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Discussion Starter #1
When we look at the improvements produced by digital room correction systems we usually look at the frequency response graphs... or at least this is the aspect most people pay attention to.

But in evaluating the performance of our system (speakers and listening room as a whole) I believe we should not oversee this comments from the REW's documentation:

"The frequency response is only half of the description of what the system is doing to signals that pass through it, the phase response is the other half. Trying to understand systems by looking at the frequency response alone is like trying to understand a book by reading only the even numbered pages. To really understand you need to look at both. That is a bit problematic, however. The frequency response is fairly easy to understand, but the phase response doesn't give up its secrets quite so easily"

The easiest way to evaluate the phase response is to look at the impulse response. I'm posting here a screenshot by a forumer with the pulse responses "before" and "after" correction in his listening room, together with an explanation on how to interpret them by Jakob Agren, one of Dirac Research engineers:



The impulse response tells us how a system behaves when excited by an impulse. An impulse in this context refers to a signal that in theory have infinite height and no width, that is, all energy comes at the same point in time, with nothing happening before this moment, nor after. For a system consisting of a speaker in a room to be "perfect" it need to do nothing to the signal, we want to hear exactly what is on the recording, nothing more, nothing less. To achieve this, the impulse response needs to be an impulse.

If it is not, something has happened to the signal along the way. Worth noting is that there are no systems with a perfect impulse response, since it would (among other things) require infinitely high frequencies to achieve.
To judge if an impulse response is better or worse than another when comparing two different systems, or in this case, the same system before and after compensation, we need decide if the impulse response looks more like an impulse after compensation, or less.

A better impulse response will have a more distinct first peak, with more distinct meaning the main impulse is higher compared to the tail than before. We also want the main impulse to be narrower. In the example plot we can see that the ratio of main impulse to tail has increased by about a factor three, the main impulse is higher while the tail is roughly the same. The interpretation of this is that the direct wave (the main impulse) from the speaker to the listener is more distinct than the reflections (the tail) after compensation.

The values on the x-axis are time in miliseconds. The delay of the compensated impulse response is a consequence of the compensation. In order to improve the impulse response you need to have some headroom in time in which to apply the compensation.
In order to achieve this, the output is delayed slightly, introducing a latency in the audio chain. This is a static delay, and is not system dependent, meaning it does not depend on the room or the speaker (the delay to the uncompensated impulse response depends on the distance from the speaker to the microphone however).

The values on the y axis are normalized and are used to compare the before and after results. In this case we can see that the direct wave from the speaker is much more distinct, as it is about three times higher while the reflections are on roughly the same level. Note that as the y values are normalized, the absolute values have no meaning.

This is certainly not all information that is available in an impulse response, but they are hard to inspect visually and most of the information are easier to interpret in the frequency domain.
The image has been created by Dirac Live®, a patented Pc and Mac DRC application that successfully applies phase correction to a listening area of arbitrary size.

I hope this has not been boring,
Flavio
 

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Interesting reading. How do you effect the impulse... With a MiniDSP?
 

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Discussion Starter #3

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I think the more traditional way of improving impulse response is through room treatment. That is, to reduce the magnitude of the reflected signal. No?
 

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That is correct. There are aspects of the impulse response mostly due to speaker contribution that can sometimes be improved somewhat by digital room correction (DRC), too, but for most users the room is by far the bigger contributor, with acoustical treatment the ideal attack. That said, pushing a few buttons to run DRC is mighty tempting, and often gives surprising improvements.
 

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Does this imply the mic was (approximately) 26 feet from the speaker? I am using Dirac on my music server and don't recall seeing that much time between the two impulses.

In a previous life, I installed the original digital room correction hardware in home around the US (SigTech). We used the impulse charts to find where to apply passive correction since the delay between the original impulse and the following ones gave us at least a hint of where to look. It usually took some trial and error but the good news is that you could see if (or if not) the passive treatment actually dealt with the problem.

Thanks for posting this.
 

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Discussion Starter #10 (Edited)
Does this imply the mic was (approximately) 26 feet from the speaker? I am using Dirac on my music server and don't recall seeing that much time between the two impulses.
Hello Audioguy,

as you say 26 milliseconds (roughly 26 feet) are a long time but you should take into account that the first impulse is placed at appx. 10 milliseconds and that means that the mic was placed at appx. 10 feet from the speaker.
The other 16 milliseconds from the first impulse to the second one are more interesting... that's the headroom in time that was required for processing to be able to apply the phase compensation by delaying those frequencies that need that (since we do not know about the future some latency is necessary)
That latency will not affect the sound quality at all... let's say that the playback will start 16 milliseconds later.

Ciao, Flavio
 
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