SIST ETS 300 969 E2:2003
(Main)Digital cellular telecommunications system (Phase 2+) (GSM); Half rate speech; Half rate speech transcoding (GSM 06.20 version 5.1.1)
Digital cellular telecommunications system (Phase 2+) (GSM); Half rate speech; Half rate speech transcoding (GSM 06.20 version 5.1.1)
This European Telecommunications Standard (ETS) specifies the speech c odec to be used for the GSM half rate channel. It also specifies the t est methods to be used to verify that the codec implementation complie s with this ETS. The requirements are mandatory for the codec to be used either in GSM Mobile Stations (MS)s or Base Station Systems (BSS)s that utilize the half rate GSM speech traffic channel.
Digitalni celični telekomunikacijski sistem (faza 2+) – Govor s pol hitrosti – Prekodiranje pri govoru s pol hitrosti (GSM 06.20, različica 5.1.1)
General Information
Standards Content (Sample)
SLOVENSKI STANDARD
SIST ETS 300 969 E2:2003
01-december-2003
'LJLWDOQLFHOLþQLWHOHNRPXQLNDFLMVNLVLVWHP ID]D ±*RYRUVSROKLWURVWL±
3UHNRGLUDQMHSULJRYRUXVSROKLWURVWL *60UD]OLþLFD
Digital cellular telecommunications system (Phase 2+) (GSM); Half rate speech; Half rate
speech transcoding (GSM 06.20 version 5.1.1)
Ta slovenski standard je istoveten z: ETS 300 969 Edition 2
ICS:
33.070.50 Globalni sistem za mobilno Global System for Mobile
telekomunikacijo (GSM) Communication (GSM)
SIST ETS 300 969 E2:2003 en
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.
---------------------- Page: 1 ----------------------
SIST ETS 300 969 E2:2003
---------------------- Page: 2 ----------------------
SIST ETS 300 969 E2:2003
EUROPEAN ETS 300 969
TELECOMMUNICATION May 1998
STANDARD Second Edition
Source: SMG Reference: RE/SMG-110620QR1
ICS: 33.020
Key words: Digital cellular telecommunications system, Global System for Mobile communications (GSM),
CODEC, GSM, speech
R
GLOBAL SYSTEM FOR
MOBILE COMMUNICATIONS
Digital cellular telecommunications system (Phase 2+);
Half rate speech;
Half rate speech transcoding
(GSM 06.20 version 5.1.1)
ETSI
European Telecommunications Standards Institute
ETSI Secretariat
Postal address: F-06921 Sophia Antipolis CEDEX - FRANCE
Office address: 650 Route des Lucioles - Sophia Antipolis - Valbonne - FRANCE
Internet: secretariat@etsi.fr - http://www.etsi.fr - http://www.etsi.org
Tel.: +33 4 92 94 42 00 - Fax: +33 4 93 65 47 16
Copyright Notification: No part may be reproduced except as authorized by written permission. The copyright and the
foregoing restriction extend to reproduction in all media.
© European Telecommunications Standards Institute 1998. All rights reserved.
---------------------- Page: 3 ----------------------
SIST ETS 300 969 E2:2003
Page 2
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
Whilst every care has been taken in the preparation and publication of this document, errors in content,
typographical or otherwise, may occur. If you have comments concerning its accuracy, please write to
"ETSI Editing and Committee Support Dept." at the address shown on the title page.
