Hello Michael,
I agree with you 100 
percent about forgetting realtime and creating the wav files that can be played 
on my existing setup. Even in the past, when I've used MATLAB to alter signals 
(for example, a MATLAB-implemented phase vocoder for 
implant simulations), I do the processing off-line, save the file(s), 
and then line them up in a DAW for reproduction in a quasi-random order. What 
you suggest is what I've intended to do all along (although not 
necessarily with Linux/AmbDec until it was suggested). Maybe I've been 
using 'real-time' in an incorrect way. Please allow me to explain...
Because some of my experiments involve adaptive test procedures, something has 
to 'give' during the test (during = the real-time part). The wav  files, 
however, do NOT undergo processing once they've been lined up to play in a 
particular sequence 
via a multi-track device or DAW. An example of an adpative procedure 
would be trying to 'force' a speech comprehension score of 50 percent 
correct, and altering the SNR to achieve this score. The background 
noise level, signal level, or both would have to change in 'real time' 
(that is, change while the test is in progress) to zero in on the SNR 
that yields a 50 percent speech score, subject by subject. Background noise is 
continuous and on its own tracks, while the speech stimuli is lined up 
sequentially on a separate track or tracks.The background noise plays through x 
number of speakers at a time (let x = 6 for horizontal-only Ambisonic surround) 
while the speech plays 
through a dedicated speaker or speakers, but always one at a time. If I want 
the speech signal to be 
maintained at, for example, 75 dB SPL and the listener isn't having 
difficulty with the speech material at + 5 dB SNR, then the background level 
(all six 
channels) has to be elevated as the test moves forward. All I really need to do 
is automate the levels, not process the wav 
files from B-format to speaker feeds. To change the background noise 
level (presented through 6 speakers) means that all six channels have to change 
in unison and precisely by the same amount, perhaps in 2 dB 
steps. This is pretty straightforward with a discrete, 6-channel preamp. With a 
DAW, as you know, controlling six channels simultaneously 
requires sending them to a buss with a minimum of six channels (a 7.1 
surround buss works nicely--just don't panpot anything and keep the 
channels totally discrete). The surround buss's master fader can adjust 
all channels simultaneously, and I do this in 'real-time' based listener 
responses. Actually, responses are recorded, and my software/hardware 
combo does the fader adjustment automatically.

Things get slightly more complicated when the research involves 
electro-acoustic simulation (EAS), which is a hybrid form of cochlear 
implantation. Again, I don't need to do 
anything with the Ambisonic processing, but 'real time' filtering comes 
into play. Briefly, the listener hears an acoustic signal at or below a 
certain f0--say 250 Hz--while the speech in the ranges of 250 Hz to 8 
kHz is presented electronically via his/her implant or via a simulation. I have 
programmable, hardware digital filters (up to 8th order) for my 
filtering needs, so I don't have to do wav file processing prior to 
presenting the stimuli, nor do I have to run all of the multitrack 
channels through a VST filter. As with the aforementioned adaptive test 
protocol, the test subject's responses are electronically recorded, 
responses are fed back (after an algorithm does the decision making), 
and signals are adjusted accordingly to achieve a certain outcome.
Ok, maybe I didn't explain this all too well, but at least it should help 
explain my definition of real-time, at least as far as my personal setup goes.
As always, many thanks to all for the feedback, suggestions, and 
questions. The questions make me think harder, and perhaps my bifurcated 
ganglion-of-a-brain will grow at some point!
Cheers!
Eric C.
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