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4.1 Functional description of audio parts
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4.2 Preparation of speech samples
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4.2.1 PCM format conversion
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4.3 Principles of the adaptive multi-rate speech encoder
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4.4 Principles of the adaptive multi-rate speech decoder
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4.5 Sequence and subjective importance of encoded parameters
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5 Functional description of the encoder
................................................................................................... 19 5.1 Pre-processing (all modes) ...............................................................................................................................19
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5.2 Linear prediction analysis and quantization
.....................................................................................................19 12.2 kbit/s mode 19 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes.........................................................................................................19 5.2.1 Windowing and auto-correlation computation............................................................................................20 12.2 kbit/s mode 20 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes.........................................................................................................21 5.2.2 Levinson-Durbin algorithm (all modes)......................................................................................................21
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5.2.3 LP to LSP conversion (all modes)
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5.2.4 LSP to LP conversion (all modes)
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5.2.5 Quantization of the LSP coefficients
..........................................................................................................24 12.2 kbit/s mode 24 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes........................................................................................................25
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5.2.6 Interpolation of the LSPs
............................................................................................................................25 12.2 kbit/s mode 25 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes........................................................................................................26 5.2.7 Monitoring resonance in the LPC spectrum (all modes).............................................................................26 5.3 Open-loop pitch analysis ..................................................................................................................................27 12.2 kbit/s mode 27 10.2 kbit/s mode 28 7.95, 7.40, 6.70, 5.90 kbit/s modes....................................................................................................................................29 5.15, 4.75 kbit/s modes......................................................................................................................................................29
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5.4 Impulse response computation (all modes)
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5.5 Target signal computation (all modes)
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5.6 Adaptive codebook
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5.6.1 Adaptive codebook search
..........................................................................................................................31 12.2 kbit/s mode 31 7.95 kbit/s mode 32 10.2, 7.40 kbit/s mode .......................................................................................................................................................33 6.70, 5.90 kbit/s modes......................................................................................................................................................33 5.15, 4.75 kbit/s modes......................................................................................................................................................34 5.6.2 Adaptive codebook gain control (all modes)............................................................................................................35
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5.7 Algebraic codebook
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5.7.1 Algebraic codebook structure
.....................................................................................................................35 12.2 kbit/s mode 35 10.2 kbit/s mode 36 7.95, 7.40 kbit/s modes......................................................................................................................................................36 6.70 kbit/s mode 36 5.90 kbit/s mode 37 (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) ETSI Page 4 TS 26.090 : December 1999 5.15, 4.75 kbit/s modes......................................................................................................................................................37
