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Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 1 ESE250: Digital Audio Basics Week 5 Feb. 9, 2012 Nyquist-Shannon Theorem
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2 Course Map Numbers correspond to course weeks 2,5 6 11 13 12 Today ESE 250 – S’12 Kod & DeHon Week 5 – Nyquist-Shannon
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Where are we ? Week 2 Received signal is sampled & quantized q = PCM[ r ] Week 3 Quantized Signal is Coded c =code[ q ] Week 4 Sampled signal first transformed into frequency domain Q = DFT[ q ] Week 5 signal oversampled & low pass filtered Q = LPF[ DFT(q+n) ] Week 6 Transformed signal analyzed Using human psychoaoustic models Week 7 Acoustically Interesting signal is “perceptually coded” C = MP3[ Q] Over Sample DFT LPF DecodeProduce r(t)r(t) p(t)p(t) q + n C Perceptual Coding Store / Transmit Q + N Q Week 4 Week 6 Week 5Week 3 [Painter & Spanias. Proc.IEEE, 88(4):451–512, 2000] Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 3
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Reconstruction Acquisition Side we’ve convinced ourselves to sample, then process, then store and transmit discretely sampled values On the other end? how should the stored sound be produced? ? send strings of sampled spl levels o off to the sound card o at what rate? ? Interpolate o with what family of interpolants? Week 4: harmonic reconstruction Gets better with more terms But need more samples to compute them and error seems unpredictable This week: general reconstruction introduce further assumptions about signal o to guarantee exact finite reconstruction o with appropriate basis functions introduce another processing step to achieve those assumptions Generic Digital Signal Processor Sample r(t)r(t) q Code c Store/ Transmit DecodeProduce p(t)p(t) Week 5 – Nyquist-Shannon 4 ESE 250 – S’12 Kod & DeHon
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Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 5 Shannon’s Theorem M. Unser. Proc. IEEE, 88(4):569–587, 2000 Questions: are the real functions countable after all? what is “frequency content”? b0(t)b0(t) Hypothesis given a (sufficiently “nice”) real function, r(t) whose frequency content does not exceed M = 2 f M Conclusion There is a set of “sampling” basis functions (we’ll not discuss) B S = { …, b -2 (t), b -1 (t), b 0 (t), b 1 (t), b 2 (t), … } which can exactly reconstruct the function r(t) = … + r -2 b -2 (t) + r -1 b -1 (t) + r 0 b 0 (t) + r 1 b 1 (t) + r 2 b 2 (t) + … from samples r k = r(k T M ) taken at sampling intervals T M = / M = 1 / (2 f M ) called the “Nyquist” rate
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Trial Application to Our Setting Psychoacoustic measurements introduced in next (week 6) lecture show human audition is bandlimited at frequency A = 2 f A for f A = 22 kHz Naïve Processing Strategy sample received signal, r(t) at intervals T A = / A = 1 /2 f A = [ 44 ¢ 10 3 ] -1 sec reconstruct exactly as needed r(t) = … + r -2 b -2 (t) + r -1 b -1 (t) + r 0 b 0 (t) + r 1 b 1 (t) + r 2 b 2 (t) + … from samples r = {…, r -2, r -1, r 0, r 1, r 2, … } where r k = r(k T A ) Why bother with DFT at all?? Week 5 – Nyquist-Shannon 6 ESE 250 – S’12 Kod & DeHon
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Poking the Trial Balloon In reality we receive a “mixed” signal r(t) = q(t) + n(t) q(t) q(t) o signal component of auditory interest o has bandwidth f A = 22 kHz n(t) n(t) o noise (uninteresting information ) o has (typically) high frequency content impurities associated with transmission and recording background sounds We only get to sample the received signal, r(t), not the desired signal, q(t) !! Week 5 – Nyquist-Shannon 7 ESE 250 – S’12 Kod & DeHon
