17 May
2018
17 May
'18
2:04 a.m.
Le 16/05/2018 à 22:45, Chris Caudle a écrit :
The use-case is about voice calls. Humans are rather tolerant about the
latency of a phone call. isn't it?
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. 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
Christophe