ELEC4620​ ​Final​ ​Exam​ ​2013

PART​ ​A:
1. ​ ​D​ ​-​ ​E,​ ​because​ ​a​ ​square​ ​wave​ ​has​ ​an​ ​infinite​ ​number​ ​of​ ​harmonics​ ​it​ ​is​ ​not​ ​band
limited​ ​and​ ​nyquist​ ​doesn’t​ ​hold.

2. ​ ​D

3. ​ ​A

4. ​ ​A​ ​-​ ​E,​ ​if​ ​you​ ​do​ ​it​ ​by​ ​hand​ ​or​ ​with​ ​matlab​ ​youll​ ​notice​ ​that​ ​its​ ​[8​ ​-2​ ​0​ ​-2]​ ​Which​ ​is​ ​A

What’s​ ​the​ ​quickest​ ​way​ ​to​ ​go​ ​about​ ​this?​ ​DFT​ ​Matrix

5. ​ ​A​ ​ ​Matlab​ ​Solution:​ ​goo.gl/Nh4PAV

It’s​ ​just​ ​DFT​ ​of​ ​[1​ ​3​ ​2​ ​3​ ​]​ ​with​ ​copies.​ ​im

6. ​ ​C​ ​-Matlab​ ​Solution:​ ​ ​goo.gl/Nh4PAV

7. ​ ​B

Lecture​ ​4​ ​slide​ ​22,​ ​[2014]

Fs AdB 10k 70
8. ​ ​C ​ ​n = Δf
· 22
= 500
· 22
= 63.63 ≈ 64

9. ​ ​E​ ​-​ ​should​ ​be​ ​10​ ​taps

Sorry,​ ​number​ ​of​ ​zeros​ ​is​ ​9.​ ​Number​ ​of​ ​taps​ ​should​ ​be​ ​N+1​ ​=​ ​10​ ​taps.
 10. ​ ​B​ ​-​ ​Low​ ​Pass

Zeros​ ​are​ ​on​ ​the​ ​circle​ ​closer​ ​to​ ​the​ ​left​ ​side.​ ​Zeros​ ​bring​ ​the​ ​response​ ​down,​ ​hence​ ​LPF

11. ​ ​A​ ​-​ ​E​ ​?​ ​Fs/2​ ​is​ ​at​ ​pi​ ​on​ ​unit​ ​circle,​ ​and​ ​there​ ​is​ ​a​ ​zero​ ​there​ ​at​ ​zero

Yeah,​ ​it​ ​should​ ​be​ ​0

12. ​ ​B​ ​ ​-​ ​Might​ ​also​ ​be​ ​D​ ​I​ ​said​ ​D.

Isn’t​ ​this​ ​method​ ​used​ ​to​ ​design​ ​linear​ ​filters?

If​ ​an​ ​FIR​ ​filter​ ​has​ ​linear​ ​response,​ ​then​ ​naturally​ ​the​ ​zeros​ ​off​ ​the​ ​unit​ ​circle​ ​must​ ​occur​ ​in
reciprocal​ ​conjugate​ ​pairs.

13. ​ ​A​ ​-​ ​[14​ ​ ​16​ ​ ​14​ ​ ​16]

Either​ ​perform​ ​a​ ​fft(DFT)​ ​and​ ​then​ ​ifft​ ​by​ ​hand,​ ​or​ ​do​ ​a​ ​linear​ ​convolution​ ​and​ ​wrap​ ​the
answer​ ​back​ ​around.

How​ ​do​ ​you​ ​wrap​ ​it​ ​back​ ​around​ ​after​ ​performing​ ​a​ ​linear​ ​convolution?

conv([1​ ​2​ ​3​ ​4],​ ​[2​ ​1​ ​2​ ​1])​ ​=​ ​[2​ ​ ​ ​ ​5​ ​ ​ ​10​ ​ ​ ​16​ ​ ​ ​12​ ​ ​ ​11​ ​ ​ ​ ​4]

for​ ​[x0​ ​x1​ ​x2​ ​x3​ ​x4​ ​x5​ ​x6]​ ​wrap​ ​the​ ​vector​ ​onto​ ​a​ ​circular​ ​plane.