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In class experiment: assume a nominal band limit at frequency N = 2 f N for f N = 5/4 Hz [samples/sec] Question: what is the Nyquist sample rate? o T N = 1/(2 f N ) = 2/5 sec [sample/sec ] -1 In this experiment, assume that q(t) ´ 0 o signal component of auditory interest o satisfies bandwidth requirement of f N = 5/4 Hz n(t) = Cos[9/5 N t] o noise (uninteresting information ) o has “high” frequency content o i.e., it is above (by 9/5) the presumed Nyquist rate Class will sample r(t) = q(t) + n(t) over “window” - T 0 /2 < t < T 0 /2 for T 0 = 4 [sec] at the presumed Nyquist rate Questions: o what is the number of Nyquist samples? n S ¢ T N = T 0 Number of samples: n S = 2 f N T 0 = 10 o what are the Nyquist sample times? t 2 {-2,-(8/5),-(6/5),-(4/5),-(2/5),0,2/5,4/5,6/5,8/5,2} q(t)q(t) n(t)n(t) r(t) = q(t) + n(t) Bursting the Trial Balloon Week 5 – Nyquist-Shannon 8 ESE 250 – S’12 Kod & DeHon
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Bursting the Trial Balloon Class will sample r(t) = q(t) + n(t) over “window” - T 0 /2 < t < T 0 /2 for T 0 = 4 at the presumed Nyquist rate o Number of samples: n S = 2 f N T 0 = 10 o Sample instants: t 2 {-2,-(8/5),-(6/5),-(4/5),- (2/5),0,2/5,4/5,6/5,8/5,2} and get an “aliased” version of the noise! r(t) looks like a sampled version of n(t) = Cos[1/5 N t] Week 5 – Nyquist-Shannon 9 ESE 250 – S’12 Kod & DeHon
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Aliasing Widely familiar phenomenon Wikipedia aliasing article Wikipedia aliasing article “wagon wheel” effect “wagon wheel” effect Wikipedia Nyquist-Shannon Theorem Wikipedia Nyquist-Shannon Theorem “Folding:” another general manifestation of aliasing premise: r(t) = q(t) + n(t) o we expect q(t) with bandwidth less than N = 2 f N o but turns out that n(t) is some “ superharmonic residue” n(t) = cos[ m N (1+ ) t ] where 0 < <1 outcome: sample at “Nyquist rate” T N = 1/(2 f N ) o r k = r( k T N ) = q(k T N ) + n(k T N ) o where the noise n( k T N ) = cos[m N (1+ ) k T N ] = cos[mk (1+ ) ] = cos[ mk ¢ cos[mk ] - sin[ mk ¢ sin[mk ] = 1 ¢ cos[mk ] - 0 ¢ sin[mk ] = cos[ N k T N ] o “folds over” to act as if it were a low-band tone of frequency f N Week 5 – Nyquist-Shannon 10 ESE 250 – S’12 Kod & DeHon
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Anti-Aliasing Given the “mixed” signal r(t) = q(t) + n(t) q(t) – auditory signal with bandwidth f A = 22 kHz n(t) – high frequency noise We require some “anti-aliasing” pre-process that “smooths away” the noise which will otherwise appear in the samples o r = {…, r -2, r -1, r 0, r 1, r 2, … } o where r k = r(k T A ) o and T A = / A = 1 /2 f A = [ 44 ¢ 10 3 ] -1 sec Question: how to do this? Week 5 – Nyquist-Shannon 11 ESE 250 – S’12 Kod & DeHon
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Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 12 Interlude: Visual Aliasing Wolfram: Drawing a Line on Digital Display Berkeley Course Demo Typography
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Back to Shannon (new today) Hypothesis given a (sufficiently “nice”) real function, r(t) whose frequency content does not exceed M = 2 f M Conclusion There is a set of “sampling” basis functions (we’ll not show) B S = { …, b -2 (t), b -1 (t), b 0 (t), b 1 (t), b 2 (t), … } which can exactly reconstruct the function r(t) = … + r -2 b -2 (t) + r -1 b -1 (t) + r 0 b 0 (t) + r 1 b 1 (t) + r 2 b 2 (t) + … from samples r k = r(k T M ) taken at sampling intervals T M = / M = 1 /(2 f M ) called the “Nyquist” rate M. Unser. Proc. IEEE, 88(4):569–587, 2000 Questions: are the real functions countable after all? what is “frequency content”? what are the “basis functions” ? Week 5 – Nyquist-Shannon 13 ESE 250 – S’12 Kod & DeHon
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Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 14 Fourier’s Theorem (review) Hypothesis given a (sufficiently “nice”) real function, r(t) which is periodic with period T 0 r(t) = r(t + T 0 ) Conclusion the set of “harmonic” basis functions B H = { …, h -2 (t), h -1 (t), h 0 (t), h 1 (t), h 2 (t), … } at frequency 0 = 2 f 0 for f 0 = 1/T 0 defined by h k (t) = cos k 0 t (k>0) h k (t) = 1 (k>0) h k (t) = sin k 0 t (k<0) can exactly reconstruct the function r(t) = … + R -2 h -2 (t) + R -1 h -1 (t) + R 0 h 0 (t) + R 1 h 1 (t) + R 2 h 2 (t) + … M. Unser. Proc. IEEE, 88(4):569–587, 2000 Question: are the real functions countable after all?
