|Implement unit tests for Audio Conference Mixer module|
|Reported by email@example.com, May 7 2014||Back to list|
The Audio Conference Mixer module lacks proper testing.
May 7 2014,
There used to be a test from GIPS which was removed in this cl: https://code.google.com/p/webrtc/source/detail?r=4511 It was never ported to using gtest so it was deleted. Don't know if the code is still usable.
May 7 2014,
We can probably use that as a starting point.
May 8 2014,
May 22 2014,
Aug 13 2014,
Oct 30 2014,
Nov 4 2015,
This bug hasn't been modified for more than a year. Is this still a valid open issue?
Nov 5 2015,
It still needs to be done if/when we start working on the mixer.
[Bulk edit] This issue hasn't been modified the last twelve months -> archiving. If this is still a valid issue that should be open, please reopen again.
aleloi@: the legacy AudioConferenceMixer is gone (https://codereview.webrtc.org/3015553002). Would you say that we have a better unit testing situation with the new one?
Yes, the testing situation is better. There are a two low-prio bugs webrtc:7364 and webrtc:8036 to add even more tests. We don't have test coverage for running more than one stream through the decoder, fake network, encoder and mixer. I looked into the possibility to extending CallTests and found it difficult. I heard rumors that someone else is working on CallTests to make that possible.
Thanks. Then this is obsolete, because the component no longer exists, and the replacement is better tested.
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