Dynamic Noise Filter
Build this noise reduction project
It is widely accepted that a double ended encode-decode process is the most effective route to tape noise reduction and this belief is wholly endorsed by the author. However, even if your recorder is endowed with such a system, you may not find it a complete solution, especially when you have bounced tracks around a few times. Moreover, these systems invariably have fixed threshold characteristics which are optimised for particular noise and signal levels. The other problem you may be faced with is what to do with the noisy recordings you made before you obtained your noise reduced equipment? Your tape recorder noise reduction system does nothing either for the noise generated by equipment further down the line, such as mixers and audio processors like flangers and echo units, etc.
Clearly what is needed is an effective single ended noise reducer which can be used at final mixdown, and is capable of being adjusted to cater for a wide range of signal and noise levels. The HSR Dynamic Noise Filter (DNF) was designed to satisfy just this requirement. It will significantly reduce any level of tape noise or equipment hiss, and yet when correctly adjusted will have no noticeable effect on the music programme, which still retains all of its original brightness and clarity.
The DNF is by no means a new idea; indeed DNF's are in common use in professional recording studios to supplement normal noise reduction and to enhance the signal-to-noise ratio's of mixers and signal processing equipment.
Although the foremost design consideration has been high quality, a reasonably low cost has also been maintained.
The DNF is effectively a self adjusting low-pass filter, whose cutoff frequency rises when high frequency signals are present above a predetermined threshold level. The operation ofthe DNF relies on the well known masking property of the human ear, where high frequency sounds will disguise noise signals of lower level but similar frequency, such as tape noise. When these high frequencies are not present to mask the noise, the cutoff frequency of the filter lowers to attenuate the noise, but since no other high frequency signals are present, the programme is not effected. The filter cutoff frequency will only rise when there is sufficient high frequency signal content to mask the noise.
The HSR DNF contains two such filters to provide two channels for stereo operation. The filters have a common control circuit to prevent stereo balance shifting, but the signal paths are quite independent. While first order 6dB/octave filters might at first be thought to be adequate for this application, the design presented here in fact uses second order 12dB/octave filters. This steeper cutoff slope enables the filters to separate the noise from the signal much more effectively. Lesser designs would either lose the clarity of the upper signal frequencies, or not attenuate the noise sufficiently or both, depending on how they are adjusted. The adjustment limits of the HSR DNF allow a wide range of signal levels and signal-to-noise ratio's to be handled with optimum performance.
Figure 1 shows the dynamic response curves for the DNF, from which it can be seen that input levels which exceed the threshold level by 20dB or more will be unaffected, but lower levels become progressively more attenuated at higher frequencies. Note that the signal levels shown are not absolute, but referenced to the threshold level.
An LED indication of the filtering operation is provided so that a check can be kept on how the programme is being affected. The DNF was designed for use with the E&MM Twinpak power supply which along with the Comp-lim and Sweep EQ, make a comprehensive signal processing set-up.
As shown in Figure 2, each audio channel comprises two cascaded current controlled low-pass filters formed by IC2 and IC3, dual transconductance amplifiers. The simple cascade connection, although not realising a perhaps more desirable Butterworth response, was chosen so that DC shift related 'thumps' are eliminated by the cancelling effect of the second inverting stage.
For control purposes, the two input signals are mixed together by R1 and R2, which present the composite signal via the high pass filter, C3 and R3 to IC1a. This high pass filter allows only high frequency signals to open the filters, while the low pass characteristic caused by C4 prevents any ultrasonic signals from opening the filters. The gain of IC1a is adjustable by means of RV1 to allow a wide range of threshold levels to be set. C5 and C6 further promote the high pass characteristic. IC1b and c form a precision full wave rectifier which eliminates the forward voltage of the diodes, so no temperature dependent terms are involved in determining the threshold level. IC1c also acts as a peak detector, which charges C7 to the peak level detected. R10 prevents any very short transient peaks from opening the filters. In the absence of a signal, C7 will discharge via R10 and R9, resulting in a decay time constant of about 100mS.
