[asterisk-bugs] [JIRA] (ASTERISK-25599) [patch] SLIN Resampling Codec only 80 msec

[Alexander Traud (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25599?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Alexander Traud updated ASTERISK-25599:
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    Comment: was deleted

(was: Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].)

> [patch] SLIN Resampling Codec only 80 msec
> ------------------------------------------
>
>                 Key: ASTERISK-25599
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25599
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/codec_resample
>    Affects Versions: 11.20.0, 13.6.0
>            Reporter: Alexander Traud
>            Severity: Minor
>         Attachments: opus_resample.patch
>
>
> {{WARNING: translate.c:402 Out of buffer space}}
> Currently, {{codecs/codec_resample.c}} uses a buffer with a fixed size of 8096 for each signed-linear resolution. Furthermore, these 8096 depend on the size of {{int16_t}} of the platform. On my machine, this results in a maximum of 4048 samples. However for example, the Opus Codec is able to create 5760 samples (6 frames a 20 milliseconds @ 48000 samples per second). Consequently, the current buffer is too small for the Opus Codec when used with a packetization time of more than 80 milliseconds, for example in chan_sip with sip.conf: {{allow=opus:120}} or {{autoframing=yes}}.
> The attached patch uses the maximum of the Opus Codec (5760) as the new minimum sample amount. Therefore, the actual amount of bytes must depend on {{int16_t}} and not the other way around. This guarantees we are able to convert at least 5760 samples, regardless of the platform.



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