SIST ETS 300 580-2 E3:2003

Digital cellular telecommunications system (Phase 2) (GSM); Full rate speech; Part 2: Transcoding (GSM 06.10 version 4.2.1)

CRs from SMG#32

Digitalni celični telekomunikacijski sistem (faza 2) – Govor s polno hitrostjo – 2. del: Prekodiranje (GSM 06.10, različica 4.2.1)

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
• Ref Project: ETSI ETS 300 580-2 ed.3 (2000-12) - Digital cellular telecommunications system (Phase 2) (GSM); Full rate speech; Part 2: Transcoding (GSM 06.10 version 4.2.1)

Standards Content (Sample)

SLOVENSKI STANDARD
SIST ETS 300 580-2 E3:2003
01-december-2003
'LJLWDOQLFHOLþQLWHOHNRPXQLNDFLMVNLVLVWHPID]D±*RYRUVSROQRKLWURVWMR±
GHO3UHNRGLUDQMH*60UD]OLþLFD
Digital cellular telecommunications system (Phase 2) (GSM); Full rate speech; Part 2:
Transcoding (GSM 06.10 version 4.2.1)
Ta slovenski standard je istoveten z: ETS 300 580-2 Edition 3
ICS:
33.070.50 Globalni sistem za mobilno Global System for Mobile
telekomunikacijo (GSM) Communication (GSM)
SIST ETS 300 580-2 E3:2003 en
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.

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SIST ETS 300 580-2 E3:2003

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SIST ETS 300 580-2 E3:2003


EUROPEAN ETS 300 580-2
TELECOMMUNICATION December 2000
STANDARD Third Edition

Source: SMG Reference: RE/SMG-110610PR2
ICS: 33.020
Key words: Digital cellular telecommunications system, Global System for Mobile communications (GSM)
R
GLOBAL SYSTEM FOR
MOBILE COMMUNICATIONS

Digital cellular telecommunications system (Phase 2);
Full rate speech;
Part 2: Transcoding
(GSM 06.10 version 4.2.1)
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.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 2000. All rights reserved.

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Whilst every care has been taken in the preparation and publication of the present 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
Intellectual Property Rights.7
Foreword.7
1 General.9
1.1 Scope .9
1.1.2 Normative references .9
1.1.3 Abbreviations.10
1.2 Outline description .10
1.3 Functional description of audio parts .10
1.4 PCM format conversion .11
1.5 Principles of the RPE-LTP Encoder.11
1.6 Principles of the RPE-LTP Decoder .11
1.7 Sequence and subjective importance of encoded parameters.12
2 Transmission characteristics.14
2.1 Performance characteristics of the analogue/digital interfaces .14
2.2 Transcoder delay .14
3 Functional description of the RPE-LTP Codec.15
3.1 Functional description of the RPE-LTP Encoder .15
3.1.1 Offset compensation.16
3.1.2 Preemphasis.16
3.1.3 Segmentation .16
3.1.4 Autocorrelation .16
3.1.5 Schur Recursion.16
3.1.6 Transformation of reflection coefficients to Log.-Area Ratios .16
3.1.7 Quantization and coding of Log.-Area Ratios .17
3.1.8 Decoding of the quantized Log.-Area Ratios .17
3.1.9 Interpolation of Log.-Area Ratios.18
3.1.10 Transformation of Log.-Area Ratios into reflection coefficients .18
3.1.11 Short Term Analysis Filtering .18
3.1.12 Sub-segmentation .18
3.1.13 Calculation of the LTP parameters.18
3.1.14 Coding/Decoding of the LTP lags.19
3.1.15 Coding/Decoding of the LTP gains.19
3.1.16 Long term analysis filtering.20
3.1.17 Long term synthesis filtering.20
3.1.18 Weighting Filter.20
3.1.19 Adaptive sample rate decimation by RPE grid selection.21
3.1.20 APCM quantization of the selected RPE sequence.21
3.1.21 APCM inverse quantization .23
3.1.22 RPE grid positioning.23
3.2 Decoder.23
3.2.1 RPE decoding section .23
3.2.2 Long Term Prediction section.23
3.2.3 Short term synthesis filtering section.23
3.2.4 Postprocessing.23
4 Computational details of the RPE-LTP Codec .27
4.1 Data representation and arithmetic operations.27
4.2 Fixed point implementation of the RPE-LTP Coder.30
4.2.0 Scaling of the input variable .30
4.2.1 Downscaling of the input signal.31
4.2.2 Offset compensation.31
4.2.3 Preemphasis.31
4.2.4 Autocorrelation .31
4.2.5 Computation of the reflection coefficients .32
4.2.6 Transformation of reflection coefficients to Log.-Area Ratios .33

