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The neptuneEQ™ is the result of more than 10 years of product development. It is a sophisticated product, made possible only by the most recent DSP technology. The neptuneEQ™ corrects for acoustical problems created by the room, furniture and speakers in a 7.1 channel or less home theater system, and has both RCA and XLR connectors.

The neptuneEQ™ performs Equalization by the use of graphic EQ techniques. It utilizes thirty 1/3 octave bands for each of the 7 full range channels (just like a professional movie theater), and twenty bands for the subwoofer channel; all sampled at 96 kHz, 24 bits. We use four top of the line stereo codecs (the part that converts the audio to digital and back) from Cirrus logic®, instead of a single, less expensive surround sound codec. The neptuneEQ™ is expressly designed for discerning high end home theater use, where no compromises are made in the design and application.

Features:

  • Fully automatic room equalization within 4 minutes
  • Balanced and Unbalanced inputs and outputs
  • System comes complete including calibrated microphone. No other test equipment is needed
  • Digitally created, 100% repeatable pseudo random signal for optimum room measurement
  • Easy to use; no special education or certification needed
  • Separate analog and digital power supplies for maximum isolation
  • Front panel vacuum fluorescent graphic display
  • ISO standard thirty 1/3 octave bands per each full range channel / twenty bands for subwoofer channel
  • Measures listening area in 5 places
  • Automatically adjusts delays and levels
  • Automatically sets subwoofer crossover including phase
  • 24 bit 96 KHz digital bitstream
  • Robust steel chassis and aluminum front panel
  • Removable IEC power cord
  • 120 / 240 Volt operation
In a nutshell, the neptuneEQ™ will output a test signal, then listen for the response using an external microphone (supplied.) Then, it will calculate the best fit for several mic positions and adjust the graphic EQ filters, Delays, Crossover, and more.

In some detail, measuring the room requires the use of test signals. In general there are three basic test signals that can be used. These are: sine wave, impulse, or noise

[img]http://www.neptuneaudio.net/v/vspfiles/images/img_technology2.gif[/img]

Sine wave:
A sine wave measurement is simply a sequence of pure tones which are stepped over the frequency response of the system (typically 20 Hz to 20 kHz) at a constant strength. A microphone measures the signal at the sitting position and the processor determines the signal strength of each band. This is a very simple approach but has the problem of reflections causing cancellations and reinforcements at the specific frequencies which greatly reduce the accuracy of the measurements. Further, the time it takes to output and then listen to each frequency is prohibitive.


[img]http://www.neptuneaudio.net/v/vspfiles/images/img_technology3.gif[/img]

Impulse:
The impulse (or ping) method is done by generating a pulse with a very fast rise time and a very short duration. The shape of this pulse is altered by the audio system and room acoustics. It is measured and recorded, and then it is analyzed to determine the characteristics of the audio system. This is theoretically straightforward but has the practical difficulty that when using signals low enough not to cause audio system distortion, obtaining enough energy over the entire frequency range to get accurate measurements is difficult.


[img]http://www.neptuneaudio.net/v/vspfiles/images/img_technology.gif[/img]

Noise:
Of the three signals, noise is the closest in character to movie sound. Pseudo random noise (or p/n)is generated digitally and is seeded to cause it to be exactly the same every time; therefore it is a knownsource, unlike true random noise. It allows the best measurement because noise can provide a large enough signal for quality measurements without distorting the audio system and it is much less susceptible to incidental reflections than a sine wave, since there are no discrete tones. This is the signal we have chosen to implement in the neptuneEQ™.


Automatic EQ
The neptuneEQ™ comes standard with a calibrated microphone and stand. The mic is plugged into the front panel and is placed in the first listening position. The processor in the neptuneEQ™ generates the test signals, sends them through all of the channels in turn, automatically adjusts the proper level for the audio system and makes the first round of measurements. Then the mic is then moved and the measurements are repeated four more times for a total of five listening positions. The EQ’s processor then automatically determines the necessary corrections (EQ, balance, delay and crossover) for each channel, and applies them. No external computer is needed. This entire process takes approximately 4 minutes. The user can easily do this job and needs no special training or equipment, thus avoiding an expensive service call. The corrections are held in the EQ’s memory permanently until a remeasurement is needed such as a change in furniture, decor or audio component.