[​ ​x0​ ​x1​ ​x2​ ​x3

​ ​x4​ ​x5​ ​x6​ ​]​ ​ ​sum​ ​the​ ​rows​ ​together.

 14. ​ ​B​ ​-​ ​Hann?​ ​Has​ ​good​ ​roll-off​ ​but​ ​ ​main​ ​lobe​ ​is​ ​narrower​ ​than​ ​blackman​ ​which​ ​might
be​ ​useful​ ​for​ ​two​ ​frequencies​ ​close​ ​to​ ​one​ ​another.

What​ ​about​ ​Rectangular​ ​window?​ ​my​ ​reasoning​ ​would​ ​be​ ​that​ ​since​ ​both​ ​frequency
has​ ​equal​ ​amplitude​ ​but​ ​are​ ​close​ ​to​ ​each​ ​other,​ ​rectangular​ ​window’s​ ​very​ ​narrow
mainlobe​ ​would​ ​help​ ​a​ ​lot​ ​better​ ​compared​ ​to​ ​Hann

^I​ ​agree​ ​with​ ​above.​ ​Rectangular​ ​windows​ ​have​ ​a​ ​generally​ ​narrower​ ​mainlobe​ ​and​ ​stepper
transition​ ​which​ ​would​ ​help​ ​if​ ​the​ ​freq​ ​are​ ​close.

Doesn’t​ ​the​ ​rectangular​ ​window​ ​have​ ​crap​ ​sidelobe​ ​suppression?​ ​Guess​ ​that​ ​doesn’t​ ​matter
though​ ​because​ ​the​ ​key​ ​words​ ​are​ ​“closely​ ​spaced”​ ​in​ ​this​ ​case?

Brian​ ​said​ ​to​ ​pick​ ​extremes​ ​in​ ​revision​ ​lecture​ ​and​ ​suggested​ ​rectangular.

15. ​ ​C​ ​-​ ​Hamming?​ ​The​ ​response​ ​is​ ​fairly​ ​flat​ ​in​ ​the​ ​stopband​ ​suggesting​ ​that​ ​if​ ​they​ ​are
far​ ​apart​ ​they​ ​should​ ​stick​ ​above​ ​the​ ​flat​ ​areas.

I​ ​think​ ​Blackman​ ​would​ ​be​ ​a​ ​better​ ​choice​ ​since​ ​the​ ​frequency​ ​both​ ​have​ ​high
amplitude​ ​and​ ​since​ ​both​ ​tones​ ​are​ ​further​ ​apart​ ​than​ ​Q14,​ ​Blackman​ ​very​ ​wide​ ​mainlobe
would​ ​be​ ​ok,​ ​also​ ​Blackman’s​ ​very​ ​low​ ​dB​ ​sidelobes​ ​would​ ​suppress​ ​the​ ​other​ ​tones​ ​better
too.

Are​ ​we​ ​assuming​ ​we​ ​want​ ​max​ ​side​ ​lobe​ ​suppression​ ​here​ ​then?​ ​So​ ​Blackman​ ​window??

16. ​ ​B​ ​-​ ​Hann?​ ​Same​ ​Reasoning​ ​as​ ​14.

I​ ​agree​ ​with​ ​you​ ​on​ ​this,​ ​because​ ​unlike​ ​Q14,​ ​the​ ​tone​ ​that​ ​we​ ​want​ ​is​ ​lower
amplitude​ ​compared​ ​to​ ​the​ ​other,​ ​and​ ​Hann’s​ ​great​ ​attenuation​ ​(especially​ ​the​ ​first
sidelobe)​ ​would​ ​help​ ​distinguish​ ​the​ ​first​ ​tone​ ​to​ ​the​ ​second​ ​tone

This​ ​was​ ​in​ ​the​ ​4600​ ​or​ ​3600​ ​exam​ ​papers..