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5.7.2 Algebraic codebook search
.........................................................................................................................37 12.2 kbit/s mode 39 10.2 kbit/s mode 39 7.95, 7.40 kbit/s modes......................................................................................................................................................40 6.70 kbit/s mode 40 5.90 kbit/s mode 40 5.15, 4.75 kbit/s modes......................................................................................................................................................41
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5.8 Quantization of the adaptive and fixed codebook gains
...................................................................................41 5.8.1 Adaptive codebook gain limitation in quantization ....................................................................................41
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5.8.2 Quantization of codebook gains
..................................................................................................................41 Prediction of the fixed codebook gain (all modes)............................................................................................................41 12.2 kbit/s mode 42 10.2 kbit/s mode 42 7.95 kbit/s mode 42 7.40 kbit/s mode 43 6.70 kbit/s mode 43 5.90, 5.15 kbit/s modes......................................................................................................................................................43 4.75 kbit/s mode 43 5.8.3 Update past quantized adaptive codebook gain buffer (all modes) ..........................................................................43
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5.9 Memory update (all modes)
..............................................................................................................................44 4.75 kbit/s mode 44
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6 Functional description of the decoder
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6.1 Decoding and speech synthesis
........................................................................................................................44 6.2 Post-processing.................................................................................................................................................48 6.2.1 Adaptive post-filtering (all modes).............................................................................................................48 12.2, 10.2 kbit/s modes......................................................................................................................................................49 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes.................................................................................................................49
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6.2.2 High-pass filtering and up-scaling (all modes)
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7 Detailed bit allocation of the adaptive multi-rate codec
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8 Homing sequences
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8.1 Functional description
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8.2 Definitions
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8.3 Encoder homing
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8.4 Decoder homing
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9 Bibliography
.......................................................................................................................................... 59 Annex A: Change history...................................................................................................................... 60 History ............................................................................................................................................................. 61 ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 5 TS 26.090 : December 1999 Foreword This Technical Specification has been produced by the 3GPP. The present document describes the detailed mapping of the narrowband telephony speech service employing the Adaptive Multi-Rate (AMR) speech coder within the 3GPP system. The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of this TS, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: Version 3.y.z where: x the first digit: 1 presented to TSG for information; 2 presented to TSG for approval; 3 Indicates TSG approved document under change control. y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. z the third digit is incremented when editorial only changes have been incorporated in the specification; ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 6 TS 26.090 : December 1999 1 Scope This Telecommunication Standard (TS) describes the detailed mapping from input blocks of 160 speech samples in 13-bit uniform PCM format to encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits and from encoded blocks of 95, 103, 118, 134, 148, 159, 204, and 244 bits to output blocks of 160 reconstructed speech samples. The sampling rate is 8 000 samples/s leading to a bit rate for the encoded bit stream of 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2 or 12.2 kbit/s. The coding scheme for the multi-rate coding modes is the so-called Algebraic Code Excited Linear Prediction Coder, hereafter referred to as ACELP. The multi-rate ACELP coder is referred to as MR- ACELP. In the case of discrepancy between the requirements described in this TS and the fixed point computational description (ANSI-C code) of these requirements contained in [4], the description in [4] will prevail. The ANSI-C code is not described in this TS, see [4] for a description of the ANSI-C code. The transcoding procedure specified in this TS is mandatory for systems using the AMR speech codec. 2 Normative references This TS incorporates by dated and undated reference, provisions from other publications. These normative references are cited in 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 TS only when incorporated in it by amendment or revision. For undated references, the latest edition of the publication referred to applies. [1] GSM 03.50: " Digital cellular telecommunications system (Phase 2); Transmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) system" [2] TS 26.101 : "AMR Speech Codec; Frame structure". [3] TS 26.094: "AMR Speech Codec; Voice Activity Detection (VAD)". [4] TS 26.073: "AMR Speech Codec; ANSI-C code". [5] TS 26.074: "AMR Speech Codec; Test sequences". [6] ITU-T Recommendation G.711 (1988): "Coding of analogue signals by pulse code modulation Pulse code modulation (PCM) of voice frequencies". [7] ITU-T Recommendation G.726: "40, 32, 24, 16 kbit/s adaptive differential pulse code modulation (ADPCM)". ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 7 TS 26.090 : December 1999 3 Definitions, symbols and abbreviations 3.1 Definitions For the purposes of this TS, 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 adaptive multi-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.. anti-sparseness processing: An adaptive post-processing procedure applied to the fixed codebook vector in order to reduce perceptual artifacts from a sparse fixed codebook vector. 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 adaptive multi- 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 adaptive multi- 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 adaptive multi-rate codec, the fixed codebook is implemented using an algebraic codebook. fractional lags: A set of lag values having sub-sample resolution. In the adaptive multi-rate codec a sub-sample resolution of 1/6th or 1/3rd 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 subsample 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 its multiple or sub-multiple. 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. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 8 TS 26.090 : December 1999 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 adaptive multi-rate codec, the length of the analysis window is always 240 samples. For each frame, two asymmetric windows are used to generate two sets of LP coefficient in the 12.2 kbit/s mode. For the other modes, only a single asymmetric window is used to generate a single set of LP coefficients. In the 12.2 kbit/s mode, no samples of the future frames are used (no lookahead). The other modes use a 5 ms lookahead. LP coefficients: Linear Prediction (LP) coefficients (also referred as Linear Predictive Coding (LPC) coefficients) is a generic descriptive term for the short term filter coefficients. mode: When used alone, refers to the source codec mode, i.e., to one of the source codecs employed in the AMR codec. 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 adaptive multi-rate codec, an open-loop pitch search is performed in every other subframe. 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 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 that the state information in the filter is all zeroes. 3.2 Symbols For the purposes of this TS, the following symbols apply: ( ) A z The inverse filter with unquantized coefficients ( ) A z The inverse filter with quantized coefficients ( ) ( ) H z A z = 1  The speech synthesis filter with quantized coefficients ai The unquantized linear prediction parameters (direct form coefficients) ai The quantified linear prediction parameters m The order of the LP model 1 B z( ) The long-term synthesis filter ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 9 TS 26.090 : December 1999 ( ) W z The perceptual weighting filter (unquantized coefficients) γ γ 1 2 , The perceptual weighting factors F z E( ) Adaptive pre-filter T The integer pitch lag nearest to the closed-loop fractional pitch lag of the subframe β The adaptive pre-filter coefficient (the quantified pitch gain) H z A z A z f n d ( ) ( / ) ( / ) = γ γ The formant postfilter γ n Control coefficient for the amount of the formant post-filtering γ d Control coefficient for the amount of the formant post-filtering ( ) H z t Tilt compensation filter γ t Control coefficient for the amount of the tilt compensation filtering µ γ = tk1' A tilt factor, with k1' being the first reflection coefficient ( ) h n f The truncated impulse response of the formant postfilter Lh The length of ( ) h n f r i h( ) The auto-correlations of ( ) h n f ( ) A z n γ The inverse filter (numerator) part of the formant postfilter ( ) 1 A z d γ The synthesis filter (denominator) part of the formant postfilter ( ) r n The residual signal of the inverse filter ( ) A z n γ ( ) h n t Impulse response of the tilt compensation filter β sc n ( ) The AGC-controlled gain scaling factor of the adaptive postfilter α The AGC factor of the adaptive postfilter ( ) H z h1 Pre-processing high-pass filter w n I ( ) , w n II ( ) LP analysis windows L I 1 ( ) Length of the first part of the LP analysis window w n I ( ) L I 2 ( ) Length of the second part of the LP analysis window w n I ( ) L II 1 ( ) Length of the first part of the LP analysis window w n II ( ) L II 2 ( ) Length of the second part of the LP analysis window w n II ( ) ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 10 TS 26.090 : December 1999 r k ac( ) The auto-correlations of the windowed speech s n '( ) ( ) w i lag Lag window for the auto-correlations (60 Hz bandwidth expansion) f0 The bandwidth expansion in Hz f s The sampling frequency in Hz r k ac ' ( ) The modified (bandwidth expanded) auto-correlations ( ) E i LD The prediction error in the ith iteration of the Levinson algorithm ki The ith reflection coefficient aj i( ) The jth direct form coefficient in the ith iteration of the Levinson algorithm ( ) ′ F z 1 Symmetric LSF polynomial ( ) ′ F z 2 Antisymmetric LSF polynomial ( ) F z 1 Polynomial ( ) ′ F z 1 with root z = −1 eliminated ( ) F z 2 Polynomial ( ) ′ F z 2 with root z = 1 eliminated qi The line spectral pairs (LSPs) in the cosine domain q An LSP vector in the cosine domain  ( ) qi n The quantified LSP vector at the ith subframe of the frame n ωi The line spectral frequencies (LSFs) T x m( ) A mth order Chebyshev polynomial f i f i 1 2 ( ), ( ) The coefficients of the polynomials F z 1( ) and F z 2( ) f i f i 1 2 ' ' ( ), ( ) The coefficients of the polynomials ( ) ′ F z 1 and ( ) ′ F z 2 f i( ) The coefficients of either ( ) F z 1 or ( ) F z 2 ( ) C x Sum polynomial of the Chebyshev polynomials x Cosine of angular frequency ω λ k Recursion coefficients for the Chebyshev polynomial evaluation fi The line spectral frequencies (LSFs) in Hz [ ] f t f f f = 1 2 10  The vector representation of the LSFs in Hz ( ) z( )1 n , ( ) z( ) 2 n The mean-removed LSF vectors at frame n ( ) r( )1 n , ( ) r( ) 2 n The LSF prediction residual vectors at frame n ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 11 TS 26.090 : December 1999 p( ) n The predicted LSF vector at frame n ( )  ( ) r 2 1 n − The quantified second residual vector at the past frame f k The quantified LSF vector at quantization index k ELSP The LSP quantization error w i i, , , , = 1 10  LSP-quantization weighting factors di The distance between the line spectral frequencies fi+1 and fi−1 ( ) h n The impulse response of the weighted synthesis filter Ok The correlation maximum of open-loop pitch analysis at delay k O i ti , , , =1 3  The correlation maxima at delays t i i, , , = 1 3  ( ) M t i i i , , , , =1 3  The normalized correlation maxima Mi and the corresponding delays t i i, , , = 1 3  H z W z A z A z A z ( ) ( ) ( / ) ( ) ( / ) = γ γ 1 2 The weighted synthesis filter ( ) A z γ 1 The numerator of the perceptual weighting filter ( ) 1 2 A z γ The denominator of the perceptual weighting filter T1 The integer nearest to the fractional pitch lag of the previous (1st or 3rd) subframe s n '( ) The windowed speech signal ( ) s n w The weighted speech signal ( ) s n Reconstructed speech signal ( ) ′s n The gain-scaled post-filtered signal ( ) s n f Post-filtered speech signal (before scaling) ( ) x n The target signal for adaptive codebook search ( ) x n 2 , x2 t The target signal for algebraic codebook search res n LP( ) The LP residual signal ( ) c n The fixed codebook vector ( ) v n The adaptive codebook vector y n v n h n ( ) = ( ) ( ) ∗ The filtered adaptive codebook vector ( ) y n k The past filtered excitation ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 12 TS 26.090 : December 1999 ( ) u n The excitation signal ( ) u n The emphasized adaptive codebook vector '( ) u n The gain-scaled emphasized excitation signal Top The best open-loop lag tmin Minimum lag search value tmax Maximum lag search value ( ) R k Correlation term to be maximized in the adaptive codebook search b24 The FIR filter for interpolating the normalized correlation term ( ) R k ( ) R k t The interpolated value of ( ) R k for the integer delay k and fraction t b60 The FIR filter for interpolating the past excitation signal ( ) u n to yield the adaptive codebook vector ( ) v n Ak Correlation term