Ni.com Data Analysis: Time and Frequency Domain. ni.com Typical Data Acquisition System.

Slides:



Advertisements
Similar presentations
| Page Angelo Farina UNIPR | All Rights Reserved | Confidential Digital sound processing Convolution Digital Filters FFT.
Advertisements

Frequency analysis.
Physical Layer: Signals, Capacity, and Coding
Copyright 2001, Agrawal & BushnellVLSI Test: Lecture 181 Lecture 18 DSP-Based Analog Circuit Testing  Definitions  Unit Test Period (UTP)  Correlation.
Analog-to-Digital Converter (ADC) And
ACHIZITIA IN TIMP REAL A SEMNALELOR. Three frames of a sampled time domain signal. The Fast Fourier Transform (FFT) is the heart of the real-time spectrum.
Speech Compression. Introduction Use of multimedia in personal computers Requirement of more disk space Also telephone system requires compression Topics.
The Illinois Society of Electroneurodiagnostic Technologists (ISET) Fall Meeting: Electronics Crash Course for Technologists Saturday, November 9, 2013.
Filtering Filtering is one of the most widely used complex signal processing operations The system implementing this operation is called a filter A filter.
Digital Fundamentals Tenth Edition Floyd Chapter 12.
Analog to Digital Conversion. 12 bit vs 16 bit A/D Card Input Volts = A/D 12 bit 2 12 = Volts = Volts = 2048 −10 Volts = 0 Input Volts.
Presented by- Md. Bashir Uddin Roll: Dept. of BME KUET, Khulna-9203.
SIMS-201 Characteristics of Audio Signals Sampling of Audio Signals Introduction to Audio Information.
IT-101 Section 001 Lecture #8 Introduction to Information Technology.
Discussion #25 – ADCECEN 3011 Conversion Mosiah 5:2 2 And they all cried with one voice, saying: Yea, we believe all the words which though has spoken.
EET260: A/D and D/A converters
CEN352, Dr. Ghulam Muhammad King Saud University
Spectrum analyser basics Spectrum analyser basics 1.
Spectrum Analyzer. Another Oscilloscope??? Like an oscilloscope Oscilloscope in time domain Spectrum analyzer in frequency domain (selectable)
JF 12/04111 BSC Data Acquisition and Control Data Representation Computers use base 2, instead of base 10: Internally, information is represented by binary.
William Stallings Data and Computer Communications 7th Edition (Selected slides used for lectures at Bina Nusantara University) Data, Signal.
EKT343 –Principle of Communication Engineering
Data Acquisition. Data Acquisition System Analog Signal Signal Conditioner ADC Digital Processing Communication.
INTRODUCTION TO DIGITAL SIGNAL PROCESSING Dr. Hugh Blanton ENTC 4347.
Digital Audio Multimedia Systems (Module 1 Lesson 1)
 Principles of Digital Audio. Analog Audio  3 Characteristics of analog audio signals: 1. Continuous signal – single repetitive waveform 2. Infinite.
Digital Communication Techniques
Department of Electrical & Computer Engineering 1 ES585a - Computer Based Power System Protection Course by Dr.T.S.Sidhu - Fall 2005 Class discussion presentation.
Digital to Analogue Conversion Natural signals tend to be analogue Need to convert to digital.
Over-Sampling and Multi-Rate DSP Systems
T Digital Signal Processing and Filtering
Spectrum Analyzer. INTRODUCTION  A spectrum in the practical sense is a collection of sine waves, when combined properly produces the required time domain.
Formatting and Baseband Modulation
DSP. What is DSP? DSP: Digital Signal Processing---Using a digital process (e.g., a program running on a microprocessor) to modify a digital representation.
Numerical algorithms for power system protection Prof. dr. sc. Ante Marušić, doc. dr. sc. Juraj Havelka University of Zagreb Faculty of Electrical Engineering.
