[asterisk-dev] [Code Review] 3793: media_formats: Add minimum bytes to builtin codecs (so that smoothers can actually be made)

rmudgett reviewboard at asterisk.org
Tue Jul 15 12:38:31 CDT 2014


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3793/#review12656
-----------------------------------------------------------

Ship it!


Ship It!

- rmudgett


On July 15, 2014, 12:04 p.m., Matt Jordan wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3793/
> -----------------------------------------------------------
> 
> (Updated July 15, 2014, 12:04 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> This patch does two things:
> (1) It makes the full value of the sample rate of 44kHz SLIN 44100 (as that is what it actually is...)
> (2) It updates the built-in codecs with their minimum_bytes values. These values were pulled from the previous static format definitions in Asterisk 12. Using this, smoothers can be created successfully.
> 
> 
> Diffs
> -----
> 
>   ./team/group/media_formats-reviewed-trunk/main/codec_builtin.c 418631 
>   ./team/group/media_formats-reviewed-trunk/codecs/codec_resample.c 418631 
> 
> Diff: https://reviewboard.asterisk.org/r/3793/diff/
> 
> 
> Testing
> -------
> 
> The sip_tls_call test previously failed without this patch due to the RTP engine failing to make the smoother. With this patch, it now passes.
> 
> 
> Thanks,
> 
> Matt Jordan
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140715/f1d84307/attachment.html>


More information about the asterisk-dev mailing list