Chris,

Actually, I too come from a broadcast background, having installed graphics 
systems into production and master controls for over 25 years. I completely 
appreciate the demand for hard real-time and zero latency.

I've tracked Opus since its earliest days in the IETF CODEC working group. The 
standard has many operative modes. It's absolutely capable of full-bandwidth, 
in both lossy and lossless modes.

You will find it both in the production/contribution side of the house (remote 
codecs, STL, etc.) and distribution. It also dominates video conference space. 

It's basically the pinnacle of audio encoding at this point, having merged the 
best ideas from CELT, Silk and a few entirely new ones. It would be hard to see 
how any proprietary codec vendor could compete except where addressing a very 
narrow niche.

Michael Graves
[email protected]
http://www.mgraves.org
o(713) 861-4005
c(713) 201-1262
sip:[email protected]
skype mjgraves

-----Original Message-----
From: Sursound <[email protected]> On Behalf Of Chris Woolf
Sent: Friday, May 31, 2019 5:45 AM
To: [email protected]
Subject: Re: [Sursound] wifi audio (was Re: Deconstructing soundbar marketing 
B.S.)

On 30/05/2019 17:51, mgraves mstvp.com wrote:
> The RF issue of range, carrier frequency, channel width is quite separate 
> from the deliverable audio path.
>
> The Opus audio codec has revolutionized audio coding. It's able to deliver 
> full-bandwidth audio at bitrates not much more than what was once typical of 
> a telephone call. This means that the RF band need not be large to deliver 
> high quality audio over a digital link.

This answer is quite revealing of the different approaches and requirements 
within our audio field. My background is broadcast audio, so for origination 
purposes any digital coding has to be lossless, and latency has to be ~very~ 
low. Lossy coding is fine as a delivery format (and so would be OK for speaker 
feeds) but if the sound has to be processed en route the psychoacoustic stuff 
doesn't stand up. Likewise latency of 5-10ms can begin to alter performance, 
depending upon how the foldback is returned to an artist.

I don't know Opus but having read up its spec (on Wikipedia) it is lossy and so 
can only be used as a delivery format. I had to smile at 30ms latency being 
reported as adequate for musicians to feel "in-time" - not for the ones I've 
ever worked with. Likewise the suggestion that 45-100ms is acceptable for 
lipsync is laughable - that's up to 5 TV frames adrift. Maybe audiences have 
become inured to low quality standards. Latency for "live interaction" at each 
end of a phone line, and face-to-face a few feet apart in a room require very 
different standards - Opus's suggestion of 150ms for VOIP might just be 
acceptable for the first, but it would destroy the second application.

I don't doubt that it is a clever and well-designed codec, and that it is 
extremely useful, but one must keep in mind what it ~actually~ is rather than 
what it sounds like. Opus doesn't deliver full bandwidth audio, any more than 
other digitally compressed systems do. It delivers something that convinces 
most ears that it is a full bandwidth, full dynamic range signal, but it must 
always be remembered what is missing. 
If you used such a system to deliver sound to speakers (assuming there is a 
technique for maintaining multichannel phase coherence) it should work 
perfectly well. If you used it for passing the output channels of a microphone 
I doubt you would not remain happy for long.

Which also means that the statement "the RF issue of range, carrier frequency, 
channel width is quite separate from the deliverable audio path" must be very 
carefully qualified - it is only correct in very specific circumstances.

Chris Woolf



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