EE421, Fall 1998 Michigan Technological University Timothy J. Schulz 08-Sept, 98EE421, Lecture 11 Digital Signal Processing (DSP) Systems l Digital processing.
Chapter 5 Frequency Domain Analysis of Systems. Consider the following CT LTI system: absolutely integrable,Assumption: the impulse response h(t) is absolutely.
Vibrationdata 1 Unit 5 The Fourier Transform. Vibrationdata 2 Courtesy of Professor Alan M. Nathan, University of Illinois at Urbana-Champaign.
DSP Techniques for Software Radio DSP Front End Processing Dr. Jamil Ahmad.
Sampling Terminology f 0 is the fundamental frequency (Hz) of the signal –Speech: f 0 = vocal cord vibration frequency (>=80Hz) –Speech signals contain.
Lab #8 Follow-Up: Sounds and Signals* * Figures from Kaplan, D. (2003) Introduction to Scientific Computation and Programming CLI Engineering.
GCT731 Fall 2014 Topics in Music Technology - Music Information Retrieval Overview of MIR Systems Audio and Music Representations (Part 1) 1.
Lecture 1 Signals in the Time and Frequency Domains
Basics of Signal Processing. SIGNALSOURCE RECEIVER describe waves in terms of their significant features understand the way the waves originate effect.
Understanding ADC Specifications September Definition of Terms 000 Analogue Input Voltage Digital Output Code FS1/2.
Lecture 18 DSP-Based Analog Circuit Testing
CSC361/661 Digital Media Spring 2002
ACOE2551 Microprocessors Data Converters Analog to Digital Converters (ADC) –Convert an analog quantity (voltage, current) into a digital code Digital.
Dual-Channel FFT Analysis: A Presentation Prepared for Syn-Aud-Con: Test and Measurement Seminars Louisville, KY Aug , 2002.
Filters and Delta Sigma Converters
Acoustic Analysis of Speech Robert A. Prosek, Ph.D. CSD 301 Robert A. Prosek, Ph.D. CSD 301.
ECE 4710: Lecture #6 1 Bandlimited Signals  Bandlimited waveforms have non-zero spectral components only within a finite frequency range  Waveform is.
1 Introduction to Information Technology LECTURE 6 AUDIO AS INFORMATION IT 101 – Section 3 Spring, 2005.
ECE 4710: Lecture #7 1 Overview  Chapter 3: Baseband Pulse & Digital Signaling  Encode analog waveforms into baseband digital signals »Digital signaling.
CS Spring 2009 CS 414 – Multimedia Systems Design Lecture 3 – Digital Audio Representation Klara Nahrstedt Spring 2009.
Fundamentals of Sampling, Data Acquisition & Digital Devices.
4.2 Digital Transmission Pulse Modulation Pulse Code Modulation
CS Spring 2014 CS 414 – Multimedia Systems Design Lecture 3 – Digital Audio Representation Klara Nahrstedt Spring 2014.
Sampling and Aliasing.
Digital Audio I. Acknowledgement Some part of this lecture note has been taken from multimedia course made by Asst.Prof.Dr. William Bares and from Paul.
ΨΗΦΙΑΚΟΣ ΕΛΕΓΧΟΣ (22Δ802) Β΄ ΕΞΑΜΗΝΟ Καθηγητής Πέτρος Π. Γρουμπός  Ώρες Γραφείου: Τετάρτη Πέμπτη Παρασκευή 11:00- 12:00 Γραφείο: 1.
By. Jadhav Avinash J Roll no - 2K13E11. Reference: Hewlett Packard Agilent Technology Wikipedia GwINSTEK.
Lecture Notes / PPT UNIT III
Lifecycle from Sound to Digital to Sound. Characteristics of Sound Amplitude Wavelength (w) Frequency ( ) Timbre Hearing: [20Hz – 20KHz] Speech: [200Hz.
Fourier Analysis Patrice Koehl Department of Biological Sciences National University of Singapore
COMPUTER NETWORKS and INTERNETS
Unit 5 The Fourier Transform.
Digital Acquisition of Analog Signals – A Practical Guide
Fourier Analyses Time series Sampling interval Total period
CEN352, Dr. Ghulam Muhammad King Saud University
Presentation transcript:

ni.com Data Analysis: Time and Frequency Domain

ni.com Typical Data Acquisition System

ni.com Digitization An analog signal is sampled at a point in time and converted to a time series

ni.com Digitization Each sampled signal value is digitized using and analog-to-digital converter Parameters: –Resolution: number of bits used to represent the analog signal –Range: min. and max. voltage ADC can span (-5V to +5V) –Gain: range scale factor (gain factor of 10 means that a range spans 1/10 of the original range). –Polarity: single (-5 to 5V) or double (0 to 10V) Each sampled signal value is digitized using and analog-to-digital converter Parameters: –Resolution: number of bits used to represent the analog signal –Range: min. and max. voltage ADC can span (-5V to +5V) –Gain: range scale factor (gain factor of 10 means that a range spans 1/10 of the original range). –Polarity: single (-5 to 5V) or double (0 to 10V)

ni.com Code Width (LSB) Number of codes is a function of resolution: #of codes = 2 Code width (vertical sensitivity is the amount of voltage corresponding to a one-bit increment in code number LSB = Number of codes is a function of resolution: #of codes = 2 Code width (vertical sensitivity is the amount of voltage corresponding to a one-bit increment in code number LSB = resolution range gain x #of codes

ni.com Code Value to Voltage Conversion : voltage = (code) x code_width + Conversion : voltage = (code) x code_width + Bottom of range gain

ni.com When to Sample? Settling time is important desired measured desired

ni.com When to Sample?

ni.com Improperly sampled Properly sampled f N = f s /2f s : sampling frequency Sampling Guidelines Nyquist Theorem sampling rate > 2 x maximum frequency of signal Nyquist Frequency (f N ) maximum frequency that can be analyzed Frequencies above Nyquist Frequency cause aliasing Nyquist Theorem sampling rate > 2 x maximum frequency of signal Nyquist Frequency (f N ) maximum frequency that can be analyzed Frequencies above Nyquist Frequency cause aliasing

ni.com What is Aliasing? (Time Domain) Samples acquired at 1 kHz 150 Hz sine tone ? 850 Hz sine tone ? (1000 Hz – 150 Hz) 1150 Hz sine tone ? (1000 Hz Hz)

ni.com  n * F sampling  150 Hz Aliasing (Frequency Domain) 150, 850, and 1150 Hz

ni.com f1f1 f1f1 f3f3 f3f3 f s /2 fsfs fsfs alias free bandwidth f1f1 f1f1 f s /2 fsfs fsfs anti-aliasing filter anti-aliasing filter f2f2 f2f2 attenuated f 2 attenuated f 2 alias f 3 alias f 3 f4f4 f4f4 RAW SIGNAL ACQUIRED SIGNAL Time Domain Considerations Alias Free Bandwidth Nyquist Frequency Sample Frequency

ni.com Removes frequencies higher than Nyquist frequency Analog low-pass filter Before sampling Removes frequencies higher than Nyquist frequency Analog low-pass filter Before sampling Time Domain Considerations Anti-Aliasing Filter Flat Frequency Response Sharp Roll-off

ni.com Anti-Aliasing Filter (Analog Only) Analog anti-aliasing filter –Passband – DC to 400 Hz –Stopband – 600 Hz  Analog anti-aliasing filter –Passband – DC to 400 Hz –Stopband – 600 Hz 

ni.com Anti-Aliasing Filter (Analog+Digital) Analog filter –Passband – DC to 400 Hz –Stopband – 1600 Hz  Analog filter –Passband – DC to 400 Hz –Stopband – 1600 Hz  Digital filter (2X decimation) Passband – DC to 400 Hz Stopband – 600 to 1400 Hz Digital filter (2X decimation) Passband – DC to 400 Hz Stopband – 600 to 1400 Hz

ni.com Sampling Methods Simultaneous Sampling Interval Sampling Continuous Sampling Random Sampling Multiplexing Simultaneous Sampling Interval Sampling Continuous Sampling Random Sampling Multiplexing