---------------------- Page: 4 ----------------------
SIST ETS 300 969 E2:2003
Page 3
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
Contents
Foreword .5
1 Scope .7
2 Normative references.7
3 Definitions, symbols and abbreviations.7
3.1 Definitions .7
3.2 Symbols .9
3.3 Abbreviations .10
4 Functional description of the GSM half rate speech codec.11
4.1 GSM half rate speech encoder .11
4.1.1 High-pass filter .13
4.1.2 Segmentation .13
4.1.3 Fixed Point Lattice Technique (FLAT).14
4.1.4 Spectral quantization.15
4.1.4.1 Autocorrelation Fixed Point Lattice Technique (AFLAT) .15
4.1.5 Frame energy calculation and quantization.17
4.1.6 Soft interpolation of the spectral parameters .17
4.1.7 Spectral noise weighting filter coefficients.18
4.1.8 Long Term Predictor lag determination.19
4.1.8.1 Open loop long term search initialization.20
4.1.8.2 Open loop lag search.21
4.1.8.3 Frame lag trajectory search (Mode „ 0) .27
4.1.8.4 Voicing mode selection.29
4.1.8.5 Closed loop lag search .29
4.1.9 Harmonic noise weighting .30
4.1.10 Code search algorithm .32
4.1.10.1 Decorrelation of filtered basis vectors .33
4.1.10.2 Fast search technique .34
4.1.11 Multimode gain vector quantization.35
4.1.11.1 Coding GS and P0.36
4.2 GSM half rate speech decoder .39
4.2.1 Excitation generation.39
4.2.2 Adaptive pitch prefilter.39
4.2.3 Synthesis Filter.40
4.2.4 Adaptive spectral postfilter .40
4.2.5 Updating decoder states .42
5 Homing sequences .42
5.1 Functional description.42
5.2 Definitions .42
5.3 Encoder homing.43
5.4 Decoder homing .43
5.5 Encoder home state.43
5.6 Decoder home state .43
Annex A (normative): Codec parameter description .44
A.1 Codec parameter description .44
A.1.1 MODE .44
A.1.2 R0 .44
A.1.3 LPC1 - LPC3.45
A.1.4 LAG_1 - LAG_4 .45
A.1.5 CODEx_1 - CODEx_4 .45
---------------------- Page: 5 ----------------------
SIST ETS 300 969 E2:2003
Page 4
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
A.1.6 GSP0_1 - GSP0_4.45
A.2 Basic coder parameters. 45
Annex B (normative): Order of occurrence of the codec parameters over Abis. 46
Annex C (informative): Bibliography . 47
History. 48
---------------------- Page: 6 ----------------------
SIST ETS 300 969 E2:2003
Page 5
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
Foreword
This European Telecommunication Standard (ETS) has been produced by the Special Mobile Group
(SMG) of the European Telecommunications Standards Institute (ETSI).
This ETS specifies the speech codec to be used for the GSM half rate channel for the digital cellular
telecommunications system. This ETS is part of ETSs series covering the half rate speech traffic channels
as described below:
GSM 06.02 ETS 300 966: "Digital cellular telecommunications system (Phase 2+); Half rate
speech; Half rate speech processing functions".
GSM 06.06 ETS 300 967: "Digital cellular telecommunications system (Phase 2+); Half rate
speech; ANSI-C code for the GSM half rate speech codec".
GSM 06.07 ETS 300 968: "Digital cellular telecommunications system (Phase 2+); Half rate
speech; Test sequences for the GSM half rate speech codec".
GSM 06.20 ETS 300 969: "Digital cellular telecommunications system (Phase 2+); Half
rate speech; Half rate speech transcoding".
GSM 06.21 ETS 300 970: "Digital cellular telecommunications system (Phase 2+); Half rate
speech; Substitution and muting of lost frames for half rate speech traffic
channels".
GSM 06.22 ETS 300 971: "Digital cellular telecommunications system (Phase 2+); Half rate
speech; Comfort noise aspects for half rate speech traffic channels".
GSM 06.41 ETS 300 972: "Digital cellular telecommunications system (Phase 2+); Half rate
speech; Discontinuous Transmission (DTX) for half rate speech traffic
channels".
GSM 06.42 ETS 300 973: "Digital cellular telecommunications system (Phase 2+); Half rate
speech; Voice Activity Detector (VAD) for half rate speech traffic channels".
The contents of this ETS may be subject to continuing work within SMG and may change following formal
SMG approval. Should SMG modify the contents of this ETS, it will be resubmitted for OAP by ETSI with
an identifying change of release date and an increase in version number as follows:
Version 5.x.y
where:
y the third digit is incremented when editorial only changes have been incorporated in the
specification;
x the second digit is incremented for all other types of changes, i.e. technical enhancements,
corrections, updates, etc.
