SIST ETS 300 726 E2:2003

Digital cellular telecommunications system (Phase 2+) (GSM); Enhanced Full Rate (EFR) speech transcoding (GSM 06.60 version 5.2.1)

SUBJECT Corrections to GSM 06.60

Digitalni celični telekomunikacijski sistem (faza 2+) – Prekodiranje izboljšanega govora s polno hitrostjo (EFR) (GSM 06.60, različica 5.2.1)

General Information

• Status: Published
• Publication Date: 30-Nov-2003
• ICS: 33.070.50 - Global System for Mobile Communication (GSM)
• Technical Committee: SS EIT - Technical Board for the electrotechnical, information technology and telecommunications fields
• Current Stage: 6060 - National Implementation/Publication (Adopted Project)
• Start Date: 01-Dec-2003
• Due Date: 01-Dec-2003
• Completion Date: 01-Dec-2003
• Mandate: T-172
• Ref Project: ETSI ETS 300 726 ed.2 (1999-07) - Digital cellular telecommunications system (Phase 2+) (GSM); Enhanced Full Rate (EFR) speech transcoding (GSM 06.60 version 5.2.1)

Standards Content (Sample)

SLOVENSKI STANDARD
SIST ETS 300 726 E2:2003
01-december-2003
'LJLWDOQLFHOLþQLWHOHNRPXQLNDFLMVNLVLVWHPID]D±3UHNRGLUDQMHL]EROMãDQHJD
JRYRUDVSROQRKLWURVWMR()5*60UD]OLþLFD
Digital cellular telecommunications system (Phase 2+) (GSM); Enhanced Full Rate
(EFR) speech transcoding (GSM 06.60 version 5.2.1)
Ta slovenski standard je istoveten z: ETS 300 726 Edition 2
ICS:
33.070.50 Globalni sistem za mobilno Global System for Mobile
telekomunikacijo (GSM) Communication (GSM)
SIST ETS 300 726 E2:2003 en
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.

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SIST ETS 300 726 E2:2003

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SIST ETS 300 726 E2:2003
EUROPEAN ETS 300 726
TELECOMMUNICATION July 1999
STANDARD Second Edition
Source: SMG Reference: RE/SMG-110660QR1
ICS: 33.020
Key words: EFR, digital cellular telecommunications system, Global System for Mobile communications
(GSM), speech
R
GLOBAL SYSTEM FOR
MOBILE COMMUNICATIONS
Digital cellular telecommunications system (Phase 2+);
Enhanced Full Rate (EFR) speech transcoding
(GSM 06.60 version 5.2.1)
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: [e-mail] - http://www.etsi.fr - http://www.etsi.org
Tel.: [phone] - Fax: [phone]
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 1999. All rights reserved.

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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 Standards Making Support Dept." at the address shown on the title page.

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Contents
Foreword .5
1 Scope .7
2 Normative references.7
3 Definitions, symbols and abbreviations .8
3.1 Definitions .8
3.2 Symbols .9
3.3 Abbreviations .15
4 Outline description.15
4.1 Functional description of audio parts .16
4.2 Preparation of speech samples .16
4.2.1 PCM format conversion.16
4.3 Principles of the GSM enhanced full rate speech encoder.17
4.4 Principles of the GSM enhanced full rate speech decoder.18
4.5 Sequence and subjective importance of encoded parameters.19
5 Functional description of the encoder .19
5.1 Pre-processing.19
5.2 Linear prediction analysis and quantization .19
5.2.1 Windowing and auto-correlation computation.19
5.2.2 Levinson-Durbin algorithm .21
5.2.3 LP to LSP conversion.21
5.2.4 LSP to LP conversion.23
5.2.5 Quantization of the LSP coefficients .24
5.2.6 Interpolation of the LSPs .25
5.3 Open-loop pitch analysis.25
5.4 Impulse response computation.26
5.5 Target signal computation .26
5.6 Adaptive codebook search .27
5.7 Algebraic codebook structure and search .28
5.8 Quantization of the fixed codebook gain.31
5.9 Memory update.32
6 Functional description of the decoder .32
6.1 Decoding and speech synthesis .32
6.2 Post-processing .34
6.2.1 Adaptive post-filtering.34
6.2.2 Up-scaling .35
7 Variables, constants and tables in the C-code of the GSM EFR codec.35
7.1 Description of the constants and variables used in the C code.36
8 Homing sequences .40
8.1 Functional description.40
8.2 Definitions .40
8.3 Encoder homing.42
8.4 Decoder homing .42
8.5 Encoder home state.42
8.6 Decoder home state .44
9 Bibliography.49
Annex A (informative): Document change history.50

