Sie sind auf Seite 1von 4

How Analog-to-Digital Converter (ADC) Works

Author: Gabriel Torres Type: Tutorials Page: 9 of 10 Last Updated: April 21, 2006

Real-time pricing for Lone Oak 1-883-47713-1 E. G. Nadeau David J. Thompson Cooperation Works How People Are Using Cooperative Action to Rebuild Communities and Revitalize the Economy Lone Oak Press Business/Economics Politics Current Events Peopl Amazon: $16.95

Sigma-Delta ADC
The sigma-delta ADC also called delta-sigma, 1-bit ADC or oversampling ADC uses a different approach. We can divide it into two major blocks: analog modulator, which takes the analog signal and converts it into a stream of bits, and digital filter, which converts the serial stream from the modulator into an usable digital number.

click to enlarge

Figure 13: Block diagram of a sigma-delta ADC.


The analog modulator is somewhat similar to a dual-slope ADC, however using a 1-bit DAC as a feedback loop. The basic sigma-delta modulator design can be found in Figure 14.

click to enlarge

Figure 14: Sigma-delta ADC analog modulator basic design.


The analog signal will make the first op amp, which is a summing integrator, to create a sawtooth waveform proportional to the analog signal voltage. This sawtooth waveform found on the integrator output is then compared with zero volts by the second op amp, which is a comparator. It can be considered a 1-bit ADC, since its output will have only two states, high or low, depending whether the integrator output is positive or negative. The comparator output is stored on the D-type flipflop, which is a one-bit static memory. This flip-flop is clocked at a very

high frequency. Then the flip-flop output is used to feedback the circuit through a one-bit DAC. This one-bit DAC will basically convert the 0 or 1 stored at the flip-flop into a positive or negative reference voltage to be added to the input of the summing integrator. So the summing integrator will sum the next sample with the result of the previous sample (a positive or a negative voltage), aiming to maintain zero at the integrator output. The result is that at the flip-flop output we will have a series of zeros and ones that correspond the sampled data: the bitstream average level represents the analog input signal average voltage. Since the clock rate used at the flip-flop is very high, data is sampled many times over, a technique known as oversampling. The higher the clock, the higher the precision of the sigma-delta ADC. In real-world applications, the flip-flop clock rate will be 64 times higher than the sampling rate (i.e., a oversampling rate of 64). So, for a 44,100 Hz sampling rate, the flip-flop clock rate will be of 2,822,400 Hz. Another way to increase the ADC precision and thus reducing noise is to implement a second summing integrator between the original integrator and the comparator. This implementation would be called second order sigma-delta ADC. Commercial hi-end ADCs for audio application can use a fifth order analog modulator. The problem with sigma-delta ADCs would be that with using just one bit the SNR would be too low. If you remember our formula SNR = 6.02 x n + 1.76 dB, sigma-delta ADCs would have a SNR of only 7.78 dB. However, due to oversampling, the quantization noise is thrown to the high frequencies of the spectrum, and not spread all over the spectrum as it occurs with other designs. This effect is called shaped noise. With all the noise concentrated in a specific portion of the spectrum on a frequency range above the sampled data , is quite easy to construct a filter to remove it, thus improving SNR. This kind of filter is known as low pass filter (which allows only frequencies below a certain trigger frequency to pass through it) and this filtering is done on the digital filtering stage. The use of this kind of filter improves the SNR in 9 dB for every doubling of the clock rate used (9 dB/octave). Without it SNR improves 3 dB/octave only. So, a first order sigma-delta ADC with an oversampling rate of 64 (2^6) would have a SNR of 54 dB (9 dB x 6). If we use the SNR formula again, we will come to the conclusion that the

effective number of bits (ENOB) can be calculated using: ENOB = (SNR 1.76 dB) / 6.02 This means that a first order sigma-delta ADC with an oversampling rate of 64 has the same performance of a multi-bit ADC with a resolution of 8 bits but using only one bit! Increasing the number of summing integrators also improves SNR. The SNR improvement can be calculated by the formula 6 x L + 3 dB, where L is the order number. So a second order sigma-delta ADC provides a SNR of 15 dB/octave, a third order provides a SNR of 21 dB/octave, a fourth order a SNR of 27 dB/octave and a fifth order a SNR of 33 dB/octave. An easier way to calculate the effective number of bits it through the formula: ENOB = ((L + 0.5) x n) + 1 Where L is the order number of the sigma-delta ADC and n is the oversampling factor given by 2^n; for a 64x oversampling, n would be 6. So a second order sigma-delta ADC with a 64x oversampling achieves an effective number of bits of 16 (a 98 dB SNR), thus achieving the same performance of a conventional 16-bit ADC with a simpler (and cheaper) construction. The next step is to know what digital number the bitstream found at the analog modulator represents, what is done by the digital filtering block, which does also the low pass filtering already explained. Since the bitstream is oversampled (i.e., its clock is higher than the sampling rate), this stage also reduces its clock rate to the sampling rate. This process is known as decimation.

click to enlarge

Figure 15: Block diagram of the sigma-delta digital filter.

Problem

When the Physiological signals are digitized it is common practice to firstly remove the high Electrode Offset Potential from the signals by means of high-pass filtering, followed by an amplifier with a gain 1000 for large signals such as EMG and ECG to 5000 for EEG signals. The following medium resolution Analog to Digital Converter now only has to comply with the maximum signal range of the specific physiological signal. If each electrode signal has to be sampled individually, the analog high-pass filters exhibit too low frequency accuracy to allow for a high common mode rejection. This is caused by the inaccuracy of the input capacitor used in the high pass filter. Also the anti-aliasing filter in the individual data path is a source of mismatch between channels leading to a low common mode rejection in subsequent stages of signal processing. Solution For best performance no analog filters are allowed in the individual signal path. Using the Reference Amplifier with a gain of 20 and to allow for some headroom in the signal, the ADC now needs a resolution of a factor of 200 to 1000 higher than the one used with high pass filters. To achieve this high resolution either a SigmaDelta ADC or an over-sampling ADC may be used. A rule of thumb is that the cut off frequency of the antialiasing filter should lay at least a factor of 100 above the main frequency in the common mode component, in Europe with a line frequency of 50 Hz about 5 kHz. In this case an over-sampling frequency of 20 to 25 kHz is sufficient. Digital decimation filtering is used to increase resolution of the digitized data. Starting with a 16 bits ADC the resolution may be increased by 5 bits at a resulting sub-sample frequency of 2 kHz. A pre-requisite is that the 16 bits data contains at least 1.5 bits random noise before decimation as shown in the figure.

Das könnte Ihnen auch gefallen