ETSI EN 300 961 V6.1.1 (2000-11)
Digital cellular telecommunications system (Phase 2+) (GSM); Full rate speech; Transcoding (GSM 06.10 version 6.1.1 Release 1997)
Digital cellular telecommunications system (Phase 2+) (GSM); Full rate speech; Transcoding (GSM 06.10 version 6.1.1 Release 1997)
REN/SMG-110610Q6R1
Digitalni celični telekomunikacijski sistem (faza 2+) – Govor s polno hitrostjo – Prekodiranje (GSM 06.10, različica 6.1.1, izdaja 1997)
General Information
Standards Content (Sample)
2003-01.Slovenski inštitut za standardizacijo. Razmnoževanje celote ali delov tega standarda ni dovoljeno.Digital cellular telecommunications system (Phase 2+) (GSM); Full rate speech; Transcoding (GSM 06.10 version 6.1.1 Release 1997)33.070.50Globalni sistem za mobilno telekomunikacijo (GSM)Global System for Mobile Communication (GSM)ICS:Ta slovenski standard je istoveten z:EN 300 961 Version 6.1.1SIST EN 300 961 V6.1.1:2003en01-december-2003SIST EN 300 961 V6.1.1:2003SLOVENSKI
STANDARD
FOR MOBILE COMMUNICATIONSR
ETSI ETSI EN 300 961 V6.1.1 (2000-11)2(GSM 06.10 version 6.1.1 Release 1997)
Reference REN/SMG-110610Q6R1 Keywords Digital cellular telecommunications system, Global System for Mobile communications (GSM) ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE
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© European Telecommunications Standards Institute 2000. All rights reserved.
ETSI ETSI EN 300 961 V6.1.1 (2000-11)3(GSM 06.10 version 6.1.1 Release 1997) Contents Intellectual Property Rights.6 Foreword.6 1 Scope.8 1.1 References.8 1.1.1 Abbreviations.9 1.2 Outline description.9 1.3 Functional description of audio parts.9 1.4 PCM Format conversion.10 1.5 Principles of the RPE-LTP encoder.10 1.6 Principles of the RPE-LTP decoder.11 1.7 Sequence and subjective importance of encoded parameters.11 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.14 3.1 Functional description of the RPE-LTP encoder.14 3.1.1 Offset compensation.15 3.1.2 Pre-emphasis.15 3.1.3 Segmentation.15 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.16 3.1.8 Decoding of the quantized Log.-Area Ratios.17 3.1.9 Interpolation of Log.-Area Ratios.17 3.1.10 Transformation of Log.-Area Ratios into reflection coefficients.17 3.1.11 Short term analysis filtering.17 3.1.12 Sub-segmentation.18 3.1.13 Calculation of the LTP parameters.18 3.1.14 Coding/Decoding of the LTP lags.18 3.1.15 Coding/Decoding of the LTP gains.19 3.1.16 Long term analysis filtering.19 3.1.17 Long term synthesis filtering.19 3.1.18 Weighting Filter.20 3.1.19 Adaptive sample rate decimation by RPE grid selection.20 3.1.20 APCM quantization of the selected RPE sequence.20 3.1.21 APCM inverse quantization.21 3.1.22 RPE grid positioning.22 3.2 Decoder.22 3.2.1 RPE decoding clause.22 3.2.2 Long Term Prediction clause.22 3.2.3 Short term synthesis filtering clause.22 3.2.4 Post-processing.22 4 Codec homing.26 4.1 Functional description.26 4.2 Definitions.26 4.3 Encoder homing.27 4.4 Decoder homing.27 4.5 Encoder home state.28 4.6 Decoder home state.28 5 Computational details of the RPE-LTP codec.28 5.1 Data representation and arithmetic operations.28 5.2 Fixed point implementation of the RPE-LTP coder.30 SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)4(GSM 06.10 version 6.1.1 Release 1997) 5.2.0 Scaling of the input variable.31 5.2.1 Downscaling of the input signal.31 5.2.2 Offset compensation.31 5.2.3 Pre-emphasis.31 5.2.4 Autocorrelation.32 5.2.5 Computation of the reflection coefficients.32 5.2.6 Transformation of reflection coefficients to Log.-Area Ratios.33 5.2.7 Quantization and coding of the Log.-Area Ratios.34 5.2.8 Decoding of the coded Log.-Area Ratios.34 5.2.9 Computation of the quantized reflection coefficients.34 5.2.9.1 Interpolation of the LARpp[1.8] to get the LARp[1.8].34 5.2.9.2 Computation of the rp[1.8] from the interpolated LARp[1.8].35 5.2.10 Short term analysis filtering.35 5.2.11 Calculation of the LTP parameters.36 5.2.12 Long term analysis filtering.37 5.2.13 Weighting filter.37 5.2.14 RPE grid selection.38 5.2.15 APCM quantization of the selected RPE sequence.38 5.2.16 APCM inverse quantization.39 5.2.17 RPE grid positioning.39 5.2.18 Update of the reconstructed short term residual signal dp[-120.