to be maximized in the algebraic codebook search at index k Ck The correlation in the numerator of Ak at index k EDk The energy in the denominator of Ak at index k d H x = t 2 The correlation between the target signal ( ) x n 2 and the impulse response ( ) h n , i.e., backward filtered target H The lower triangular Toepliz convolution matrix with diagonal ( ) h 0 and lower diagonals ( ) ( ) h h 1 39 , ,  Φ = H H t The matrix of correlations of ( ) h n d n ( ) The elements of the vector d φ( , ) i j The elements of the symmetric matrix Φ ck The innovation vector C The correlation in the numerator of Ak mi The position of the ith pulse ϑ i The amplitude of the ith pulse N p The number of pulses in the fixed codebook excitation ED The energy in the denominator of Ak ( ) res n LTP The normalized long-term prediction residual ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 13 TS 26.090 : December 1999 ( ) b n The signal used for presetting the signs in algebraic codebook search ( ) s n b The sign signal for the algebraic codebook search ( ) ′ d n Sign extended backward filtered target φ' ( , ) i j The modified elements of the matrix Φ , including sign information zt , ( ) z n The fixed codebook vector convolved with ( ) h n ( ) E n The mean-removed innovation energy (in dB) E The mean of the innovation energy ( ) ~E n The predicted energy [ ] b b b b 1 2 3 4 The MA prediction coefficients ( ) R k The quantified prediction error at subframe k EI The mean innovation energy R n ( ) The prediction error of the fixed-codebook gain quantization EQ The quantization error of the fixed-codebook gain quantization e n ( ) The states of the synthesis filter ( ) 1 A z ( ) e n w The perceptually weighted error of the analysis-by-synthesis search η The gain scaling factor for the emphasized excitation gc The fixed-codebook gain ′ gc The predicted fixed-codebook gain gc The quantified fixed codebook gain g p The adaptive codebook gain g p The quantified adaptive codebook gain γ gc c c g g = ′ A correction factor between the gain gc and the estimated one ′ gc γ gc The optimum value for γ gc γ sc Gain scaling factor ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 14 TS 26.090 : December 1999 3.3 Abbreviations For the purposes of this TS, the following abbreviations apply. ACELP Algebraic Code Excited Linear Prediction AGC Adaptive Gain Control AMR Adaptive Multi-Rate CELP Code Excited Linear Prediction EFR Enhanced Full Rate 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 TS 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 or µ -law samples. Sections 4.3 and 4.4 present a simplified description of the principles of the AMR codec encoding and decoding process respectively. In subclause 4.5, the sequence and subjective importance of encoded parameters are given. Section 5 presents the functional description of the AMR codec encoding, whereas clause 6 describes the decoding procedures. In section 7, the detailed bit allocation of the AMR codec is tabulated. 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 ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 15 TS 26.090 : December 1999 − 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 or µ -law compounded format, based on a standard A-law or µ -law codec/filter according to ITU-T Recommendations G.711 [6] 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 TS does not on its own ensure sufficient out-of-band attenuation. The specification of out-of-band signals is defined in [1] 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 or µ -law compressed data and linear data with 13-bit resolution at the speech encoder input shall be as defined in ITU-T Rec. G.711 [6]. ITU-T Rec. G.711 [6] specifies the A-Law or µ -law to linear conversion and vice versa by providing table entries. Examples on how to perform the conversion by fixed-point arithmetic can be found in ITU-T Rec. G.726 [7]. Section 4.2.1 of G.726 [7] describes A-Law or µ -law to linear expansion and subclause 4.2.8 of G.726 [7] provides a solution for linear to A-Law or µ -law compression. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 16 TS 26.090 : December 1999 4.3 Principles of the adaptive multi-rate speech encoder The AMR codec consists of eight source codecs with bit-rates of 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbit/s. The codec is based on the code-excited linear predictive (CELP) coding model. A 10th order linear prediction (LP), or short-term, synthesis filter is used which is given by: ( ) ( ) H z A z a z i i i m = = + − = ∑ 1 1 1 1   , (1) where  , , , , a i m i = 1  are the (quantified) linear prediction (LP) parameters, and m = 10 is the predictor order. The long-term, or pitch, synthesis filter is given by: ( ) 1 1 1 B z g z p T = − − , (2) where T is the pitch delay and g p is the pitch gain. The pitch synthesis filter is implemented using the so-called adaptive codebook approach. The CELP speech synthesis model is shown in figure 2. In this model, the excitation signal at the input of the short-term LP synthesis filter is constructed by adding two excitation vectors from adaptive and fixed (innovative) codebooks. The speech is