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4.2.7 Quantization and coding of the Log.-Area Ratios. 33
4.2.8 Decoding of the coded Log.-Area Ratios. 33
4.2.9 Computation of the quantized reflection coefficients. 34
4.2.9.1 Interpolation of the LARpp[1.8] to get the LARp[1.8] . 34
4.2.9.2 Computation of the rp[1.8] from the interpolated
LARp[1.8] . 34
4.2.10 Short term analysis filtering . 34
4.2.11 Calculation of the LTP parameters . 35
4.2.12 Long term analysis filtering . 36
4.2.13 Weighting filter . 36
4.2.14 RPE grid selection . 37
4.2.15 APCM quantization of the selected RPE sequence . 37
4.2.16 APCM inverse quantization . 38
4.2.17 RPE grid positioning . 38
4.2.18 Update of the reconstructed short term residual signal dp[-120.-1] . 38
4.3 Fixed point implementation of the RPE-LTP Decoder . 39
4.3.1 RPE decoding section . 39
4.3.2 Long term synthesis filtering . 39
4.3.3 Computation of the decoded reflection coefficients. 40
4.3.4 Short term synthesis filtering section. 40
4.3.5 Deemphasis filtering . 40
4.3.6 Upscaling of the output signal. 41
4.3.7 Truncation of the output variable . 41
4.4 Tables used in the fixed point implementation of the rpe-ltp coder and decoder . 41
5 Digital test sequences. 43
5.1 Input and output signals . 43
5.2 Configuration for the application of the test sequences . 43
5.2.1 Configuration 1 (encoder only) . 43
5.2.2 Configuration 2 (Decoder only). 43
5.3 Test sequences . 44
5.3.1 Test sequences for configuration 1. 44
5.3.2 Test sequences for configuration 2. 44
Annex 1 (informative): Codec Performance. 48
A1.1 Introduction. 48
A1.2 Speech performance. 48
A1.2.1 Single encoding . 48
A1.2.2 Speech performance when interconnected with coding systems on an
analogue basis. 49
A1.2.2.1 Performance with 32 kbit/s ADPCM (G.721, superseded
by G.726). 49
A1.2.2.2 Performance with another RPE-LTP codec . 49
A1.2.2.3 Performance with encoding other than RPE-LTP and 32
kbit/s ADPCM (G.721, superseded by G.726). 49
A1.3 Non-speech performance. 49
A1.3.1 Performance with single sine waves. 49
A1.3.2 Performance with DTMF tones . 50
A1.3.3 Performance with information tones . 50
A1.3.4 Performance with voice-band data . 50
A1.4 Delay . 50
A1.5 References . 52
Annex 2 (informative): Subjective relevance of the speech coder output bits . 53
Annex 3 (informative): Format for test sequence distribution . 55
A3.1 Type of files provided . 55
A3.2 File format description . 55
Annex 4 (informative): Test sequence diskette. 56
Annex 5 (informative): Change Request History. 57

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History.58

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Intellectual Property Rights
IPRs essential or potentially essential to the present document may have been declared to ETSI. The
information pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-
members, and can be found in ETSI SR 000 314: "Intellectual Property Rights (IPRs); Essential, or
potentially Essential, IPRs notified to ETSI in respect of ETSI standards", which is available from the ETSI
Secretariat. Latest updates are available on the ETSI Web server (http://www.etsi.org/ipr).
Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI.
No guarantee can be given as to the existence of other IPRs not referenced in ETSI SR 000 314 (or the
updates on the ETSI Web server) which are, or may be, or may become, essential to the present
document.
Foreword
This European Telecommunication Standard (ETS) has been produced by the Special Mobile Group
(SMG) Technical Committee of the European Telecommunications Standards Institute (ETSI).
The present document specifies the full rate speech transcoding within the digital cellular
telecommunications system.
NOTE: The present document is a reproduction of recommendation T/L/03/11 "13 kbit/s
Regular Pulse Excitation - Long Term Prediction - Linear Predictive Coder for use in
the digital cellular telecommunications system".
A 3,5 inch diskette (Annex 4 informative) is supplied with the present document, the diskette contains test
sequences, as described in clause 5.
The diskette contain files labelled as follows:
Diskette 1 ETS 300 580-2, annex 4: Test sequences for the GSM Full Rate speech codec;
Test sequences SEQ01.xxx to SEQ05.xxx.
The specification from which the present document has been derived was originally based on CEPT
documentation, hence the presentation of the present document may not be entirely in accordance with
the ETSI/PNE Rules.