The​ ​unwanted​ ​signal​ ​is​ ​close​ ​to​ ​the​ ​signal​ ​of​ ​interest​ ​and​ ​higher​ ​power..​ ​So​ ​we​ ​need​ ​large
suppression​ ​and​ ​smaller​ ​transition​ ​widths?​ ​For​ ​a​ ​signal​ ​that’s​ ​reasonably​ ​close​ ​then
Hamming​ ​seems​ ​to​ ​provide​ ​the​ ​most?​ ​Hanning​ ​does​ ​a​ ​slightly​ ​worse​ ​job​ ​initially??

I​ ​said​ ​Hamming.
 17. ​ ​ ​C​ ​?-​ ​looking​ ​at​ ​attenuation​ ​of​ ​sidelobes​ ​on​ ​the​ ​frequency​ ​response

Here​ ​do​ ​they​ ​mean​ ​40​ ​or​ ​more?​ ​Because​ ​Hamming​ ​has​ ​40dB​ ​but​ ​Blackman​ ​for​ ​example​ ​has
much​ ​more​ ​and​ ​it​ ​falls​ ​off​ ​too..

i​ ​think​ ​there​ ​is​ ​no​ ​harm​ ​designing​ ​something​ ​better.​ ​There​ ​is​ ​also​ ​no​ ​option​ ​for​ ​hamming
only.

18. ​ ​D​ ​Blackman?

Strictly​ ​speaking,​ ​the​ ​answer​ ​should​ ​be​ ​none​ ​of​ ​above​ ​because​ ​the​ ​main​ ​lobe​ ​of​ ​a​ ​Blackman
is​ ​only​ ​57dB​ ​attenuation​ ​for​ ​the​ ​first​ ​sidelobe.

19. ​ ​A​ ​-​ ​Rectangular

Rectangular​ ​has​ ​the​ ​most​ ​narrow​ ​main​ ​lobe​ ​and​ ​hence​ ​narrowest​ ​transition.

20. ​ ​C​ ​-​ ​Blackman

21. ​ ​B​ ​-​ ​A​ ​?:​ ​lecture​ ​10​ ​/​ ​27

22. ​ ​E

I​ ​think​ ​Good​ ​Thomas​ ​is​ ​the​ ​least​ ​suitable​ ​since​ ​Good​ ​Thomas​ ​needs​ ​m​ ​x​ ​n​ ​where​ ​m
and​ ​n​ ​are​ ​relatively​ ​prime,​ ​and​ ​since​ ​N​ ​=​ ​128,​ ​there​ ​is​ ​no​ ​way​ ​you​ ​can​ ​make​ ​GCD(m,n)​ ​=​ ​1,
therefore​ ​it​ ​is​ ​not​ ​possible​ ​to​ ​use​ ​Good​ ​Thomas​ ​in​ ​this​ ​case,​ ​so​ ​I​ ​think​ ​the​ ​answer​ ​is​ ​B

He​ ​mentioned​ ​in​ ​revision​ ​lecture​ ​it​ ​is​ ​indeed​ ​Good​ ​thomas

23. ​ ​C​ ​-​ ​B?​ ​Constant​ ​geometry​ ​has​ ​2​ ​arrays​ ​and​ ​ping​ ​pongs​ ​data​ ​between​ ​(Lecture​ ​10/27)

Cooley-Tukey​ ​is​ ​an​ ​in-place​ ​algorithm(​ ​with​ ​bit-reversal​ ​and​ ​twiddle​ ​factor),​ ​not​ ​sure​ ​about
Good-Thomas,​ ​Constant​ ​Geo​ ​defs​ ​need​ ​2​ ​arrays.

