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6.1 Introduction
1
Ts
2W
1
Ts
2W
1
Ts
2W
1. Notation
- g (t ) : original analog signal
- Ts : sampling period
1 : sampling rate
- fs
Ts
2. Sampling
Sampling with (t)
gδ f g nT δ t nT s ; ideal sampled signal
n
gδ f fs G f nTs Gδ ( f )
let
n
where g (t ) G ( f )
Gδ f g nT exp j 2π nfT
s s
n
1
If Ts
2W
n j 2nf
G f g exp
n 2W W
G f f s G f f s G f mf s
m
m0
G f 0 for f W
Under following conditions
f s 2W
1
G f G f W f W
2W
1
n j 2nf
G f g 2W exp , W f W ; D.F.T.
2W n W
3. Reconstruction
n
Reconstruction g (t ) from g
2W
g (t ) G( f ) exp( j 2π ft )dt
n
W 1 n jπ nf
W 2W
n
g 2W exp
exp( j 2π ft )df
W
n 1 W n
g W exp[ j 2π f (t )]df
n 2W 2W 2W
n sin(2π Wt nπ )
g
n 2W ( 2π Wt nπ )
n
So, g (t ) g 2W sinc 2Wt - n
n
4.Sampling Theorem
5. Aliasing
6. Reconstruction filter
1. PAM
2. PAM signal
PAM signal
s (t ) m(nT )h(t nT )
n
s s
1, 0 t T
1
where h (t ) , t 0,t T
2
0 , otherwise
Instantaneoussampled version of m(t )
m(t ) m(nT
n
s )(t nTs )
m(t ) h(t ) m()h(t )d
m(nT )( nT )h(t )d
n
s s
m(nT )
s
( nTs ) h(t )d
m(nT
n
s ) h(t nTs )
So, s (t ) mδ (t ) h(t )
S ( f ) M ( f ) H ( f )
SKKU 7 J.D. Cho
Lecture on Communication Theory
여기서 Mδ ( f ) f s M( f
k
kf s )
S( f ) fs M( f
k
kf s ) H ( f )
3. S f 로 부터m(t ) 를 recover 하는 방법
Reconstruction Equalizer 특성
filter output
Equalizer
1 1 πf sin 1 (π fT )
H ( f ) T sinc( fT ) sin(π fT ) T
T
0.1, or 10 times over sampling
Ts
amplitude variation 0.005[0.5%]
can omit equalizer
4.PAM 특징
6.4 TDM
Total BW 는 N 배 필요
Sensitivity of PPM
Standard Pulse
Delayed Version
of sliver output
5. Noise in PPM
Vn
A sin(2π f / BT )
G( f )
2π f (1 4 f 2 / BT2 )
; g(t) 의 slope
; error
2 4Vn
2
4 K 2 E Vn2 4 K 2 N 0
K E 2 2 2 2 2 2 2
π BT A π BT A π BT A2
E Vn2 BT N 0 ( 2 BT
N0
2
)
π 2 BT TS2 A2
(SNR)o
32 N 0
TS
1 3A2
P g (t ) dt
2 2
T
T S
2
4 BT TS
3 A2
( SNR)C
4TS BT WN o
( SNR ) O π 2 BT2Ts3W
FOM
( SNR ) C 24
1
T
Assume s ; Nyquist sampling rate
2W
2
π 2 BT
FOM
199 W
3) 결론
1. PPM 과 FM 의 유사한 점
1) FOM 가 (BT/W)2 에 비례
2) Threshold Effect
1) Quantizer
index
Signal amplitude m k if k : mk m mk 1
k = 1,2,…..,L
Midthread
Uniform Quantizer
2) Midrise
Non-uniform Quantizer
2. Quantization Noise
m v
g
r.v. M r.v. V
(continuous) (discrete)
2) p.d.f of Q
1
q
fQ (g ) 2 2
0 otherwise
Δ
σ 2
Q 2
Δ q 2 f Q q dq E Q 2
2
Δ
1 Δ2
q dq
2 2
Δ
Δ
2
12
3. Bit 할당 및 SNR
L 2 R , R log 2 L
2mmax
Δ
2R
1 2 1 2
σ Q
2
Δ m max 2 2 R
12 3
P 3P 2 R
(SNR)o 2
2
2
σ Q mmax
Am2
P mmax Am
2
Am2 2 R
3 2
3
(SNR)O 2 2 (2 2 R )
Am 2
32 5 31.8
64 6 37.8
128 7 43.8
256 8 49.8
12 73.8
16 97.8
6.8 PCM
1. PCM 구조
1) Linear Quantizer
2) Nonlinear Quantizer
Voice signal 에서 사용하는 이유 :
loud talk : weak talk = 100 : 1
weak talk small step-size
loud talk large step-size
1 3 7 15 31
- = 0 Uniform Quantizer
dm log(1 μ )
(1 μ m )
dv μ
dv 1 μ
dm
log(1 μ ) 1 μ m
μ m 1 linear
m 1 logarithmic
A - law
Am 1
, 0 m
1 log A A
v
1 log( A m ) 1
, m 1
1 log A A
- A = 1 Uniform Quantizer
- Practical value of A A 100
- Reciprocal slope
1 log A 1
dm , 0 m
A A
dv 1
(1 log A) m , m 1
A
4. Encoding
Differential Encoding
1 : no transition
0 : transition
8. Multiplexing ; TDM
9. Synchronization
Example 3. T1 System
64
128
256
512
1024
2048
4096
MSB = 1 if +
8bit word MSB = 0 if -
Next 3 bits : segments
LSB 4 bits : amplitude within a segment
S/N db (RMS)
SKKU 32 J.D. Cho
Lecture on Communication Theory
1E+01
1E+00
1E-01
Probability of Error
16-VSB
1E-02
Symbol Error Rate
1E-03
1E-04 Segment Error
Rate After Reed-
1E-05 Solomon FEC
1E-06
1E-07
1E-08
8 12 16 20 24 28 32 36
S/N db (RMS)
2. 단점
data compression
1) Increased channel BW
wide band : satellite,optic fiber
2) Increased system complexity VLSI or Delta modulation
1. DM System
Oversampling to increase the correlation between adjacent
samples.
Staircase approximation ±
2. DM Quantization error
2) Granular Noise
dm(t )
max
Ts dt
하게 결정
2. DPCM System
eq (( n 1)Ts
Ts
mq ( n 1)Ts
mq (nTs )
m( nTs ) q ( nTs )
3. Processing Gain
2
σ
M
( SNR )O
2
σ
Q
2 2
( SNR )O E Gp ( SNR )Q
M
2 2
E Q
2
where : Variance of prediction error
E
( SNR)Q : signal to quantization noise ratio
- Processing Gain ; Predictor 를 사용해서 생기는 Gain
2
M
Gp
2
E
1. Design Philosophy
2. 오디오 압축의 원리
Psychoacoustic model
1KHz sinewave masker
70
60
Threshold in quiet
Sound pressure level (dB)
50 Masking threshold
40
30
20
10
0
frequency
0.02 0.05 0.1 0.2 0.5 1 2 5 10 20KHz
- Masking curve 이하는 감지 불능 .
- 큰 소리는 주변의 작은 소리를 Mask 함 .
where : constant
ˆ M (nTs ) : estimate of M (nTs )
Adaptive quantization with forward estimation (AQF)
unquantized samples of the input signal
forward estimates of M ( nTs )
Buffer 필요 : (nTs) 를 정할 동안 unquantized sample 저장
TX of level information
Delay in encoding(16msec) AQB 를 더 많이 사용
Buffer 없음 , delay 없음
2) Adaptive Prediction
time-varying