Datasheet

ADC08831, ADC08832
SNAS015C –SEPTEMBER 1999–REVISED MARCH 2013 www.ti.com
The zero error of the A/D converter relates to the location of the first riser of the transfer function and can be
measured by grounding the V
IN
(−) input and applying a small magnitude positive voltage to the V
IN
(+) input.
Zero error is the difference between the actual DC input voltage which is necessary to just cause an output
digital code transition from 0000 0000 to 0000 0001 and the ideal ½ LSB value (½ LSB = 9.8mV for V
REF
=
5.000V
DC
).
Full Scale
The full-scale adjustment can be made by applying a differential input voltage which is 1½ LSB down from the
desired analog full-scale voltage range and then adjusting the magnitude of the V
REF
input (or V
CC
for the
ADC08832) for a digital output code which is just changing from 1111 1110 to 1111 1111.
Adjusting for an Arbitrary Analog Input
Voltage Range
If the analog zero voltage of the A/D is shifted away from ground (for example, to accommodate an analog input
signal which does not go to ground), this new zero reference should be properly adjusted first. A V
IN
(+) voltage
which equals this desired zero reference plus ½ LSB (where the LSB is calculated for the desired analog span,
using 1 LSB = analog span/256) is applied to selected “+” input and the zero reference voltage at the
corresponding “−” input should then be adjusted to just obtain the 00
HEX
to 01
HEX
code transition.
The full-scale adjustment should be made [with the proper V
IN
(−) voltage applied] by forcing a voltage to the V
IN
(+) input which is given by:
where
• V
MAX
= the high end of the analog input range
• V
MIN
= the low end (the offset zero) of the analog range
• (Both are ground referenced.) (2)
The V
REF
IN (or V
CC
) voltage is then adjusted to provide a code change from FE
HEX
to FF
HEX
. This completes the
adjustment procedure.
DYNAMIC PERFORMANCE
Dynamic performance specifications are often useful in applications requiring waveform sampling and digitization.
Typically, a memory buffer is used to capture a stream of consecutive digital outputs for post processing.
Capturing a number of samples that is a power of 2 (ie, 1024, 2048, 4096) allows the Fast Fourier Transform
(FFT) to be used to digitally analyze the frequency components of the signal. Depending on the application,
further digital filtering, windowing, or processing can be applied.
Sampling Rate
The Sampling Rate, sometimes referred to as the Throughput Rate, is the time between repetitive samples by an
Analog-to-Digital Converter. The sampling rate includes the conversion time, as well as other factors such a MUX
setup time, acquisition time, and interfacing time delays. Typically, the sampling rate is specified in the number of
samples taken per second, at the maximum Analog-to-Digital Converter clock frequency.
Signals with frequencies exceeding the Nyquist frequency (1/2 the sampling rate), will be aliased into frequencies
below the Nyquist frequency. To prevent signal degradation, sample at twice (or more) than the input signal
and/or use of a low pass (anti-aliasing) filter on the front-end. Sampling at a much higher rate than the input
signal will reduce the requirements of the anti-aliasing filter.
Some applications require under-sampling the input signal. In this case, one expects the fundamental to be
aliased into the frequency range below the Nyquist frequency. In order to be assured the frequency response
accurately represents a harmonic of the fundamental, a band-pass filter should be used over the input range of
interest.
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