-1].40 5.3 Fixed point implementation of the RPE-LTP decoder.40 5.3.1 RPE decoding clause.40 5.3.2 Long term synthesis filtering.40 5.3.3 Computation of the decoded reflection coefficients.41 5.3.4 Short term synthesis filtering clause.41 5.3.5 De-emphasis filtering.42 5.3.6 Upscaling of the output signal.42 5.3.7 Truncation of the output variable.42 5.4 Tables used in the fixed point implementation of the RPE-LTP coder and decoder.42 6 Digital test sequences.44 6.1 Input and output signals.44 6.2 Configuration for the application of the test sequences.44 6.2.1 Configuration 1 (encoder only).44 6.2.2 Configuration 2 (decoder only).45 6.3 Test sequences.45 6.3.1 Test sequences for configuration 1.45 6.3.2 Test sequences for configuration 2.46 6.3.3 Additional Test sequences for Codec Homing.50 6.3.3.1 Codec homing frames.50 6.3.3.2 Sequence for an extensive test of the decoder homing.50 6.3.3.3 Sequences for finding the 20 ms framing of the GSM full rate speech encoder.50 6.3.3.4 Formats and sizes of the synchronization sequences.51 Annex A (informative): Codec performance.53 A.1 Performance of the RPE-LTP.53 A.1.1 Introduction.53 A.1.2 Speech performance.53 A.1.2.1 Single encoding.53 A.1.2.2 Speech performance when interconnected with coding systems on an analogue basis.54 A.1.2.2.1 Performance with 32 kbit/s ADPCM (G.721, superseded by G.726).54 A.1.2.2.2 Performance with another RPE-LTP codec.54 A.1.2.2.3 Performance with encoding other than RPE-LTP and 32 kbit/s ADPCM (G.721, superseded by G.726).54 A.1.3 Non-speech performance.55 A.1.3.1 Performance with single sine waves.55 A.1.3.2 Performance with DTMF tones.55 A.1.3.3 Performance with information tones.55 A.1.3.4 Performance with voice-band data.55 A.1.4 Delay.55 A.1.5 Bibliography.57 SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)5(GSM 06.10 version 6.1.1 Release 1997) A.2 Subjective relevance of the speech coder output bits.57 A.3 Format for test sequence distribution.59 A.3.1 Type of files provided.59 A.3.2 File format description.60 Annex B (informative): Test sequence disks.62 Annex C (informative): Change Request History.63 History.64
ETSI ETSI EN 300 961 V6.1.1 (2000-11)6(GSM 06.10 version 6.1.1 Release 1997) 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 Standard (Telecommunications series) has been produced by ETSI Technical Committee Special Mobile Group (SMG). 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". Archive en_300961v060101p0.ZIP which accompanies the present document, contains test sequences, as described in clause 6 and annex A.3. The archive contains the following: Disk1.zip Annex B: Test sequences for the GSM Full Rate speech codec; Test sequences SEQ01.xxx to SEQ05.xxx. (Disk1.zip contains LHA compressed files.) Disk2.zip Annex B: Test sequences for the GSM Full Rate speech codec with homing frames; Test sequences SEQ01H.* to SEQ02H.*. Disk3.zip Annex B: Test sequences for the GSM Full Rate speech codec with homing frames; Test sequences SEQ03H.* to SYNC159.COD. Disk4.zip Annex B: 8 bit A-law test sequences for the GSM Full Rate speech codec with and without homing frames (Disk4.zip contains self-extracting files). Disk5.zip Annex B: 8 bit µ-law test sequences for the GSM Full Rate speech codec with and without
homing frames (Disk5.zip contains self-extracting files). The contents of the present document is subject to continuing work within SMG and may change following formal SMG approval. Should SMG modify the contents of the present document it will be re-released with an identifying change of release date and an increase in version number as follows: Version 6.x.y where: 6 indicates Release 1997 of GSM Phase 2+. x the second digit is incremented for all 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.
