When considering wireless links, there are three domains possible: - Analog - Digital, not IP passed; DECT, BT, proprietary. - Digital, IP-based; Wi-Fi, LTE, Wimax.
If latency is the concern I think that non-IP based approaches may have an inherent advantage. 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 Augustine Leudar Sent: Thursday, May 30, 2019 4:43 AM To: Surround Sound discussion group <sursound@music.vt.edu> Subject: Re: [Sursound] wifi audio (was Re: Deconstructing soundbar marketing B.S.) Leaving aside the power issue for now and with regard to multichannel audio for shows as opposed to home use. I had some walkie talkies that had a range of one KM with admitedly terrible audio (surely this could be improved) . Whereas Senheiser in ear monitors have a really short distance range of around 40 metres and use much higher electromagnetic frequencies ((863 mhz) . Why is it something cant be done with the same sort of range as the walkie talkies but for.multichammel audio (according to wikipedia 30 - 400 mhz) ? On Wednesday, 29 May 2019, Wim <object...@gmail.com> wrote: > Dante/AVB have a latency under 5 ms, transporting many channels, even > @96 kHz. It can be done. Just not wireless. > > The major problem with wireless lays in the re-authentication that > occurs after a preset period. That takes up to several hundred > millisecs. Not a problem for a download, or viewing a webpage. Big > problem for low-latency streaming. Running without any encryption > makes it less, but then you also need a good S/R on the wireless side > to stop it from having other problems, like switching channels, or speed. > > Apple's solution for AirPlay is having a big buffer in their devices. > I believe the old Airport Express has 1 to 4 MB allocated for > streaming buffer, resulting in seconds of latency. Not a problem for playback. > > BT is even far worse, and the range is too limited. > > I've tried most of the possibilities, with Apple devices, Raspberry Pi > and ESP8266. It works. It's just not reliable. I've used it for a > little while, for recording in forests, where there's no neighbouring wifi to > be found. > I've reverted back to VHF wireless mics. Less of a hassle. > > Just my 2 eurocents. > > Wim > > Op wo 29 mei 2019 om 17:41 schreef mgraves mstvp.com <mgra...@mstvp.com>: > > > 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 > > > > > > > > --- > > This email has been checked for viruses by Avast antivirus software. > > https://www.avast.com/antivirus > > > > _______________________________________________ > > Sursound mailing list > > Sursound@music.vt.edu > > https://mail.music.vt.edu/mailman/listinfo/sursound - unsubscribe > > here, edit account or options, view archives and so on. > > _______________________________________________ > > Sursound mailing list > > Sursound@music.vt.edu > > https://mail.music.vt.edu/mailman/listinfo/sursound - unsubscribe > > here, edit account or options, view archives and so on. > > > -------------- next part -------------- An HTML attachment was > scrubbed... > URL: < > https://mail.music.vt.edu/mailman/private/sursound/attachments/2019052 > 9/54bacb40/attachment.html > > > _______________________________________________ > Sursound mailing list > Sursound@music.vt.edu > https://mail.music.vt.edu/mailman/listinfo/sursound - unsubscribe > here, edit account or options, view archives and so on. > -------------- next part -------------- An HTML attachment was scrubbed... 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