Transposition dates
Date of adoption of this ETS: 1 May 1998
Date of latest announcement of this ETS (doa): 31 August 1998
Date of latest publication of new National Standard
or endorsement of this ETS (dop/e): 28 February 1999
Date of withdrawal of any conflicting National Standard (dow): 28 February 1999
---------------------- Page: 7 ----------------------
SIST ETS 300 969 E2:2003
Page 6
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
Blank page
---------------------- Page: 8 ----------------------
SIST ETS 300 969 E2:2003
Page 7
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
1 Scope
This European Telecommunications Standard (ETS) specifies the speech codec to be used for the GSM
half rate channel. It also specifies the test methods to be used to verify that the codec implementation
complies with this ETS.
The requirements are mandatory for the codec to be used either in GSM Mobile Stations (MS)s or Base
Station Systems (BSS)s that utilize the half rate GSM speech traffic channel.
2 Normative references
This ETS incorporates by dated and undated reference, provisions from other publications. These
normative references are cited at the appropriate places in the text and the publications are listed
hereafter. For dated references, subsequent amendments to or revisions of any of these publications
apply to this ETS only when incorporated in it by amendment or revision. For undated references, the
latest edition of the publication referred to applies.
[1] GSM 06.02 (ETS 300 966): "Digital cellular telecommunications system (Phase
2+); Half rate speech; Half rate speech processing functions".
[2] GSM 06.06 (ETS 300 967): "Digital cellular telecommunications system (Phase
2+); Half rate speech; ANSI-C code for the GSM half rate speech codec".
[3] GSM 06.07 (ETS 300 968): "Digital cellular telecommunications system (Phase
2+); Half rate speech; Test sequences for the GSM half rate speech codec".
3 Definitions, symbols and abbreviations
3.1 Definitions
For the purposes of this ETS, the following definitions apply:
adaptive codebook: The adaptive codebook is derived from the long term filter state. The lag value can
be viewed as an index into the adaptive codebook.
adaptive pitch prefilter: In the GSM half rate speech decoder, this filter is applied to the excitation signal
to enhance the periodicity of the reconstructed speech. Note that this is done prior to the application of the
short term filter.
adaptive spectral postfilter: In the GSM half rate speech decoder, this filter is applied to the output of
the short term filter to enhance the perceptual quality of the reconstructed speech.
allowable lags: The set of lag values which may be coded by the GSM half rate speech encoder and
transmitted to the GSM half rate speech decoder. This set contains both integer and fractional values (see
table 3).
analysis window: For each frame, the short term filter coefficients are computed using the high pass
filtered speech samples within the analysis window. The analysis window is 170 samples in length, and is
centered about the last 100 samples in the frame.
basis vectors: A set of M, M1, or M2 vectors of length Ns used to generate the VSELP codebook
vectors. These vectors are not necessarily orthogonal.
closed loop lag search: A process of determining the near optimal lag value from the weighted input
speech and the long term filter state.
closed loop lag trajectory: For a given frame, the sequence of near optimal lag values whose elements
correspond to each of the four subframes as determined by the closed loop lag search.
codebook: A set of vectors used in a vector quantizer.
---------------------- Page: 9 ----------------------
SIST ETS 300 969 E2:2003
Page 8
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
Codeword (OR Code): An M, M1, or M2 bit symbol indicating the vector to be selected from a VSELP
codebook.
Delta (LAG) code: A four bit code indicating the change in lag value for a subframe relative to the
previous subframe's coded lag. For frames in which the long term predictor is enabled (MODE 1, 2, or 3),
the lag for subframe 1 is independently coded using eight bits, and delta codes are used for subframes 2,
3, and 4.
direct form coefficients: One of the formats for storing the short term filter parameters. All filters which
are used to modify speech samples use direct form coefficients.
fractional lags: A set of lag values having sub-sample resolution. Note that not every fractional lag value
considered in the GSM half rate speech encoder is an allowable lag value.
frame: A time interval equal to 20 ms, or 160 samples at an 8 kHz sampling rate.
harmonic noise weighting filter: This filter exploits the noise masking properties of the spectral peaks
which occur at harmonics of the pitch frequency by weighting the residual error less in regions near the
pitch harmonics and more in regions away from them. Note that this filter is only used when the long term
filter is enabled (MODE = 1, 2 or 3).
high pass filter: This filter is used to de-emphasize the low frequency components of the input speech
signal.
integer lags: A set of lag values having whole sample resolution.
interpolating filter: An FIR filter used to estimate sub-sample resolution samples, given an input sampled
with integer sample resolution.
lag: The long term filter delay. This is typically the pitch period, or a multiple or sub-multiple of it.
long term filter: This filter is used to generate the periodic component in the excitation for the current
subframe. This filter is only enabled for MODE = 1, 2 or 3.
