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29 Sept '09Comp30291 Section 11 Comp30291 Digital Media Processing Barry Cheetham barry@man.ac.uk www.cs.man.ac.uk/~barry/mydocs/Comp30291

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29 Sept '09Comp30291 Section 12 Signal: time-varying measurable quantity whose variation normally conveys information. Quantity often a voltage obtained from some transducer e.g. a microphone. Define two types of signal: Continuous time (analogue) Discrete time Section 1 Introduction

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29 Sept '09Comp30291 Section 13 Analogue signals Continuous variations of voltage. Exist for all values of t in range - to + . Examples: (i) 5sin(62.82t) : sine-wave of frequency 62.82 radians/second ( 10 Hz) t Voltage 0.1 -0.1 5 t Voltage 1 Graph of voltage against time gives continuous 'waveform' :

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29 Sept '09Comp30291 Section 14 ‘Sound’ Variation in air pressure May be generated by musical instruments or a human voice. E.g. by a piano string vibrating or your vocal cords vibrating Local pressure variation travels thro’ the air to your ear or a microphone. There it causes a ‘diaphragm’ to vibrate. Microphone produces a continuous voltage proportional to the variation in air pressure. Graph of voltage against time is ‘sound waveform’.

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29 Sept '09Comp30291 Section 15 Segment of telephone speech: ‘Its close to Newcastle’ 00.511.5 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 closetoNewcastle.pcm time in seconds voltage

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29 Sept '09Comp30291 Section 16 16 s segment of violin music

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29 Sept '09Comp30291 Section 17 50ms segment of music: violin note 00.0050.01 0.0150.020.0250.030.035 0.04 0.045 0.05 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 Violin note before sampling time in seconds voltage What is frequency of note being played? Ans 16 cycles in 0.05 s 320 Hz

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29 Sept '09Comp30291 Section 18 Discrete-time signal Exists only at discrete points in time. Often obtained by sampling an analogue signal, i.e. measuring its value at discrete points in time. Sampling points separated by equal intervals of T seconds. Given analogue signal x(t), x[n] = value of x(t) when t = nT. Sampling process produces a sequence of numbers: {..., x[-2], x[-1], x[0], x[1], x[2],..... } Referred to as {x[n]} or ' the sequence x[n] '. Sequence exists for all integer n in the range - to .

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29 Sept '09Comp30291 Section 19 Examples of discrete time signals: (i) {..., -4, -2, 0, 2, 4, 6,....} sequence whose nth element, x[n], is defined by : x[n] = 2n. Underline sample corresponding to n = 0. (ii) {..., -4.75, -2.94, 0, 2.94, 4.75, 4.76,...} sequence with x[n] = 5 sin(62.82t) with t=nT and T=0.01. (iii) {..., 0,..., 0. 0, 1, 1, 1,..., 1,...} “ unit step ” sequence whose nth element is: 0 : n < 0 u[n] = 1 : n 0

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29 Sept '09Comp30291 Section 110 n x[n] 1 2 3 4 -3 -2 -1 2 - -2- Graphs of discrete time signals For Example (i):

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29 Sept '09Comp30291 Section 111 n Graphs of discrete time signals x[n] For Example (ii): 1 2 3 4 -3 -2 -1 3 - -3

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29 Sept '09Comp30291 Section 112 n x[n] 1 2 3 4 -3 -2 -1 1 - -1- Graphs of discrete time signals For Example (iii):

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29 Sept '09Comp30291 Section 113 Speech sampled at 8kHz with 16 bits per sample 020004000600080001000012000 -3 -2.5 -2 -1.5 -0.5 0 0.5 1 1.5 2 x 10 4closetoNewcastle.pcm sample index n 16-bit binary number

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29 Sept '09Comp30291 Section 114 45 ms segment of music sampled at 22.05 kHz Frequency of note 19/1000 = 0.019 cycles/sample = 0.019 *22050 = 419 Hz

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29 Sept '09Comp30291 Section 115 50 ms segment of music sampled at 22.05 kHz 010020030040050060070080090010001100 -3 -2 0 1 2 3 x 10 4 Violin note sampled at 22050 Hz sample index n 16-bit binary number What’s the frequency now, folks?