Transposition dates
Date of adoption of this ETS: 1 December 2000

Date of latest announcement of this ETS (doa):
31 March 2001

Date of latest publication of new National Standard
or endorsement of this ETS (dop/e): 30 September 2001

Date of withdrawal of any conflicting National Standard (dow): 30 September 2001

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1 General
1.1 Scope
The transcoding procedure specified in the present document is applicable for the full-rate traffic channel
(TCH) in digital cellular telecommunications system. The use of this transcoding scheme for other
applications has not been considered.
In recommendation GSM 06.01, a reference configuration for the speech transmission chain of the digital
cellular telecommunications 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
Rec.GSM 05.03. In the receive direction, the inverse operations take place.
The present document describes the detailed mapping between input blocks of 160 speech samples in
13 bit uniform PCM format to encoded blocks of 260 bits and from encoded blocks of 260 bits to output
blocks of 160 reconstructed speech samples. The sampling rate is 8000 sample/s leading to an average
bit rate for the encoded bit stream of 13 kbit/s. The coding scheme is the so-called Regular Pulse
Excitation - Long Term prediction - Linear Predictive Coder, here-after referred to as RPE-LTP.
The recommendation 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. The recommendation also describes the codec down to the bit level,
thus enabling the verification of compliance to the recommendation to a high degree of confidence by use
of a set of digital test sequences. These test sequences are also described and are available in the
diskette (informative annex 4) which is attached to the back cover of the present document.
1.1.2 Normative references
The present document 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 the present document 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);
Definitions, abbreviationsand acronyms".
[2] GSM 05.03 (ETS 300 575): " Digital cellular telecommunications system
(Phase 2); Channel coding".
[3] GSM 06.01 (ETS 300 580-1): " Digital cellular telecommunications system
(Phase 2); Full rate speech processingfunctions".
[4] GSM 11.10 (ETS 300 607): " Digital cellular telecommunications system
(Phase 2); Mobile Station (MS) conformity specification".
[5] ETS 300 085: "Integrated Services Digital Network (ISDN);3,1kHz telephony
teleservice Attachment requirements for handset terminals (Candidate
NET 33)".
[6] ITU-T Recommendation G.711: "Pulse code modulation (PCM) of voice
frequencies".
[7] ITU-T Recommendation G.712: "Transmission performance characteristics of
pulse code modulation".
[8] ITU-T Recommendation G.726: "40, 32, 24, 16 kbit/s adaptive differential pulse
code modulation (ADPCM)".

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[9] ITU-T Recommendation Q.35: "Technical characteristics of tones for the
telephone service".
[10] ITU-T Recommendation V.21: "300 bits per second duplex modem
standardised for use in the general switched telephone network".
[11] ITU-T Recommendation V.23: "600/1 200-band modem standardised for use in
the general switched telephone network".
1.1.3 Abbreviations
Abbreviations used in the present document are listed in GSM 01.04.
1.2 Outline description
The specification is structured as follows:
Section 1.3 contains a functional description of the audio parts including the A/D and D/A functions.
Section 1.4 describes the conversion between 13 bit uniform and 8 bit A-law samples. Sections 1.5 and
1.6 present a simplified description of the principles of the RPE-LTP encoding and decoding process
respectively. In section 1.7, the sequence and subjective importance of encoded parameters are given.
Section 2 deals with the transmission characteristics of the audio parts that are relevant for the
performance of the RPE-LTP codec.
Some transmission characteristics of the RPE-LTP codec are also specified in section 2. Section 3
presents the functional description of the RPE-LTP coding and decoding procedures, whereas section 4
describes the computational details of the algorithm. Procedures for the verification of the correct
functioning of the RPE-LTP are described in section 5.
Performance and network aspects of the RPE-LTP codec are contained in annex 1.
1.3 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
- microphone,
- input level adjustment device,
- input anti-aliasing filter,
- sample-hold device sampling at 8 kHz,
- analogue-to-uniform digital conversion to 13 bits representation.

