Ground SoundDK DCN28 Digital Crossover Filter, Pre-Amplifier and Room Correction in one unit.
Dansk firma med en masse DIY.
DSP-Digital crossover filter.
DCN28 Digital Crossover Filter, Pre-Amplifier and Room Correction in one unit.
Ground Sound DCN23
DCN23 is a high performance digital crossover filter with equalization and delay. It features 2 balanced analogue inputs and 3 single-ended analogue outputs
(of which the third has an inverted output for easy bridging at e.g. low frequencies). The usage of high performance Burr-Brown converters gives DCN23 crystal clear sound and very low noise floor.
The sample rate is 96 kHz to extend bandwidth and have low order analogue low pass output filtering. The resolution of converters is 24 bit and the internal signal processor resolution is 76 bit.
The signal processor holds 3 channels of 16 biquads each for filtering and equalization. DCN23 has an optical isolated USB interface to avoid hum and noise from the PC.
The circuit board is a high quality 4 layer type which prevents noise and hum. The layout features a lot of decoupling - at supply entry and very close to all active components.
DCN23 is the heart of an active speaker management system offering a superb sound, easy installation and easy use in daily life.
The XOverWizard program is a graphical tool to manipulate with inputs, outputs, gain, crossover frequencies, crossover slopes, equalization and delay.
The XOverWizard has a very significant feature: The ability to import a text file containing measured driver data of frequency, sound pressure level and phase.
With these data the XOverWizard are able to display frequency response and more, while design are in progress. Compared to this method the “old” trial and error method seems obsolete.
The new XOverWizard II will also be able to progarm DCN23. XOverWizard II has additional features - especially the advanced version with the integrated measurement system.
Ground Sound DCN23-BOX
Ground Sound DCN24 er med 4 udgange.
Every listening space introduces errors that can be in the magnitude of dozen decibels or even more.
These errors are most pronounced at the bass and lower midrange regions, where the classic acoustic treatment loses its effectiveness.
Anti-Mode 2.0 Dual Core is DSPeaker's new generation loudspeaker optimization system capable of perfecting the acoustic performance of any full-range stereo audio system.
It is based on our new Anti-Mode 2.0 algorithm which refines the popular, award-winning Anti-Mode algorithm even further.
The first version of the Owner's Manual is now available: AntiMode20DualCoreEng.pdf .
New, improved Anti-Mode 2.0 automatic room correction algorithm
Can be used as a locally clocked high quality jitter-free DAC
Supports USB audio as well as S/PDIF input/output via optical toslink
Dual balanced ADCs to deliver the highest level of audio performance that meets the expectations of most demanding audiophiles
Dual per channel balanced DACs with jitter free operation (local crystal), oversampling to 6144 kHz, DSP buffering and filtering.
Can be used as a high quality pre-amp with remote volume control
Extensive sound customization options, including quick house curve design tool, linear-phase full spectrum tilt, several parametric EQs etc
Supports several sound profiles (e.g. independent ”sweet spots” and/or equalizer settings), quickly switchable with the remote controller
Friendly graphical user interface, very easy to set-up and operate
Ability to measure and display room responses on a built-in 262k-color TFT display
”Pure digital” mode using S/PDIF input & output: signal is processed in the digital domain all the way (S/PDIF output resolution is independent from input)
Digital signals are re-aligned for jitter-free operation
Versatile connectivity: insert Anti-Mode 2.0 Dual Core between your existing pre-amp and power amplifier or powered speakers or utilize the processor loop or tape loop of your integrated amplifier (or use it as a standalone DAC / preamp!)
A typical listening room resonates in low frequencies. Because of this, even the best subwoofer may not sound good enough.
Much of the frequency content get masked by the resonances, introducing unwanted characteristics (such as "slowness" and "boominess") to the sound.
This problem is most pronounced when subwoofer is placed close to walls or corners of the room.
The Golden Ear awarded Anti-Mode™ technology eliminates the resonances of the speaker and the room by equalizing
both amplitude and time domain responses using very accurate digital signal processing filter structures
and anti-phasing technology. This way, the listener can hear frequencies down to the cut-off frequency of the subwoofer.
Transient response is also drastically improved in the process, making bass sound faster and more controlled.
Anti-Mode™ 8033 is very simple to use. The calibration process is completely automatic:
it generates frequency sweeps to the desired calibration point (or multiple points) and
measures the combined transfer function of the subwoofer-room system.
Anti-Mode™ 8033 uses regular RCA connectors, which can be found from vast majority of home theater equipment.
Input to the device is taken from the line level subwoofer output of the (pre-)amplifier. Output is connected to the line level input of the subwoofer.
på dansk :
DSPeaker Anti-Mode 8033 Automatic Subwoofer Equalizer
ved udligning af både amplitude og tids domæne med et præcist digitalt filter i signalbehandlingen og anti-udfasnings teknologi.