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29 Sept '09Comp30291 Section 116 Voiced telephone speech sampled at 16000 Hz Short ( 100ms) segment of a vowel. [1ms = 1/1000 second] Vowels are approximately periodic. What is the pitch of the voice here? Ans: 9 cycles in 1/10 s i.e. 90 Hz - probably male speech 050010001500 -2 -1.5 -0.5 0 0.5 1 1.5 2 x 10 4 n 16-bit binary number

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29 Sept '09Comp30291 Section 117 Discrete time signals often generated by ADC devices. Produce binary numbers from sampled voltages or currents. Accuracy determined by 'word-length ' of ADC device, i.e. number of bits available for each binary number. Quantisation: Truncating or rounding sampled value to nearest available binary number Resulting sequence of quantised numbers is a digital signal.. i.e. discrete time signal with each sample digitised for arithmetic processing. Digital signal

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29 Sept '09Comp30291 Section 118 Signal Processing Analogue signals "processed" by circuits consisting of resistors, capacitors, transistors etc. Digital signals "processed" using programmed computers, microcomputers or special purpose digital hardware. Examples of the type of processing that may be carried out are: (i) amplification or attenuation. (ii) filtering out some unwanted part of the signal. (iii) rectification : making waveform purely positive. (iv) multiplying by another signal.

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29 Sept '09Comp30291 Section 119 ‘Real time’ & ‘non-real time’ processing ’Real time' processing: A mobile phone contains 'DSP' processor fast & powerful enough to perform the mathematical operations required to process digitised speech as it is being received. ‘Non-real time’ processing: A PC can perform DSP processing on a stored recording of music. It can take as much time as it needs to complete this processing. Useful in its own right; e.g for MP3 compression. Also used to 'simulate' software for real time DSP systems before they are built into special purpose hardware. Simulated DSP systems tested with stored segments of speech/music.

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29 Sept '09Comp30291 Section 120 Introduction to ‘MATLAB’ High level computing language for matrix operations. Widely used for studying signal processing & comms. No need to declare variables in advance. Each variable may be array with real or complex elements. No distinction between reals & integers

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29 Sept '09Comp30291 Section 121 Examples of MATLAB statements A = [ 1 2.8 3.2 4 5 6 7.7 8 ] ; X = [ 1 2 3 4 ] ; Y = [ 1 2 3 4 ] ; Z = 3 ; W = 4 + 5.5j ;% Comment

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29 Sept '09Comp30291 Section 122 clear all; T = 0.01; % sampling interval (seconds) for n=1:80 s(n) = 5*sin(2*pi*10*n*T); end; plot (s); Generating, storing & plotting signals Generate & plot 80 samples of a 10 Hz sine-wave of amplitude 5, sampled at 100 Hz: 01020304050607080 -5 0 5

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29 Sept '09Comp30291 Section 123 Signals in MATLAB Segments in row or column ‘vector’ arrays: n x 1 or 1 x n matrices. Previous example produces row vector s To produce column vector s containing sine-wave segment for n = [1:100]’ s(n) = sin(n*pi/10) end; Another (preferred) way of defining column vector s: n=[1:100]’ ; % 100x1 col vector with entries 1,2,...,100 s(n) = sin(n*pi/10)% For each entry of n, calculate value.

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29 Sept '09Comp30291 Section 124 x = [ 1.1, 2, 3.3, 4, 5 ] defines row vector x = [ 1.1; 2; 3.3; 4; 5 ] defines column vector. Easier to write than: x = [ 1.1 2 3.3 4 5 ] Row-vectors & column-vectors

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29 Sept '09Comp30291 Section 125 Transpose x.' is transpose of x. (Note the.' symbols). If x is col vector, x.' will be row vector. Note dot apostrophe (.') rather than simply apostrophe ('). Omitting dot will give “complex conjugate transpose”. Index of first element of a row or column vector is always one. Sometimes inconvenient since a signal starts at time zero. Frequency spectrum normally starts at frequency zero. Consider adopting following statements: for n = 0 : 1023 x(1+n) = sin(n*pi/10) end; Preserves natural definition of n as time-index.

