ETSI ETS 300 961 ed.3 (2000-12)
Digital cellular telecommunications system (Phase 2+) (GSM); Full rate speech; Transcoding (GSM 06.10 version 5.2.1 Release 1996)
Digital cellular telecommunications system (Phase 2+) (GSM); Full rate speech; Transcoding (GSM 06.10 version 5.2.1 Release 1996)
RE/SMG-110610QR2
Digitalni celični telekomunikacijski sistem (faza 2+) – Govor s polno hitrostjo – Prekodiranje (GSM 06.10, različica 5.2.1, izdaja 1996)
General Information
Standards Content (Sample)
SLOVENSKI STANDARD
01-december-2003
'LJLWDOQLFHOLþQLWHOHNRPXQLNDFLMVNLVLVWHP ID]D ±*RYRUVSROQRKLWURVWMR±
3UHNRGLUDQMH *60UD]OLþLFDL]GDMD
Digital cellular telecommunications system (Phase 2+) (GSM); Full rate speech;
Transcoding (GSM 06.10 version 5.2.1 Release 1996)
Ta slovenski standard je istoveten z: ETS 300 961 Edition 3
ICS:
33.070.50 Globalni sistem za mobilno Global System for Mobile
telekomunikacijo (GSM) Communication (GSM)
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.
EUROPEAN ETS 300 961
TELECOMMUNICATION December 2000
STANDARD Third Edition
Source: SMG Reference: RE/SMG-110610QR2
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;
Transcoding
(GSM 06.10 version 5.2.1 Release 1996)
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.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 2000. All rights reserved.
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
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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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
Contents
Intellectual Property Rights.7
Foreword.7
1 Scope .9
1.1 Normative references.9
1.1.1 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 .12
1.7 Sequence and subjective importance of encoded parameters.12
2 Transmission characteristics.15
2.1 Performance characteristics of the analogue/digital interfaces .15
2.2 Transcoder delay .15
3 Functional description of the RPE-LTP codec .16
3.1 Functional description of the RPE-LTP encoder.16
3.1.1 Offset compensation.16
3.1.2 Pre-emphasis .16
3.1.3 Segmentation .17
3.1.4 Autocorrelation .17
3.1.5 Schur Recursion.17
3.1.6 Transformation of reflection coefficients to Log.-Area Ratios .17
3.1.7 Quantization and coding of Log.-Area Ratios .18
3.1.8 Decoding of the quantized Log.-Area Ratios .18
3.1.9 Interpolation of Log.-Area Ratios.18
3.1.10 Transformation of Log.-Area Ratios into reflection coefficients .19
3.1.11 Short term analysis filtering .19
3.1.12 Sub-segmentation .19
3.1.13 Calculation of the LTP parameters.19
3.1.14 Coding/Decoding of the LTP lags.20
3.1.15 Coding/Decoding of the LTP gains.20
3.1.16 Long term analysis filtering.20
3.1.17 Long term synthesis filtering.21
3.1.18 Weighting Filter.21
3.1.19 Adaptive sample rate decimation by RPE grid selection.21
3.1.20 APCM quantization of the selected RPE sequence.22
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.24
3.2.4 Post-processing.24
4 Codec homing .28
4.1 Functional description .28
4.2 Definitions .28
4.3 Encoder homing.29
4.4 Decoder homing.29
4.5 Encoder home state.30
4.6 Decoder home state.30
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5 Computational details of the RPE-LTP codec . 31
5.1 Data representation and arithmetic operations . 31
5.2 Fixed point implementation of the RPE-LTP coder. 33
5.2.0 Scaling of the input variable. 33
5.2.1 Downscaling of the input signal . 33
5.2.2 Offset compensation. 33
5.2.3 Pre-emphasis. 34
5.2.4 Autocorrelation. 34
5.2.5 Computation of the reflection coefficients. 35
5.2.6 Transformation of reflection coefficients to Log.-Area Ratios. 36
5.2.7 Quantization and coding of the Log.-Area Ratios. 36
5.2.8 Decoding of the coded Log.-Area Ratios. 37
5.2.9 Computation of the quantized reflection coefficients. 38
5.2.9.1 Interpolation of the LARpp[1.8] to get the LARp[1.8] . 38
5.2.9.2 Computation of the rp[1.8] from the interpolated
LARp[1.8] . 38
5.2.10 Short term analysis filtering . 39
5.2.11 Calculation of the LTP parameters . 39
5.2.12 Long term analysis filtering . 40
5.2.13 Weighting filter . 42
5.2.14 RPE grid selection . 42
5.2.15 APCM quantization of the selected RPE sequence . 43
5.2.16 APCM inverse quantization . 44
5.2.17 RPE grid positioning . 44
