bitflipper
Just once I'd like to see somebody publish before-and-after waterfall plots. It'll never happen, though.

 
:-)
 
 
 
The vendors should do it. In the meantime I'm left to assume that the flat white line in one app is every bit as a good as the flat white line in another app.

bitflipper
Just once I'd like to see somebody publish before-and-after waterfall plots. It'll never happen, though.

Never is a long time. :)

I'd expect that the perfect flat response is at the mic location, with the same measurement mic unmoved from the spot, and probably using only one mic location.  Move the mic a few inches (or maybe less) and I believe it will look considerably different.  They do point out in their lit that perfect flatness is the secondary objective, and if I read correctly, you can set a target response for the amount of correction (or maybe limiting over-correction).  Not sure if ARC offers that, but still worth a mention. 
 
I agree that these apps are not a panacea, but can help 1) if you already have corrected the big stuff with traps, diffusers, etc. and want that "last few dB" of clarity and focus, or 2) you are stuck with a lousy environment and need whatever help that you can get, limitations or not.  All told, if it can tame fluctuations down to even a couple (or few dB) over a sweet spot bigger than the size of my head, then I think I'll take it happily.
 
Response attenuation in 64-bit floating-point samples is not a big deal - you can easily correct for 20-30dB with insignificant impact on the audible dynamic range at the final rendering to linear bits. You only really need to compensate for the gain reduction before or after the fact (metering with pink noise or your favorite program material should do the trick).

 
Bose uses an active technology to modify environmental sound conditions.
 
Here's an example: Cadillac
 
"Active noise cancelation also contributes to the Cadillac CTS sedan’s refined performance, reducing noise levels by up to 20 decibels in certain conditions. It is enabled by the Bose sound system and a unique processor that takes input from several interior-mounted microphones and sends opposite-tuned frequencies through the sound system’s speakers to cancel out undesirable sounds."
 
I am trying to understand how a electronics engineer can *flat line* a room without using some sort of dynamic correction. That is why I keep, optimistically, imagining the use of dynamic convolution.
 
I suspect it is easier to imagine that a flat line response is possible if you are imagining the *wiggly line* is a static or averaged response. In real time the response experienced, in a enclosed space with reflections, is wiggling up and down all over the place over the course of time.
 
If you use a Real Time Analyzer in a room you see how the response is wiggling all over the place through the course of time and the idea of a single drawing of a *wiggly line* begins to seem like an overly simplified abstraction of what is really being experienced at the listening position.
 
 
I think I understand how a FIR filter can use a single impulse response to alter the phase of the output compared to the input. I struggle to imagine how any technology that uses a single impulse can be adroit, and offer the sort of constant corrections, that I suspect are necessary, to approach an ideal such as a flat line graphic represents. I keep imagining that the circumstance demands some sort of dynamic application of a series of rapidly applied corrections.
 
The BOSE technology such as the example of it cited above makes sense to me. It seems relatively simple. Real time sampling and an active application of near real time corrections resulting in near real time control.
 
 
?
 
 
 
 
 

Been off reading for a few moments...
 
Continuous Time systems?

mike_mccue
 I am trying to understand how a electronics engineer can *flat line* a room without using some sort of dynamic correction. That is why I keep, optimistically, imagining the use of dynamic convolution.
 
I suspect it is easier to imagine that a flat line response is possible if you are imagining the *wiggly line* is a static or averaged response. In real time the response experienced, in a enclosed space with reflections, is wiggling up and down all over the place over the course of time.
 
If you use a Real Time Analyzer in a room you see how the response is wiggling all over the place through the course of time and the idea of a single drawing of a *wiggly line* begins to seem like an overly simplified abstraction of what is really being experienced at the listening position.
 
 
I think I understand how a FIR filter can use a single impulse response to alter the phase of the output compared to the input. I struggle to imagine how any technology that uses a single impulse can be adroit, and offer the sort of constant corrections, that I suspect are necessary, to approach an ideal such as a flat line graphic represents. I keep imagining that the circumstance demands some sort of dynamic application of a series of rapidly applied corrections.
 

