http://www.asecuritysite.com/subjects/chapter71

1. CSN08704
Data, Audio, Video and Images
http://asecuritysite.com/comms
Telecommunications
Prof Bill Buchanan
Audio and Speech

2. Nyquist Sampling
• Nyquist defined that we
can reconstruct a signal if
we sample at twice the
highest frequency.
• Speech: 4kHz – One
sample every 125 μS.
• Audio: 20kHz - One
sample every 25 μS.

3. Sampling and Quantisation
• Sample at twice the
highest frequency
of the signal.
• N bits gives 2N
levels.
• Quality defined by
SNR and Dynamic
Range.
• Max error =
+/- Full_scale/2N
000
001
010
011
100
101101
110
111
3 bits -> 8 levels
N bits -> 2N
levels N
scaleFull
2
1
=errorMax 
+ ADC 111 010 110 000
Clock (Twice
highest frequency
of signal)
Samples

4. Dynamic Range
min
max
rangeDynamic
V
V

12levelsofNumber  n
dB)12log(20
12
log20rangeDynamic
max
max


 n
n
V
V
if 2n
is much greater that 1, then
dB02.62log202log20rangeDynamic nnn


5. Dynamic Range
Number of bits DR (dB) [ratio] Number of bits DR (dB) [ratio]
1 6.02 [2] 11 66.23 [2048]
2 12.04 [4] 12 72.25 [4096]
3 18.06 [8] 13 78.27 [8192]
4 24.08 [16] 14 84.29 [16384]
5 30.10 [32] 15 90.31 [32768]
6 36.12 [64] 16 96.33 [65536]
7 42.14 [128] 17 102.35 [131072]
8 48.16 [256] 18 108.37 [262144]
9 54.19 [512] 19 114.39 [524288]
10 60.21 [1024] 20 120.41 [1048576]

6. Signal-to-Noise Ratio
dB6.02+1.76=SNR n
Number of bits SNR (dB) [ratio] Number of bits SNR (dB) [ratio]
7 43.90 [156.68] 14 86.04 [20044.72]
8 49.92 [313.33] 15 92.06 [40086.67]
9 55.94 [626.61] 16 98.08 [80167.81]
10 61.96 [1253.14] 17 104.10 [160324.5]
11 67.98 [2506.11] 18 110.12 [320626.9]
12 74.00 [5011.87] 19 116.14 [641209.6]
13 80.02 [10023.05] 20 122.16 [1282331]
Link

7. Delta Modulation
• 1 bit used to code.
• Faster sampling rate.
• Tracks signal.
• Slope overload. This occurs when
the signal changes too fast for the
modulator to keep up. It is possible
to overcome this problem by
increasing the clock frequency or
increasing the step size.
• Granular noise. This occurs when
the signal changes slowly in
amplitude. The reconstructed signal
contains a noise which is not
present at the input.
DAC
+
-
Clock
Up/Down
Input
Output
Sample
and hold
Up/down
counter
1111111000100011000010101
Analogue
Signal
Decoded output
Code:
Analogue
signal
DAC output
PCM
Slope overload
Input signal
PCM
Reconstructed signal

8. ADM and DPCM
• Adaptive Delta Modulation.
Change bit change to keep up
with slope.
• Differential PCM. Quantise
within the maximum change
in level.
Analogue
signal
m levels
n levels
coding region
Current
sample
Next
sample
Input
n-bit bus
Differential
PCM
Differential
PCM
Analogue
output
Low-pass
filter
+
-
DAC Clock
delay
ADC
Low-pass
filter
Sample and
hold
+
-
DAC

9. CSN08704
Data, Audio, Video and Images
http://asecuritysite.com/comms
Telecommunications
Prof Bill Buchanan
Speech Encoding

10. Speech Encoding
• Subjective and system
tests have found that 12-
bit coding is required to
code speech signals,
which gives 4096
quantization levels.
• Noise in speech more
noticeable on low
volumes.
Soft
speech
Loud
speech
Quantization noise
Quantization noise
Quantization noise less
noticeable because signal
strength swamps the quantization
noise
Quantization noise noticeable

11. A-Law and μ-Law Encoding
• Compander used to convert
12-bit samples into 8 bits.
• Expander used to convert 8
bits into 12-bits.
000000000000
11111111111
11111111
00000000
Input co
Output code
Low-pass
filter
Sampler 12-bit
ADC
Compander
8 kHz
Low-pass
filter
12-bit
DAC
Expander
64 kbps
Input
Output
12-bit
samples
Output
Input
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
0 0.
1
0.
2
0.
3
0.
4
0.
5
0.
6
0.
7
0.
8
0.
9
1
A=1
A=100
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
0 0.
1
0.
2
0.
3
0.
4
0.
5
0.
6
0.
7
0.
8
0.
9
1
Output
Input
=1
=50=255
0for
)1log(
)1log(



 x
x
x
y













1
1
for
1
0for
log1
)log(1
log1
x
A
A
x
A
Ax
A
Ax
y

12. Piecewise Linear Companding
16
15.5
32
47.5
64
11.5
2049 A-Law
4079.5 -Law
Input
Output
16
32
48
128
Segment 0
Segment 1
Segment 2
Segment 7
Input Companded Decoder level Decoded level
number
Step
size
0–1
…
15–16
000 0000
…
000 1111
0
…
15
0.5
…
15.5
1
16–17
…
31–32
001 0000
…
001 1111
16
…
31
16.5
…
31.5
1
32–34
…
62–64
010 0000
…
010 1111
32
…
47
33
…
63
2
64–68
…
124–128
011 0000
…
011 1111
48
…
63
66
…
126
4
128–136
…
248–256
100 0000
…
100 1111
64
…
79
132
…
252
8
256–272
…
496–512
101 0000
…
101 1111
80
…
95
264
…
504
16
512–544
…
992–1024
110 0000
…
110 1111
96
…
111
528
…
1008
32
1024–1088
…
1984–2048
111 0000
…
111 1111
112
…
127
1056
…
2016
64

13. Audio Encoding Standards
ITU standard Technology Bit rate Description
G.711 PCM 64 kbps Standard PCM
G.721 ADPCM 32 kbps Adaptive delta PCM where each
value is coded with 4 bits
G.722 SB-ADPCM 48, 56 and 64 kbps Subband ADPCM allows for higher-
quality audio signals with a sampling
rate of 16 kHz
G.728 LD-CELP 16 kbps Low-delay code excited linear
prediction for low bit rates
+ ADC Rate = 8 bits x 8
kHz= 64 kbps
8kHz
Samples

14. Time Division Multiplexing
Bits per time slot = 8
Number of time slots = 32
Time for frame = 125s
kbps2048
10125
832
Time
bitsofNo
rateBit 6



 
30
0 1 2 3 14 15
0 1 2 3 16 31
Speech 0 Speech 30
One multiframe every 2 ms
Time slot 0 - Frame word alignment
Time slot 16 - Signalling information
125 s

15. CSN08704
Data, Audio, Video and Images
http://asecuritysite.com/comms
Telecommunications
Prof Bill Buchanan
Audio and Speech