ETSI ETSI EN 300 961 V6.1.1 (2000-11)7(GSM 06.10 version 6.1.1 Release 1997) National transposition dates Date of adoption of this EN: 24 November 2000 Date of latest announcement of this EN (doa): 28 February 2001 Date of latest publication of new National Standard or endorsement of this EN (dop/e):
31 August 2001 Date of withdrawal of any conflicting National Standard (dow): 31 August 2001
ETSI ETSI EN 300 961 V6.1.1 (2000-11)8(GSM 06.10 version 6.1.1 Release 1997) 1 Scope The transcoding procedure specified in the present document 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 present document also describes the codec down to the bit level, thus enabling the verification of compliance to the present document to a high degree of confidence by use of a set of digital test sequences. These test sequences are described and are contained in archive en_300961v060101p0.ZIP which accompanies the present document. 1.1 References The following documents contain provisions which, through reference in this text, constitute provisions of the present document. • References are either specific (identified by date of publication, edition number, version number, etc.) or non-specific. • For a specific reference, subsequent revisions do not apply. • For a non-specific reference, the latest version applies. • A non-specific reference to an ETS shall also be taken to refer to later versions published as an EN with the same number. • For this Release 1997 document, references to GSM documents are for Release 1997 versions (version 6.x.y). [1] GSM 01.04: "Digital cellular telecommunications system (Phase 2+); Abbreviations and acronyms". [2] GSM 05.03: "Digital cellular telecommunications system (Phase 2+); Channel coding". [3] GSM 06.01: "Digital cellular telecommunications system (Phase 2+); Full rate speech; Processing functions". [4] GSM 11.10: "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". SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)9(GSM 06.10 version 6.1.1 Release 1997) [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". [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: "Digital cellular telecommunications system (Phase 2+); Voice Activity Detector (VAD)". 1.1.1 Abbreviations Abbreviations used in the present document are listed in GSM 01.04. 1.2 Outline description The present document 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 clause 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; SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)10(GSM 06.10 version 6.1.1 Release 1997) - 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 ITU-T Recommendation G.711/714, followed by the 8-bit to 13-bit conversion according to the procedure specified in clause 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 clause 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), clause 4.2.1, sub-block EXPAND and clause 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.
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. SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)11(GSM 06.10 version 6.1.1 Release 1997) 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, clause 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) ==================================================================
==================================================================
LAR 1
b1
- b6
LAR 2
b7
- b12 FILTER
Log. Area
LAR 3
b13 - b17
ratios
LAR 4
b18 - b22 PARAMETERS
1 - 8
LAR 5
b23 - b26
LAR 6
b27 - b30
LAR 7
b31 - b33
LAR 8
b34 - b36 ==================================================================
Sub-frame no.1 ================================================================== LTP
LTP lag
N1
b37 - b43 PARAMETERS
LTP gain
b1
b44 - b45 ------------------------------------------------------------------
RPE grid position
M1
b46 - b47 RPE
Block amplitude
Xmax1
b48 - b53 PARAMETERS
RPE-pulse no.1
x1(0)
b54 - b56
RPE-pulse no.2
x1(1)
b57 - b59
..
...
...