24. ​ ​D

25. ​ ​E​ ​-​ ​How​ ​do​ ​we​ ​recognise​ ​this​ ​in​ ​exam?

26. ​ ​B​ ​-​ ​does​ ​0+0i​ ​count​ ​as​ ​imaginary??​ ​Yes​ ​it​ ​does,​ ​since​ ​“purely​ ​imaginary”​ ​is​ ​defined​ ​as
having​ ​Real​ ​Part​ ​=​ ​0,​ ​0+0i​ ​satisfies​ ​this,​ ​therefore​ ​is​ ​purely​ ​imaginary.

Is​ ​purely​ ​real​ ​defined​ ​as​ ​having​ ​0​ ​imaginary​ ​part?​ ​or​ ​it’s​ ​real​ ​part​ ​has​ ​an​ ​absolute​ ​value
greater​ ​than​ ​0​ ​as​ ​well??

Because​ ​without​ ​defining​ ​something​ ​about​ ​the​ ​magnitudes​ ​of​ ​the​ ​real/complex​ ​part​ ​being
greater​ ​than​ ​0​ ​couldn’t​ ​0​ ​+​ ​0i​ ​be​ ​both​ ​purely​ ​real​ ​or​ ​purely​ ​imaginary..?
 27. ​ ​B​ ​-​ ​Has​ ​anyone​ ​tested​ ​this?​ ​Yes,​ ​c​ ​gives​ ​inf​ ​where as​ ​b​ ​doesn’t

28. ​ ​A​ ​-​ ​are​ ​the​ ​trailing​ ​zeros​ ​not​ ​a​ ​problem?​ ​It​ ​shouldn’t​ ​be​ ​a​ ​problem​ ​since​ ​the​ ​trailing
zeros​ ​are​ ​more​ ​like​ ​placeholder​ ​so​ ​that​ ​it​ ​ensures​ ​the​ ​answer​ ​has​ ​a​ ​length​ ​of​ ​7

But​ ​matlab​ ​gives​ ​a​ ​vector​ ​of​ ​length​ ​12?​ ​Isn’t​ ​that​ ​wrong?

ifft(fft([1​ ​2​ ​3​ ​4],7).*fft([5​ ​6​ ​7​ ​8],7))​ ​=​ ​conv([1​ ​2​ ​3​ ​4],[5​ ​6​ ​7​ ​8])
the​ ​tailing​ ​0s​ ​just​ ​mean​ ​that​ ​the​ ​coefficients​ ​are​ ​0

29. ​ ​A​ ​-​ ​Kaiser

Kaiser​ ​is​ ​love,​ ​kaiser​ ​is​ ​life.​ ​Jokes​ ​aside​ ​Kaiser​ ​is​ ​really​ ​a​ ​family​ ​of​ ​window​ ​functions,​ ​where
\beta​ ​is​ ​used​ ​as​ ​a​ ​tradeoff​ ​to​ ​determine​ ​an​ ​optimal​ ​window.​ ​\beta​ ​=​ ​0​ ​is​ ​rectangular
window,​ ​while​ ​when​ ​\beta​ ​increases​ ​mainlobe​ ​width​ ​increases​ ​and​ ​sidelobe​ ​height
decreases.

30. ​ ​C​ ​ ​Can​ ​someone​ ​elaborate​ ​on​ ​why​ ​C​ ​is​ ​the​ ​answer?

Overlap​ ​method​ ​L​ ​=​ ​M​ ​+​ ​N​ ​+​ ​1,​ ​Where​ ​L​ ​=​ ​256​ ​of​ ​hardware,​ ​N​ ​=​ ​64​ ​filter​ ​length.​ ​M​ ​is​ ​block
length​ ​and​ ​should​ ​be​ ​191​ ​though………….

264​ ​-​ ​64​ ​+​ ​1​ ​=​ ​191​ ​(PLEASE​ ​check​ ​this​ ​math)

Is​ ​the​ ​formula​ ​definitely​ ​...+1​ ​and​ ​not​ ​...-1?

