BOX CLEVER RME ADI96 Pro 24-bit/96KHz A-D Converter Published in SOS April 2000 Reviews : A-D/D-A Converter The ADI96 Pro is a 96kHz-capable A-D converter featuring both mic and line inputs, as well as a whole host of onboard DSP functions. Hugh Robjohns tries it out.
The new machine is a fairly conventional-looking 1U rackmount box measuring a modest 483 x 44 x 205 mm, and weighing just 2kg. Like most A-D converters, it has few operational controls, but appearances are deceptive, for there is far more under the skin than is obvious at first glance. Aside from the expected 24-bit/96kHz stereo A-D converter technology, the unit boasts electronically balanced line and microphone inputs, together with a wealth of DSP-based signal-processing functions including an expander, a de-esser, phase reversal, high-pass filtering, automatic level control, time delay, M DSP Technology To The Fore Most state-of-the-art A-D converters these days incorporate a DSP because it is the most pragmatic means of decimating high-resolution digital signals from 96kHz to the more mundane sample rates of today's commercial releases. However, the boffins at RME have extended the functionality of the DSP in the ADI96 to condition and process the converted signal completely in the digital domain. The ADI96 employs an 80MHz Motorola DSP 56303 at its heart, working in conjunction with the latest 24/96 A-D converter chip from Crystal Semiconductors. The DSP system is capable of maintaining a full 48-bit internal resolution for all computations even at a 96kHz sample rate. The ADI96 is also unusual in that it samples and processes the analogue input at 96kHz regardless of the required output rate. RME have labelled this approach 'CDS' or 'Constant Double Speed' and they claim it has sonic benefits over sampling at the required output base rate. This seems entirely plausible to me, as the analogue anti-alias filtering is far easy to design with fewer audible artefacts at this elevated rate and, assuming the down-sampling and decimation algorithms have been designed carefully, the number-crunching should maintain the best possible output quality at the desired rate. The other side to this CDS approach is that the initial conversion and subsequent digital signal processing is always performed with 24-bit resolution, the DSP re-calculating the output for 8-, 16-, 20- or 24-bit word lengths, as required, after any other signal processing has been completed. Word-length reduction is provided with three dithering options: off, or with triangular dithering added at an amplitude of 0.5 or 1 LSB (least significant bit). With the appropriate mic or line selection, input signals ranging between -50 and +22dBu can be accommodated, and the input circuitry automatically compensates for balanced and unbalanced sources. Apparently, the line input remains completely balanced all the way up to the A-D converter. Back To Front Microphone inputs are catered for on the rear panel, with a pair of 3-pin XLR connectors (phantom power is switchable via the menu screens), while line-level inputs are provided with both XLRs and TRS quarter-inch jack sockets. All analogue inputs are electronically balanced and the ADI's digital output is simultaneously available at three different ports (although all carry the same data). A software menu selects the data format of the output between consumer (with or without copy protection) or professional. There is an XLR connector for AES-EBU equipment, while S/PDIF interfaces are available on both a co-axial (phono) socket and a TOSlink optical port. In 96kHz sampling mode, the ADI96's outputs produce a 'double-fast' data stream, which is to say that each digital interface operates at twice its normal speed to accommodate the necessary data flow. This places higher technical demands on the transmitting and receiving devices, as well as the cable (or optical fibre) used to carry the signal. The alternative high sample-rate interface format, not supported by the ADI96, is called 'double-wide' and splits the data stream across a pair of standard digital output connectors, thereby maintaining the original interface specifications in terms of bandwidth and data rates. Both of these solutions A single BNC socket at the centre of the rear panel can be configured either to transmit or receive word clock information, allowing the ADI96 to be used as a master or slave device respectively. The machine has a low-jitter clock design, quoted as better than 3nS. The unit will generate or slave to any of 32, 44.1, 48, 64, 88.2 and 96kHz sample rates. Rounding off the rear panel facilities, a 9-pin D-sub connector provides an RS232 interface for remote control purposes. Mains power is catered for with the ubiquitous IEC mains connector and integral fuse holder. The front panel is equally simple. A pair of uncalibrated input level controls occupy the leftmost portion of the panel, with a clear 2 x 20-character backlit LCD and seven associated push buttons next. A stereo bar-graph PPM fills out the remaining space to the right and is augmented with a separate correlation bar-graph meter. The liquid crystal display is easy to read with plain text messages and fairly intuitive mnemonics. The menu structure is navigated via a diamond cluster of buttons, to the right of the display, with the two middle buttons acting as left/right cursors while the top and bottom buttons provide increment and decrement functions. An Enter button positioned just off to the right confirms selections, and the two remaining buttons provide a complete DSP bypass, and switch the level and correlation bar-graphs pre (input) or post (output) the DSP. The last switch on the right-hand side is the mains power switch. What's On the Menu? After its initial boot-up test the machine recalls the last used settings, although 16 user memories are available for specific setups. Pressing the Enter button recalls an overview display of the status of the DSP processing, with each function represented by a three-letter mnemonic with a pair of dott The Input module selects mic or line inputs, phantom power and phase reverse for each channel independently. An MS decoder can also be introduced to convert an MS stereo signal to conventional left-right. The next module (Low Cut) provides high-pass filtering with adjustable turnover frequencies between 2 and 250Hz. The slope is unfortunately fixed at only 12dB/octave -- 18 and 24dB/octave options would have made this facility more useful. The Delay function provides independent channel delays between 0 and 170mS for general time-alignment or special effects. It also provides the dynamic processors with a 