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An ad for a piece of audio gear got my attention and prompted me to post here. Basically, in describing the functionality of this item, the ad describes how the product helps to "time align frequencies so that the highs reach our ears first, which provides superior spatial imaging and richer harmonic content." So here are my questions.

1. Is it true that its better (or clearer, anyways) when the highs reach the listeners ears just before the lows? I'm a moderate newbie so this was sort of news to me.

2. Other than just buying this thing, what sort of things can be done in the recording and mixing process to make the highs arrive earlier? I'm of course familiar with EQing, but I am not aware of how to move certain frequencies ahead or behind in time.

3. I have a very modest recording setup - I've got decent mics, but they're just running into a Firepod and Digital performer 5. I still take recording very seriously, but I'm wondering if this is an issue that I should even bother being concerned with?

Thanks a lot folks!

*The product in question is the sonic maximizer by BBE Sound. The ad I saw was on page 128 of the March 2009 Musicians Friend catalog.

Comments

BobRogers Sun, 03/01/2009 - 19:42

The basic theory is that the speed of sound is not frequency dependent. Reality is almost certainly more complex (science always is). But I don't believe any difference in sound speed isn't going to be noticed in any room that I'm ever going to own. Heck, sound guys don't compensate for sound frequency dependent sound speed in stadiums. (At least none that I know.)

So count me a big time skeptic (even without having seen the ad you are talking about).

Space Sun, 03/01/2009 - 20:47

I have read others, mostly bass players, say that the Sonic Maximizer adds some clarity to their live rig setups.

An RFZ(reflection free zone) also comes to mind.

I think moving frequency around in a time/space domain is mostly what acoustical treatments are all about. Even then it isn't so much moving the frequency(s) as much as it is taking power away from certain parts of the spectrum as it interacts with the room to give the ear a better overall perception of what it should be hearing, less the early reflections, poor frequency distribution, etc.

Even if high frequency does get to the ear first, the question now wants the answer to, is this best for the piece of music I am working on? Is this an accurate recreation of what a listener could expect as it pertains to a specific piece of music? Is this an "always on" product or will it have minimum use in my recording? Will this guy ever stop asking me questions?

That's as close to an answer as I can come with my ninth grade education.

natural Sun, 03/01/2009 - 21:02

The BBE theory is basically this.
The hi Freq drivers of your typical speaker system are slightly inefficient and drag behind mid and lo freq drivers.
So BBE delays the lo and mids until it catches up to the hi end.
This recreates the sound as it would normally have been heard, and would be brighter without being necessarily louder.
Attempting to do the same with EQ ends up boosting the level as well as creating phase anomolies.
Used sparingly, it's quite effective.

RemyRAD Mon, 03/02/2009 - 00:22

Most equalizers are really speed control devices. Altering phase is not simply an in or out thing, a 0° or a 180° thing. Phase is the speed at which electrons travel at. A lot of this is marketing blah blah hype with a subjective interpretation. I've been told by certain experts that I cannot perceive things that I know I can perceive. It's all related to politics. Only you can decide what sounds good not what some experts tell you you should be hearing first or last. That's like telling you blue is better than red, Ford is better than Chevy, Digital is better than Analog. We all know that's not true. The bit about Digital that is. Get it? The bit...?

Digital microphones are not Digital
Ms. Remy Ann David

Boswell Mon, 03/02/2009 - 06:25

Any acoustic transmission medium (including air) is dispersive, resulting in the velocity of sound being a function of frequency as well as other factors such as temperature, pressure, humidity etc. So yes, it is true that different frequencies originating at the same point and time reach the ear at different times, but the magnitude of the effect under normal music listening conditions is so small that the ear does not interpret it as a difference of arrival times.

Don't confuse this arrival time variation over frequency with the brain's sensitivity to difference in arrival times between the two ears. This is the principal way that we localise sound in the angular dimension. Dispersive effects here are one of the reasons that a distant broad-spectrum sound such as a gunshot can be difficult to localise, even in the absence of echoes off buildings and other structures.

The ear is relatively insensitive to phase. For example, if you take a sustained note played on an instrument that is rich in harmonics (such as an oboe) and randomise the phases of the harmonics, the waveform becomes unrecognizable visually as the same sound, but the ear hears no difference. However, the ear is probably more sensitive to phase effects of shorter sounds (transients), and I believe the heavy phase distortion of the anti-aliasing filters contributed to the unpleasant "digital sound" of early CDs.

When it comes to multi-transducer loudspeaker systems, there is a need to account for the difference in time delay through the crossover networks for the high and low frequency units. This would show up as an unnatural phase change when sweeping a sinewave through the crossover frequency. The better-designed units employ Linkwitz-Riley crossovers (or similar), in which the resulting amplitude is approximately constant over the transition range, but the phase still needs attention. One of the ways the designer can attend to the phase is to incorporate a positional fore-and-aft displacement of the transducer units in his design of the cabinets, but it is not a simple calculation as he has also to take into account the phase of sound emanating from the bass reflex port.

To my mind, the marketing claims made for signal processors that play with phase in order to make significant audio improvements are dubious at best.

dvdhawk Mon, 03/02/2009 - 09:48

I can't believe Max hasn't been in here yet to offer his BBE up as a doorstop.

First off since we're in the DAW dept. I should say, I have ZERO experience with the plug-in version.

But as I said in another post regarding the hardware, we have an old blackface 802 (balanced ins and outs / maybe 20 years old or more) that makes a subtle, but still distinct, improvement to just about any signal that passes through it - whether it's being played through any of the studio monitors (KRK 7000, Event 6, JBL 4328) or a finely tuned 4-way live PA system.

I can't testify to the theory behind it, or their marketing claims. I don't care about that. All I want to know is, 'how does it sound?' I know that's all very subjective, but all I can tell you for sure is this. As far as recording goes, we do not use it during tracking. In the final-mix stage or mastering stage once we have a mix we're very happy with, we'll add the 802 and Avalon 747 before the master recorder and monitor the recorder outputs. The client, without fail, will prefer the sound with the BBE engaged. We almost never dial it in more than 2 or 3 clicks and can generally leave the Lo Contour alone. If you take it too far, it can sound over-hyped and artificial in the highs. But properly used, we find it opens up the sound with more clarity and detail especially noticeable in acoustic instruments and the air it gives a good reverb. If you're using it to EQ, you've got other problems, and hopefully you've got better tools for that job. *sidenote: another click or two and it does an awesome job breathing life back into some old archived cassette recordings that have lost their luster (and magnetic particles) over the years.

Maybe they don't make 'em like they used to, but the old one we have does the job very convincingly. The original reviews by people we respected in Mix magazine and Electronic Musician were gushing with praise at the time, and I'd have to say we've been very happy with the one we have. I never liked the synthesized 'aural exciter' sound of other devices, but this was something very different.

In the interest of remaining 'fair & balanced' or maybe 'unfair & unbalanced' I will say one of the churches I service has a red-faced unbalanced model that the previous installer shoved inline with their mains. Not surprisingly, it was doing more harm that good and we removed it from the signal chain - but the semi-pro model was never intended to be run inline after a 40 channel Soundcraft console.

It's about putting it in the right place at the right time......

So bottom line: We really like the one we have. But don't buy one anywhere that doesn't have a reasonable return policy. Not everyone can hear what it does, and of those that can hear it -- not everyone likes what it does.

Can you hear the difference between a studio-quality Mogami or Canare quad cable compared to a garden variety mic cable?

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