Is​ ​this​ ​method​ ​in​ ​the​ ​notes?​ ​Where​ ​did​ ​you​ ​get​ ​L​ ​=​ ​M+N+1

N​ ​=​ ​L​ ​+​ ​M​ ​-​ ​1​ ​?​ ​ ​ ​so​ ​rearrange​ ​for​ ​L?

31. ​ ​D,​ ​confirmed

Answer​ ​is​ ​[1087​ ​1]

Stopband​ ​attenuations​ ​A​ ​=​ ​-20​ ​log​ ​_{10}​ ​\delta,​ ​where​ ​\delta​ ​=​ ​peak​ ​approximation​ ​error

At​ ​passband​ ​1-​ ​\delta​ ​=​ ​at​ ​1dB​ ​(ripple)

at​ ​stopband​ ​\delta​ ​=​ ​at​ ​80dB​ ​(attenuation)​ ​=

Weighting​ ​is​ ​ratio​ ​of​ ​passband​ ​to​ ​stopband​ ​=​ ​Pass/Stop​ ​=​ ​1087
δp 1−10−1/20
1 − δ p = 10−1/20 , δ s = 10−80/20 ⇒ = = 1087
δs 10−80/20

32. ​ ​D

33. ​ ​Maybe​ ​A
B​ ​-​ ​Defs​ ​True

C​ ​is​ ​false​ ​rigth???

34. ​ ​D​ ​,​ ​D​ ​is​ ​defs​ ​true.​ ​I​ ​believe​ ​(a)​ ​because​ ​finite​ ​sequences​ ​should​ ​always​ ​converge.

What​ ​about​ ​(b)?​ ​Shouldn’t​ ​an​ ​infinite​ ​sequence​ ​always​ ​diverge​ ​and​ ​hence​ ​isn’t​ ​it
false?

Answer​ ​should​ ​be​ ​A,​ ​since​ ​finite​ ​sequence​ ​always​ ​converge.​ ​ ​+1
http://courses.media.mit.edu/2012spring/mas160/zI.pdf​​ ​page​ ​7

35. ​ ​C

Analogue​ ​filtering​ ​is​ ​considerably​ ​more​ ​expensive.

36. ​ ​C,​ ​maybe​ ​should​ ​be​ ​B?

Brian​ ​mentioned​ ​it’s​ ​estimated​ ​to​ ​be​ ​5dB​ ​/bit​ ​SNR​ ​for​ ​arithmetic​ ​operations​.​ ​Agree​ ​as​ ​the
question​ ​doesn't​ ​mention​ ​anything​ ​about​ ​quantizers​ ​which​ ​you​ ​use​ ​6dB/bit​ ​for​ ​SNR.

I​ ​got​ ​B

37. D

From​ ​the​ ​lecture​ ​slides​ ​“It​ ​can​ ​add​ ​about​​ ​12​ ​dB​​ ​to​ ​the​ ​dynamic​ ​range”​ ​in​ ​reference​ ​to
dithering.​ ​Also,​ ​from​ ​google,​ ​“Digital​ ​audio​ ​at​ ​16-bit​ ​resolution​ ​has​ ​a​ ​theoretical​ ​dynamic
range​ ​of​ ​96​ ​dB​”.​ ​96​ ​+​ ​12​ ​=​ ​108​ ​(d)

Does​ ​the​ ​4x​ ​oversampling​ ​have​ ​no​ ​effect?​ ​From​ ​the​ ​notes,​ ​each​ ​time​ ​you​ ​double​ ​the
sampling​ ​rate​ ​gives​ ​+3db​ ​so​ ​shouldnt​ ​the​ ​answer​ ​be​ ​108+6​ ​=​ ​114?​ ​This​ ​3dB​ ​would​ ​refer​ ​to
the​ ​noise​ ​wouldn’t​ ​it?​ ​Not​ ​the​ ​dynamic​ ​range