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29 Sept '09Comp30291 Section 126 Command line mode Run MATLAB by clicking on icon & type in statements. If in doubt about any command, e.g. ‘plot’ type: help plot

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29 Sept '09Comp30291 Section 127 More convenient to have statements in script file e.g. myprog.m. Just a text file often referred to as an "M-file". To execute the statements in "myprog.m" just type "myprog" MATLAB has a text editor. Paths must be set to a "work" directory where the "M-files" will be stored. Scripting ‘m-files’

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29 Sept '09Comp30291 Section 128 Digital media file formats Confusing number of different ways of storing speech, music, images & video in digital form. Consider just six: ‘*.pcm’ or ‘*.raw’: binary file without header often used for uncompressed ‘narrowband’speech sampled at 8 or 16 kHz. ‘*.wav’: binary file with header often used uncompressed high- fidelity stereo sound (music or speech) sampled at 44.1 kHz. ‘*.mp3’ for mp3 compressed music or hi-fi speech. ‘*.tif’ files for images with lossless compression. ‘*jpg’ files for images with lossy compression ‘*.avi’ audio/video interleaved files.

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29 Sept '09Comp30291 Section 129 MATLAB functions for reading/writing *.pcm % Open *.pcm file of 8 kHz sampled speech,16 bits/sample IFid=fopen('operamp.pcm','rb'); inspeech = fread(IFid, 'int16'); L = length(inspeech); % number of samples scale = max(abs(inspeech)) + 1; sound(outspeech/scale,8000,16); % listen to the speech outspeech = inspeech / 2; % reduce by 6 dB OFid = fopen(‘opout.raw’, ‘wb’); fwrite(OFid,outspeech, ‘int16’); fclose('all');

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29 Sept '09Comp30291 Section 130 MATLAB functions for reading/writing *.wav clear all; close all; % closes all graphs etc. [inmusic, Fs, Nbits] = wavread('cap4th.wav'); S=size(inmusic); % inmusic may be mono or stereo L=S(1); % number of mono or stereo samples if S(2) ==2 disp(‘music is stereo’); end; sound(inmusic,Fs,Nbits); outmusic = inmusic /sqrt(2); % reduce amplitude by 3 dB wavwrite(outmusic, Fs, Nbits,’outfile.wav’); fclose (‘all’); % closes all files

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29 Sept '09Comp30291 Section 131 Another way of saving/loading arrays Does not produce files that can be read by wave-editors, though it can produce text files. To save an array called "b" in a disk file called "bits.dat": save bits.dat b -ascii Omitting ‘-ascii’ creates a more compact ‘non-ascii’ binary file. To read contents of ascii file ‘bits.dat’ into array called ‘bits’: load bits.dat -ascii

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29 Sept '09Comp30291 Section 132 Frequency spectrum & sampling A sine-wave has a ‘frequency’( f ) in Hz or 2 f radians/second. Sine-waves do not sound very nice. Speech & music need of lots of sine-waves added together. Harmonically related, i.e. f, 2f, 3f, 4f, 5f,... Other sounds approximated as sum of lots of very small sine- waves. Any type of signal has a ‘frequency spectrum’. Most energy in speech is within 300 Hz to 3400 Hz (3.4 kHz). Energy in music that we can hear is in range 50 Hz to 20 kHz.

