006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2274 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-23 02:47:58 +00:00
andrew@webrtc.org
b06db96840
Add a framework for audio end-to-end quality testing.
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The quality comparison step is still to be done.
BUG=issue502
TEST=manual
Review URL: https://webrtc-codereview.appspot.com/577005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@2220 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-05-10 18:45:11 +00:00