synthesized by feeding the two properly chosen vectors from these codebooks through the short-term synthesis filter. The optimum excitation sequence in a codebook is chosen using an analysis-by-synthesis search procedure in which the error between the original and synthesized speech is minimized according to a perceptually weighted distortion measure. The perceptual weighting filter used in the analysis-by-synthesis search technique is given by: ( ) ( ) ( ) W z A z A z = γ γ 1 2 , (3) where ( ) A z is the unquantized LP filter and 0 1 2 1 < < ≤ γ γ are the perceptual weighting factors. The values γ 1 0 9 = . (for the 12.2 and 10.2 kbit/s mode) or 94 .0 1 = γ (for all other modes) and γ 2 0 6 = . are used. The weighting filter uses the unquantized LP parameters. The coder operates on speech frames of 20 ms corresponding to 160 samples at the sampling frequency of 8 000 sample/s. At each 160 speech samples, the speech signal is analysed to extract the parameters of the CELP model (LP filter coefficients, adaptive and fixed codebooks' indices and gains). These parameters are encoded and transmitted. At the decoder, these parameters are decoded and speech is synthesized by filtering the reconstructed excitation signal through the LP synthesis filter. The signal flow at the encoder is shown in figure 3. LP analysis is performed twice per frame for the 12.2 kbit/s mode and once for the other modes. For the 12.2 kbit/s mode, the two sets of LP parameters are converted to line spectrum pairs (LSP) and jointly quantized using split matrix quantization (SMQ) with 38 bits. For the other modes, the single set of LP parameters is converted to line spectrum pairs (LSP) and vector quantized using split vector quantization (SVQ). The speech frame is divided into 4 subframes of 5 ms each (40 samples). The adaptive and fixed codebook parameters are transmitted every subframe. The quantized and unquantized LP parameters or their interpolated versions are used depending on the subframe. An open-loop pitch lag is estimated in every other subframe (except for the 5.15 and 4.75 kbit/s modes for which it is done once per frame) based on the perceptually weighted speech signal. Then the following operations are repeated for each subframe: The target signal ( ) x n is computed by filtering the LP residual through the weighted synthesis filter ( ) ( ) W z H z with the initial states of the filters having been updated by filtering the error between LP ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 17 TS 26.090 : December 1999 residual and excitation (this is equivalent to the common approach of subtracting the zero input response of the weighted synthesis filter from the weighted speech signal). The impulse response, ( ) h n of the weighted synthesis filter is computed. Closed-loop pitch analysis is then performed (to find the pitch lag and gain), using the target ( ) x n and impulse response ( ) h n , by searching around the open-loop pitch lag. Fractional pitch with 1/6th or 1/3rd of a sample resolution (depending on the mode) is used. The target signal ( ) x n is updated by removing the adaptive codebook contribution (filtered adaptive codevector), and this new target, ( ) x n 2 , is used in the fixed algebraic codebook search (to find the optimum innovation). The gains of the adaptive and fixed codebook are scalar quantified with 4 and 5 bits respectively or vector quantified with 6-7 bits (with moving average (MA) prediction applied to the fixed codebook gain). Finally, the filter memories are updated (using the determined excitation signal) for finding the target signal in the next subframe. The bit allocation of the AMR codec modes is shown in table 1. In each 20 ms speech frame, 95, 103, 118, 134, 148, 159, 204 or 244 bits are produced, corresponding to a bit-rate of 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2 or 12.2 kbit/s. More detailed bit allocation among the codec parameters is given in tables 9a-9h. Note that the most significant bits (MSB) are always sent first. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 18 TS 26.090 : December 1999 Table 1: Bit allocation of the AMR coding algorithm for 20 ms frame Mode Parameter 1st subframe 2nd subframe 3rd subframe 4th subframe total per frame 2 LSP sets 38 12.2 kbit/s Pitch delay 9 6 9 6 30 (GSM EFR) Pitch gain 4 4 4 4 16 Algebraic code 35 35 35 35 140 Codebook gain 5 5 5 5 20 Total 244 LSP set 26 10.2 kbit/s Pitch delay 8 5 8 5 26 Algebraic code 31 31 31 31 124 Gains 7 7 7 7 28 Total 204 LSP sets 27 7.95 kbit/s Pitch delay 8 6 8 6 28 Pitch gain 4 4 4 4 16 Algebraic code 17 17 17 17 68 Codebook gain 5 5 5 5 20 Total 159 LSP set 26 7.40 kbit/s Pitch delay 8 5 8 5 26 (TDMA EFR) Algebraic code 17 17 17 17 68 Gains 7 7 7 7 28 Total 148 LSP set 26 6.70 kbit/s Pitch delay 8 4 8 4 24 (PDC EFR) Algebraic code 14 14 14 14 56 Gains 7 7 7 7 28 Total 134 LSP set 26 5.90 kbit/s Pitch delay 8 4 8 4 24 Algebraic code 11 11 11 11 44 Gains 6 6 6 6 24 Total 118 LSP set 23 5.15 kbit/s Pitch delay 8 4 4 4 20 Algebraic code 9 9 9 9 36 Gains 6 6 6 6 24 Total 103 LSP set 23 4.75 kbit/s Pitch delay 8 4 4 4 20 Algebraic code 9 9 9 9 36 Gains 8 8 16 Total 95 4.4 