It just depends on the degree to which the room's response is linear. If your signal is wiggling on its own, the response will also wiggle.
 
I believe that, ignoring things like absorption* and speaker distortion, the response due to reflected sound is linear. Note that the impulse response captures response of the reflections until they fall below the noise floor.
 
 
 
*Hmmm... 

 
More like DSP, along the lines of Finite Impulse Response (FIR) multipath channel compensators:
http://dsp-book.narod.ru/spra140.pdf
http://www.cplire.ru/rus/informchaoslab/papers/iccsc04ak.pdf
 
A multipath channel has a direct path and other paths (echoes) with different delays and attenuations, where the attenuations may themselves be frequency-dependent.  For studios, this translates to the idea of the mirror technique in your room, tracing out the line path that waves take from your speakers to your ear(s).  The compensator for multipath uses the information in the main and echo paths to create a filter (a convolution operation) whose coefficients equalize out the echo paths and leave the main path as much by itself as possible. In the second paper above, the multipath channel's impulse response is in Figure 5, and the compensated result is in Figure 7.  The spike in the second figure is what you want, and the junk along the baseline is what's left.
 
Techniques like these can actually compensate for echoes that produce complete nulls in the frequency response.  It sounds weird, I know, but it's because they operate in the time domain.  The lit from IKM and MathAudio don't emphasize this, but this has been "the way to do it" for, like 20+ years now.  Perhaps they emphasize the frequency domain because they're using it to apply a type of constraint that the user can specify over the process, to limit over-compensation at certain frequencies, for example.
 
Edit: What's really new to what they are doing, that I can tell, is applying this tech to the case of studio acoustics, accommodating and optimizing over multiple mic positions, and adding the constraint of "target responses" specified in the frequency domain *amplitude* response, and of course, packaging it as a VST plugin.
 
On waterfall plots: As long as the room is not too reverberant, the above techniques act to convert the cluster of reflection impulses to a single main-path spike, but only at the measurement point (the mic). Widening that point out to the size of a watermelon is challenge enough, going wider than that is where I'd like to see the proof in the pudding.

If I had the money I'd buy two ARC 2s.
 
Gotta be twice as good.
 
Or....... three. LRC.

Good thing I don't have a 5.1 setup.
 
I'm sure IKM will not sell me .1 ARC 2.
 

losguy
 
More like DSP, along the lines of Finite Impulse Response (FIR) multipath channel compensators:
http://dsp-book.narod.ru/spra140.pdf
http://www.cplire.ru/rus/informchaoslab/papers/iccsc04ak.pdf
 
A multipath channel is on that has a direct path and other paths (echoes) with different delays and attenuations, where the attenuations may themselves be frequency-dependent.  It translate to the idea of the mirror technique in your room, tracing out the line path that waves take from your speakers to your ear(s).  The compensator uses the information in the main and echo paths to create a filter (a convolution operation) whose coefficients equalize out the echo paths and leave the main path as much by itself as possible. In the second paper above, the multipath channel's impulse response is in Figure 5, and the compensated result is in Figure 7.  The spike is what you want, and the junk is what's left.
 
Techniques like these can actually compensate for echoes that produce complete nulls in the frequency response.  It sounds weird, I know, but it's because they operate in the time domain.  The lit from IKM and MathAudio don't emphasize this, but this has been "the way to do it" for, like 20+ years now.  Perhaps they describe their use of the frequency domain as a type of constraint that the user can apply to the process, to limit over-compensation at certain frequencies, for example.

 
The first link or example, http://dsp-book.narod.ru/spra140.pdf, seems to be focused on radio transmissions where there is dsp operating at the receiving end to help facilitate the ability to resolve to the center tap. 
 
I don't have a dsp chip running in my ears... yet.