Technical Articles and Editorials
- Written by Brian Florian and Colin Miller
- Published on 30 November 2007
Page 1 of 2
Many surround sound receivers and processors these days offer an "Auto" set-up routine that attempts to configure the basic set-up for the consumer, including whether a speaker should be high-passed or not, which frequency to high-pass it at if applicable, the distance/delay setting for each speaker, and of course the calibrated level (loudness) for each one.
The process often includes EQ for some or all of the channels, usually with the tag that doing so "corrects the room". Seem like a good idea? Yes, a great idea, but when it comes to the practical application in a real room, the results can leave much to be desired.
In such systems, the manufacturer supplies a microphone with known anomalies that can be accounted for in the calibration software. You plug in the microphone, place it at a listening position, the system cycles test signals from different speakers, comparing what the microphone picks up to what it sent to the respective speakers, in terms of time arrivals, frequency response, phase response, polarity, etc., and then applies 'correction' in the form of compensatory delays, level-matching, polarity swaps (if necessary), and frequency response equalization as best it can to make up the difference.
So what's the problem? In terms of setting delay and matching channel levels, if the software/microphone combination works correctly, nothing. It should still be verified with manual calibration (because sometimes these systems produce inaccurate adjustments, or you might want to fine tune it after the fact), but in handling the basics, there's no reason why this portion of the auto set'up can't work just fine. The problems really arise in "Auto EQ."
"Auto EQ" (a.k.a. "Room Correction"), in many cases, ranges from a marginally helpful 'band-aid' to useless or severely detrimental alteration.
All rooms start as inherently flawed in terms of acoustic transparency. If they've got walls, floors, and ceilings, and most do, they have surfaces that offer reflections in additional to the direct sound that, assuming a reasonably good loudspeaker, we want to hear primarily, if not almost exclusively. Getting a room that controls these reflections through deliberate absorption and diffusion (dispersed reflection) treatments (use of wall and ceiling panels, i.e., Room Treatment) is a mixture of science and art. Note that, for the sake of discussion, we differentiate here between "Room Treatment", which we define as the use of physical objects in the room (absorption and diffusion panels), and "Room Correction", which we define as the DSP applied in SSPs and receivers to change the sound that you eventually hear. We have a previous article on the types of materials used in room treatments.
There are many consumers who have actually spent time and effort addressing acoustic issues by using room treatments in a sound system who will testify that it's a far more fruitful upgrade than the time and money spent swapping out cables, components, or even loudspeakers. Of course, actually doing this, compared to playing audiophile nervosa, is a real effort, an effort most consumers, and even hobbyists, will not tolerate, and are eager to rationalize into the lowest possible priority. In extreme cases, some would rather debate about refining their sound with digital audio cables than talk about slapping up some rigid fiberglass panels on reflective surfaces.
Enter "Auto EQ", or "Room Correction".
If you could electronically correct for room acoustics, that would be great. Unfortunately, the fact is, you simply can't.
The acoustic character of a room is imparted when the loudspeaker (or instrument, or person) creates a sound. That sound then travels in multiple directions. Some of the sound travels directly at the listener(s). Most of it travels somewhere else, bounces off of surfaces, and much of it eventually reaches the listeners after the 'original' sound that took the direct path. No matter what you do to the sound before it leaves a loudspeaker, you can't undo what happens to the sound after it leaves the loudspeaker.
This reverberant sound field, the sound that reaches the listener through reflection, is fundamentally different from the direct sound field that travels without the alteration of reflected surfaces. It's not only delayed from the direct sound field, but also delayed among itself, in that there is a gradual decay, and instead of instantly stopping, gradually fades away. The rate of decay is described as its T-60 time, or the time that it takes for a sound to drop 60 dB, or a factor of 1 million (for each 10 dB, the sound decreases by a factor of 10, and 106 = 1,000,000).
When this reverberant sound field recombines with the direct sound field at the listening position, the multiple delays as well as the differences of reflection characteristics of the various surfaces, add relative time arrivals, stretch out the duration of sound, add to the directionality of the sound, and alter the frequency response of the system.
If it's done in a controlled manner so that nothing extreme happens, the effect can be pleasant, and even more dimensional or believable. After all, this process happens in the real world, and our cognitive systems process this information to provide information about the location of the sound, and the environment in which the sound occurred.
That doesn't seem so bad, does it? Maybe not, from a subjective standpoint, if you like the effect, but it masks any acoustics actually recorded, and in any case, diminishes the ability to resolve the original content. We're not advocating that people listen to music in an anechoic chamber. After all, the recording engineers anticipated some reverberant character in playback systems. But, for optimal results in serious listening applications, the 'live' character that's so nice to enhance your singing in the shower is the worst case scenario for an audio system, and if you want something resembling accurate playback, the typical living room is a nightmare.