Finally, your system sounds right!
The processor in the neptuneEQ™ automatically calculates the corrections needed to remove the room's effects. It does so by determining how to condition the signals so that they arrive at the listening position sounding like the moviemaker intended. No additional input is needed to make these corrections. In addition to this, there are user preference presets and adjustments to tailor the sound to your preferences; thus giving you the ultimate control of your sound system.

Specifications

Audio Inputs:
  • 8 Analog input channels accommodating: Center, Left and right front, Left and Right Surround, Left and Right rear and Sub channels
  • XLR and RCA connectors all channels
  • Input impedance: 10 k Ohms unbalanced 20k balanced
  • Input sensitivity: 1.0 Volt rms for full output
  • Microphone input connector 3.5 mm jack
Audio Outputs:
  • 8 Analog output channels accommodating Center, Left and Right main, Left and Right surround, Left and Right rear and Sub channel
  • XLR and RCA connectors all channels
  • Input impedance: 10 k Ohms unbalanced 20k balanced
  • Output impedance: 560 Ohms unbalanced 1120 Ohms balanced
  • Output level max (all channels) Selectable 1.0, 1.4, 2.0, 2.8 Volts rms
Performance:
  • Frequency response: 10 Hz to 20 kHz +/- 0.3 dB
  • Distortion: Less than .0028%
  • Dynamic range: 105 dB
  • Signal to Noise ratio: 109 dB "A" weighted
  • Crosstalk -100.5 dB at 1kHz any channel to any other channel
General:
  • Analog to digital conversion: 24 bit 96 kHz sampling frequency all channels
  • Digital to analog conversion: 24 bit 96 kHz sampling frequency all channels
  • Bands per channel: 30 for each of seven full range channels, 20 for one sub channel
  • Digital Processing: 32 bit 50 mflops / sec.
  • Power requirements: 120V/240V 50/60 Hz +/- 20 % universal line cord
  • Operating temperature range: 0 C to +45 C (32 F to 113F)
  • Relative Humidity: 91% max
  • H. W. D. 3.5"x17"x9"
  • Weight 11 Lbs, 4.5 Oz
 

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I was reading in one of the other forums about a device that SVS is coming out with that will do an analysis and correction for subwoofers. One of their main points is that it will also correct the frequency response in the time domain.

I never thought I needed anything corrected in the time domain, but now that they brought it to my attention, maybe I do.

Does the Neptune perform corrections in the time domain? I don't think this is the same as speaker distances or polarity.
 

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Discussion Starter #3
Hi Jim, thanks for your inquiry!

The short answer is yes, we are doing the correction in the time domain. Whether that means much to you is uncertain. You are right, it doesn't refer to loudspeaker distance or phase, but rather how the signal is processed internally. Time alignment™ which is a trademarked term (not by us, but by Ed Long) is what would be corrected for with speaker distance compensation.

Like everything in electronics (or the world for that matter), everything has advantages and disadvantages. Doing correction in the time domain is more or less just different than doing it in the frequency domain. It has the advantage of speed and accuracy at the expense of processing power, and thus, cost (at least in our implementation). Since we made this product for the high end market, we decided that tradeoff was worth it. Time will tell if that's so. :ponder:

What we do as well as the EQ (amongst other things) is the above mentioned time alignment™. That is not trivial, and it cannot be properly done by a simple measurement with a tape measure. Due to the nature of subwoofers (the large voice coils, the long excursion requirements, the internal equalization etc), they have HUGE amounts of group delay. Often the internal delay in a subwoofer is greater than the delay caused by it's distance. Our neptuneEQ™ measures the actual delay, and therefore provides excellent time alignment™.

I have been totally unaware of SVS's correction device. I looked on their website and couldn't find anything about it. If you can remember where you saw this in the forum (I looked there too, with no success), I'd love to read more.

Ken
 

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The SVS EQ was displayed at CES recently. It is being finalized for production currently, I think.

The SVS PDF on the subject can be found here.
 

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Discussion Starter #7
Hello Bill, thank you for your interest.

Kal pretty much hit it on the head (thanks Kal). Not having tried the SVS unit, but only read about it, I have no doubt it's a very nice capable unit. It is, however, subwoofer only. All I can do is extol the benefits of our product:

The neptuneEQ was designed as a high end accessory from the ground up. (please keep in mind that I'm not saying the SVS unit wasn't). Since it is for multiple channels (7.1) AND full range, the components used are many and pricey.