'Look Ahead' facility so that they can react in advance of abrupt dynamic changes in the programme material. The next module is an Expander function operating over thresholds ranging from -30 to -110dBFS, with ratios between 1.2 and 5 (the latter providing a gating facility). Release times can be adjusted between 0.1 and 25.5 seconds, with a choice of Fast (1mS) or Slow (25mS) attack times. Stereo linking is available and if the Delay algorithm is set over 0.1mS, the expander automatically uses the 'Look Ahead' facility. The De-esser function uses a dynamic band-pass filter which automatically adapts its threshold parameter based on a comparison of the overall level with the level of the filtered signal -- the idea being to maintain a consistent action regardless of the signal amplitude. The filter centre frequency is adjustable (from 700Hz to 20kHz), as is the Q (0.2 to 10) and Ratio or attenuation (0 to -50dB). There is also a 'Key Listen' mode to help fine-tune the parameters. Next is a 'Non Linear Compressor' (a soft-limiter) intended to control only the highest level peaks in a similar manner to tape saturation. The only adjustment is of the required increase in loudness from 1 to 5dB. The handbook recommends that the process is not used on delicate solo instruments, but harmonically complex sources; com Moving on through the processing chain, the next device is an 'Auto Level Control' which pulls the level of the audio signal up to a defined headroom point (between -1 and -12dBFS). The attack time is determined by the available headroom -- low settings use a 3mS attack time to ensure every peak is caught, while greater headroom margins (over 7dB) enable a slower 6mS attack time to be used. If the Delay mode is active and greater than 4mS the ALC automatically operates in the 'Look Ahead' mode and thus avoids peak clipping altogether. The maximum gain applied during levelling is adjustable between 0 and 24dB as is the rise time (the time needed for a 20dB gain change), from 0.1 to 25.5 seconds. Long settings providing a transparent smoothing of level whereas shorter times allow the algorithm to operate more like a conventional compressor. A stereo link facility is available to avoid image shifts. At the end of the signal path the Output function determines the digital output format, selecting sample frequency, word length, dithering options, and word clock output on/off. It also allows the channel status to be selected between Professional, Consumer with copyright, or Consumer without copyright. The Setup page provides control over various global parameters such as the peak hold time of the bar-graph metering (0.1 up to 9.9 seconds) and the meter fall-back speed. The latter ranges from an incredibly leisurely 1dB/S to a positively manic 100dB/S. The LCD contrast can also be adjusted and all current processing functions saved to, or recalled from, any of the 16 memory presets. A further function here ('Boot Other Option') allows alternative DSP core programs to be loaded. The d The final menu page, Information, is the default display on power-up, providing the DSP software name and version number. The window is also displayed automatically 100 seconds after the last button press and shows the selected audio input, sample rate and word length. Putting It To Use The ADI96 is very straightforward to set up and use, with everything operating in a very logical manner. The menu functions are slightly fiddly to navigate -- dedicated hardware controls always win as far a The remote control software also provides enhanced versions of the input and output signal level bar-graph displays, covering an 80dB range in 1dB steps. Gain-reduction meters improve the expander and de-esser sections, and there is a 'gain-enhancement' meter for the auto level control function. The Delay section provides status information about the Look Ahead mode for the expander and ALC too. I made some initial comparisons between the RME A-D converter (with all of the signal processing options disabled) and my reference Apogee unit, the PSX100. The asking price for the RME is a little above that of the Apogee Rosetta A-D (effectively the A-D half of the PSX100) and at the base rates of 44.1 and 48kHz, the RME compared favourably with the Apogee, although I think it lacked some of its clarity and precision. Against this, it must be pointed out that the Rosetta can only cope with 44.1 and 48kHz sample rates. There is a Rosetta 96kHz model, but this costs several hundred pounds more than the RME. High-frequency detail and stereo imaging were both to a high standard, lending weight to the low-jitter claims for the RME, and the sound was always natural and open -- a benefit of the CDS technology I presume. Switching between 16-, 20- and 24-bit resolutions gave progressively greater realism and depth to analogue source material, as one would expect, and seemed to match the resolution of the PSX100 closely. Unfortunately, I was unable to make direct use of the 96kHz sample rate because none of my equipment would accept the double-fast output format (although I am awaiting an update for the Apogee to accommodate this increasingly common mode). However, given its strong performance at the base sample rates, and assuming it was partnered with suitable equipment, I have no doubts the machine would serve well at the higher sampling rates too. As a converter alone, then, the ADI96 appears to more than justify its asking price -- but that is without considering the added value of the signal processing and microphone inputs. I found all the DSP processes to work very well, allowing a wide range of effects to be applied from the extremely subtle to the positively heavy-handed. The unit can be used as a mastering processor for analogue sources very effectively indeed, or as a specialist recording preamp for solo instruments. All the processes work well and provide an excellent range of tools (although steeper high-pass filtering would have been useful on occasion). The remote control software (see the box below) also adds a powerful dimension which, whilst not suiting every application, will be an attractive feature to many users. The ability to reconfigure the unit's core DSP programs for alternative functions such as the spectrum analyser will also extend this product's appeal and provides even greater value for money. The bottom line is that the RME ADI96 is a good quality A-D converter which also includes a wealth of signal-processing tools and enormous flexibility for an attractive price. Published in SOS April 2000 | Saturday 11th October 2008 October 2008
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