38. ​ ​Maybe​ ​C​ ​(Should​ ​be​ ​E)

By​ ​Harris​ ​approx.​ ​N​ ​~​ ​872.72

If​ ​sampling​ ​is​ ​24kHz,​ ​then​ ​every​ ​second​ ​there​ ​are​ ​24k​ ​data​ ​points.​ ​If​ ​normal​ ​FIR​ ​filter,​ ​then​ ​it
is​ ​a​ ​conv​ ​operation​ ​of​ ​sample​ ​data​ ​and​ ​filter,​ ​an​ ​O(NM)​ ​operation.​ ​Hence,​ ​multiplications
~20Million.

Not​ ​close​ ​enough,​ ​has​ ​to​ ​be​ ​within​ ​10%.​ ​Or​ ​email​ ​brian.

39. ​ ​B​ ​-​ ​Fs/(PassEdge+StopEdge)​ ​seems​ ​to​ ​work​ ​here​ ​and​ ​on​ ​assignment​ ​q

Or​ ​simply​ ​moving​ ​Nyuist​ ​Sampling​ ​to​ ​between​ ​the​ ​pass​ ​edge​ ​and​ ​stop​ ​edge.​ ​So​ ​yes​ ​(B)

40. e,

I​ ​think​ ​D.​ ​If​ ​I​ ​calculate​ ​from​ ​Q38,​ ​then​ ​my​ ​answer​ ​of​ ​~20Million/40​ ​~​ ​523636
PART​ ​B:
Question​ ​1

Given​ ​the​ ​usage​ ​of​ ​CD​ ​players​ ​in​ ​high​ ​fidelity​ ​audio​ ​systems,​ ​justify​ ​why​ ​the​ ​sampling​ ​rate
of​ ​44khz​ ​may​ ​have​ ​been​ ​chosen​ ​for​ ​CDs​ ​as​ ​opposed​ ​to,​ ​say,​ ​20​ ​kHz​ ​or​ ​100kHz?

Optimised​ ​solution​ ​for​ ​both​ ​nyquist​ ​and​ ​storage​ ​space.​ ​Human​ ​hearing​ ​<20kHz,​ ​so​ ​require
sampling​ ​of​ ​>40kHz.​ ​But​ ​having​ ​100kHz​ ​would​ ​half​ ​the​ ​amount​ ​of​ ​music​ ​that​ ​could​ ​be​ ​stored
on​ ​a​ ​single​ ​disk.

I​ ​don’t​ ​think​ ​the​ ​above​ ​is​ ​what​ ​he​ ​want​ ​from​ ​this​ ​question,​ ​the​ ​sampling​ ​rate​ ​of​ ​44khz​ ​was
chosen​ ​because​ ​first​ ​of​ ​all​ ​with​ ​this​ ​sampling​ ​rate,​ ​the​ ​nyquist​ ​frequency​ ​would​ ​fall​ ​at​ ​22khz.
This​ ​is​ ​good​ ​because​ ​a​ ​cheaper​ ​analog​ ​filter​ ​would​ ​be​ ​needed​ ​(with​ ​a​ ​cutoff​ ​at​ ​20khz)​ ​since
the​ ​required​ ​filter​ ​cutoff​ ​frequency​ ​would​ ​not​ ​need​ ​to​ ​be​ ​as​ ​steep​ ​as​ ​if​ ​the​ ​sampling
frequency​ ​is​ ​at​ ​20khz.​ ​If​ ​the​ ​sampling​ ​is​ ​at​ ​40khz​ ​(nyquist​ ​of​ ​20khz),​ ​the​ ​filter​ ​required​ ​to
avoid​ ​aliasing​ ​would​ ​be​ ​very​ ​steep​ ​(ideal​ ​filter).​ ​100khz​ ​sampling​ ​frequency​ ​is​ ​obviously​ ​not
chosen​ ​because​ ​this​ ​is​ ​an​ ​overkill,​ ​more​ ​expensive,​ ​and​ ​as​ ​the​ ​previous​ ​answer​ ​said,​ ​would
half​ ​the​ ​amount​ ​of​ ​music​ ​stored.

