Hi Richard, I’ve changed the title of this topic to something more relevant.
I still prefer the term Delta Stereophony to describe this. It seems to date back to the mid 1980’s, and was described by Gerhard Steinke and Wolfgang Ahnert. They were working in East Germany behind the Iron Curtain, reputedly working with Sinclair ZX Spectrum computers and expensive AKG delay lines somehow imported from Austria. It does make a great deal of sense. When digital delay lines became more generally available and affordable (1990’s ???) they were increasingly used in public address systems to improve coverage over a greater area, using speakers down the length of an auditorium to augment the usual left/right or LCR main frontal system. The feed to these was delayed by an amount that caused the time of arrival of sound from them to match that of the main frontal system . Sometimes the feed to a "front fill” system, arrayed along the front of the stage to increase clarity in the rows of seating near the stage, was also delayed to match the time of arrival of sound from its source. Amplitudes were usually adjusted by ear, as indeed were delay times after an initial calculation. These systems were more “appropriately distributed mono” than spatial. It is impossible to get the delay/amplitude combination correct for every position in the space with a finite number of speakers and output channels, so compromises are inevitable. This became common practice, especially for large scale stadium events. Digital mixing desks now commonly incorporate delays on each output, making this simpler to implement. Current products do not allow progress to true Delta Stereophony (DBADP), as the architecture does not provide delay as well as amplitude control on each matrix crosspoint, and the market doesn’t expect or demand it. DSP chips are now capable of providing it, as proved by TiMax, LISA, d&b’s Soundscape, Iosono , Astro, and Meyer’s relaunched system. The market is small, and the DSP boxes pricey. It becomes relatively more affordable for large multi-speaker systems with large budgets. For the rest of us, it’s down to software. Ircam have a basic implementation of DBAP in Spat~ for Max/MSP (or you can roll your own), and adding the delay component is relatively simple. You can then scale the amplitude and delay separately for each source, as seems appropriate. Using delay alone is surprisingly effective. The variation of amplitude between widely spaced speakers can be excessive. Of course you need a fast and powerful computer, and efficient programming to do this, but that is also true with any of the alternative algorithms (ambisonics, VBAP, DBAP, WFS etc.). None of these are perfect for every situation, and it is hard to envisage a combination of them that would work. Ciao, Dave Hunt > On 14 Nov 2020, at 17:00, sursound-requ...@music.vt.edu wrote: > > From: Richard Foss <rich...@immersivedsp.com> > Subject: Re: [Sursound] Was: Recorder for ORTF-3D OUTDOOR SET > Date: 14 November 2020 at 16:48:36 GMT > To: sursound@music.vt.edu > > > Dave, I have meant to follow up on your message for some time, because your > ideas match what I am currently busy with - at last getting to it! > > Our first immersive audio implementation uses networked PoE miniDSP speakers > which each incorporate a matrix mixer with volume and delay control at the > cross points. The delays were a later addition, and I certainly found that > the localization was enhanced by incorporating delays. We implemented DBAP > for the amplitude panning, but we have implemented and experimented with > VBAP. Given that our targeted applications will need irregular speaker > configurations, we have settled on DBAP for now. > > We had an idea, similar to yours, to utilize the signal processing > capabilities of audio interfaces/mixers. Because we owned MOTU devices, we > tried this first on three of the MOTU devices, and have updated our ImmerGo > software to work with these interfaces. However, it was not possible to > implement a delay matrix on the MOTU devices, so they just have a DBAP > implementation, not DBADP (your innovative label:)). > > I am now working on a mixing console implementation where I believe I can > have delay/EQ at matrix cross points for a few channels, where there is an > inverse relationship between number of speakers and number of channels with > delay/EQ, although all channels can have DBAP. One does need to have mix > buses to enable this, and also there often is the timing constraint, because > a lot of messages go to the mixer as the sound sources are spatialized. I > have found that the MOTU devices are very responsive in this regard. > > Anyway, good to have a fellow DBADP enthusiast . > > Regards, > > Richard. > > — > Richard Foss (PhD) > Software engineer/director > > ImmersiveDSP > 46 Silver Ranch Estate > Keurbooms River Road > Plettenberg Bay 6600 > South Africa _______________________________________________ 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.