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29 Sept '09Comp30291 Section 133 Spectral analysis of a sine-wave Power spectral density f Hz F x(t)=Asin(2 Ft) is sine-wave of frequency F Hz All its power is concentrated at F Hz. Its power spectrum is infinitely high line at f=F. Draw upward arrow & call it an ‘impulse’. Its ‘strength’ is A 2 /2 watt/Hz for Asin(2 Ft) Can make length of line strength but must put arrow on top. A 2 /2

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29 Sept '09Comp30291 Section 134 Spectral analysis of musical note Power spectral density f Hz F ‘Fundamental’ sine-wave at F Hz & lots of sine-wave ‘harmonics’ at 2F, 3F, etc. F is ‘note’: 440 Hz for ‘A’, 266 Hz for ‘mid C’ What is the note being played here? 910 Hz ( Just above ‘top A’). 20 kHz

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29 Sept '09Comp30291 Section 135 Spectral analysis of analogue signals Spectral analysis of periodic sound before it is sampled is tricky because of the upward arrows. They are infinitely high!! We plotted their ‘strengths’, not their amplitudes. Things become easier when we spectrally analyse digitised sound (later)!

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29 Sept '09Comp30291 Section 136 ‘Sampling Theorem’ If a signal has all its spectral energy below B Hz, & is sampled at Fs Hz where Fs ≥ 2B, it can be reconstructed exactly from the samples. Speech below 3400 Hz can be sampled at 6800 Hz (in practice 8 kHz). Music can be sampled at 40 kHz (in practice 44.1 kHz). When we process a sampled signal in MATLAB & sampling rate was Fs Hz, we can only observe frequencies in range 0 to Fs/2.

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29 Sept '09Comp30291 Section 137 ‘Aliasing’ What happens if you sampled a speech or music signal at Fs & it has some spectral energy above Fs/2 ? Leads to a form of distortion known as ‘aliasing’ which sounds very bad. Must filter off all spectral energy at & above Fs/2 Hz before sampling at Fs.

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29 Sept '09Comp30291 Section 138 MATLAB Signal Processing toolbox Set of functions for: (i) carrying out operations on signal segments stored in vectors (ii) evaluating the effect of these operations, and (iii) computing parameters of digital filters etc. Two commonly used operations are: Digital filtering Spectral analysis

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29 Sept '09Comp30291 Section 139 Digital filtering Low-pass digital filters, high-pass and other categories Remove or enhance frequency components of a sampled signal. Low-pass filter aims to ‘filter out’ spectral energy above cut-off frequency Fc, keeping all spectral energy below Fc unchanged. Only consider frequencies up to Fs/2. Gain f Fc 1 Fs/2 ‘Gain’ frequency-response graph for ideal low-pass digital filter.

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29 Sept '09Comp30291 Section 140 Ideal high-pass digital filter Aims to ‘filter out’ spectral energy below cut-off frequency Fc, keeping all spectral energy above Fc unchanged. Only consider frequencies up to Fs/2. Gain f Fc 1 Fs/2 ‘Gain’ frequency-response graph for ideal high-pass digital filter:

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29 Sept '09Comp30291 Section 141 Ideal band-pass digital filter Aims to ‘filter out’ spectral energy below frequency F L, & above F H keeping all spectral energy between F L & F H unchanged. Only consider frequencies up to Fs/2. ‘Gain’ frequency-response graph for ideal band-pass digital filter: Gain f FLFL 1 Fs/2 FHFH

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29 Sept '09Comp30291 Section 142 Ideal band-stop digital filter Aims to ‘filter out’ spectral energy between F L, & F H keeping all spectral energy below F L & above F H unchanged. Only consider frequencies up to Fs/2. ‘Gain’ frequency-response graph for ideal band-stop digital filter:

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29 Sept '09Comp30291 Section 143 Ideal band-stop digital filter Aims to ‘filter out’ spectral energy between F L, & F H keeping all spectral energy below F L & above F H unchanged. Only consider frequencies up to Fs/2. Gain f FLFL 1 Fs/2 ‘Gain’ frequency-response graph for ideal band-stop digital filter: FHFH

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29 Sept '09Comp30291 Section 144 IIR & FIR digital filters There are 2 types of digital filter: Infinite impulse response (IIR) type. Finite impulse response (FIR) type. Either can be low-pass, high-pass, band-pass etc. MATLAB can design both types by many different methods.

