Centrad GF266 Manuel D'instructions page 21

After the signal has been "cut" into the two dimensions (time and amplitude), it can be depicted as a puzzle or mosaic.
If the elementary pieces are sufficiently small compared to the distance of the observer, this will see only the main
picture.
In the sphere of sound, a trained human ear (music lover or musician) is capable of distinguishing a purely analogue
signal from a signal which has been sampled at less than 100 000 levels. This explains while the audio CD, with its
coding over 16 bits (65 536 levels), heralded as a revolution with reference to the vinyl record because of its amazing
dynamic range, was not well received by many music lovers, and not only through nostalgia... New, high-performance
formats have appeared since: SACD, etc.
Digital information can be transferred in two ways:
- Parallel mode: over several wires (Fig. 2)
- Serial mode: over only one wire, one bit at a time (Fig. 3)
fig. 2
fig. 3
In its digital form, the signal can go through processing comparable to that available for its analogue counterpart
(filtering, gain control, etc.). For this purpose, dedicated circuits, called Digital Signal Processors (DSPs) are used.
The digital form also facilitates data storage on hard disks, CDs, DVDs, etc. ). The data stored in analogue form always
degrade with time. In contrast, digital information can be detected and fully regenerated as long as it is not below
the intelligibility threshold (reliable recognition of the 1 and 0 levels).
Digital-to-analogue conversion
This operation is generally performed in two steps. The first step consists of producing a voltage with an amplitude
corresponding to the quantized values as they arrive, i.e. at the sampling rate ; this signal still consists of discrete
steps ("staircase shape"). The second step consists of "smoothing" this signal using an appropriate low-pass filter
to restore a continuously variable signal.
Introduction to digital signal synthesis
Basically, direct digital synthesis (DDS) is not very far from the principle of operation of an audio CD. In the CD, the
values corresponding to the audio signal are frozen on the medium and therefore must be read at a determined speed
to feed the analogue-to-digital converter, as for a direct transfer, to reconstruct the original message.
In this function generator, the input signal is always periodic, i.e. it repeats exactly with time.
The principle of direct digital synthesis therefore consists of storing all samples corresponding to a signal period in
a read only memory (ROM) and reading them in a loop.
Detailed theory of DDS: (see functional diagram No. 1)
The purpose of this system is to provide a given periodic signal over a wide frequency range. In our example, the
signal will be a sine wave.
All amplitude values over one period are stored sequentially into a special ROM called a Loop Up Table (LUT). To
generate the sine wave, all that is needed is to continuously step through this ROM at a clock frequency (FMCLK).
There is still to convert the ROM output using a digital-to-analogue converter (DAC).
n
The ROM is stepped through using a 2 bit accumulator which is incremented every DDS clock cycle (MCLK) by a
number Delta (∆), which will therefore be proportional to the output sine wave frequency
(Fout), such that : Fout = ∆ x FMCLK/2
n
Example with a delta of 269: if FMCLK = 50MHz, ∆ = 269 and n = 228, then Fout = 50.105Hz
C l o c k
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A d d e r
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R e g i s t e r
R O M
P h a s e A c c u m u l a t o r
P h a s e R e g i s t e r
F u n c t i o n a l d i a g r a m n ° 1 : D i r e c t D i g i t a l S y n t h e s i s
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