[Jack-Devel] How to use jackd as a 'system-wide' server?

Christophe Lohr christophe.lohr at cegetel.net
Thu May 17 09:03:58 CEST 2018

Le 16/05/2018 à 22:45, Chris Caudle a écrit :
> I would set to the same sample rate that asterisk expects so that no
> rate conversion is needed.

Well, in fact Asterisk does not expect a specific sample rate. This
depends of the codecs that have been negotiated with the peer for the
ongoing call...
By default, the sip trunk uses  ilbc g729 gsm g723 ulaw codecs. The
sampling rate can be 8kHz, 16kHz, or 13.3...
Maybe I can tell jack to use 16kHz sampling rate, and let it convert
when needed.

>> Linux xaal-c 4.15.17-1-MANJARO #1 SMP PREEMPT Thu Apr 12 17:29:48 UTC
>> 2018 x86_64 GNU/Linux
> That has the low-latency configuration set (PREEMPT), but not the full
> realtime patch (that would show PREEMPT-RT).  Should be fine for most use,
> but you may not be able to set to the very lowest latency settings. The
> dummy backend uses 1024 sample period size by default.

The use-case is about voice calls. Humans are rather tolerant about the
latency of a phone call. isn't it?

> What app do you have connected? 

The app is the speech-to-text Pocketsphinx (the code is slightly
modified for my needs)
Pocketsphinx uses pulseaudio as audio interface. So I need its jack
source/sink plugin.

So, there is no physical audio device in my system: the audio of the
caller is to be passed to the STT recognition system via the audio server.

Best regards

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