Professional Documents
Culture Documents
David Tipper
Associate Professor
Graduate Program of Telecommunications and
Networking
University of Pittsburgh
Telcom 2720
Slides 7
http://www.sis.pitt.edu/~dtipper/tipper.html
http://www.sis.pitt.edu/~dtipper/tipper.html
Applications
Telephony: (cellular, wired and Internet- VoIP)
Speech Storage (Automated call-centers)
High-Fidelity recordings/voice
Text-to-speech (machine generated speech)
Issues
Efficient use of bandwidth
Compress to lower bit rate per user => more users
Speech Quality
Want tollgrade or better quality in a specific transmission environment
Environment ( BER, packet lost, packet out of order, delay, etc.)
Hardware complexity
Speed (coding/decoding delay), computation requirement and power
consumption
Telcom 2720
Channel
Encoder
Modulator
Destination
Source
Decoder
Channel
Decoder
Demod
-ulator
Channel
Source
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Characteristics of Speech
Bandwidth
Time Signal
High correlation
Short term stationary
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Characteristics of Speech
Typical
Voiced
speech
Typical
Unvoiced
speech
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Digital Speech
Waveform coders
Hybrid Coders
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Mean Opinion
Score is a
subjective measure
of quality
Tradeoff in quality
vs. data rate vs.
complexity
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Analog
input
Bandpass
filter
Analog
compressor
Sampleand-hold
circuit
PAM
Analog-toDigital
converter
PCM
law compander
Transmission
medium
Analog
output
Bandpass
filter
Analog
expander
Hold
circuit
PAM
Digital-to
Analog
converter
law expander
Pulse code modulation (PCM) system with analog companding then digital
conversion
ITU G.700 standard basis for speech coding In PSTN in 60s
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Companding
-Law companding
1
0.9
15
0.8
0.7
= 255
Analog Compander
emphasizes small values,
de-emphasizes large
values in-order to
equalize SNR across
samples.
100 40
Output
0.6
0.5
0.4
0 (no compression)
0.3
0.2
0.1
0
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
Input
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Bandpass
filter
Sampleand-hold
circuit
PAM
Analog-toDigital
converter
Linear
PCM
Digital
compressor
Compressed
PCM
PCM transmitter
Transmission
medium
Analog
output
Bandpass
filter
Hold
circuit
PAM
Digital-to
Analog
converter
Linear
PCM
Digital
expander
PCM receiver
13
Low-pass
filter
Differentiator
(summer)
Encoded
difference
samples
Analog-toDigital
converter
Accumulated
signal level
Digital-to
Analog
converter
Integrator
DPCM transmitter
DPCM
input
Digital-to
Analog
converter
Hold
circuit
Integrator
Low-pass
filter
Analog
output
DPCM receiver
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Analog
speech
Bandpass
Filter 1
A/D 1
Bandpass
Filter 2
A/D 2
Bandpass
Filter 3
A/D 3
Mux
Channel
encoder
band Range
encoding
---------------------------------------1
50- 700 Hz 4 bits
2
700-2000 Hz 3 bits
3
2000-3400Hz 2 bits
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Vocoders
Vocoders (Parametric Coders)
Filter excitation
Transmitted parameters:
gain, voiced/unvoiced decision, pitch (if voiced), LPC
parameters
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Vocoders
Linear Predictive Coders (LPC)
Excitation
periodic pulse (voiced speech) or noise (unvoiced speech)
Transmitted parameters: gain, voiced/unvoiced decision, pitch (if
voiced), LPC parameters
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Vocoders
Example Tenth Order Linear Predictive Coder
Samples Voice at 8000 Hz buffer 240 samples => 30 msec
Filter Model
(M=10 is order, G is gain, z-1 unit delay, bk are filter coefficients)
H (z) =
1+
k =1
bk z
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Vocoders
LPC coders can achieve low bit rates 1.2 4.8 Kbps
Characteristics of LPC
Quality Low
Complexity Moderate
Bit Rate Low
Delay Moderate
Robustness Low
Quality of pure LPC vocoder to low for cellular telephony try to improve quality by using hybrid coders
Try to improve the quality by
refining model of speech,
improve accuracy of model
improve input to speech coder
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Vocoders
Hybrid Coders
Combine Vocoder and Waveform Coder concept
Residual LPC (RELP)
Codebook excited LPC (CELP)
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RELP Vocoder
Residual Excited LPC
improve quality of LPC by transmitting error (residue)
along with LPC parameters
residue
Buffer/
Window
ST-LP
Analysis
LP parameters
v/u decision
gain, pitch
ENCODER
s(n)
Encoded
output
LPC
Synthesis
21
10
104 kbps
13 kbps
RPE-LTP
Channel
A/D
speech
encoder
encoder
Low-pass
filter
8000 samples/s,
13 bits/sample
GSM uses Regular Pulse Excited -- Linear Predictive Coder (RPE-LPC) for speech
Basically combine DPCM concept with LPC
Information from previous samples used to predict the current sample.
The LPC coefficients, plus an encoded form of the residual (predicted actual sample = error), represent the signal.
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160 samples/
20 ms from A/D
(= 2080 bits)
RPE-LTP
speech
encoder
36 LPC bits/20 ms
9 LTP bits/5 ms
47 RPE bits/5 ms
260 bits/20 ms
to channel
encoder
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11
53 bits
3-bit
CRC
(2,1,5)
convolution
coder
78 class 2 bits
470
bits
456
bits/
Bit
inter- 20 ms
leaver = 22.8
kb/s
24
Hybrid Vocoders
Codebook Excited LPC
Problem with simple LPC is U/V
decision and pitch estimation
doesnt model transitional speech
well, and not always accurate
Codebook approach pass
speech through an analyzer to find
closest match to a set of possible
excitations (codebook)
Transmit codebook pointer +
LPC parameters
NA-TDMA standard, IS-95, 3G,
ITU G.729 standard
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12
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13
Code Book
#1
Code Book
#2
Code Book
#3
LPC Synthesis
Filter
Output
Filter Coefficients
28
Major Procedures
Absolute Category Rating
Subjects listen to samples
and rank them on an
absolute scale - result is a
mean opinion score (MOS)
Comparison Category Rating
Subjects listen to coded
samples and original uncoded sample (PCM or
analog), the two are
compared on a relative scale
result is a comparison
mean opinion score (CMOS)
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4
3
2
1
3
2
1
0
-1
-2
-3
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14
Speech coder
PCM
CT2
DECT
NA-TDMA
Waveform
ADPCM
ADPCM
Hybrid VSELPC
Bit rate
64 Kbps
32 Kbps
32 Kbps
8Kbps
MOS
4.3
4.1
4.1
3.0
GSM
QCELP
Hybrid RELPC
Hybrid CELP
13 kbps
14.4 Kbps
3.54
3.4 4.0
QCELP
LPC
ITU G.729
Hybrid CELP
Vocoder
Hybrid CELP
9.6 Kbps
2.4 Kbps
8.Kbps
3.4
2.5
3.9
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31
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