IC1d and TR1 convert the voltage on C7 into a filter control current which is injected into the transconductance amplifiers via R16 and R17. When the control current is low and the filters are closed, then LED D5 will illuminate, showing that the programme is being filtered.
C14 - C17 take care of any supply-borne noise or transients that might otherwise find their way onto the supply rails.
As with the other projects in this 'audio processing' series, construction is simplified by containing all the components on a single PCB, which should be assembled according to the component overlay. First insert and solder the seven veropins and the wire link, then insert and solder progressively larger components; resistors, diodes (but not the LED), transistor, capacitors and finally 1C sockets. Don't fit the ICs themselves at this stage, though. Be careful with the orientation of the diodes and electrolytic capacitors.
Before soldering the pot in position, trim the shaft to length and make sure it is seated firmly down onto the PCB. Next prepare the panels as shown in Figure 4 and mount all the sockets. Before proceeding any further, check that the assembly of the PCB is complete and correct, and that the soldered joints all look healthy and free from solder splashes or whiskers between adjacent joints. This is best done with an eyeglass or at least, a magnifying glass. If all is well load the ICs into their appropriate sockets, taking care with orientation.
Fix the PCB assembly to the front panel by means of the pot nut, making sure that the PCB is parallel with the panel. To fit the LED, bend its leads down at 90° about 7mm from the body and pass the leads through the PCB, checking polarity. Now push the LED clip bezel into the front of the panel and slip the ring over the LED. Offer the LED into the clip and push the ring over to secure. The LED can now be soldered in place.
Construction is completed by dropping the panel assemblies into the case base, and wiring up as shown in Figure 3. The PCB can be held firmly in place by 'nipping' the back edges with 2 of the self tapping screws supplied with the case.
Although designed for use with the E&MM Twinpak featured in the September 1982 issue, the DNF can be powered by any regulated ±15V twin rail DC supply. The DC power is fed in via one of the rear DIN sockets, the other socket being intended to extend the DC power to another audio processing unit.
The unit will accept a wide range of input levels, but levels much lower than -12dBm will lead to a degraded signal-to-noise ratio, which rather defeats the object! The DNF is best placed right at the end of the playback chain on final mixdown, after all mixing and processing. This way, not only is the tape noise reduced, but so is the mixing and processing equipment noise with it. It also means that no matter how many tracks you mix down, a single DNF will suffice, so long as the mixdown is only for two channel stereo.
Your old noisy recordings can be cleaned up too, either by playing them back via the DNF to your amplifier, or for a more permanent solution, re-record via the DNF onto another machine (with noise reduction, of course!). Disc surface noise reduction can be affected most conveniently by feeding the tape output of your amplifier to the DNF, from where the signal returns to the tape monitor input.
The correct adjustment of the DNF threshold control will depend very much on the level and nature of the programme and on the noise level, but it is in any case vital to the performance of the unit. Adjustment is best done during periods of complete silence, or during quiet passage. Start with the control fully clockwise, and gradually adjust it anticlockwise until the noise just disappears. The LED will be glowing by this time. See if the control can be turned back clockwise slightly without re-introducing the noise.
Again depending on the nature of the noise and programme, it may be necessary to adjust the control a little more anticlockwise if during moderately quiet passages, the noise starts to 'peep' through again. When the LED extinguishes completely, then the filters are open fully, and the programme is being passed unaffected. You will see the LED flashing on and off continuously as the music plays, indicating that the DNF is cutting the noise, but its effect on the music will usually be imperceptible.
A complete kit of parts for the DNF, including PCB and case, is available from (Contact Details), priced at £29.95 inc VAT and P&P. Please order as DNF kit.
PARTS COST GUIDE £29.95.
Feature by Paul Williams
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