5.2.18 Update of the reconstructed short term residual signal dp[-120.-1] . 44
5.3 Fixed point implementation of the RPE-LTP decoder. 44
5.3.1 RPE decoding section . 44
5.3.2 Long term synthesis filtering . 45
5.3.3 Computation of the decoded reflection coefficients. 45
5.3.4 Short term synthesis filtering section. 45
5.3.5 De-emphasis filtering . 46
5.3.6 Upscaling of the output signal. 47
5.3.7 Truncation of the output variable . 47
5.4 Tables used in the fixed point implementation of the RPE-LTP coder and decoder . 47
6 Digital test sequences. 48
6.1 Input and output signals . 49
6.2 Configuration for the application of the test sequences . 49
6.2.1 Configuration 1 (encoder only) . 49
6.2.2 Configuration 2 (decoder only) . 49
6.3 Test sequences . 50
6.3.1 Test sequences for configuration 1. 50
6.3.2 Test sequences for configuration 2. 51
6.3.3 Additional Test sequences for Codec Homing . 55
6.3.3.1 Codec homing frames. 55
6.3.3.2 Sequence for an extensive test of the decoder homing. 55
6.3.3.3 Sequences for finding the 20 ms framing of the GSM full
rate speech encoder . 55
6.3.3.4 Formats and sizes of the synchronization sequences . 56
Annex A (informative): Codec performance . 58
A.1 Performance of the RPE-LTP. 58
A.1.1 Introduction. 58
A.1.2 Speech performance. 58
A.1.2.1 Single encoding . 58
A.1.2.2 Speech performance when interconnected with coding systems on an
analogue basis. 59
A.1.2.2.1 Performance with 32 kbit/s ADPCM (G.721, superseded
by G.726). 59
A.1.2.2.2 Performance with another RPE-LTP codec . 59
A.1.2.2.3 Performance with encoding other than RPE-LTP and 32
kbit/s ADPCM (G.721, superseded by G.726). 59
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A.1.3 Non-speech performance .59
A.1.3.1 Performance with single sine waves .59
A.1.3.2 Performance with DTMF tones.60
A.1.3.3 Performance with information tones.60
A.1.3.4 Performance with voice-band data.60
A.1.4 Delay.60
A.1.5 Bibliography .61
A.2 Subjective relevance of the speech coder output bits.62
A.3 Format for test sequence distribution.63
A.3.1 Type of files provided.63
A.3.2 File format description.64
Annex B (informative): Test sequence diskettes.66
Annex C (informative): Change Request History .67
History.68
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
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.
The associated files (annex B) are supplied with this ETS, the files contain test sequences, as described
in clause 6 and annex A.3.
The diskettes are labelled as follows:
Diskette 1 ETS 300 961, annex B: Test sequences for the GSM Full Rate speech codec;
Test sequences SEQ01.xxx to SEQ05.xxx. (Diskette 1 contains LHA
compressed files.)
Diskette 2 ETS 300 961, annex B: Test sequences for the GSM Full Rate speech codec
with homing frames; Test sequences SEQ01H.* to SEQ02H.*.
Diskette 3 ETS 300 961, annex B: Test sequences for the GSM Full Rate speech codec
with homing frames; Test sequences SEQ03H.* to SYNC159.COD.
Diskette 4 ETS 300 961, annex B : 8 bit A-law test sequences for the GSM Full Rate
speech codec with and without homing frames (Diskette 4 contains self-
extracting files).
Diskette 5 ETS 300 961, annex B: 8 bit μ-law test sequences for the GSM Full Rate
speech codec with and without homing frames (Diskette 5 contains self-
extracting files).
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:
5 GSM Phase 2+ Release 1996.
x the second digit is incremented for changes of substance, i.e. technical enhancements,
corrections, updates, etc.;
y the third digit is incremented when editorial only changes have been incorporated in the
specification.
The specification from which this ETS has been derived was originally based on CEPT documentation,
hence the presentation of this ETS may not be entirely in accordance with the ETSI/PNE rules.
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
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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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
1 Scope
The transcoding procedure specified in this European Telecommunication Standard (ETS) is applicable
for the full-rate Traffic Channel (TCH) in the digital cellular telecommunications system. The use of this
transcoding scheme for other applications has not been considered.