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History. 51

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Foreword
This European Telecommunication Standard (ETS) has been produced by the Special Mobile Group
(SMG) of the European Telecommunications Standards Institute (ETSI).
This ETS describes the detailed mapping between input blocks of 160 speech samples in 13-bit uniform
PCM format to encoded blocks of 244 bits and from encoded blocks of 244 bits to output blocks of 160
reconstructed speech samples within the digital cellular telecommunications system.
Transposition dates
Date of adoption of this ETS: 6 November 1998
Date of latest announcement of this ETS (doa):
28 February 1999
Date of latest publication of new National Standard
or endorsement of this ETS (dop/e): 31 August 1999
Date of withdrawal of any conflicting National Standard (dow): 31 August 1999

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1 Scope
This European Telecommunication Standard (ETS) describes the detailed mapping between input blocks
of 160 speech samples in 13-bit uniform PCM format to encoded blocks of 244 bits and from encoded
blocks of 244 bits to output blocks of 160 reconstructed speech samples. The sampling rate is
8 000 sample/s leading to a bit rate for the encoded bit stream of 12,2 kbit/s. The coding scheme is the
so-called Algebraic Code Excited Linear Prediction Coder, hereafter referred to as ACELP.
This ETS also specifies the conversion between A-law PCM and 13-bit uniform PCM. Performance
requirements for the audio input and output parts are included only to the extent that they affect the
transcoder performance. This part also describes the codec down to the bit level, thus enabling the
verification of compliance to the part to a high degree of confidence by use of a set of digital test
sequences. These test sequences are described in GSM 06.54 [7] and are available on disks.
In case of discrepancy between the requirements described in this ETS and the fixed point computational
description (ANSI-C code) of these requirements contained in GSM 06.53 [6], the description in
GSM 06.53 [6] will prevail.
The transcoding procedure specified in this ETS is applicable for the enhanced full rate speech traffic
channel (TCH) in the GSM system.
In GSM 06.51 [5], a reference configuration for the speech transmission chain of the GSM enhanced full
rate (EFR) system is shown. According to this reference configuration, the speech encoder takes its input
as a 13-bit uniform PCM signal either from the audio part of the Mobile Station or on the network side,
from the PSTN via an 8-bit/A-law to 13-bit uniform PCM conversion. The encoded speech at the output of
the speech encoder is delivered to a channel encoder unit which is specified in GSM 05.03 [3]. In the
receive direction, the inverse operations take place.
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 01.04 (ETR 100): "Digital cellular telecommunications system (Phase 2);
Abbreviations and acronyms".
[2] GSM 03.50 (ETS 300 540): "Digital cellular telecommunications system
(Phase 2); Transmission planning aspects of the speech service in the GSM
Public Land Mobile Network (PLMN) system".
[3] GSM 05.03 (ETS 300 575): "Digital cellular telecommunications system
(Phase 2); Channel coding".
[4] GSM 06.32 (ETS 300 580-6): "Digital cellular telecommunications system
(Phase 2); Voice Activity Detection (VAD)".
[5] GSM 06.51 (ETS 300 723): "Digital cellular telecommunications system;
Enhanced Full Rate (EFR) speech processing functions General description".
[6] GSM 06.53 (ETS 300 724): "Digital cellular telecommunications system; ANSI-C
code for the GSM Enhanced Full Rate (EFR) speech codec".
[7] GSM 06.54 (ETS 300 725): "Digital cellular telecommunications system; Test
vectors for the GSM Enhanced Full Rate (EFR) speech codec".
[8] ITU-T Recommendation G.711 (1988): "Coding of analogue signals by pulse
code modulation Pulse code modulation (PCM) of voice frequencies".