Principles of the adaptive multi-rate speech decoder The signal flow at the decoder is shown in figure 4. At the decoder, based on the chosen mode, the transmitted indices are extracted from the received bitstream. The indices are decoded to obtain the coder parameters at each transmission frame. These parameters are the LSP vectors, the fractional pitch lags, the innovative codevectors, and the pitch and innovative gains. The LSP vectors are converted to the LP filter coefficients and interpolated to obtain LP filters at each subframe. Then, at each 40-sample subframe: - the excitation is constructed by adding the adaptive and innovative codevectors scaled by their respective gains; - the speech is reconstructed by filtering the excitation through the LP synthesis filter. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 19 TS 26.090 : December 1999 Finally, the reconstructed speech signal is passed through an adaptive postfilter. 4.5 Sequence and subjective importance of encoded parameters The encoder will produce the output information in a unique sequence and format, and the decoder must receive the same information in the same way. In table 9a-9h, the sequence of output bits 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. The output and input frame formats for the AMR speech codec are given in [2], where a reordering of bits take place. 5 Functional description of the encoder In this clause, the different functions of the encoder represented in figure 3 are described. 5.1 Pre-processing (all modes) Two pre-processing functions are applied prior to the encoding process: high-pass filtering and signal down-scaling. Down-scaling consists of dividing the input by a factor of 2 to reduce the possibility of overflows in the fixed-point implementation. The high-pass filter serves as a precaution against undesired low frequency components. A filter with a cut off frequency of 80 Hz is used, and it is given by: 2 1 2 1 1 911376953 .0 906005859 .1 1 927246903 .0 8544941 .1 927246093 .0 ) ( − − − − + − + − = z z z z z H h . (4) Down-scaling and high-pass filtering are combined by dividing the coefficients at the numerator of ( ) H z h1 by 2. 5.2 Linear prediction analysis and quantization 12.2 kbit/s mode Short-term prediction, or linear prediction (LP), analysis is performed twice per speech frame using the auto-correlation approach with 30 ms asymmetric windows. No lookahead is used in the auto-correlation computation. The auto-correlations of windowed speech are converted to the LP coefficients using the Levinson-Durbin algorithm. Then the LP coefficients are transformed to the Line Spectral Pair (LSP) domain for quantization and interpolation purposes. The interpolated quantified and unquantized filter coefficients are converted back to the LP filter coefficients (to construct the synthesis and weighting filters at each subframe). 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes Short-term prediction, or linear prediction (LP), analysis is performed once per speech frame using the auto-correlation approach with 30 ms asymmetric windows. A lookahead of 40 samples (5 ms) is used in the auto-correlation computation. The auto-correlations of windowed speech are converted to the LP coefficients using the Levinson-Durbin algorithm. Then the LP coefficients are transformed to the Line Spectral Pair (LSP) domain for quantization and ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 20 TS 26.090 : December 1999 interpolation purposes. The interpolated quantified and unquantized filter coefficients are converted back to the LP filter coefficients (to construct the synthesis and weighting filters at each subframe). 5.2.1 Windowing and auto-correlation computation 12.2 kbit/s mode LP analysis is performed twice per frame using two different asymmetric windows. The first window has its weight concentrated at the second subframe and it consists of two halves of Hamming windows with different sizes. The window is given by: w n n L n L n L L n L L L I I I I I I I I ( ) . .46 , , , , . .46 ( ) , , , . ( ) ( ) ( ) ( ) ( ) ( ) ( ) = − −     = − + − −     = + −       0 54 0 1 0 1 0 54 0 1 1 1 1 1 2 1 1 2 cos cos π π   (5) The values L I 1 160 ( ) = and L I 2 80 ( ) = are used. The second window has its weight concentrated at the fourth subframe and it consists of two parts: the first part is half a Hamming window and the second part is a quarter of a cosine function cycle. The window is given by: w n n L n L n L L n L L L II II II II II II II II ( ) . .46 , , , , ( ) , , , ( ) ( ) ( ) ( ) ( ) ( ) ( ) = − −     = − − −     = + −       0 54 0 2 2 1 0 1 2 4 1 1 1 1 1 2 1 1 2 cos cos π π   (6) where the values L II 1 232 ( ) = and L II 2 8 ( ) = are used. Note that both LP analyses are performed on the same set of speech samples. The windows are applied to 80 samples from past speech frame in addition to the 160 samples of the present speech frame. No samples from future frames are used (no lookahead). A diagram of the two LP analysis windows is depicted below. 