FYS3240 PC-based instrumentation and microcontrollers. Signal sampling. Spring 2017 Lecture #5

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FYS3240 PC-based instrumentation and microcontrollers Signal sampling Spring 2017 Lecture #5 Bekkeng, 30.01.2017

Content Aliasing Sampling Analog to Digital Conversion (ADC) Filtering Oversampling Triggering

Analog Signal Information Three types of information: Level Shape Frequency

Sampling Considerations An analog signal is continuous A sampled signal is a series of discrete samples acquired at a specified sampling rate The faster we sample the more our sampled signal will look like our actual signal If not sampled fast enough a problem known as aliasing will occur Actual Signal Sampled Signal

Aliasing Adequately Sampled Signal Signal Aliased Signal

Bandwidth of a filter The bandwidth B of a filter is defined to be between the -3 db points

Sampling & Nyquist s Theorem Nyquist s sampling theorem: The sample frequency should be at least twice the highest frequency contained in the signal Or, more correctly: The sample frequency f s should be at least twice the bandwidth Δf of your signal In mathematical terms: f s 2 *Δf, where Δf = f high f low 0 Δf f However, to accurately represent the shape of the signal, or to determine peak maximum and peak locations, a higher sampling rate is required ECG signal Typically a sample rate of 10 times the bandwidth of the signal is required. Illustration from wikipedia

Sampling Example Aliased Signal 100Hz Sine Wave Sampled at 100Hz Adequately Sampled for Frequency Only (Same # of cycles) 100Hz Sine Wave Sampled at 200Hz 100Hz Sine Wave Sampled at 1kHz Adequately Sampled for Frequency and Shape

Hardware Filtering Filtering To remove unwanted signals from the signal that you are trying to measure Analog anti-aliasing low-pass filtering before the A/D converter To remove all signal frequencies that are higher than the input bandwidth of the device. If the signals are not removed, they will erroneously appear as signals within the input bandwidth of the device (known as aliasing) Frequency Domain:

Idealized Filter Responses

Filter parameters A filter will affect the phase of a signal, as well as the amplitude!

Filtering example In post-processing (non-real time) a zero-phase digital filter can be used, by processing the input data in both the forward and reverse directions Example from MathWorks

Analog filters Passive filters: RC, LCR (often inductors L are avoided, but they are needed for high Q-factor) Bessel Active filters opamp + R and C Some common filter characteristics Butterworth Chebyshev Bessel (constant group delay in pass band) Elliptic

Sallen-Key - Active analog filter Structure WEBENCH LP HP

Switched-Capacitor Filter Can be suitable as an ADC anti-aliasing filter if you build your own electronics Be aware of possible clock noise (add RC-filters before and after) The corner frequency (cut-off) fc is programmable using an external clock Example: MAX7400 8th-order,lowpass, elliptic filter MAX7400 has a transition ratio (fs/fc) of 1.5 and a typical stop band rejection of 82dB

ADC architectures Multiplexed sampling Gives a time delay between channel sampling Simultaneous sampling One ADC, multiple Sample-and-Hold registers Multiple ADCs Important for phase measurements

ADC resolution The number of bits used to represent an analog signal determines the resolution of the ADC Larger resolution = more precise representation of your signal The resolution determine the smallest detectable change in the input signal, referred to as code width or LSB (least significant bit) Example: 10.00 8.75 7.50 6.25 Amplitude 5.00 (volts) 3.75 2.50 1.25 0 0 16-Bit Versus 3-Bit Resolution (5kHz Sine Wave) 111 110 101 100 011 010 001 000 50 16-bit resolution 3-bit resolution 100 150 200 Time (ms)

ADC accuracy Common ADC errors: Noise Linearity error Gain error Offset error Quantization (resolution error) Less than LSB/2

Digital signals: Bits, dynamic range, and SNR SNR = signal to noise ratio The number of bits used determines the maximum possible signal-to-noise ratio Using the entire ADC range (using an amplifier) increases the SNR The minimum possible noise level is the error caused by the quantization of the signal, referred to as quantization noise.

ADC oversampling Oversampling means to sample faster than the Nyquist rate, which is given by fs = 2 *Δf The SNR of an ideal N-bit ADC (due to quantization effects) is: SNR(dB) = 6.02*N + 1.76

ADC oversampling II If the sampling rate is increased, we get the following SNR: SNR(dB) = 6.02*N + 1.76 + 10* log 10 (OSR) OSR = f s /f nyquist Oversampling makes it possible to use a simple RC anti-aliasing filter before the ADC After A/D conversion, perform digital low-pass filtering and then down sampling to f nyquist Effective resolution with oversampling N eff = N + 1/2 *log 2 (f s /f nyquist ), where N is the resolution of an ideal N-bit ADC at the Nyquist rate If OSR = f s /f nyquist = 1024, an 8-bit ADC gets and effective resolution equal to that of a 13-bit ACD at the Nyquist rate

Trigger (from hardware or software) A trigger is a signal that causes a device to perform an action, such as starting a data acquisition. You can program your DAQ device to react on triggers such as: a software command (software trigger) a condition on an external digital signal a condition on an external analog signal E.g. level triggering

Important trigger types Start trigger start data acquisition when an external digital signal have e.g. a rising edge. Pre-trigger Uses a data buffer (circular buffer) Can include a specified number of samples before the trigger event. Useful for e.g. high speed imaging.