LPC coefficients: Linear Predictive Coding (LPC) coefficients is a generic descriptive term for describing
the short term filter coefficients.
open loop lag search: A process of estimating the near optimal lag directly from the weighted speech
input. This is done to narrow the range of lag values over which the closed loop lag search shall be
performed.
open loop lag trajectory: For a given frame, the sequence of near optimal lag values whose elements
correspond to the four subframes as determined by the open loop lag search.
reflection coefficients: An alternative representation of the information contained in the short term filter
parameters.
residual: The output signal resulting from an inverse filtering operation.
short term filter: This filter introduces, into the excitation signal, short term correlation which models the
impulse response of the vocal tract.
soft interpolation: A process wherein a decision is made for each frame to use either interpolated or
uninterpolated short term filter parameters for the four subframes in that frame.
soft interpolation bit: A one bit code indicating whether or not interpolation of the short term parameters
is to be used in the current frame.
spectral noise weighting filter: This filter exploits the noise masking properties of the formants (vocal
tract resonances) by weighting the residual error less in regions near the formant frequencies and more in
regions away from them.
---------------------- Page: 10 ----------------------
SIST ETS 300 969 E2:2003
Page 9
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
subframe: A time interval equal to 5 ms, or 40 samples at an 8 kHz sampling rate.
vector quantization: A method of grouping several parameters into a vector and quantizing them
simultaneously.
GSP0 vector quantizer: The process of vector quantization, its intermediate parameters (GS and P0) for
the coding of the excitation gains b and g.
VSELP codebook: Vector-Sum Excited Linear Predictive (VSELP) codebook, used in the GSM half rate
speech coder, wherein each codebook vector is constructed as a linear combination of the fixed basis
vectors.
zero input response: The output of a filter due to all past inputs, i.e. due to the present state of the filter,
given that an input of zeros is applied.
zero state response: The output of a filter due to the present input, given that no past inputs have been
applied, i.e. given the state information in the filter is all zeroes.
3.2 Symbols
For the purposes of this ETS, the following symbols apply:
A(z) Short term spectral filter.
aThe LPC coefficients.
i
b (n) The output of the long term filter state (adaptive codebook) for lag L.
L
bThe long term filter coefficient.
C(z) Second weighting filter.
e(n) Weighted error signal
th
f (i) The coefficients of the j phase of the 10th order interpolating filter used to
j
evaluate candidate fractional lag values; i ranges from 0 to P -1.
f
th
g (i) The coefficients of the j phase of the 6th order interpolating filter used to
j
interpolate C's and G's as well as fractional lags in the harmonic noise weighting;
i ranges from 0 to Pg-1.
gThe gain applied to the vector(s) selected from the VSELP codebook(s).
H A M2 bit code indicating the vector to be selected from the second VSELP
codebook (when operating in mode 0).
I A M or M1 bit code indicating the vector to be selected from one of the two first
VSELP codebooks.
L The long term filter lag value.
L 142 (samples), the maximum possible value for the long term filter lag.
max
L 21 (samples), the minimum possible value for the long term filter lag.
min
M 9, the number of basis vectors, and the number of bits in a codeword, for the
VSELP codebook used in modes 1, 2, and 3.
M1 7, the number of basis vectors, and the number of bits in a codeword, for the
first VSELP codebook used in mode 0.
M2 7, the number of basis vectors, and the number of bits in a codeword, for the
second VSELP codebook used in mode 0.
MODE A two bit code indicating the mode for the current frame (see annex A).
N 170, the length of the analysis window. This is the number of high pass filtered
A
speech samples used to compute the short term filter parameters for each
frame.
N 160, the number of samples per frame (at a sampling rate of 8 kHz).
F
N 10, the short term filter order.
p
N 40, the number of samples per subframe (at a sampling rate of 8 kHz).
s
P1 6, the number of bits in the prequantizer for the r1 - r3 vector quantizer.