In 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 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 present document 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 ETS also describes the codec down to the bit level, thus enabling
the verification of compliance to the ETS to a high degree of confidence by use of a set of digital test
sequences. These test sequences are described and are included in this ETS on diskette.
1.1 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 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 350): "Digital cellular telecommunications system (Phase 2+);
Abbreviations and acronyms".
[2] GSM 05.03 (ETS 300 909): "Digital cellular telecommunications system
(Phase 2+); Channel coding".
[3] GSM 06.01 (ETS 300 960): "Digital cellular telecommunications system (Phase
2+); Full rate speech; Processing functions".
[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)".
[9] ITU-T Recommendation Q.35: "Technical characteristics of tones for the
telephone service".
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
[10] ITU-T Recommendation V.21: "300 bits per second duplex modem
standardized for use in the general switched telephone network".
[11] ITU-T Recommendation V.23: "600/1 200-band modem standardized for use in
the general switched telephone network".
[12] GSM 06.32 (ETS 300 965): "Digital cellular telecommunications system (Phase
2+); Voice Activity Detector (VAD)".
1.1.1 Abbreviations
Abbreviations used in this ETS are listed in GSM 01.04.
1.2 Outline description
The ETS is structured as follows:
Subclause 1.3 contains a functional description of the audio parts including the A/D and D/A functions.
Subclause 1.4 describes the conversion between 13 bit uniform and 8 bit A-law samples. Subclauses 1.5
and 1.6 present a simplified description of the principles of the RPE-LTP encoding and decoding process
respectively. In subclause 1.7, the sequence and subjective importance of encoded parameters are
given.
Clause 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 clause 2. Clause 3
presents the functional description of the RPE-LTP coding and decoding procedures, whereas clause 4
describes the computational details of the algorithm. Procedures for the verification of the correct
functioning of the RPE-LTP are described in clause 5.
Performance and network aspects of the RPE-LTP codec are contained in annex A.
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 /8 kHz uniform PCM to analogue;
- a hold device;
- reconstruction filter including x/sin x correction;
- output level adjustment device;
- earphone or loudspeaker.
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
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 ITU-T Recommendation G.711/714, followed by the 8-bit to 13-bit conversion
according to the procedure specified in subclause 1.4.
For the D/A operation, the inverse operations take place.
In the latter case it should be noted that the specifications in ITU-T 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-bit uniform format shall be as defined
in ITU-T Recommendation G.721 (superseded by G.726), subclause 4.2.1, sub-block EXPAND and
subclause 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 figure 1.1. 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 analysed 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.
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
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.
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 shall 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 A, subclause A.2.
Table 1.1: Encoder output parameters in order of occurrence and
bit allocation within the speech frame of 260 bits/20 ms
==================================================================
Parameter Parameter Parameter Var. Number Bit no.
number name name of bits (LSB-MSB)
==================================================================
==================================================================
1 LAR 1 6 b1 - b6
2 LAR 2 6 b7 - b12
FILTER 3 Log. Area LAR 3 5 b13 - b17
4 ratios LAR 4 5 b18 - b22
PARAMETERS 5 1 - 8 LAR 5 4 b23 - b26
6 LAR 6 4 b27 - b30
7 LAR 7 3 b31 - b33
8 LAR 8 3 b34 - b36
==================================================================
Sub-frame no.1
==================================================================
LTP 9 LTP lag N1 7 b37 - b43
PARAMETERS 10 LTP gain b1 2 b44 - b45
------------------------------------------------------------------
11 RPE grid position M1 2 b46 - b47
RPE 12 Block amplitude Xmax1 6 b48 - b53
PARAMETERS 13 RPE-pulse no.1 x1(0) 3 b54 - b56
14 RPE-pulse no.2 x1(1) 3 b57 - b59
.. ... ...
25 RPE-pulse no.13 x1(12) 3 b90 - b92
==================================================================
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Sub-frame no.2
==================================================================
LTP 26 LTP lag N2 7 b93 - b99
PARAMETERS 27 LTP gain b2 2 b100- b101
------------------------------------------------------------------
28 RPE grid position M2 2 b102- b103
RPE 29 Block amplitude Xmax2 6 b104- b109
PARAMETERS 30 RPE-pulse no.1 x2(0) 3 b110- b112
31 RPE-pulse no.2 x2(1) 3 b113- b115
.. ... ...