På denne måde kan lytteren høre frekvenser ned til cut-off frekvensen af subwooferen.
Transient respons er også drastisk forbedret i processen, hvilket gør subwooferen hurtigere og mere kontrollerede.
Normalt er det ikke optimalt at placere en subwoofer i et hjørne, men med Anti-Mode korrektion kan der ske underværker!
Ved kalibrering placerer man den medfølgende mikrofon ved lytte positionen, trykker på en knap og venter et par minutter.
Man har mulighed for flerpunkts kalibrering.
Pris i Norge kr. 1.999.-
Pris i Sverige kr.2.890 kr
Pris i Tyskland €. 275.-
These measurements are displaying the data measured in a typical 4m x 5m listening room.
The subwoofer is placed in the front corner during first three cases, and the listening position to which the 8033 has been calibrated is varied.
The first two cases (1B and 2B) have been corrected by 1-point measurement.
The third case (3B) is calibrated by sophisticated multi-point Gradient method built in 8033 Anti-Mode for more global results.
In near-wall situations the gradient method is recommended if first calibration does not yield satisfying results.
Listener in the opposite corner
In the first case, the position of calibration is in the opposite corner from the subwoofer.
The figure 1A presents the situation before calibration.
There is the strongest modal peak at 53 Hz, and slowly decaying low frequency resonance at 34 Hz.
Also the frequencies are emphasized toward 0Hz because of the narrow radiation angle caused by corner placement.
After-calibration measurement is in the figure 1B.
The strongest peak at 53Hz has been countered quite succesfully, and the slow decay mode of 34 Hz has been improved for the first 200ms.
Audible difference is obvious, although the frequencies above 80Hz remain slightly below the average level compared to range betwnn 16 and 80Hz.
In this case, the listener is in the midde of the room where low frequencies are usually cancelled by destructive reflections.
The energy is shifted toward heavy modal concentration within midbass range at 63 Hz.
Listening to the music, this standing wave will blur almost all transients,
as it is narrow and has louder contribution still after 200ms compared to initial level of other frequencies.
The corrected plot in figure 2B shows that this devastating room mode has been counter-modelled efficiently
and the decay is now actually faster all the way to 350ms, which is perhaps already better than neccessary for the global point of view.
Perhaps one of the most typical subwoofer-listener alignments is illustrated in the figures 3A and 3B.
In the initial situation, the whole bass range is infested with several high-Q asymmetric room-modes.
Proximity to the back wall is treacherous for automatic calibration algorithms, as the response will change
rapidly for even the slightest deviations from one point. Therefore this situation is advised to be calibrated using the clever
"Gradient method" which is also implemented in 8033 Anti-Mode.
The figure 3B shows that although the respone decay is still not completely uniform,
a great improvement compared to the original has been achieved.
The result is audibly now better for the whole region of the backwall except right in the corners.
The gradient point for second phase was chosen 20cm toward the side wall, 10 cm toward the back wall and 10cm toward the floor.
Anti-Mode 8033 has three special "Lift" modes that allow the user to apply a digital low-frequency boosting filter together
with a protecting steep digital subsonic filter. The maximum boost of Lift is around +7dB (peak vs. 80Hz).
In the following measurements, also the noise-shaping Bessel lowpass is active at output, so the level of boost
virtually looks ~3dB higher because of the output-LPF attenuation toward higher frequencies.
Here are the different frequency responses of the three different Lifting states of 8033.
(Click to enlarge)
(Click to enlarge)
DSP Linear Phase Crossovers
Linear Phase Crossovers
Bessel, Butterworth and Linkwitz-Riley crossovers are provided, selectable up to 48 dB per octave.
The crossover interface also allows you to quickly add low shelf, high shelf, and parametric filters to each output.
These filters use the latest in digital signal processing technology, maintaining their symmetrical shape at high frequencies.
You can also build your own custom crossover functions by enabling or disabling high pass and low pass filters on any output channel.
Linear Phase Crossovers
Linear phase crossovers are another feature separating Lake Processing technology from ordinary processors.
Linear phase crossovers can match traditional crossover slopes, such as 24 dB per octave and 48 dB per octave, when desired.
But they are also capable of transition slopes exceeding 180 dB per octave.
These slopes offer dramatic benefits when applied to various types of loudspeaker arrays. Off-axis lobing and cancellation between loudspeakers are dramatically reduced.
Greater control over slopes means that different speaker cabinet types can be more easily mixed and matched.
Also, a loudspeaker’s impulse response is significantly improved, providing a time-coherent wavefront.
Additionally, improvements of up to 3 dB or more in acoustic output power may be expected in some frequency ranges.
Multi-way speaker processing
Surround Sound processors
Custom/ DIY Audio processors
Mobile Audio processing
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