Agreed. Most of what I think of as the "local signal processing" is quite 
speedy. Packetization delay is never less than 20 ms. Transmission delay 
dependent upon the network and distance. Poorly designed network elements lead 
to buffer bloat, which increases latency dramatically.

The very latest DECT chipsets are able to deliver a 12.5 kHz audio path from a 
microphone. Not sure how that's done. DECT is quite opaque. It remains the most 
common approach to a real-time wireless link built specifically for streaming 
audio.

Michael Graves
mgra...@mstvp.com
http://www.mgraves.org
o(713) 861-4005
c(713) 201-1262
sip:mgra...@mjg.onsip.com
skype mjgraves

-----Original Message-----
From: Sursound <sursound-boun...@music.vt.edu> On Behalf Of Chris Woolf
Sent: Wednesday, May 29, 2019 10:29 AM
To: sursound@music.vt.edu
Subject: Re: [Sursound] wifi audio (was Re: Deconstructing soundbar marketing 
B.S.)


On 28/05/2019 19:47, Marc Lavallée wrote:
> Le 28/05/2019 à 13:48, mgraves mstvp.com a écrit :
>
> ....
> The latency is not only caused by the packetization; the transmission 
> chain looks like:
>
> (microphone -> ADC -> encoding -> BT transmission) -> (BT reception ->
> decoding) -> (SIP + encoding -> IP transmission) -> (IP reception -> 
> SIP + decoding) -> (DAC -> loudspeaker)
>
True enough, but the ADC, encoding, decoding and DAC elements can be reduced to 
<3ms (as happens with some of the best recent digital radio mics), which does 
indeed indicate that the intermediate stages are the ones that really do the 
harm.

A while back I had to make a short range speech reinforcer for a friend with a 
damaged larynx. It had to use an analogue pathway because no (affordable at the 
time) digital path had anything like low enough latency to permit normal, 
unstilted conversation. A target figure ~has~ to be <10ms to avoid disturbing 
speech, and for most people/environments must be <<5ms. I find it laughable 
that "low latency" frequently seems to mean 30-50ms.

Chris Woolf



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