CSN08704
Data, Audio, Video and Imageshttp://asecuritysite.com/comms
Telecommunications
Prof Bill Buchanan
Audio and Speech
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.
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
001010011100101101110111
3 bits -> 8 levelsN bits -> 2N levels N2
scale Full21 =error Max
+ ADC 111 010 110 000
Clock (Twice highest frequency of signal)
Samples
Dynamic Range
minmaxrange Dynamic
VV
12levels ofNumber n
dB )12log(20 12
log20range Dynamicmax
max
nnV
V
if 2n is much greater that 1, then dB 02.6 2log20 2log20range Dynamic nnn
Dynamic Range
Number of bits DR (dB) [ratio] Number of bits DR (dB) [ratio] 1 6.02 [2] 11 66.23 [2 048] 2 12.04 [4] 12 72.25 [4 096] 3 18.06 [8] 13 78.27 [8 192] 4 24.08 [16] 14 84.29 [16 384] 5 30.10 [32] 15 90.31 [32 768] 6 36.12 [64] 16 96.33 [65 536] 7 42.14 [128] 17 102.35 [131 072] 8 48.16 [256] 18 108.37 [262 144] 9 54.19 [512] 19 114.39 [524 288] 10 60.21 [1 024] 20 120.41 [1 048 576]
Signal-to-Noise Ratio
dB 6.02+1.76=SNR nNumber of bits SNR (dB) [ratio] Number of bits SNR (dB) [ratio] 7 43.90 [156.68] 14 86.04 [20 044.72] 8 49.92 [313.33] 15 92.06 [40 086.67] 9 55.94 [626.61] 16 98.08 [80 167.81] 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 [10 023.05] 20 122.16 [1 282 331]
Link
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
InputOutputSample
and hold
Up/downcounter
1111111000100011000010101
AnalogueSignal
Decoded output
Code:1111111000100011000010101
Analoguesignal
DAC output
PCM
Slope overload
Input signalPCM
Reconstructed signal
ADM and DPCM• Adaptive Delta Modulation.
Change bit change to keep up with slope.
• Differential PCM. Quantise within the maximum change in level.
Analoguesignal
m levels
n levels
coding region
Currentsample
Nextsample
Input
n-bit bus
DifferentialPCM
DifferentialPCM
Analogueoutput
Low-passfilter
+
-
DAC Clockdelay
ADC
Low-passfilter
Sample andhold
+
-DAC
CSN08704
Data, Audio, Video and Imageshttp://asecuritysite.com/comms
Telecommunications
Prof Bill Buchanan
Speech Encoding
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.
Softspeech
Loudspeech
Quantization noise
Quantization noise
Quantization noise less noticeable because signalstrength swamps the quantizationnoise
Quantization noise noticeable
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
111111111111
11111111
00000000
Input code
Output code
Low-passfilter Sampler 12-bit
ADC Compander
8 kHz
Low-passfilter
12-bitDAC Expander
64 kbps
Input
Output
12-bitsamples
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(
xxxy
11for
10for
log1)log(1
log1x
A
Ax
AAxA
Ax
y
Piecewise Linear Companding
1615.5
3247.5
6411.5
2049 A-Law4079.5 -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
Audio Encoding StandardsITU 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
Time Division MultiplexingBits per time slot = 8 Number of time slots = 32 Time for frame = 125s
kbps 204810125832
Timebits of NorateBit 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 alignmentTime slot 16 - Signalling information
125 s
CSN08704
Data, Audio, Video and Imageshttp://asecuritysite.com/comms
Telecommunications
Prof Bill Buchanan
Audio and Speech