ni.com Simultaneous Sampling Critical time relation btw. signals Requires: –Sample-and-hold circuits OR –Individual ADC’s Critical time relation btw. signals Requires: –Sample-and-hold circuits OR –Individual ADC’s

ni.com Interval Sampling Simulate simultaneous sampling for low- frequency signals

ni.com Continuous Sampling Sampling multiplexed channels at constant rate. Causes phase skew btw. Channels –Use only if time relation btw. Channels is not important Sampling multiplexed channels at constant rate. Causes phase skew btw. Channels –Use only if time relation btw. Channels is not important

ni.com Classic Multiplexed MIO Low cost/flexible –No anti-aliasing filters –Only one A/D converter for all channels Conflicts with some common requirements of many applications that require dynamic signal acquisition –Aliasing protection –Simultaneous sampling Low cost/flexible –No anti-aliasing filters –Only one A/D converter for all channels Conflicts with some common requirements of many applications that require dynamic signal acquisition –Aliasing protection –Simultaneous sampling

ni.com Multiplexing: Some Definitions Channels – the actual number of input channels scanned by the board Scan clock – the output data rate for each channel Decimation factor (D) – the acquisition over- sampling factor for each channel A/D clock – the actual sample rate of the multiplexing A/D converter A/D clock = channels * decimation * scan clock Channels – the actual number of input channels scanned by the board Scan clock – the output data rate for each channel Decimation factor (D) – the acquisition over- sampling factor for each channel A/D clock – the actual sample rate of the multiplexing A/D converter A/D clock = channels * decimation * scan clock

ni.com Multiplexing Identical Input 4 channels (same input signal on all channels) Scan clock = 1 kHz A/D clock = 4 kHz 4 channels (same input signal on all channels) Scan clock = 1 kHz A/D clock = 4 kHz

ni.com Resulting Delayed Acquisitions Our four channels appear to have different phases even though we input the same signal to each Scan clock = 1 kHz A/D clock = 4 kHz Our four channels appear to have different phases even though we input the same signal to each Scan clock = 1 kHz A/D clock = 4 kHz

ni.com Relative Phase Responses: Skew 4 channels Scan clock = 1 kHz A/D clock = 16 kHz (over-sampled 4X) 4 channels Scan clock = 1 kHz A/D clock = 16 kHz (over-sampled 4X)

ni.com Additional Time Domain Considerations  analog to digital converter –High resolution –Built-in anti-aliasing filters –Suited for sound and vibration measurements Simultaneous sampling and triggering –Phase relationship between signals Programmable gain Overload detection  analog to digital converter –High resolution –Built-in anti-aliasing filters –Suited for sound and vibration measurements Simultaneous sampling and triggering –Phase relationship between signals Programmable gain Overload detection

ni.com Time Domain Considerations Smoothing Windows Nonintegral number of cycles Reduces spectral leakage Window selection depends on the application PC Based instruments greatly facilitate transient analysis Reduces spectral leakage Window selection depends on the application PC Based instruments greatly facilitate transient analysis No windowing Windowing Window

ni.com Time vs Frequency Domain

ni.com Sample Time Domain Signal Sample Time Domain Signal FFT Anti-Alias Filter Anti-Alias Filter Octave Acquire Waveform Acquire Waveform Basics of Frequency Measurements Signal Conditioning Signal Conditioning Frequency Analysis Frequency Analysis

ni.com Frequency Domain Analysis FFT analysis Octave analysis Swept sine analysis FFT analysis Octave analysis Swept sine analysis

ni.com FFT Analysis Time domain in discrete values Use Discrete Fourier Transform (DFT) Fast Fourier Transform (FFT) Optimized version of DFT Highest frequency that can be analyzed Frequency resolution Time domain in discrete values Use Discrete Fourier Transform (DFT) Fast Fourier Transform (FFT) Optimized version of DFT Highest frequency that can be analyzed Frequency resolution f s : sampling frequency T : total acquisition time N : FFT block size T : total acquisition time N : FFT block size