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29 Sept '09Comp30291 Section 145 GAIN FREQUENCY 1 fCfC Gain-response of typical IIR low-pass filter in practice Previous graph shown ‘ideal response’. Referred to as ‘brick-wall’. In practice cut-off is more gradual as illustrated here. Fs/2

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29 Sept '09Comp30291 Section 146 GAIN (dB) FREQUENCY 0 fCfC Same gain-response now in deciBels Plot 20*log10(GAIN) against freq Absolute gain of 0 is 0 dB. Absolute gain of 0 is - dB. -10 -20 -40 -30 Fs/2

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29 Sept '09Comp30291 Section 147 Decibel scale Gain is often converted to deciBels (dB) Gain in dB = 20 log 10 (Absolute Gain) dB Absolute 01 62 -61/2 3 2 2010 40100

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29 Sept '09Comp30291 Section 148 y = filter(a, b, x) ; % Pass x thro’ digital filter to produce y. Coefficients in vectors a and b determine the filter’s effect. For an FIR filter, b=1. The filter has a ‘transfer function’ which is: a(1) + a(2)z -1 + a(3)z -2 +... H(z) = 1 + b(2)z -1 + b(3)z -2 +... freqz(a,b); % Display gain & phase responses of digital filters:- % Note the 'zoom' function. [a,b] = butter(n, f C /(Fs/2)); % Get a & b for nth order low-pass filter. Some of the signal processing functions provided

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29 Sept '09Comp30291 Section 149 [a,b] = butter(n, fc/(Fs/2), ‘high’); % High-pass Butt IIR. [a,b] = butter(n, [f L f U ]/(Fs/2) ); % Band-pass Butt IIR. [a,b] = butter(n, [f L f U ]/(Fs/2),’stop’); % Band-stop Butt IIR [a,b] = cheby1(n, Rp,fc/(Fs/2) ); % Low-pass Chebychev type 1. [a,b] = cheby2(n, Rs,.. ); % Low-pass Chebychev type 2. a = fir1(n, fc/(Fs/2));% Get 'a' coeffs of FIR low-pass filter order n. % Cut-off is fc. a = fir1(n, (f L, f U ]/(Fs/2) ); % FIR band-pass filter design. Many more exist. More functions for designing digital filters

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29 Sept '09Comp30291 Section 150 Specifying the cut-off frequencies This is a bit cumbersome. If the cut-off is Fc, must enter: Fc / (Fs/2) i.e. divide by half the sampling frequency!

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29 Sept '09Comp30291 Section 151 To illustrate for IIR : Design 4 th order "Butterworth type" IIR low-pass filter with Fc = 1000, Fs=8kHz. [a,b] = butter(4, 1000/4000 ) ; freqz(a,b); % plot gain & phase responses Implement it by applying it to signal in array x : y = filter(a,b,x);

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29 Sept '09Comp30291 Section 152 To illustrate for FIR: Design & implement 40 th order FIR low-pass digital filter with cut-off frequency 1 kHz Apply it to signal in array x. %To design: a = fir1(40, 1 / 4 ) ; freqz ( a, 1) ; % plot gain & phase % To implement: y = filter(a, 1, x );

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29 Sept '09Comp30291 Section 153 clear all; %Input speech from a file:- fs = 8000; % sampling rate in Hz IFid=fopen('operamp.pcm','rb'); Inspeech = fread(IFid, 'int16'); %Design FIR digital filter:- fc = 1000; % cut-off frequency in Hz [a b] = fir1(20, fc/(0.5*fs) ); freqz(a,b); %Process speech by filtering:- Outspeech = filter(a, b, Inspeech); % Output speech to a file OFid=fopen('newop.pcm','wb'); fwrite(OFid, Outspeech, 'int16'); fclose('all'); MATLAB Demo1: Filtering speech

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29 Sept '09Comp30291 Section 154 Speech file: ‘operamp.pcm’ This may be downloaded from www.cs.man.ac.uk/~barry/mydocs/Comp30192

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29 Sept '09Comp30291 Section 155 Graphs produced by ‘freqz’ in previous slide 4kHz

