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Digital Sampler/Delay (Part 1)

Tantek's most ambitious project to date.


The Modular Effects Rack takes a significant step forward this month with it's first digital module; the Digital Sampler/Delay. Paul Williams presents this high spec device which incorporates full sample editing, CV pitch control, 'velocity sensitive' dynamics and a bandwidth/delay product up to 15kHz and 1.4 seconds.


It is not without good reason that the Modular Effects Rack has been lacking a digital module for so long. Having evolved it over a long period since the sampler has 'come of age' gives it the advantage of incorporating all the facilities the modern musician looks for today, together with an uncompromised specification. An affordable price has only recently been made possible as new silicon devices have become available.

The fact that both samplers and DDLs are based around fairly large memories into which digitised audio is stored and manipulated makes it almost illogical to produce one of these devices without incorporating the facilities for the other. This module is thus a dual personality digital unit which can function as either a delay line or a sampler at the touch of a button. In the delay mode, full bandwidth delays of up to 1.4 seconds can be produced, and this figure is stretchable (at the expense of bandwidth) up to 8 seconds.

In the sample mode, a sample is recorded into the memory where it's 'frozen'. The sample can then be played back either at the command of an external trigger pulse, by pushing the 'Trig' switch, or continuously by setting the 'Loop' switch. The pitch of the sample can be varied using the 'Trim' control, or played via the CV input by an analogue synthesiser. The start and finish of the sample can be edited using the 'Start' and 'Length' controls so that any presample silence or after-sample noise can be eliminated, since the editing controls also effect the recording start and finish points, several samples can be stored for playing separately, or in a chain.

A 'Decay' control helps to tidy up any samples that end abruptly, as well as allowing looped samples to be envelope shaped. Complex overlays can be built up using the 'Overdub' switch for 'bouncing' samples. Of great advantage when playing percussive samples is the velocity sensitivity, which when used with an electronic drum trigger pad, makes the amplitude of the regurgitated sound proportional to the violence with which the pad is assaulted!

The sound quality? Well, a companding Digital-to-Analogue Converter (DAC) with pre- and de-emphasis, and a built-in limiter ensure a healthy dynamic range which, together with the 15kHz bandwidth add up to a sound quality significantly better than most cassette players.

To allow for the inevitable future progress in silicon memory, the plug-in memory card occupies an expansion socket, which would allow either a larger memory card to be plugged in, or connection to an expansion module, depending on user demand and future advances in technology.

For those handy with a soldering iron, the kit is an excellent way of stretching your tight budget a little further, but it's not recommended for the complete novice. If you have no experience of constructing electronic kits, then this is probably not the one to start with. However, this need not stop you from owning a module though, since they are also readily available in an assembled form too.

Figure 1


Architecture



Figure 1 shows the overall structure of the device. Noteworthy are the tunable low pass filters which at the input provide anti-aliasing with the clock frequency, and reconstitution and clock removal at the output. By using switched capacitor filters, the cutoff frequency of both is maintained at a set multiple of the main clock. The bandwidth of the system is thus optimised for whatever clock frequency is in use. The aforementioned clock is actually a Voltage Controlled Oscillator (VCO) which, apart from manual Tune and Trim controls, can be controlled by a voltage applied at the CV input. The filter clock frequency is maintained at sixteen times the VCO rate by means of a Phase Locked Loop (PLL) frequency multiplier.

Before the input signal can be converted into it's digital form, it must be broken up into 'samples' of instantaneous voltage (not to be confused with the more general term 'sampling' of musical sounds). Each sample is, for a short time, held constant in a Sample and Hold (S & H) amplifier. The Analogue-to-Digital (AD) converter sub-system then goes about it's task in producing a digital word corresponding to each sample.

The all-important memory has sufficient capacity for over 65,000 samples. The operational location, or address is determined by the address counter which is incremented by the VCO. Each sample is thus allocated a different address. In the delay mode, the existing contents of each new address are read and converted back to analogue by the Digital to Analogue (DA) converter. A further S & H is used to fill in the gaps between the read samples, after which the signal is passed via the reconstitution filter to the output. The Regen control allows some of the delayed signal to be re-cycled for echo repeat effects. In the sample mode, writing new data and subsequently reading is performed as two separate tasks. Once a memory full of data has been written, it will remain in the memory indefinitely while the power remains on. The trigger input controls the onset of sample playing, unless the loop mode is selected where playing is continuous. The Voltage Controlled Amplifier (VCA) allows the amplitude of the sample to be controlled by a trigger source applied to the audio input.

The editing controls both produce an analogue voltage corresponding to the editing point chosen which is given digital words during a special conversion cycle. The same converter is used for editing and audio by the use of an analogue multiplexer (MUX). The editing data is used to preset two counters. The address counter is preset with the start point address, and the down counter with the length data. Each run through the sample therefore, initiates at the start address, and terminates when the number set in the down counter has been exceeded. In the delay mode, the start address is always zero, leaving the length control to set the amount of memory being used, and thus the delay time.