P2 5, the number of bits in the prequantizer for the r4 - r6 vector quantizer.
P3 4, the number of bits in the prequantizer for the r7 - r10 vector quantizer.
P The order of one phase of an interpolating filter used to evaluate candidate
f
fractional lag values. P equals 10 for j „ 0 and equal to 1 for j = 0.
f
P The order of one phase of an interpolating filter, f (n), used to interpolate C's and
g j
G's as well as fractional lags in the harmonic noise weighting, P equals 6.
g
---------------------- Page: 11 ----------------------
SIST ETS 300 969 E2:2003
Page 10
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
pitch The time duration between the glottal pulses which result when the vocal chords
vibrate during speech production.
Q1 11, the number of bits in the r1 - r3 reflection coefficient vector quantizer.
Q2 9, the number of bits in the r4 - r6 reflection coefficient vector quantizer.
Q3 8, the number of bits in the r7 - r10 reflection coefficient vector quantizer.
R0 A five bit code used to indicate the energy level in the current frame.
r(n) The long term filter state (the history of the excitation signal); n < 0
r (n) The long term filter state with the adaptive codebook output for lag L appended.
L
s'(n) Synthesized speech.
W(z) Spectral weighting filter.
lThe harmonic noise weighting filter coefficient.
hnw
xThe adaptive pitch prefilter coefficient.
ØxøCeiling function: the largest integer y where y < x + 1,0.
ºxßFloor function: the largest integer y where y £ x.
K
∑xi() Summation: x(j)+x(j+1)+.+x(K).
ij=
K
16
xi Product: x(j)(x(j+1)).(x(K))
∏
ij=
max(x,y) Find the larger of two numbers x and y.
min(x,y) Find the smaller of two numbers x and y.
round(x) Round the non-integer x to the closest integer yy:,=+x 05 y: y=x+0,5.
3.3 Abbreviations
For the purposes of this ETS, the following abbreviations apply:
AFLAT Autocorrelation Fixed point LAttice Technique
CELP Code Excited Linear Prediction
FLAT Fixed Point Lattice Technique
LTP Long Term Predictor
SST Spectral Smoothing Technique
VSELP Vector-Sum Excited Linear Prediction
---------------------- Page: 12 ----------------------
SIST ETS 300 969 E2:2003
Page 11
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
4 Functional description of the GSM half rate speech codec
The GSM half rate codec uses the VSELP (Vector-Sum Excited Linear Prediction) algorithm. The VSELP
algorithm is an analysis-by-synthesis coding technique and belongs to the class of speech coding
algorithms known as CELP (Code Excited Linear Prediction).
The GSM half rate codec's encoding process is performed on a 20 ms speech frame at a time. A speech
frame of the sampled speech waveform is read and based on the current waveform and the past history of
the waveform, the codec encoder derives 18 parameters that describe it. The parameters extracted are
grouped into the following three general classes:
- energy parameters (R0 and GSP0);
- spectral parameters (LPC and INT_LPC);
- excitation parameters (LAG and CODE).
These parameters are quantized into 112 bits for transmission as described in annex A and their order of
occurrence over Abis is given in annex B.
The GSM half rate codec is an analysis-by-synthesis codec, therefore the speech decoder is primarily a
subset of the speech encoder. The quantized parameters are decoded and a synthetic excitation is
generated using the energy and excitation parameters. The synthetic excitation is then filtered to provide
the spectral information resulting in the generation of the synthesized speech (see figure 1).
GSM half rate speech codec
transm itted speech
speech
param eters
Speech
encoder
synthesised received speech
speech param eters
Speech
decoder
Figure 1: Block diagram of the GSM half rate speech codec
The ANSI-C code that describes the GSM half rate speech codec is given in GSM 06.06
(ETS 300 967) [2] and the test sequences in GSM 06.07 (ETS 300 968) [3] (see clause 5 for the codec
homing test sequences).