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29 Sept '09Comp30291 Section 156 Normalized Frequency (1 Fs/2) 00.10.20.30.40.50.60.70.80.91 -100 -80 -60 -40 -20 0 Magnitude (dB) Gain-response from previous slide 4kHz

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29 Sept '09Comp30291 Section 157 IFid=fopen('newop.pcm','rb'); speech = fread(IFid,'int16'); scale = max(abs(speech))+1; SOUND(speech/scale,8000,16); fclose('all'); MATLAB Demo2: Listen to speech in a file To listen to the filtered speech as produced by MATLAB Demo1, run the following program. Effect of the lowpass filtering should be to make speech sound ‘muffled’ Higher frequencies have been filtered out.

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29 Sept '09Comp30291 Section 158 Spectral analysis Fast Fourier Transform (FFT) is famous algorithm for determining frequency content of sampled signals. Given a signal segment of N samples stored in array x: X = fft(x) ; gives complex vector X of N samples of frequency spectrum of x. plot ( abs(X[1:N / 2] ) ; plots 'magnitude spectrum' of x from 0 Hz to Fs/2.

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29 Sept '09Comp30291 Section 159 FFT analyse 200 samples of 5Hz sine-wave clear all; close all; N=200; Fs = 100; % sampling frequency (Hz) for n=0:N-1 x(1+n) = 100*cos(2*pi*5*n/Fs) ; end; figure(1); plot(x); X=fft(x)/(N/2); % Dividing by N/2 gives correct scaling for graph. figure(2); plot(abs(X(1:N/2)),'*-'); grid on;

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29 Sept '09Comp30291 Section 160 Plot of 5Hz sine-wave segment sampled at 100 Hz Sample index n Value

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29 Sept '09Comp30291 Section 161 Spectral graph from fft Frequency-index (0:100 0:Fs/2) FFT magnitude 50Hz 5 Hz

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29 Sept '09Comp30291 Section 162 Analyse sum of 2 sine-waves clear all; N=200; Fs = 100; for n=0:N-1 x(1+n) = 100*cos(2*pi*5*n/Fs) + 50*sin(2*pi*10*n/Fs); end; figure(1); plot(x); X=fft(x)/(N/2); figure(2); plot(abs(X(1:N/2)),'*-'); grid on;

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29 Sept '09Comp30291 Section 163 Plot of sum of 2 sine-waves Sample-index n Voltage

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29 Sept '09Comp30291 Section 164 Spectral graph from FFT for 2 sine-waves Frequency index (100 Fs/2) 50Hz FFT magnitude

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29 Sept '09Comp30291 Section 165 50 ms segment of music sampled at 22.05 kHz

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29 Sept '09Comp30291 Section 166 FFT magnitude spectrum of music segment 550 Fs/2 = 11025 Hz

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29 Sept '09Comp30291 Section 167 Same FFT spectrum in dB against frequency - also frequency axis scaled to show Hz

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29 Sept '09Comp30291 Section 168 Decimation (down sampling) If a signal is sampled at 44.1 kHz, FFT gives spectrum with frequency range 0 to 22.05 kHz. You may only want to see spectrum from 0 to 2kHz. Down-sampling by a factor 10 makes Fs = 4.41 kHz. FFT then gives spectrum in range 0 to 2.205Hz. To down-sample x by factor 10: xd = decimate(x, 10); Creates new array xd of length 1/10 th of that of x. FFT analyse xd rather than x. ‘decimate’ works by low-pass filtering then re-sampling.

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29 Sept '09Comp30291 Section 169 Understanding decimation [a b] = butter(8, (Fs/24)/(Fs/2) ); xf = filter(a,b,x); %filter off all components above Fs/20 Hz % Now re-sample at Fs/10 Hz: for n=1:length(xf)/10 xd(n) = xf(n*10); % Omit 9 out of 10 samples end; We can easily write our own decimation procedure: But it’s quicker to use the MATLAB function ‘decimate’

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29 Sept '09Comp30291 Section 170 Spectrograms FFT can be used to analyse 20 to 30 ms of speech or music. Over longer periods of time, spectral distribution will be changing. We often want to analyse this change. A related function is ‘specgram’ Plots a whole succession of FFT spectral graphs showing how the distribution of frequencies is changing over time. Result is a ‘spectrogram’ which plots spectral distribution (using colour) against time. Demonstrated on next slide.

