Many argue that correcting for room characteristics which affect the frequency response of home audio systems is not only vital – but that it may be the single best upgrade you can make to improve the sound quality of your system.

Although it is clear that physical means using absorbent wall panels or inserts is the best way to achieve room correction, the vast majority of us do not have dedicated listening rooms for our home audio, and so physical correction is difficult or impossible. As a result, there has been an increasing number of devices marketed for room correction. They range from the relatively simple (eg., DSPeaker Anti-Mode Dual Core, which analyzes the room response and corrects for it via a built-in but configurable parametric equalizer) to the highly complex (eg., Trinnov ST2, which offers sophisticated multi-point analysis and an equally complex set of optimization algorithms based on finite impulse response (FIR) filters).

What these devices have in common is that they act as analogue-to-digital converters (ADCs) for audio signals coming from analogue sources like turntables, before applying the room correction algorithms in the digital domain.

What they also have in common is that the better devices are (to me!) quite expensive and probably unaffordable!

So, I have been trying for some time to find a cheaper alternative for digital signal processing from my vinyl source that would allow simple room correction or at least equalization in the digital domain, without losing what for me are essential characteristics of the analogue source sound.

The increasing sophistication and falling cost of professional devices used as routers, mixers, and digital signal processors (DSP) in recording studios made this a natural place to start looking. Many home audio enthusiasts seem to have tried pro devices like the Behringer DEQ2496 for DSP – but there seems to be a consensus that sound quality is not very good, particularly when converting from analogue sources.

Helpful feedback from forum members at HFW led me to consider gain structure as a potential issue. There is lots of helpful information about this here http://www.diyaudio.com/forums/diyau...structure.html. Put simply, pro devices expect much higher input voltages than home audio devices usually deliver, so the ‘noise floor’ during ADC is sub-optimal, after which the pro devices output higher-level signals following DAC conversion that must be attenuated considerably before feeding into home audio amplification stages. So, the gain structure is poor, resulting in higher noise and potentially lower resolution.

This thread is to share my early experience with an inexpensive solution that seems to work in my system (kit listed below) in case this is of interest to other forum members. I was able to find all the additional items needed (ADC/router device, software, connectors, cables etc) quite cheaply. The key item is a Focusrite Pro14 router/mixer device. This device has multiple analogue and digital inputs/outputs, including phantom-powered microphone connectors, a high quality 24 bit, 96 kHz ADC which can also operate at lower resolutions, and is sturdy and well-designed for busy pro musicians. (I already have a PC suitable for the DSP step.)

I have tried the following setup (approximate gain/attenuation at each step is noted):

Icon phono preamp (~1.3V output, no analogue attenuation) -> Focusrite Pro14 (line level input +16 dBu = 0 dBFS, 24/96 ADC, digital output) -> Audiolab 8200CDQ (digital input, attenuation for playback exclusively in the digital domain, DAC via four multibit array DACs per channel operating at 32 bit depth) -> power amps

Key points are:

• Gain structure: Seems to work very well without any noticeable fidelity or noise floor problems. The phono preamp output is fed without analogue attenuation to the Focusrite. The Focusrite line input to the ADC allows two different gain settings (+16 dBu ~4.5V(rms) = 0 dBFS, or -6 dBu ~0.35V(rms) = 0 dBFS) to prevent clipping. (The phono preamp signal works well with the low gain setting.)

Any further signal attenuation is exclusively in the digital domain in the Audiolab unit. Because the Audiolab unit uses the ESS Sabre 9018 32-bit chip (which contains 8 discrete 32-bit DACs), digital attenuation can be carried out within this enormous headroom with – theoretically at least – no discernible effect on sound quality. This is consistent with what I heard.


• DSP and room correction: The Focusrite unit is bundled with a pretty sophisticated Digital Audio Workstation (DAW) software suite (Ableton Live 8 Lite) that is compatible with a wide range of AU (Mac) or VST (PC or Mac) plugins for DSP. Also, the Focusrite routing software allows the connection of an outboard DSP unit (eg., miniDSP or the like) to process the digitized signal (more on this below).

I have not yet played around much with DSP settings or software options, but find I can already make meaningful full spectrum changes (for eg. using the multi-band parametric equalizers in the DAW) to the output that should eventually allow room correction.

The availability of phantom-powered mic sockets in the Focusrite unit means that a calibrated dynamic condenser mic can be used to make accurate measurements of the room frequency response to pink noise. There is (quite sophisticated) freeware around (like Room Equalization Wizard http://www.hometheatershack.com/roomeq/) that can analyze the room response and create FIR filters to correct it.

As I said, the Focusrite unit allows the connection of a hardware device for DSP or room correction in the digital domain between the ADC and digital output. So – it should be possible to insert a small programmable DSP unit (like the miniDSP http://www.minidsp.com/applications/auto-eq-with-rew) to load the FIR correction filters generated with REW or similar software.


I have tried A/B listening with and without the ADC/DAW routing, and using different ADC/DAW settings, and to my ears find no alteration in noise or sound quality. It still sounds like a vinyl sourced analogue system!

For those who may be interested – this setup will also allow vinyl-> digital recording at up to 24/96 quality using the Focusrite unit’s ADC to feed to a Mac or PC via FireWire. The setup should also allow active crossover configuration via the DAW assuming you have the right downstream kit.

I hope that this thread will provide useful information for anyone else who may be contemplating a simple solution for DSP and room correction in an analogue-sourced system using widely available and inexpensive pro audio devices.


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Benz Glider SL - Audio Origami 12" PU7 arm with Cardas wiring - Scheu Analog Diamond TT
RFC MC1 SUT
Icon Audio PS1.2 Mk1 Signature version valve phonostage / 1x Mullard ECC81 / 2x Mazda ECC83
Audiolab CDQ8200
Bel Canto EvoII Gen 2 Tripath-based Class D power amplifiers running in monobloc
Audio Physic Tempo III speakers