In order to convert all those channels with the quality needed, multiple top quality audiophile grade stereo codecs were employed (the best we could find). Also, the amount of processing power needed to perform the functions of the neptuneEQ is staggering. Our 32 bit floating point processor is performing 400 MMACS (400 Million Multiply Accumulate Cycles per Second; think about that).

It also has split power supplies (digital and analog) which share only the power cord, balanced XLR connections with gold contacts (and gold RCAs too), and includes a second separate processor to do the job of the laptop. Even the chassis which is made of 18ga steel and machined aluminum is very costly.

We are aware that 4 grand is a lot of money. I would never refer to this product as "affordable", because that is a relative term, but it was created for those who can afford it and want the best sound money can buy.

Ken
 

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Where do you find that the real costs come into play? These units are, what, about $4k? I suspect the volume is low (compared to what sony can expect from a given AVR), so the development costs and overhead must be divided among fewer units: but speaking of parts and manufacturing: where is the biggest expense?
 

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Discussion Starter #9
Hi Jerry,

Thank you for asking! Actually there are many small costs which add up, but the breakdown is that the biggest cost items are the front panel, the printed circuit boards, the chassis and (amazingly) the front panel display. Of course the codecs (there are 4 of them) and DSPs (there are 2 of them) are (were?) the highest end units available at the time of design.

The panel is CNC machined out of a custom aluminum extrusion here in San Diego, then hand finished (masking, sandblasting and painting) at another San Diego business as well. The PCBs, (though as an OEM amplifier manufacturer we had used overseas and across the border manufacturers in the past), are made in Colorado by the best PCB manufacturer we have been able to find. They turn out consistently great (albeit expensive) boards, so we stick with them. Of course they are multi-layer boards.

The chassis is made by a sheet metal fabricator in Iowa, because again they turn out a consistently great and well finished product. The display costs more than I think it should, but we wanted a certain appearance and resolution, and this one fit the bill.

And, of course, our production IS small, I won't try to deny that. We don't amortize the development cost amongst the product at an alarming rate, because it's all about the future, but could Sony make something like this less expensively? Absolutely! they would make everything "in house" for one thing, and likely wouldn't even dream of using a CNC machined front panel; but fortunately for us, there's plenty of room in the industry for "niche" manufacturers (especially if they build in the USA? :sn:)

Ken
 

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Hi there,
I'm new in the room-correction world but I know that It's very very important!

This neptune is very interestinf for a newbie like me.

So my main questions are:

1) This neptune can do a better job compared to my actual audyssey inside my onkyo 905?
2) Why?
3) If i buy a Neptune I've to turn off all the audyssey procedure?

Thank you so much,

francesco
 

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Discussion Starter #11
Hi Francesco,

Sorry for my late reply; we have been reorganizing and our system has been completely upgraded.

To answer your questions, we believe the neptuneEQ system offers the most natural sounding room correction available. What is offered within AV receivers is scaled down compared to stand alone units necessarily, due to the amount of processing power needed to fully EQ 7 full range channels plus sub.

If you were to use the neptuneEQ in your system, you would need to bypass the audyssey built into your system, or if they operate in tandem, one could do the course correction, and the other do the fine tuning. Of course which does which depends upon the order.

Ken
 

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If you were to use the neptuneEQ in your system, you would need to bypass the audyssey built into your system, or if they operate in tandem, one could do the course correction, and the other do the fine tuning. Of course which does which depends upon the order.
Can you explain how this might be? Since any built-in eq would have to precede any external EQ in terms of signal flow, the latter cannot know anything about the former.

In other words, if you try to run the built-in one first, it will attempt to correct as much as possible. If you run the external one, before or after, it will also attempt to correct as much as possible since it cannot "hear" the corrections of the built-in. Using one or the other works but using both can only superimpose excessive EQ on the system.

As I see it, if one wants to use both, the only option is to use the external EQ first to do the heavy lifting and, then, run the built-in one for tweaking (as its output will work through the external one).

OTOH, considering how competent the NeptuneEQ is (as is Audyssey), I cannot see why one would want to use both. In addition, NeptuneEQ is "tweakable."
 

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Is anyone able to get in touch with Neptune? Their site has disappeared. I've found their equipment excellent, but feel that Audyssey may have marketed their product far more effectively, leaving Neptune out in the cold.
 
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