The uniform format shall be represented in two’s complement.

2) Uniform digital to analogue
- conversion from 13 bit /8kHz 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 companded format, based on a standard A-law codec/filter
according to CCITT rec. G.711/714, followed by the 8-bit to 13-bit conversion according to
the procedure specified in section 1.4.

For the D/A operation, the inverse operations take place.

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In the latter case it should be noted that the specifications in CCITT recommendation 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, the present document does not on its own ensure sufficient out-of-band
attenuation.
The specification of out-of-band signals is defined in section 2 between the acoustic signal and the digital
interface to take into account that the filtering in the terminal can be achieved both by electronic and
acoustical design.
1.4 PCM format conversion
The conversion between 8 bit A-law companded format and the 13-bituniform format shall be as defined
in CCITT Recommendation G.721 (superseded by G.726), section 4.2.1, sub-block EXPAND and section
4.2.7, sub-block COMPRESS. The parameter LAW = 1 should be used.
1.5 Principles of the RPE-LTP Encoder
A simplified block diagram of the RPE-LTP encoder is shown in Fig1.1. In this diagram the coding and
quantization functions are not shown explicitly.
The input speech frame, consisting of 160 signal samples (uniform 13 bit PCM samples), is first pre-
processed to produce an offset-free signal, which is then subjected to a first order pre-emphasis filter.
The 160 samples obtained are then analyzed to determine the coefficients for the short term analysis filter
(LPC analysis). These parameters are then used for the filtering of the same 160 samples. The result is
160 samples of the short term residual signal. The filter parameters, termed reflection coefficients, are
transformed to log.area ratios, LARs, before transmission.
For the following operations, the speech frame is divided into 4 sub-frames with 40 samples of the short
term residual signal in each. Each sub-frame is processed blockwise by the subsequent functional
elements.
Before the processing of each sub-block of 40 short term residual samples, the parameters of the long
term analysis filter, the LTP lag and the LTP gain, are estimated and updated in the LTP analysis block,
on the basis of the current sub-block of the present and a stored sequence of the 120 previous
reconstructed short term residual samples.
A block of 40 long term residual signal samples is obtained by subtracting 40 estimates of the short term
residual signal from the short term residual signal itself. The resulting block of 40 long term residual
samples is fed to the Regular Pulse Excitation analysis which performs the basic compression function of
the algorithm.
As a result of the RPE-analysis, the block of 40 input long term residual samples are represented by one
of 4 candidate sub-sequences of 13 pulses each. The subsequence selected is identified by the RPE grid
position (M). The 13 RPE pulses are encoded using Adaptive Pulse Code Modulation (APCM) with
estimation of the sub-block amplitude which is transmitted to the decoder as side information.
The RPE parameters are also fed to a local RPE decoding and reconstruction module which produces a
block of 40 samples of the quantized version of the long term residual signal.
By adding these 40 quantized samples of the long term residual to the previous block of short term
residual signal estimates, a reconstructed version of the current short term residual signal is obtained.
The block of reconstructed short term residual signal samples is then fed to the long term analysis filter
which produces the new block of 40 short term residual signal estimates to be used for the next sub-block
thereby completing the feedback loop.
1.6 Principles of the RPE-LTP Decoder
The simplified block diagram of the RPE-LTP decoder is shown in fig 1.2. The decoder includes the same
structure as the feed-back loop of the encoder. In error-free transmission, the output of this stage will be
the reconstructed short term residual samples. These samples are then applied to the short term
synthesis filter followed by the de-emphasis filter resulting in the reconstructed speech signal samples.

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1.7 Sequence and subjective importance of encoded parameters
As indicated in fig 1.1 the three different groups of data are produced by the encoder are:
- the short term filter parameters,
- the Long Term Prediction (LTP) parameters
- the RPE parameters.

The encoder will produce this information in a unique sequence and format, and the decoder must
receive the same information in the same way. In table 1.1, the sequence of output bits b1 to b260 and
the bit allocation for each parameter is shown.
The different parameters of the encoded speech and their individual bits have unequal importance with
respect to subjective quality. Before being submitted to the channel encoding function the bits have to be
rearranged in the sequence of importance as given in GSM 05.03. The ranking has been determined by
subjective testing and the procedure used is described in annex 2.
==================================================================
Parameter Parameter  Parameter     Var. Number  Bit no.
      number   name        name of bits (LSB-MSB)
================================================================
...