4.1 GSM half rate speech encoder
The GSM half rate speech encoder uses an analysis by synthesis approach to determine the code to use
to represent the excitation for each subframe. The codebook search procedure consists of trying each
codevector as a possible excitation for the Code Excited Linear Predictive (CELP) synthesizer. The
synthesized speech s'(n) is compared against the input speech and a difference signal is generated. This
difference signal is then filtered by a spectral weighting filter, W(z), (and possibly a second weighting filter,
C(z)) to generate a weighted error signal, e(n). The power in e(n) is computed. The codevector which
generates the minimum weighted error power is chosen as the codevector for that subframe. The spectral
weighting filter serves to weight the error spectrum based on perceptual considerations. This weighting
filter is a function of the speech spectrum and can be expressed in terms of the a parameters of the short
term (spectral) filter.
N
p
-i
1-az
∑
i
i=1
Wz() = (1)
N
p
-i
~
1-az
∑
i
i=1
---------------------- Page: 13 ----------------------
SIST ETS 300 969 E2:2003
Page 12
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
�
The computation of the acoefficients is described in subclause 4.1.7.
i
The second weighting filter C(z), if used, is a harmonic weighting filter and is used to control the amount of
error in the harmonics of the speech signal. If the weighting filter(s) are moved to both input paths to the
subtracter, an equivalent configuration is obtained as shown in figure 2.
y(n)
Input speech s(n)
C(z)
W(z)
LPC 1
LPC 2
Determ ine LPC
LPC 3
.
coefficients
R0
IN T_LP C
y'(n)
VSELP
_
C(z)
B(z) H(z)
X
Codebook
e(n)
2
S( )
gI
bL
total weighted
error
MODE
Find M inim um
LAG-1
over L and all I
CODE_1
LAG_2
CODE_2
LAG_3
CODE_3
LAG_4
find optim al
CODE_4
b
gains andg
GSP 0-1
GSP 0_2
GSP 0_3
GSP 0_4
Figure 2: Block diagram of the GSM half rate speech encoder (MODE = 1,2 and 3)
Here H(z) is the combination of A(z), the short term (spectral) filter, and W(z), the spectral weighting filter.
These filters are combined since the denominator of A(z) is cancelled by the numerator of W(z).
1
Hz() = (2)
N
p
~-i
1-az
∑
i
i=1
---------------------- Page: 14 ----------------------
SIST ETS 300 969 E2:2003
Page 13
ETS 300 969 (GSM 06.20 version 5.1.1): May 1998
There are two approaches that can be used for calculating the gain, g. The gain can be determined prior to
codebook search based on residual energy. This gain would then be fixed for the codebook search.
Another approach is to optimize the gain for each codevector during the codebook search. The
codevector which yields the minimum weighted error would be chosen and its corresponding optimal gain
would be used for g. The latter approach generally yields better results since the gain is optimized for each
codevector. This approach also implies that the gain term needs to be updated at the subframe rate. The
optimal code and gain for this technique can be computed as follows:
The input speech is first filtered by a high pass filter as described in subclause 4.1.1. The short term filter
parameters are computed from the filtered input speech once per frame. A fast fixed point covariance
lattice technique is used. Subclauses 4.1.3 and 4.1.4 describes in detail how the short term parameters
are determined and quantized. An overall frame energy is also computed and coded once per frame.
Once per frame, one of the four voicing modes is selected. If MODE„0, the long term predictor is used
and the long term predictor lag, L, is updated at the subframe rate. L and a VSELP codeword are selected
sequentially. Each is chosen to minimize the weighted mean square error. The long-term filter coefficient,
b, and the codebook gain, g, are optimized jointly. Subclause 4.1.8 describes the technique for selecting
from among the voicing modes and, if one of voiced modes is chosen, determining the long-term filter lag.
Subclause 4.1.10 describes an efficient technique for jointly optimizing b, g and the codeword selection.
Subclause 4.1.10 also includes the description of the fast VSELP codebook search technique. The b and
g parameters are transformed to equivalent parameters using the frame energy term, and are vector
quantized every subframe. The coding of the frame energy and the b and g parameters is described in
subclause 4.1.11.
4.1.1 High-pass filter
The 13 bit linear Pulse Code Modulated (PCM) input speech, x(n), is filtered by a fourth order pole-zero
high pass filter. This filter suppresses the frequency component
...
Questions, Comments and Discussion
Ask us and Technical Secretary will try to provide an answer. You can facilitate discussion about the standard in here.