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Clipping a Non-Periodic Function If we are willing to restrict attention to a specific observation window -T 0 /2 · t · T 0 /2 Then Fourier’s Theorem yields exact reconstruction of the periodic extension of any function Answer: apparently, “periodic” (or, more practically, finite time) real functions are countable after all! - T 0 /2T 0 /2T0T0 -T 0 - T 0 /2T 0 /2T0T0 -T 0 Week 5 – Nyquist-Shannon 15 ESE 250 – S’12 Kod & DeHon
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Fourier + Shannon Fourier: period T 0 ) exact harmonic reconstruction at frequency 0 = 2 f 0 for f 0 = 1/T 0 r(t) = … + R -2 h -2 (t) + R -1 h -1 (t) + R 0 h 0 (t) + R 1 h 1 (t) + R 2 h 2 (t) + … Shannon: bandwidth M ) exact “sampled” reconstruction at frequency M = 2 f M for f M = 1/(2T M ) r(t) = … + r -2 b -2 (t) + r -1 b -1 (t) + r -0 b 0 (t) + r 1 b 1 (t) + r 2 b 2 (t) + … clipped frequency forces finite harmonic series clipped time forces finite samples Week 5 – Nyquist-Shannon 16 ESE 250 – S’12 Kod & DeHon
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Fourier + Shannon: period T 0 & bandwidth M = 2 f M ) finite exact harmonic reconstruction with n M = Round[T 0 / T M ] samples at “Maximal Rate” T M = 1 /(2 f M ) Algebra: k 0 > M ) R -k sin k 0 t ´ R k cos k 0 t ´ 0 (otherwise high frequency) ) R -k = R k = 0 (sin & cos never identically zero) k > n M ) r (t) = R -n M h -n M (t) + … + R -2 h -2 (t) + R -1 h -1 (t) + R 0 h 0 (t) + R 1 h 1 (t) + R 2 h 2 (t) + … + R n M h n M (t) Now we’re ready to solve the anti-aliasing problem Fourier + Shannon ) Finite Series Week 5 – Nyquist-Shannon 17 ESE 250 – S’12 Kod & DeHon
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Now postulate: New signal, q(t), satisfying o period T 0 o psychoacoustic bandwidth A = 2 f A for f A = 22 kHz o bounded by maximal bandwidth A < M sampled with o n A = Round[T 0 / T A ] samples o at “Auditory Nyquist Rate” T A = 1 /(2 f A ) Fourier + Shannon: period T 0 & bandwidth A ) finite exact harmonic reconstruction q(t) = Q -n A h -n A (t) + … + Q -2 h -2 (t) + Q -1 h -1 (t) + Q 0 h 0 (t) + Q 1 h 1 (t) + Q 2 h 2 (t) + … + Q n A h n A (t) vector representation: o Q = (Q -n A, …, Q -2, Q -1, Q 0, Q 1, Q 2, …, Q n A ) Shannon + Fourier bandwidth A & period T 0 ) finite exact “sampled” reconstruction o q(t) = q -n A b -n A (t) + … + q -2 b -2 (t) + q -1 b -1 (t) + q -0 b 0 (t) + q 1 b 1 (t) + q 2 b 2 (t) + … + q n A b n A (t) vector representation: o q = (q -n A, …, q -2, q -1, q 0, q 1, q 2, …, q n A ) Question: what is the relationship between the representations frequency domain, Q time domain, q Answer: Q = DFT(q) Psychoacoustics meets Nyquist Week 5 – Nyquist-Shannon 18 ESE 250 – S’12 Kod & DeHon