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[9] ITU-T Recommendation G.726: "40, 32, 24, 16 kbit/s adaptive differential pulse
code modulation (ADPCM)".
3 Definitions, symbols and abbreviations
3.1 Definitions
For the purposes of this ETS, the following definitions apply:
adaptive codebook: The adaptive codebook contains excitation vectors that are adapted for every
subframe. 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 postfilter: This filter is applied to the output of the short term synthesis filter to enhance the
perceptual quality of the reconstructed speech. In the GSM enhanced full rate
codec, the adaptive postfilter is a cascade of two filters: a formant postfilter and
a tilt compensation filter.
algebraic codebook: A fixed codebook where algebraic code is used to populate the excitation
vectors (innovation vectors).The excitation contains a small number of nonzero
pulses with predefined interlaced sets of positions.
closed-loop pitch analysis: This is the adaptive codebook search, i.e., a process of estimating the pitch
(lag) value from the weighted input speech and the long term filter state. In the
closed-loop search, the lag is searched using error minimization loop
(analysis-by-synthesis). In the GSM enhanced full rate codec, closed-loop pitch
search is performed for every subframe.
direct form coefficients: One of the formats for storing the short term filter parameters. In the GSM
enhanced full rate codec, all filters which are used to modify speech samples
use direct form coefficients.
fixed codebook: The fixed codebook contains excitation vectors for speech synthesis filters. The
contents of the codebook are non-adaptive (i.e., fixed). In the GSM enhanced
full rate codec, the fixed codebook is implemented using an algebraic codebook.
fractional lags: A set of lag values having sub-sample resolution. In the GSM enhanced full rate
codec a sub-sample resolution of 1/6th of a sample is used.
frame: A time interval equal to 20 ms (160 samples at an 8 kHz sampling rate).
integer lags: A set of lag values having whole sample resolution.
interpolating filter: An FIR filter used to produce an estimate of sub-sample resolution samples,
given an input sampled with integer sample resolution.
inverse filter: This filter removes the short term correlation from the speech signal. The filter
models an inverse frequency response of the vocal tract.
lag: The long term filter delay. This is typically the true pitch period, or a multiple or
sub-multiple of it.
Line Spectral Frequencies: (see Line Spectral Pair)
Line Spectral Pair: Transformation of LPC parameters. Line Spectral Pairs are obtained by
decomposing the inverse filter transfer function A(z) to a set of two transfer
functions, one having even symmetry and the other having odd symmetry. The
Line Spectral Pairs (also called as Line Spectral Frequencies) are the roots of
these polynomials on the z-unit circle).

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LP analysis window: For each frame, the short term filter coefficients are computed using the high
pass filtered speech samples within the analysis window. In the GSM enhanced
full rate codec, the length of the analysis window is 240 samples. For each
frame, two asymmetric windows are used to generate two sets of LP
coefficients. No samples of the future frames are used (no lookahead).
LP coefficients: Linear Prediction (LP) coefficients (also referred as Linear Predictive Coding
(LPC) coefficients) is a generic descriptive term for describing the short term
filter coefficients.
open-loop pitch search:A process of estimating the near optimal lag directly from the weighted speech
input. This is done to simplify the pitch analysis and confine the closed-loop
pitch search to a small number of lags around the open-loop estimated lags. In
the GSM enhanced full rate codec, open-loop pitch search is performed every
10 ms.
residual: The output signal resulting from an inverse filtering operation.
short term synthesis filter: This filter introduces, into the excitation signal, short term correlation which
models the impulse response of the vocal tract.
perceptual weighting filter: This filter is employed in the analysis-by-synthesis search of the codebooks.
The filter exploits the noise masking properties of the formants (vocal tract
resonances) by weighting the error less in regions near the formant frequencies
and more in regions away from them.
subframe: A time interval equal to 5 ms (40 samples at an 8 kHz sampling rate).
vector quantization: A method of grouping several parameters into a vector and quantizing them
simultaneously.
zero input response: The output of a filter due to 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:
Az The inverse filter with unquantized coefficients
()
�
Az The inverse filter with quantified coefficients
()
1
Hz = The speech synthesis filter with quantified coefficients
()
�
Az
()
a The unquantized linear prediction parameters (direct form coefficients)
i
�
a The quantified linear prediction parameters
i
m
The order of the LP model
1
The long-term synthesis filter
Bz()

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Wz The perceptual weighting filter (unquantized coefficients)
()
gg, The perceptual weighting factors
12
Fz() Adaptive pre-filter
E
T The nearest integer pitch lag to the closed-loop fractional pitch lag of the
subframe
bThe adaptive pre-filter coefficient (the quantified pitch gain)
�
Az(/g)
n
Hz() = The formant postfilter
f
�
Az(/g)
d
gControl coefficient for the amount of the formant post-filtering
n
gControl coefficient for the amount of the formant post-filtering
d
Hz() Tilt compensation filter
t
gControl coefficient for the amount of the tilt compensation filtering
t
mg= k ' A tilt factor, with k ' being the first reflection coefficient
t 1 1
hn() The truncated impulse response of the formant postfilter
f
L The length of hn()
h f
ri() The auto-correlations of hn()
h f
�
Az(/g) The inverse filter (numerator) part of the formant postfilter
n
�
1 /Az(/g) The synthesis filter (denominator) part of the formant postfilter
d
�
�
rn() The residual signal of the inverse filter Az(/g)
n
hz() Impulse response of the tilt compensation filter
t
b()n The AGC-controlled gain scaling factor of the adaptive postfilter
sc
The AGC factor of the adaptive postfiltera
Hz Pre-processing high-pass filter
()
h1
wn , wn LP analysis windows
() ()
I II
()I
L wn()
1 I
Length of the first part of the LP analysis window