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29 Sept '09Comp30291 Section 171 Spectrograms for 4 Chinese words

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29 Sept '09Comp30291 Section 172 clear all; fs = 8000; %sampling rate in Hz T = 1/fs; % sampling interval (seconds) % Generate 10000 samples of 500 Hz sine-wave:- for n=1:10000 s(n) = 4000 * sin(2 * pi * 500 * n * T); end; % Store in a non-formatted (binary) file:- OFid=fopen('newsin.pcm','wb'); fwrite(OFid, s, 'int16'); fclose('all'); MATLAB demo: Stores sine-wave segment in pcm file

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29 Sept '09Comp30291 Section 173 for n=1:10000 Outsin(n) = 2*Insin(n); end; for n=1:10000 Outsin(n)=abs(Insin(n)); end; Outsin = abs(Insin); Outsin = 2*Insin; Using MATLAB more efficiently SlowFaster

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29 Sept '09Comp30291 Section 174 MATLAB Exercise 1: Modulation Generate 320 samples of a 50 Hz sine wave sampled at 8kHz & multiply this by a 1kHz sine-wave sampled at 8kHz. Plot resulting waveform MATLAB Exercise 2: Simulate POTS speech quality A ‘plain old fashioned telephone’ transmits speech between 300Hz & 3 kHz. Design & implement a digital filter to allow effect of this band-width restriction to be assessed by listening.

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29 Sept '09Comp30291 Section 175 MATLAB Exercise 3 File ‘noisyviolin.wav’ contains music sampled at 22.05 kHz. It is corrupted by: ‘high-pass’ random noise above 6 kHz and a sine-wave around 4 kHz. Design & implement 2 digital filters to remove the random noise and the tone. Listen to the ‘cleaned up music’

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29 Sept '09Comp30291 Section 176 clear all; fs = 8000; T=1/fs; for n=1:320 s(n) = sin(2*pi*50*n*T); end; figure(1); plot (s); for n=1:320 smod(n) = s(n)*sin(2*pi*1000*n*T); end; figure(2); plot (smod); Solution to MATLAB Exercise 1 050100150200250300350 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1

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29 Sept '09Comp30291 Section 177 % A solution for MATLAB Exercise 2 clear all; %Input speech from a file:- fs = 8000; % sampling rate in Hz IFid=fopen('operamp.pcm','rb'); Inspeech = fread(IFid, 'int16'); %Design FIR band-pass digital filter:- fU = 3000; % upper cut-off freq Hz fL = 300; % lower cut-off freq [a b] = fir1(100,[fL/(fs/2) fU/fs/2]); freqz(a,b); %Process speech by filtering:- Outspeech = filter(a, b, Inspeech); % Output speech to a file OFid=fopen('newop.pcm','wb'); fwrite(OFid, Outspeech, 'int16'); fclose('all');

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29 Sept '09Comp30291 Section 178 Solution to MATLAB exercise 3 clear all; close all; [noisymusic,Fs,bits] = wavread('noisyviolin.wav'); S=size(noisymusic); disp(sprintf('Size of array = %d, %d', S(1),S(2))); L=S(1); % length of vector wavplay(noisymusic,Fs); % Listen to noisy violin figure(1); plot(noisymusic); title('noisyviolin.wav (mono)'); [a1 b1] = butter(10,(Fs/4)/(Fs/2)); figure(2); freqz(a1,b1); cleanmusic=filter(a1,b1,noisymusic); % Low-pass filter [a2 b2] = butter(4,[3800 4200]/(Fs/2),'stop'); figure(3); freqz(a2,b2); cleanmusic = filter (a2,b2,cleanmusic); %Band-stop filter scale=max(abs(cleanmusic)); wavplay(cleanmusic/scale,Fs); figure(4); plot(cleanmusic/scale);