ni.com FFT Analysis FFT gives magnitude and phase information –Magnitude = sqrt(Real^2 + Imag^2) –Phase = Tan -1 (Imag / Real) Power Spectrum reflects the energy content –Power Spectrum = Mag^2 Applications Vibration analysis Structural dynamics testing Preventative maintenance Shock testing FFT gives magnitude and phase information –Magnitude = sqrt(Real^2 + Imag^2) –Phase = Tan -1 (Imag / Real) Power Spectrum reflects the energy content –Power Spectrum = Mag^2 Applications Vibration analysis Structural dynamics testing Preventative maintenance Shock testing

ni.com Concentrates (“zooms”) FFT on a narrow band of frequencies Improves frequency resolution Distinguishes between closely-spaced frequencies Baseband analysis requires longer acquisition time for better resolution – requires more computation Concentrates (“zooms”) FFT on a narrow band of frequencies Improves frequency resolution Distinguishes between closely-spaced frequencies Baseband analysis requires longer acquisition time for better resolution – requires more computation Zoom FFT Analysis

ni.com Zoom FFT Analysis Baseband FFT Analysis Zoom FFT Analysis Zoom FFT Analysis

ni.com Zoom FFT Analysis – How It’s Done f1f1 f1f1 f2f2 f2f2 f 1 + f 2 f 1 - f 2 f1f1 f2f2 f 1 – f 2

ni.com Octave Analysis Analysis performed through a parallel bank of bandpass filters –One octave corresponds to the doubling of the frequency –Reference frequency is 1 kHz (audio domain) Analysis performed through a parallel bank of bandpass filters –One octave corresponds to the doubling of the frequency –Reference frequency is 1 kHz (audio domain) 220 Hz 440 Hz 880 Hz A A A A A A

ni.com Octave Analysis Octave analysis gives log-spaced frequency information –Similar to human perception of sound –1/1, 1/3, 1/12, and 1/24 octave analysis FFT gives linearly-spaced frequency information Applications –noise emissions testing –acoustic intensity measurement –sound power measurement –audio equalization Octave analysis gives log-spaced frequency information –Similar to human perception of sound –1/1, 1/3, 1/12, and 1/24 octave analysis FFT gives linearly-spaced frequency information Applications –noise emissions testing –acoustic intensity measurement –sound power measurement –audio equalization

ni.com Swept Sine Analysis Source steps through a range of frequencies Analyzer measures frequency amplitude and phase at each step Non-FFT based Source steps through a range of frequencies Analyzer measures frequency amplitude and phase at each step Non-FFT based Source Device Under Test Frequenc y Respon se

ni.com Auto-ranging: dynamic range optimized at each frequency Adjust source amplitude Adjust input range Both improve dynamic range at particular frequencies –Can get 140 dB effective dynamic range Auto-ranging: dynamic range optimized at each frequency Adjust source amplitude Adjust input range Both improve dynamic range at particular frequencies –Can get 140 dB effective dynamic range Swept Sine Analysis Channel B Channel A

ni.com Swept Sine Analysis Auto-resolution –Sweep optimized - more time at lower frequencies, less time at higher –Increases frequency resolution on rapidly changing responses Applications –Speaker testing –Cell phone testing –Electronic equipment characterization Auto-resolution –Sweep optimized - more time at lower frequencies, less time at higher –Increases frequency resolution on rapidly changing responses Applications –Speaker testing –Cell phone testing –Electronic equipment characterization

ni.com Comparison of Frequency Analysis Methods FFT analysis –Very fast –Linear frequency scale –Based on discrete Fourier transform Octave analysis –Logarithmic frequency scale –Set of filters dividing frequency into bands –Similar to how human ear perceives sound Swept sine analysis –Good dynamic range –Source and analyzer step across frequency range –Slower response FFT analysis –Very fast –Linear frequency scale –Based on discrete Fourier transform Octave analysis –Logarithmic frequency scale –Set of filters dividing frequency into bands –Similar to how human ear perceives sound Swept sine analysis –Good dynamic range –Source and analyzer step across frequency range –Slower response

ni.com Next Lecture Output signals Servo-control systems Output signals Servo-control systems