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()I
L wn()
2 I
Length of the second part of the LP analysis window
()II
L wn()
1 II
Length of the first part of the LP analysis window
()II
L ()
wn
2
Length of the second part of the LP analysis window II
rk() The auto-correlations of the windowed speech sn'( )
ac
wi() Lag window for the auto-correlations (60 Hz bandwidth expansion)
lag
f
0
The bandwidth expansion in Hz
f The sampling frequency in Hz
s
rk'()
ac
The modified (bandwidth expanded) auto-correlations
Ei() The prediction error in the ith iteration of the Levinson algorithm
LD
k The ith reflection coefficient
i
()i
a The jth direct form coefficient in the ith iteration of the Levinson algorithm
j
'
Fz() Symmetric LSF polynomial
1
'
Fz()
2
Antisymmetric LSF polynomial
¢
Fz() Polynomial Fz() with root z =- 1 eliminated
1 1
¢
Fz() Fz() with root z = 1 eliminated
2 2
Polynomial
q
i
The line spectral pairs (LSPs) in the cosine domain
q An LSP vector in the cosine domain
()n
�
q The quantified LSP vector at the ith subframe of the frame n
i
w
i
The line spectral frequencies (LSFs)
Tx() A mth order Chebyshev polynomial
m
fi(),f (i) The coefficients of the polynomials Fz() and Fz()
12 1 2
''¢¢
fi(),f (i) The coefficients of the polynomials Fz() and Fz()
12 1 2
fi() The coefficients of either Fz() or Fz()
1 2

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Cx() Sum polynomial of the Chebyshev polynomials
x Cosine of angular frequency w
lRecursion coefficients for the Chebyshev polynomial evaluation
k
f The line spectral frequencies (LSFs) in Hz
i
t
f = ff �f The vector representation of the LSFs in Hz
[]
12 10
()1 ()2
z ()n , z ()n The mean-removed LSF vectors at frame n
()1 ()2
r ()n , r ()n The LSF prediction residual vectors at frame n
()
p n The predicted LSF vector at frame n
2
()
�
r ()-n 1 The quantified second residual vector at the past frame
k
�
f The quantified LSF vector at quantization index k
E The LSP quantization error
LSP
wi,,=11�,0, LSP-quantization weighting factors
i
d The distance between the line spectral frequencies f and f
i i+1 i-1
hn() The impulse response of the weighted synthesis filter
O The correlation maximum of open-loop pitch analysis at delay k
k
Oi,,=13�, The correlation maxima at delays ti,,=13�,
t i
i
Mt,,i =13,�, The normalized correlation maxima M and the corresponding delays
()
ii i
ti,,=13�,
i
Az(/g)
1
Hz()W()z = The weighted synthesis filter
�
Az()A(z/g)
2
Az(/g) The numerator of the perceptual weighting filter
1
1/(Az/g) The denominator of the perceptual weighting filter
2
T The nearest integer to the fractional pitch lag of the previous (1st or 3rd)
1
subframe
sn'( ) The windowed speech signal
sn() The weighted speech signal
w

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�
sn() Reconstructed speech signal
�¢sn() The gain-scaled post-filtered signal
�
sn() Post-filtered speech signal (before scaling)
f
xn() The target signal for adaptive codebook search
t
xn() x The target signal for algebraic codebook search
2 2
,
res ()n The LP residual signal
LP
cn() The fixed codebook vector
vn() The adaptive codebook vector
yn()=v(n)*h(n) The filtered adaptive codebook vector
yn() The past filtered excitation
k
un() The excitation signal
�
un() The emphasized adaptive codebook vector
�
un'( ) The gain-scaled emphasized excitation signal
T The best open-loop lag
op
t Minimum lag search value
min
t Maximum lag search value
max
Rk() Correlation term to be maximized in the adaptive codebook search
b The FIR filter for interpolating the normalized correlation term Rk()
24
Rk() The interpolated value of Rk() for the integer delay k and fraction t
t
b The FIR filter for interpolating the past excitation signal un() to yield the
60
adaptive codebook vector vn()
A Correlation term to be maximized in the algebraic codebook search at index k
k
C The correlation in the numerator of A at index k
k k
E The energy in the denominator of A at index k
D k
k