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29 Sept '09Comp30291 Section 179 Real time DSP often uses 'fixed point' microprocessors since they consume less power & are less expensive than 'floating point' devices. A fixed point processor deals with all numbers as integers. Numbers often restricted to a word-length of only 16 bits. Overflow can occur with disastrous consequences to sound quality. If we avoid overflow by scaling numbers to keep them small in amplitude, we lose accuracy as quantisation error is larger proportion of the value. Fixed point DSP programming can be quite a difficult task. Fixed point arithmetic

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29 Sept '09Comp30291 Section 180 When we use a PC for non real time DSP, we have floating point operations available with word-lengths that are larger than 16 bits. This makes the task much easier. When simulating fixed point real time DSP on a PC, we can restrict programs to integer arithmetic, & this is useful for testing. Can do this with MATLAB Floating point arithmetic

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29 Sept '09Comp30291 Section 181 Advantages of digital signal processing Signals are generally transmitted & stored in digital form so it makes sense to process them in digital form also. DSP systems can be designed & tested in simulation using PCs. Accuracy pre-determined by word-length & sampling rate. Reproducible: every copy of system will perform identically. Characteristics will not drift with temperature or ageing. Availability of advanced VLSI technology. Systems can be reprogrammed without changing hardware. Products can be updated via Internet. DSP systems can perform highly complex functions such as adaptive filtering speech recognition.

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29 Sept '09Comp30291 Section 182 Disadvantages of DSP Can be expensive especially for high bandwidth signals where fast analogue/digital conversion is required. Design of DSP systems can be extremely time-consuming & a highly complex and specialised activity. Power requirements for digital processing can be high in battery powered portable devices. Fixed point processing devices are less power consuming but ability to program such devices is valued & difficult skill. There is a shortage of computer science & electrical engineering graduates with the knowledge & skill required.

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29 Sept '09Comp30291 Section 183 Summary of Course Objectives 1. Significance of DSP for multi-media, storage and comms. Introduction to MATLAB 2. Fundamental concepts: linearity etc 3. Digital filters & their application to sound & images Low-pass digital filters etc. Butterworth low-pass gain response approximation. 4.FIR & IIR type digital filter design & implementation. 5.MATLAB for design, simulation & implementation of DSP. 6. Real time implementation of DSP (fixed point arith). 7.A/D conversion for DSP 8.FFT and its applications 9.Speech and music compression (CELP, MP3, etc)

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29 Sept '09Comp30291 Section 184 Background knowledge needed Basic maths ( complex numbers, Argand diagram, modulus & argument, differentiation, sum of an arithmetic series, factorisation of quadratic equation, De Moivre's Theorem, exponential function, natural logarithms). Experience with any high-level computer programming language.

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29 Sept '09Comp30291 Section 185 Recommended Book 1. S.W. Smith, "Scientist and Engineer's Guide to Digital Signal Processing " California Tech. Publishing, 2nd ed., 1999, available complete at: http://www.dspguide.com/http://www.dspguide.com/

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29 Sept '09Comp30291 Section 186 Problems 1. Why do we call continuous time signals ‘analogue’ ? 2. When could you describe a SIGNAL as being linear? 3 Give a formula for an analogue signal which is zero until t=0 and then becomes equal to a 50 Hz sine wave of amplitude 1 Volt. 4. Given that u(t) is an analogue step function, sketch the signal u(t)sin(2.5 t). 5. If u(t)sin(5 t) is sampled at intervals of T=0.1 seconds, what discrete time signal is obtained? What is the sampling frequency? 6. What is difference between a discrete time signal & a digital signal? 7.Define (i) amplification, (ii) filtering, (iii) rectification and (iv) modulation. 8. What is meant by “fixed point” & “floating point”? 9. Explain the terms “analogue filter” and “digital filter”. 10. What is meant by the term “real time” in DSP? 11. There are many applications of DSP where the digital signal being processed is not obtained by sampling an analogue signal. Give an example.

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