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DSP Sampling Basics

Digital signal processing provides an alternative method for processing analog signals digitally. It allows flexibility, accuracy, and implementation of sophisticated algorithms at a lower cost compared to analog processing. For digital processing, an analog-to-digital converter samples the analog signal at twice its highest frequency per the Nyquist sampling theorem. It then quantizes and assigns a binary code to each sample. The digital-to-analog converter reconstructs the analog signal by interpolating between samples.

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78 views12 pages

DSP Sampling Basics

Digital signal processing provides an alternative method for processing analog signals digitally. It allows flexibility, accuracy, and implementation of sophisticated algorithms at a lower cost compared to analog processing. For digital processing, an analog-to-digital converter samples the analog signal at twice its highest frequency per the Nyquist sampling theorem. It then quantizes and assigns a binary code to each sample. The digital-to-analog converter reconstructs the analog signal by interpolating between samples.

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atik al mustahid
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Digital Signal Processing

Digital signal processing provides alternative method for analog signal


Why DSP?

➢ Allows flexibility in reconfiguring


operations
➢ Accuracy
To perform the processing ➢ Allows implementation of more
digitally, need interface
between analog signal & sophisticated algorithm.
digital processor ▪ Programmable Digital computer ➢ Digital implementation of system is
▪ Microprocessor
cheaper.
Digital Signal Processing
A/D Conversion
3 steps

1. Sampling
2.Quantization
3.Coding
Sampling Theorem
• To reconstruct accurately an analog signal at receiver output, it must
be sampled at twice of its highest frequency content.
• If g(t) contains highest frequency content B Hz, then the minimum
sampling rate or Nyquist rate, fs ≥2B
• The sampling interval Ts = 1/fs. Therefore Ts < 1/2B
Aliasing Effect
Aliasing Effect
• All practical signals are non-band limited, they have infinite band width.
• If the sampled signal is passed through an ideal low pass filter, the output is not
G(ω) but a version of G(ω) distorted as a result of two separate causes:-
1. The loss of the tail of G(ω) beyond |f| > fs/2 Hz
2. The reappearance of this tail inverted or folded on to the spectrum.
• This tail inversion know as spectral folding or aliasing.
Aliasing Effect: Solution
• We should eliminate this components beyond fs/2 = 1/2Ts Hz from g(t) before
sampling g(t). This suppression of higher frequencies can be accomplished by an
ideal low pass filter of bandwidth fs/2 Hz. This filter is called anti-aliasing filter.
Quantization
• The process of converting discrete- time continuous- amplitude signal into digital
signal by expressing each sample value as a finite number of digit.
• The error introducing in representing the continuous valued signal by a finite set
of discrete value levels is called quantization error.
Quantization
Coding
• The coding process in an A/D converter assigns a unique binary number to each
quantization level. If L is the number of levels, then L levels can be represented by
b bits
where 2𝑏 = 𝐿 𝑜𝑟 𝑏 = log 2 𝐿
D/A Conversion

• The task of D/A converter is to interpolate between samples.

Ref Book: Digital Signal Processing- J.G.Prokis

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