An inbuilt limiter permanently safeguards any signal being recorded from becoming clipped in an overload situation. This applies both to the delay mode, and to the sample record mode. An indicator associated with the limiter thus indicates when the limiter is in operation, not when you have just ruined your irretrievable sample with clipping distortion.

Figure 2
(Click image for higher resolution version)


VCO



Of paramount importance to the musicality of the device is the VCO circuit, which is shown in Figure 2. IC1a sums the various control and offset voltages applied to it's input. The voltages from the CV input socket and Tune preset are only allowed to effect the VCO when a sample is being played and a jack inserted in the CV socket, as determined by the CMOS switch IC10a. The summed voltage at pin 1 of IC1a is passed to the logarithmic amplifier formed around IC1b and IC2. This uses the well known logarithmic relationship between transistor Vbe and Ic in the devices Q2 & 3 in IC2. The summed voltage drives the emitters of Q2 & 3 negatively, with a temperature compensating offset produced by Q1. The current thus produced is drawn via D4 & 5 and, due to current mirroring in Q4 & 5, is also pushed into D2 & 3.

The oscillator itself is based around the fast comparator IC3 whose output at pin 7 'steers' current of the appropriate polarity from the diode ring into the integrating capacitor, C3. When the trigger threshold is reached, the comparator toggles whereupon R13 shifts the threshold level and the diode ring reverses the current polarity, charging C3 towards the other threshold and so on. Since IC3 is the only active device concerned with oscillation, operation extends up to about 1.5MHz, although normal CV control would be within the range 10kHz to 200kHz, centering around a nominal figure of 46.875kHz. IC10b merely buffers the VCO output so that external noise cannot effect stability.

IC4 is a complete monolithic Phase Locked Loop (PLL) system whose inbuilt VCO is controlled so as to keep the input on pin 3 at the same frequency as that on pin 4, the main VCO. Since IC5 divides the PLL VCO by 16, the output on IC4 pin 4 is 16 times the main VCO frequency. This new clock frequency is used to control the turnover frequency of the switched capacitor filters, of which more next month.

Figure 3
(Click image for higher resolution version)


Converter



Figure 3 shows what is probably the most critical part of the system: the converter section. Since Digital-to-Analogue Converters are relatively expensive devices, only one is used here: IC11, which is multiplexed for audio D-A, and along with the Successive Approximation Register (SAR) IC12, for A-D conversion of audio, Start pot and Length pot voltages. The DAC used has non-linear converting characteristics which gives the effect of an inbuilt compander to enhance the dynamic range.

D-A conversion is a simple matter where data put on the data buss D0-D7 by the memory enters the DAC, causing it's current decode outputs on pins 16 and 17 to produce a corresponding voltage at the output of the current/voltage converter IC9. The voltage for each successive digital word is sampled onto, and held by C8 by one element of the analogue switch IC6. This signal is by now recognisable audio which is then passed to the audio section for reconstitution filtering.

A-D conversion is a rather more complicated affair. Firstly, the analogue switch, IC6 selects whether the conversion is for audio, Start pot or Length pot, gating the appropriate signal into the S & H amplifier, IC7. Once conversion is initiated, IC7 freezes the voltage presented to it at that instant, and keeps it constant throughout conversion. Under the control of IC14 & 15 and a crystal derived high frequency clock, the SAR IC12 builds up a digital word bit by bit. The data produced by the SAR is put on the data bus by the tri-state buffer IC13 and simultaneously collected by the DAC IC11. At each step, the comparator IC8 signals to the SAR whether the next bit needs to be high or low to get the analogue equivalent of the assembled word closer to the sampled voltage. When the digital word is fully assembled, the SAR notifies the control logic with the EOC signal so that the data is written into memory.

The Digital Sampler/Delay module is available from Tantek, (Contact Details). The price inclusive of VAT and postage (within the UK) is £219.95 in kit form, or £299.95 ready assembled and tested. Further information on the modular rack system can be obtained from the above address, or by 'phoning (Contact Details).

To be continued next month.


Series - "Digital Sampler/Delay"

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All parts in this series:

Part 1 (Viewing) | Part 2 | Part 3


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Analogue Equipment Design

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Equipment Guide


Home & Studio Recording - Copyright: Music Maker Publications (UK), Future Publishing.

 

Home & Studio Recording - Feb 1986

Donated & scanned by: Mike Gorman

Topic:

Electronics / Build


Series:

Digital Sampler/Delay

Part 1 (Viewing) | Part 2 | Part 3


Feature by Paul Williams

Previous article in this issue:

> Analogue Equipment Design

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> Equipment Guide


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