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Q+N = DFT(q) + DFT(n) = DFT(q+n) Assume we receive mixed signal r(t) = q(t) + n(t) q(t) q(t) o signal component of interest o has period T 0 & bandwidth A n(t) n(t) o noise (uninteresting information ) o has (for now) only high frequency content, , i.e., satisfies: A < · M Problem: how to remove noise? Frequency Domain approach Take the received time-sampled representation, r = q + n r = (r -n M, …, r -2, r -1, r 0, r 1, r 2, …, r n M ) = (q -n M + n -n M, …, q -2 + n -2, q -1 + n -1, q 0 + n 0, q 1 + n 1, q 2 + n 2, …, q n M + n -n M ) Compute frequency domain representation o R = DFT(r) = DFT(q+n) = DFT(q)+ DFT(n) = Q + N o R = (R -n M, …, R -2, R -1, R 0, R 1, R 2, …, R n M ) = (Q -n M + N -n M, …, Q -2 + N -2, Q -1 + N -1, Q 0 + N 0, Q 1 + N 1, Q 2 + N 2, …, Q n M + N -n M ) Recall the meaning of this representation: for h k (t) = trig[k 0 t] and trig 2 {cos, sin} we have r(t) = (Q -n M + N -n M ) h -n M (t) + … + (Q -n A + N -n A ) h -n A (t) + … + (Q -n M + N -n M ) h -2 (t) + (Q -n M + N -n M ) h -1 (t) + (Q -n M + N -n M ) h 0 (t) + (Q -n M + N -n M ) h 1 (t) + (Q -n M + N -n M ) h 2 (t) + … + (Q -n A + N -n A ) h n A (t) + … + (Q -n M + N -n M ) h n M (t) Now apply assumptions about frequency content! Week 5 – Nyquist-Shannon 19 ESE 250 – S’12 Kod & DeHon
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R = DFT(r), r = DFT -1 (R ) Compute frequency domain representation R = (R -n M, …, R -2, R -1, R 0, R 1, R 2, …, R n M ) = (Q -n M + N -n M, …, Q -2 + N -2, Q -1 + N -1, Q 0 + N 0, Q 1 + N 1, Q 2 + N 2, …, Q n M + N -n M ) Now introduce assumptions about frequency content: k > n A = A / 0 ) Q k = 0 k < n A = A / 0 ) N k = 0 Interpret in entries of R = (Q -n M + N -n M, …, Q -2 + N -2, Q -1 + N -1, Q 0 + N 0, Q 1 + N 1, Q 2 + N 2, …, Q n M + N -n M ) = (0 + N -n M, …, Q n A + N -n A, …, Q -2 + 0, Q -1 + 0, Q 0 + 0, Q 1 + 0, Q 2 + 0, …, Q n A + N -n M, …, 0 + N -n M ) = (0,…, 0, Q n A …, Q -2, Q -1, Q 0, Q 1, Q 2, …, Q n A,,, 0,…, 0 ) + (N -n M,…, N -n A, 0, …, 0, 0, 0, 0, 0, …, 0, N n A,…, N n M ) Implement Algorithmically Zero out the “higher integer” entries of S = ZERO n A (R) Obtain the inverse transform, s = DFT -1 (S) Conclude that s = q Week 5 – Nyquist-Shannon 20 ESE 250 – S’12 Kod & DeHon
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Visualizing a Low Frequency Subspace Low Frequency Signal Corrupted by High frequency noise Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 21 h0(t)h0(t) h1(t)h1(t) h -1 (t)
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Oversampled Pre-Filter Anti-aliasing requires“oversampling” o n M = Round[T 0 / T M ] samples o at “Maximal Rate” T M = 1 /(2 f M ) o Where f M represents our best estimate of o the highest noise frequency but how should we model o “noise” o and its likely bandwidth? Analog pre-processing physical electro-mechanical world o traditionally piezo-electric microphone o increasingly MEMS Has its own limited bandwidth Electronics need merely sample up to the microphone “cut off” frequency Low-Pass Filter FilterTransform r Q q ADXL001 Accelerometer Pre-Sampling r Low-Pass Week 5 – Nyquist-Shannon 22 ESE 250 – S’12 Kod & DeHon
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Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 23 To Read Further Tutorial on Shannon’s Theorem M. Unser. Sampling - 50 years after shannon. Proceedings of the IEEE, 88(4):569–587, 2000
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Week 5 – Nyquist-Shannon ESE 250 – S’12 Kod & DeHon 24 ESE250: Digital Audio Basics Week 5 Feb. 9, 2012 Nyquist-Shannon Theorem
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