[asterisk-bugs] [JIRA] (ASTERISK-25684) noise instead of audio on slin codec with translation

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25684?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-25684:
-----------------------------------

    Assignee: Victor Sverdlin
      Status: Waiting for Feedback  (was: Triage)

What byte order is the signed linear being sent in? If this is wrong then you would get exactly what you are hearing.

> noise  instead of audio on slin codec with translation
> ------------------------------------------------------
>
>                 Key: ASTERISK-25684
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25684
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Codecs/General
>    Affects Versions: 13.1.0, 13.4.0, 13.6.0
>         Environment: Ubuntu 14.4 i686 3.19.0-42 on VMware WS 10
> FreeBSD 10.2 i386 on VirtualBox 5.0.10
> Ubuntu 14.4 i686 3.19.0 on bare hw (intel core i5)
>            Reporter: Victor Sverdlin
>            Assignee: Victor Sverdlin
>            Severity: Minor
>         Attachments: slin_noise_01.pcap
>
>
> Then one peer is slin (L16/8000) or slin16 (L16/16000) and second peer is other codec (tested with ulaw, gsm and speex) slin peer listen only noise. Noise disappear if other peer muted. Slin peer can listen voice if signal is gained down by AGC set to 10 and less (anyway sound is distorted).
> Both peers use sip channel.
> Testes with several peers:
> - slin: MicroSIP 3.10.9, custom HW device
> - other: antiSIP 4.2.9 (Android), SFLphone 1.3.0, Zoiper 3.3.25608
> Asterisk compiled with gcc 4.8.5(FreeBSD) and 4.8.4(Ubuntu).
> On Ubuntu Asterisk compiled with DONT_OPTIMIZE option.
> RTP capture will be attached.



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