42 RPE-pulse no.13 x2(12) 3 b146- b148
==================================================================
Table 1.1: Encoder output parameters in order of occurrence and
bit allocation within the speech frame of 260 bits/20 ms
Sub-frame no.3
==================================================================
LTP 43 LTP lag N3 7 b149- b155
PARAMETERS 44 LTP gain b3 2 b156- b157
------------------------------------------------------------------
45 RPE grid position M3 2 b158- b159
RPE 46 Block amplitude Xmax3 6 b160- b165
PARAMETERS 47 RPE-pulse no.1 x3(0) 3 b166- b168
48 RPE-pulse no.2 x3(1) 3 b169- b171
.. ... ...
59 RPE-pulse no.13 x3(12) 3 b202- b204
==================================================================
Sub-frame no.4
==================================================================
LTP 60 LTP lag N4 7 b205- b211
PARAMETERS 61 LTP gain b4 2 b212- b213
------------------------------------------------------------------
62 RPE grid position M4 2 b214- b215
RPE 63 Block amplitude Xmax4 6 b216- b221
PARAMETERS 64 RPE-pulse no.1 x4(0) 3 b222- b224
65 RPE-pulse no.2 x4(1) 3 b225- b227
.. ... ...
76 RPE-pulse no.13 x4(12) 3 b258- b260
==================================================================
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
Reflection
coefficients coded as
Log. - Area Ratios
Short term
(36 bits/20 ms)
LPC
analysis
RPE parameters
(1) (2)
Input RPE grid (47 bits/5 ms)
Short term
Pre-
+ selection
analysis
processing
signal and coding
filter
-
(3)
(4) (5)
RPE grid
Long term
+
decoding and
analysis
positioning
filter
LTP parameters
LTP
(9 bits/5 ms)
analysis
(1) Short term residual
To
(2) Long term residual (40 samples)
(3) Short term residual estimate (40 samples)
radio
(4) Reconstructed short term residual (40 samples)
(5) Quantized long term residual (40 samples)
subsystem
Figure 1.1: Simplified block diagram of the RPE - LTP encoder
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ETS 300 961: December 2000 (GSM 06.10 version 5.2.1)
Reflection coefficients coded
as Log. - Area Ratios
(36 bits/20 ms)
RPE grid
Short term
Output
Post-
decoding and
+ synthesis
processing
signal
positioning filter
RPE
parameters Long term
(47 bits/5 ms) synthesis
filter
LTP
parameters
(9 bits/5 ms)
From
radio
subsystem
Figure 1.2: Simplified block diagram of the RPE - LTP decoder
2 Transmission characteristics
This clause specifies the necessary performance characteristics of the audio parts for proper functioning
of the speech transcoder. Some transmission performance characteristics of the RPE-LTP transcoder are
also given to assist the designer of the speech transcoder function. The information given here is
redundant and the detailed specifications are contained in recommendation GSM 11.10.
The performance characteristics are referred to the 13 bit uniform PCM interface.
NOTE: To simplify the verification of the specifications, the performance limits may be
referred to an A-law measurement interface according to ITU-T Recommendation
G.711. In this way, standard measuring equipments for PCM systems can be utilized
for measurements. The relationship between the 13 bit format and the A-law
companded shall follow the procedures defined in subclause 1.4.
2.1 Performance characteristics of the analogue/digital interfaces
Concerning 1) discrimination against out-of-band signals (sending) and 2) spurious out-of-band signals
(receiving), the same requirements as defined in ETSI standard TE 04-15 (digital telephone, candidate
NET33) apply.
2.2 Transcoder delay
Consider a back to back configuration where the parameters generated by the encoder are delivered to
the speech decoder as soon as they are available.
The transcoder delay is defined as the time interval between the instant a speech frame of 160 samples
has been received at the encoder input and the instant the corresponding 160 reconstructed speech
samples have been out-put by the speech decoder at an 8 kHz sample rate.
The theoretical minimum delay which can be achieved is 20 ms. The requirement is that the transcoder
delay should be less than 30 ms.
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3 Functional description of the RPE-LTP codec
The block diagram of the RPE-LTP-coder is shown in figure 3.1. The individual blocks are described in
the following subclauses.
3.1 Functional description of the RPE-LTP encoder
The Pre-processing section of the RPE-LTP encoder comprises the following two sub-blocks:
- Offset compensation (3.1.1);
- Pre-emphasis (3.1.2).