On​ ​top​ ​of​ ​this​ ​here’s​ ​what​ ​else​ ​wiki​​ ​says:

Early​ ​digital​ ​audio​ ​was​ ​recorded​ ​to​ ​existing​ ​analog​ ​video​ ​cassette​ ​tapes,​ ​as​ ​VCRs​ ​were​ ​the​ ​only​ ​available
transports​​ ​with​ ​sufficient​ ​capacity​ ​to​ ​store​ ​meaningful​ ​lengths​ ​of​ ​audio.​ ​To​ ​enable​ ​reuse​ ​with​ ​minimal​ ​modification
of​ ​the​ ​video​ ​equipment,​ ​these​ ​ran​ ​at​ ​the​ ​same​ ​speed​ ​as​ ​video,​ ​and​ ​used​ ​much​ ​of​ ​the​ ​same​ ​circuitry.​ ​44.1​ ​kHz
was​ ​deemed​ ​the​ ​highest​ ​usable​ ​rate​ ​meeting​ ​the​ ​following​ ​criteria

● Compatible​ ​with​ ​both​ ​PAL​​ ​and​ ​NTSC​​ ​video

● Requires​ ​encoding​ ​no​ ​more​ ​than​ ​3​ ​samples​ ​per​ ​video​ ​line​ ​per​ ​audio​ ​channel

Question​ ​2

Explain​ ​the​ ​need​ ​for​ ​digital​ ​upsampling​ ​in​ ​CD​ ​players?

The​ ​data​ ​has​ ​been​ ​compressed​ ​and​ ​digitally​ ​stored.​ ​So​ ​when​ ​the​ ​audio​ ​signal​ ​needs​ ​to​ ​be
played,​ ​upsampling​ ​is​ ​used​ ​to​ ​generate​ ​a​ ​smoother​ ​signal.​ ​Such​ ​as​ ​SINC​ ​interpolation,​ ​when
the​ ​signal​ ​has​ ​been​ ​upsampled.
Question​ ​3

Explain​ ​the​ ​advantage​ ​of​ ​using​ ​dither​ ​signals​ ​in​ ​CD​ ​mastering

From​ ​Lecture​ ​slides

The​ ​dither​ ​signal​ ​improves​ ​the​ ​SNR​ ​by​ ​breaking​ ​up​ ​the​ ​spurious​ ​line​ ​spectra​ ​that​ ​may​ ​occur

The​ ​dither​ ​has​ ​the​ ​greatest​ ​effect​ ​on​ ​small​ ​signals​ ​near​ ​the​ ​quantization​ ​noise​ ​floor.​ ​It​ ​can
add​ ​about​ ​12​ ​dB​ ​to​ ​the​ ​dynamic​ ​range​ ​and​ ​signals​ ​below​ ​the​ ​noise​ ​floor​ ​can​ ​be​ ​represented.

Although​ ​a​ ​quantizer​ ​is​ ​non-linear,​ ​dither​ ​makes​ ​it​ ​behave​ ​linearly.

Question​ ​4

Explain​ ​the​ ​advantages​ ​of​ ​Sigma​ ​Delta​ ​converters​ ​for​ ​ADC​ ​and​ ​DAC?

Oversampling,​ ​to​ ​spread​ ​the​ ​images​ ​in​ ​the​ ​frequency​ ​domain​ ​and​ ​make​ ​for​ ​less​ ​reliance​ ​on
expensive​ ​analogue​ ​filtering.

It​ ​is​ ​also​ ​very​ ​cheap​ ​since​ ​it​ ​relies​ ​on​ ​mostly​ ​digital​ ​algorithm