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t
dH= x The correlation between the target signal xn and the impulse response
()
2 2
hn , i.e., backward filtered target
()
H The lower triangular Toepliz convolution matrix with diagonal h 0 and lower
()
diagonals hh13,,� 9
() ( )
t
F= HH The matrix of correlations of hn
()
dn() The elements of the vector d
f(,ij) The elements of the symmetric matrix F
c The innovation vector
k
C The correlation in the numerator of A
k
m The position of the i th pulse
i
JThe amplitude of the i th pulse
i
N The number of pulses in the fixed codebook excitation
p
E The energy in the denominator of A
D k
res ()n The normalized long-term prediction residual
LTP
bn() The sum of the normalized dn vector and normalized long-term prediction
()
residual res ()n
LTP
sn() The sign signal for the algebraic codebook search
b
'
dn()
Sign extended backward filtered target
'
f(,ij) The modified elements of the matrix F, including sign information
t
z , zn() The fixed codebook vector convolved with hn()
En() The mean-removed innovation energy (in dB)
E The mean of the innovation energy
~
En() The predicted energy
bb b b The MA prediction coefficients
[]
12 3 4
�
Rk() The quantified prediction error at subframe k

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E The mean innovation energy
I
Rn() The prediction error of the fixed-codebook gain quantization
E The quantization error of the fixed-codebook gain quantization
Q
�
en() The states of the synthesis filter 1/Az()
en() The perceptually weighted error of the analysis-by-synthesis search
w
hThe gain scaling factor for the emphasized excitation
g The fixed-codebook gain
c
'
g The predicted fixed-codebook gain
c
�
g The quantified fixed codebook gain
c
g
The adaptive codebook gain
p
g�
The quantified adaptive codebook gain
p
' '
ggg/ g g
= A correction factor between the gain and the estimated one
gc c c
c c
�gThe optimum value for g
gc gc
gGain scaling factor
sc
3.3 Abbreviations
For the purposes of this ETS, the following abbreviations apply. Further GSM related abbreviations may
be found in GSM 01.04 [1].
ACELP Algebraic Code Excited Linear Prediction
AGC Adaptive Gain Control
CELP Code Excited Linear Prediction
FIR Finite Impulse Response
ISPP Interleaved Single-Pulse Permutation
LP Linear Prediction
LPC Linear Predictive Coding
LSF Line Spectral Frequency
LSP Line Spectral Pair
LTP Long Term Predictor (or Long Term Prediction)
MA Moving Average
4 Outline description
This ETS is structured as follows:
Section 4.1 contains a functional description of the audio parts including the A/D and D/A functions.
Section 4.2 describes the conversion between 13-bit uniform and 8-bit A-law samples. Sections 4.3 and
4.4 present a simplified description of the principles of the GSM EFR encoding and decoding process
respectively. In subclause 4.5, the sequence and subjective importance of encoded parameters are given.

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Section 5 presents the functional description of the GSM EFR encoding, whereas clause 6 describes the
decoding procedures. Section 7 describes variables, constants and tables of the C-code of the GSM EFR
codec.
4.1 Functional description of audio parts
The analogue-to-digital and digital-to-analogue conversion will in principle comprise the following
elements:
1) Analogue to uniform digital PCM
-microphone;
-input level adjustment device;
-input anti-aliasing filter;
-sample-hold device sampling at 8 kHz;
-analogue-to-uniform digital conversion to 13-bit representation.
The uniform format shall be represented in two's complement.
2) Uniform digital PCM to analogue
-conversion from 13-bit/8 kHz uniform PCM to analogue;
-a hold device;
-reconstruction filter including x/sin( x ) correction;
-output level adjustment device;
-earphone or loudspeaker.
In the terminal equipment, the A/D function may be achieved either
-by direct conversion to 13-bit uniform PCM format;
-or by conversion to 8-bit/A-law compounded format, based on a standard A-law codec/filter
according to ITU-T Recommendations G.711 [8] and G.714, followed by the 8-bit to 13-bit
conversion as specified in subclause 4.2.1.
For the D/A operation, the inverse operations take place.
In the latter case it should be noted that the specifications in ITU-T G.714 (superseded by G.712) are
concerned with PCM equipment located in the central parts of the network. When used in the terminal
equipment, this ETS does not on its own ensure sufficient out-of-band attenuation. The specification of
out-of-band signals is defined in GSM 03.50 [2] in clause 2.
4.2 Preparation of speech samples
The encoder is fed with data comprising of samples with a resolution of 13 bits left justified in a 16-bit
word. The three least significant bits are set to '0'. The decoder outputs data in the same format. Outside
the speech codec further processing must be applied if the traffic data occurs in a different representation.
4.2.1 PCM format conversion
The conversion between 8-bit A-Law compressed data and linear data with 13-bit resoluti
...