The LPC analysis section of the RPE-LTP encoder comprises the following five sub-blocks:
- Segmentation (3.1.3);
- Auto-Correlation (3.1.4);
- Schur Recursion (3.1.5);
- Transformation of reflection coefficients to Log.-Area Ratios (3.1.6);
- Quantization and coding of Log.-Area Ratios (3.1.7).
The Short term analysis filtering section of the RPE-LTP comprises the following four sub-blocks:
- Decoding of the quantized Log.-Area Ratios (LARs) (3.1.8);
- Interpolation of Log.-Area Ratios (3.1.9);
- Transformation of Log.-Area Ratios into reflection coefficients (3.1.10);
- Short term analysis filtering (3.1.11).
The Long Term Predictor (LTP) section comprises 4 sub-blocks working on subsegments (3.1.12) of
the short term residual samples:
- Calculation of LTP parameters (3.1.13);
- Coding of the LTP lags (3.1.14) and the LTP gains (3.1.15);
- Decoding of the LTP lags (3.1.14) and the LTP gains (3.1.15);
- Long term analysis filtering (3.1.16), and Long term synthesis filtering (3.1.17).
The RPE encoding section comprises five different sub-blocks:
- Weighting filter (3.1.18);
- Adaptive sample rate decimation by RPE grid selection (3.1.19);
- APCM quantization of the selected RPE sequence (3.1.20);
- APCM inverse quantization (3.1.21);
- RPE grid positioning (3.1.22).
Pre-processing section
3.1.1 Offset compensation
Prior to the speech encoder an offset compensation, by a notch filter is applied in order to remove the
offset of the input signal s to produce the offset-free signal s .
o of
s (k) = s (k) - s (k-1) + alpha*s (k-1) (3.1.1)
of o o of
-15
alpha = 32735*2
3.1.2 Pre-emphasis
The signal s is applied to a first order FIR pre-emphasis filter leading to the input signal s of the analysis
of
section.
s(k) = s (k) - beta*s (k-1) (3.1.2)
of of
-15
beta= 28180*2
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LPC analysis section
3.1.3 Segmentation
The speech signal s(k) is divided into non-overlapping frames having a length of T = 20 ms
(160 samples). A new LPC-analysis of order p=8 is performed for each frame.
3.1.4 Autocorrelation
The first p+1 = 9 values of the Auto-Correlation function are calculated by:
ACF(k)= ∑ s(i)s(i-k) ,k = 0,1.,8 (3.2)
i=k
3.1.5 Schur Recursion
The reflection coefficients are calculated as shown in figure 3.2 using the Schur Recursion algorithm. The
term "reflection coefficient" comes from the theory of linear prediction of speech (LPC), where a vocal
tract representation consisting of series of uniform cylindrical sections is assumed. Such a representation
can be described by the reflection coefficients or the area ratios of connected sections.
3.1.6 Transformation of reflection coefficients to Log.-Area Ratios
The reflection coefficients r(i), (i=1.8), calculated by the Schur algorithm, are in the range:
-1 <= r(i) <= + 1
Due to the favourable quantization characteristics, the reflection coefficients are converted into Log.-Area
Ratios which are strictly defined as follows:
1 + r(i)
Logarea(i) = log (----------) (3.3)
1 - r(i)
Since it is the companding characteristic of this transformation that is of importance, the following
segmented approximation is used.
r(i) ; |r(i)| < 0.675
LAR(i) = sign[r(i)]*[2|r(i)|-0.675] ; 0.675 <= |r(i)| < 0.950
sign[r(i)]*[8|r(i)|-6.375] ; 0.950 <= |r(i)| <= 1.000
(3.4)
with the result that instead of having to divide and obtain the logarithm of particular values, it is merely
necessary to multiply, add and compare these values.
The following equation (3.5) gives the inverse transformation.
LAR’(i) ; |LAR’(i)|<0.675
r’(i)=sign[LAR’(i)]*[0.500*|LAR’(i)|
+0.337500] ; 0.675<=|LAR’(i)|<1.225
sign[LAR’(i)]*[0.125*|LAR’(i)|
+0.796875] ; 1.225<=|LAR’(i)|<=1.625
(3.5)
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3.1.7 Quantization and coding of Log.-Area Ratios
The Log.-Area Ratios LAR(i) have different dynamic ranges and different asymmetric distribution
densities. For this reason, the transformed coefficients LAR(i) are limited and quantized differently
according to the following equation (3.6), with LAR (i) denoting the quantized and integer coded version
c
of LAR(i).
LAR (i) = Nint{A(i)*LAR(i) + B(i)}
...
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