Dbx Silver Series 386

Hugh Robjohns tries out a mic preamp from Dbx with a built-in A-D converter that is supposed to be impossible to clip...

There will always be demand for high-quality mic preamps, but the competition is so intense in this sector of the marketplace that manufacturers have to find something to give their product an edge over their rivals. The solution Dbx have adopted is a hybrid of valve and solid-state audio circuitry combined with their unique Type IV A-D conversion system.

The 386 is a two-channel device, with identical controls for each section spaced either side of the central 24-bit 96kHz-compatible Type IV converter facilities in a silver-fronted 1U metal case. A set of rackmount bolts are included with the unit, as is an excellent manual.

The Dbx 386 is actually derived from its older and bigger sibling, the Dbx 586, which costs around twice as much. The 586 is a 2U unit incorporating a three-band equaliser section in each of its two channels; the Type IV conversion stage (see box, right) is an optional extra on the 586. Both systems share a common topology, though, where the front end of the signal path is entirely solid-state with separate electronically balanced input buffers for the microphone and line inputs. Following the input selector, 'Drive' amplifier and high-pass filter, the valve stage is based around a familiar 12AU7 double triode.

DBX 386 £499 pros Flexible. Good ergonomics. Excellent, robust A-D conversion. Controllable valve sound. cons None. summary Simple but well-designed two-channel hybrid preamp with mic, line and instrument inputs and the excellent Type IV A-D converter offering full 24/96 compatibility.

The rear panel is well organised (see page 122), with all the analogue connections gathered towards the right-hand side (when viewed from the rear) and the digital interfacing and mains power facilities to the left. Each channel is equipped with an XLR for a microphone input along with a TRS quarter-inch jack for a line input, both of which are electronically balanced. Another TRS jack socket provides the unbalanced insert send/receive (post- valve stage). The output section is, again, furnished with both XLR and TRS sockets, wired in parallel.

The digital interfacing consists of both AES-EBU and S/PDIF output sockets (XLR and phono, respectively) along with a word clock input and output, both on BNC connectors. Mains power enters through the ubiquitous IEC socket, with a miniature rocker switch alongside.

The front panel is a model of good design, with a very clear and simple control surface -- every button is lit when active, often with different colours to indicate specific conditions. Starting at the left, a quarter-inch jack socket provides an unbalanced instrument input offering a 470k(omega) impedance instead of the 20k(omega) of the rear-panel line input. A rectangular push button selects between mic and line inputs, illuminating green for the latter (the instrument input overrides the rear-panel line input). The first rotary control is labelled 'Drive' and adjusts the input gain stage ahead of the valve buffer. The available gain ranges between +30 and +60dB for microphone inputs and +/-15dB for line and instrument inputs. The character of sound changes, as you would expect, by altering the drive level, yet overall unity gain through the unit can be assured by appropriate adjustment of the output level controls, described below. A red peak LED above the Drive control illuminates 3dB before clipping.

Four more rectangular buttons lie under a set of vertical ventilation slots between the Drive knob and two more rotary controls. These buttons select phantom power, a 20dB input pad, a polarity reversal, and a low-cut filter. The first three affect only the microphone input, while the last introduces a second-order (12dB/octave) high-pass filter from 75Hz for mic, line and instrument inputs.

Five more illuminated rectangular buttons are positioned between the two input control sections at the centre of the front panel, and these configure the A-D converter. The first selects the type of dithering applied when outputting 16 or 20-bit word lengths. Repeated presses of the button cycle through the three options, the illumination indicating the selection: green for TPDF, red for SNR² and off when dither is deactivated. The adjacent button selects a noise-shaping curve: green for a modest curve, red for a more aggressively shaped dither function, and no illumination when the noise-shaping is off.

The sample rate is determined by the third button in the row. Once again, multiple button presses cycle through the options: no illumination for 44.1kHz, green on 48, red for 88.2 and yellow for 96. High sample rates are transmitted in the 'double-fast' format via the single S/PDIF or AES-EBU connector. If an external clocking signal is input via the rear panel word clock BNC, the sample-rate button automatically displays the colour corresponding to the appropriate detected incoming sample rate. Should the external clock disappear the button will flash by way of warning -- pressing the button then reinstates the relevant internal sample rate clock. The word clock input is terminated in 75(omega) by default, but a jumper on the main circuit board can be moved to unterminate the clock input, necessary when daisy-chaining several converters, for example.

The output word length is established by the next button: red for 24 bits, green for 20 and no illumination for 16 bits; dithering is naturally recommended for outputs lower than 24 bits. The final button sets the digital output format between S/PDIF (red) and AES-EBU (green), which is provided to both physical connectors simultaneously (via isolating buffers).

The external word clock input and output are handled through a custom Dbx 'Pulsar' chip which contains high-performance Voltage Controlled Crystal Oscillators (VCXO) designed for very low-jitter performance (although no specification details are given). However, the handbook suggests using the Dbx 386 as a clock master in appropriate situations and I found it functioned perfectly well in that situation alongside my usual Apogee PSX100 converter.

The 386 is a very fine tool. The specs appear to be nothing special -- the microphone EIN is only -120dBu with a 150(omega) source, for example -- but the machine has a quality and character which makes such technical mumbo-jumbo all but meaningless. It might not be the quietest or cleanest preamp available, but that is not the point, is it?

This is a no-frills hybrid preamp with only mic, line and instrument inputs, very basic processing facilities (essentially just a high-pass filter) and a built-in A-D converter. At first sight, its main strength might appear to be the inclusion of a valve stage for that infamous analogue warmth. Well, the expected sonic attributes are certainly present and can be managed nicely through the Drive control for a broad range of sounds: the analogue outputs convey all the 'warmth' and character you could want.

However, I would suggest that the 386 has an even stronger card up its sleeve in the form of the Type IV converter. Though the notion of a non-linear converter might raise a few eyebrows, the system works extremely well with real-world signals. Amongst other trials, the review model was employed in a live situation to interface a Neumann KM84 vocal mic and the piezo output of an acoustic guitar with a Yamaha O3D mixer, using the Dbx's digital output. The results were excellent, far better than anything that could have been achieved with the Yamaha's internal converters -- fundamentally because the Type IV system handles transients and unexpected peak levels very well, without the obvious clipping distortion that would beset a more traditional A-D converter system.

Transient overloads in A-Ds are a real problem in unpredictable situations such as live sound or 'as live' recording. Normally, the only solutions are either to leave lots of headroom at the input to the A-D, which risks imposing a poor noise floor on the resulting recording, or to place analogue compression ahead of the converter and risk howlrounds as the gain structure changes. With a digital desk such as the O3D, the latter is not something that can be done without external dynamics units in any case.

The advantage of the Type IV system is that it remains totally linear below -4dBFS and only soft-limits signal peaks above that in a relatively benign and even flattering manner. The 'valve warmth' inherent in the hybrid design of the 386 also played its own characteristic part in creating a very appealing, usable but, above all, stable sound and all who heard the results were impressed.

I believe the Dbx 386 represents good value for money -- after all, the Aphex 107 is a similarly priced, decent-sounding two-channel tube preamp, but doesn't have a built-in A-D converter, let alone one offering 24/96 operation with a selection of sample rates, bit-resolutions, dithering and noise-shaping options. The Dbx 386 is definitely one worth putting near the top of your list when shopping for a new preamp.

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