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Overview
Sampling
Speech and Audio Signal Processing
jaj
ECE554
Nikesh Bajaj
nikesh.14730@lpu.co.in
Asst. Prof. DSP, SECE
Lovely Professional University
2 By: Nikesh Bajaj
Sampling Theorem
Ni
xa t
T t nT
sin
x nT
n
a
T t nT
Note that at the original sample instances (t = nT), the
reconstructed analog signal is equal to the value of the original
analog signal. At times between the sample instances, the
signal is the weighted sum of shifted sinc functions.
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5 By: Nikesh Bajaj 6 By: Nikesh Bajaj
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9/25/2013
jaj
7 By: Nikesh Bajaj 8 By: Nikesh Bajaj
Sampling
Prove the Theorem
Fs = 2*Fc
Ba
sh
9 By: Nikesh Bajaj 10 By: Nikesh Bajaj
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8 kHz: Popular in digital telephony. Provides Sampling theorem for bandlimited signals
coverage of first three formants for most speakers and How to change the sample rate of a signal?
most sounds. How this can be implemented using time
16 kHz: Popular in speech research. domain interpolation (based on the Sampling
Sub 8 kHz Sampling: Theorem)?
How this can be implemented efficiently using
digital filters?
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11 By: Nikesh Bajaj 12 By: Nikesh Bajaj
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9/25/2013
PCM
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13 By: Nikesh Bajaj 14 By: Nikesh Bajaj
Speech Probability Density
Function
Probability density function for x(n) is the same as for xa(t) since
x(n)=xa(nt) the mean and variance are the same for both x(n) and xa(t).
Need to estimate probability density and power spectrum from speech
waveforms
probability density estimated from long term histogram of amplitudes
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Measured Speech Densities
Distribution normalized so mean
is 0 and variance is 1(x=0, x=1)
Gamma density more closely
approximates measured
distribution for speech than
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Laplacian
good approximation is of a gamma distribution of the form:
Laplacian is still a good model
and is used in analytical studies
Small amplitudes much more
Simpler approximation is Laplacian density, of the form: likely than large amplitudes by
100:1 ratio.
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jaj
quantization process: x( n)x (n) Information Rate of Coder: I=n FS= total bit rate in
bits/second
encoding process: x (n) c(n)
n=16, FS= 8 kHz => I=128 kbps
where is the (assumed fixed) quantization step size
n=8, FS= 8 kHz => I=64 kbps
Decoding is a single-stage process
n=4, FS= 8 kHz => I=32 kbps
decoding process:c(n) x(n)
Goal of waveform coding is to get the highest quality at a
if c(n)=c(n), (no errors in transmission) then x(n) =x(n) fixed value of I (kbps), or equivalently to get the lowest
x(n) x(n) coding and quantization loses information.
value of I for a fixed quality.
Since FS is fixed, need most efficient quantization methods
to minimize I.
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Quantization Basics
Assume |x(n)| Xmax (possibly )
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jaj
if |x(n)| Xmax and x(n) is a symmetric density, then
2n =2Xmax
= 2Xmax/ 2n
if we let
x(n)=x(n) + e(n)
with x(n) the unquantized speech sample, and e(n) the
quantization
- /2 e(n) /2
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25 By: Nikesh Bajaj
Quantization Noise Model Ba
quantization noise is a zero-mean, stationary white noise
process.
E[e(n)e(n+m)]=2e, m=0
= 0 otherwise
SNR for Quantization Ref:5.3.1
sh
quantization noise is uncorrelated with the input signal
E[x(n)e(n+m)]=0 m
Distribution of quantization errors is uniform over each
quantization interval
pe(e)=1/ - /2 e /2 =0, 2e = 2/12
=0 otherwise
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Not Practical
29 By: Nikesh Bajaj 30 By: Nikesh Bajaj
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9/25/2013
- Law : Companding/Exp.
Cases
y(n) =0 for x(n) =0
jaj
- Law : Example
= 40 and L=8
Ba A-Law
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33 By: Nikesh Bajaj 34 By: Nikesh Bajaj
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Delta Modulation
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jaj
Characteristics ??
LDM
Slope overload
Avoid
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Slope overload
distortion (noise)
LDM
When input is zero or constant
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9/25/2013
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Simplicity
No sync. Req.
Simple circuit cond.
Bit-Pattern
43 By: Nikesh Bajaj 44 By: Nikesh Bajaj
ADM Comparison
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PQ <=1
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9/25/2013
jaj
6dB improvement on SNR 5 dB + 6dB improvement
No single predictor can be optimal
Need of Adaptive DPCM
ADPCM
Decoder
Ba ADPCM -Feedback
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51 By: Nikesh Bajaj 52 By: Nikesh Bajaj
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ADPCM
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55 By: Nikesh Bajaj 56 By: Nikesh Bajaj
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PCM Speech(2)
Companding Example: 5-bit per sample(1-bit polarity, 2-bit segment code,
& 2-bit quantization code)
Linear
quantization
intervals
11
10
01
00
+V signal
Ba 11
PCM Speech(3)
Companding Example: 5-bit per sample(1-bit polarity, 2-bit segment code,
& 2-bit quantization code)
Linear
quantization
intervals
11
10
01
00
+V signal
Polarity: 1
Polarity: 1
11 11
Segment 10 10
01 Segment 10 10
01
00 00
codes(+) 11 codes(+) 11
01 10
01 01 10
01
00 00
11 11
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00 10
01 00 10
01
00 00
-V 00
01 +V
00
01
10 00 10 00
11 11
00 00
Polarity: 0
Polarity: 0
Narrower 01
10 01 Wider 01
10 01
intervals 11
Segment intervals 11
Segment
for smaller 00
for smaller 00
01
10 codes(-) 01
10 codes(-)
amplitude 10
11 amplitude 10
11
00 00
01 01
10 11 10 11
11 11
-V -V
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Ni
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