RPE-pulse no.13
x1(12)
b90 - b92 ==================================================================
ETSI ETSI EN 300 961 V6.1.1 (2000-11)12(GSM 06.10 version 6.1.1 Release 1997) Sub-frame no.2 ================================================================== LTP
LTP lag
N2
b93 - b99 PARAMETERS
LTP gain
b2
b100- b101 ------------------------------------------------------------------
RPE grid position
M2
b102- b103 RPE
Block amplitude
Xmax2
b104- b109 PARAMETERS
RPE-pulse no.1
x2(0)
b110- b112
RPE-pulse no.2
x2(1)
b113- b115
..
...
...
RPE-pulse no.13
x2(12)
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
LTP lag
N3
b149- b155 PARAMETERS
LTP gain
b3
b156- b157 ------------------------------------------------------------------
RPE grid position
M3
b158- b159 RPE
Block amplitude
Xmax3
b160- b165 PARAMETERS
RPE-pulse no.1
x3(0)
b166- b168
RPE-pulse no.2
x3(1)
b169- b171
..
...
...
RPE-pulse no.13
x3(12)
b202- b204 ==================================================================
Sub-frame no.4 ================================================================== LTP
LTP lag
N4
b205- b211 PARAMETERS
LTP gain
b4
b212- b213 ------------------------------------------------------------------
RPE grid position
M4
b214- b215 RPE
Block amplitude
Xmax4
b216- b221 PARAMETERS
RPE-pulse no.1
x4(0)
b222- b224
RPE-pulse no.2
x4(1)
b225- b227
..
...
...
RPE-pulse no.13
x4(12)
b258- b260 ==================================================================
ETSI ETSI EN 300 961 V6.1.1 (2000-11)13(GSM 06.10 version 6.1.1 Release 1997) InputPre-processingsignalShort termanalysisfilterShort termLPCanalysis+RPE gridselectionand coding(1)(2)LTPanalysisLong termanalysisfilter+RPE griddecoding andpositioning(4)(5)(3)-LTP parameters(9 bits/5 ms)Reflectioncoefficients coded asLog. - Area Ratios(36 bits/20 ms)RPE parameters(47 bits/5 ms)Toradiosubsystem(1) Short term residual(2) Long term residual (40 samples)(3) Short term residual estimate (40 samples)(4) Reconstructed short term residual (40 samples)(5) Quantized long term residual (40 samples) Figure 1.1: Simplified block diagram of the RPE - LTP encoderRPE griddecoding andpositioningReflection coefficients coded as Log. - Area Ratios(36 bits/20 ms)RPE parameters(47 bits/5 ms)FromradiosubsystemLTP parameters(9 bits/5 ms)+Short termsynthesisfilterLong termsynthesisfilterPost-processingOutputsignal Figure 1.2: Simplified block diagram of the RPE - LTP decoder SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)14(GSM 06.10 version 6.1.1 Release 1997) 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 clause 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. 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 clauses. 3.1 Functional description of the RPE-LTP encoder The Pre-processing clause of the RPE-LTP encoder comprises the following two sub-blocks: - Offset compensation (3.1.1); - Pre-emphasis (3.1.2). The LPC analysis clause 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). SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)15(GSM 06.10 version 6.1.1 Release 1997) The Short term analysis filtering clause 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) clause 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 clause 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 clause 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 so to produce the offset-free signal sof.
sof(k) = so(k) - so(k-1) + alpha*sof(k-1)
(3.1.1)
alpha = 32735*2-15
3.1.2 Pre-emphasis The signal sof is applied to a first order FIR pre-emphasis filter leading to the input signal s of the analysis clause.
s(k) = sof(k) - beta*sof(k-1)
(3.1.2)
beta= 28180*2-15
LPC analysis clause 3.1.3 Segmentation The speech signal s(k) is divided into non-overlapping frames having a length of T0 = 20 ms (160 samples). A new LPC-analysis of order p=8 is performed for each frame. SIST EN 300 961 V6.1.1:2003
ETSI ETSI EN 300 961 V6.1.1 (2000-11)16(GSM 06.10 version 6.1.1 Release 1997) 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 clauses is assumed. Such a representation can be described by the reflection coefficients or the area ratios of connected clauses. 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) = log10
(----------)
(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 <= |
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