20 ms 5 ms frame (160 samples) sub frame (40 samples) frame n-1 frame n t I w (n) II w (n) Figure 1: LP analysis windows ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 21 TS 26.090 : December 1999 The auto-correlations of the windowed speech ( ) ′ = s n n , ,0 239  , are computed by: r k s n s n k k ac n k ( ) ' ( ) '( ) , , , , = − = =∑ 239 0 10  (7) and a 60 Hz bandwidth expansion is used by lag windowing the auto-correlations using the window: ( ) w i f i f i lag s = −             = exp , , 1 2 2 1 10 0 2 π  , (8) where f0 60 = Hz is the bandwidth expansion and fs = 8000 Hz is the sampling frequency. Further, rac( ) 0 is multiplied by the white noise correction factor 1.0001 which is equivalent to adding a noise floor at -40 dB. 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes LP analysis is performed once per frame using an asymmetric window. The window has its weight concentrated at the fourth subframe and it consists of two parts: the first part is half a Hamming window and the second part is a quarter of a cosine function cycle. The window is given by equation (6) where the values 200 1 = L and 40 2 = L are used. The auto-correlations of the windowed speech ( ) ′ = s n n , ,0 239  , are computed by equation (7) and a 60 Hz bandwidth expansion is used by lag windowing the auto-correlations using the window of equation (8). Further, rac( ) 0 is multiplied by the white noise correction factor 1.0001 which is equivalent to adding a noise floor at -40 dB. 5.2.2 Levinson-Durbin algorithm (all modes) The modified auto-correlations r r ac ac ' ( ) . ( ) 0 10001 0 = and r k r k w k k ac ac lag ' ( ) ( ) ( ), , , = = 1 10  are used to obtain the direct form LP filter coefficients a k k , , , , =1 10  by solving the set of equations. ( ) a r i k r i i k ac k ac ' ' ( ) , , , . − = − = =∑ 1 10 1 10  (9) The set of equations in (9) is solved using the Levinson-Durbin algorithm. This algorithm uses the following recursion: [ ] E r i a k a r i j E i a k j i a a k a E i k E i LD ac i i j i ac j i LD i i i j i j i i i j i LD i LD ( ) ' ( ) ' ( ) / ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) 0 0 1 10 1 1 1 1 1 1 0 1 1 0 1 1 1 2 = = = = − − − = = − = + = − − − − = − − − − ∑ for to do for to do end end The final solution is given as a a j j j = = ( ), , , 10 1 10  . ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 22 TS 26.090 : December 1999 The LP filter coefficients are converted to the line spectral pair (LSP) representation for quantization and interpolation purposes. The conversions to the LSP domain and back to the LP filter coefficient domain are described in the next clause. 5.2.3 LP to LSP conversion (all modes) The LP filter coefficients a k k , , , =1 10  , are converted to the line spectral pair (LSP) representation for quantization and interpolation purposes. For a 10th order LP filter, the LSPs are defined as the roots of the sum and difference polynomials: ( ) ( ) ( ) ′ = + − − F z A z z A z 1 11 1 (10) and ( ) ( ) ( ) ′ = − − − F z A z z A z 2 11 1 , (11) respectively. The polynomial ( ) ′ F z 1 and ( ) ′ F z 2 are symmetric and anti-symmetric, respectively. It can be proven that all roots of these polynomials are on the unit circle and they alternate each other. ( ) ′ F z 1 has a root z = −1 (ω π = ) and ( ) ′ F z 2 has a root z = 1 (ω = 0 ). To eliminate these two roots, we define the new polynomials: ( ) ( ) ( ) F z F z z 1 1 1 1 = ′ + − (12) and ( ) ( ) ( ) F z F z z 2 2 1 1 = ′ − − (13) Each polynomial has 5 conjugate roots on the unit circle ( ) e j i ± ω , therefore, the polynomials can be written as ( ) ( ) F z q z z i i 1 1 2 1 3 9 1 2 = − + − − =∏ , , ,  (14) and ( ) ( ) F z q z z i i 2 1 2 2 4 10 1 2 = − + − − = ∏ , , ,  , (15) where ( ) qi i = cos ω with ω i being the line spectral frequencies (LSF) and they satisfy the ordering property 0 1 2 10 < < < < < ω ω ω π  . We refer to qi as the LSPs in the cosine domain. Since both polynomials ( ) F z 1 and ( ) F z 2 are symmetric only the first 5 coefficients of each polynomial need to be computed. The coefficients of these polynomials are found by the recursive relations (for i = 0 to 4): ( ) ( ) ( ) ( ) f i a a f i f i a a f i i m i i m i 1 1 1 2 1 2 1 1 + = + − + = − + + − + − (16) where m = 10 is the predictor order. The LSPs are found by evaluating the polynomials ( ) F z 1 and ( ) F z 2 at 60 points equally spaced between 0 and π and checking for sign changes. A sign change signifies the existence of a root and the sign change interval is then divided 4 times to better track the root. The Chebyshev polynomials are used to evaluate ( ) F z 1 and ( ) F z 2 . In this ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 23 TS 26.090 : December 1999 method the roots are found directly in the cosine domain { } qi . The polynomials ( ) F z 1 or ( ) F z 2 evaluated at z e j = ω can be written as: ( ) ( ) F e C x j ω ω = − 2 5 , with: ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) C x T x f T x f T x f T x f T x f = + + + + + 5 4 3 2 1 1 2 3 4 5 2 , (17) where ( ) ( ) T x m m = cos ω is the mth order Chebyshev polynomial, and ( ) f i i, , , = 1 5  are the coefficients of either ( ) F z 1 or ( ) F z 2 , computed using the equations in (16). The polynomial ( ) C x is evaluated at a certain value of ( ) x = cos ω using the recursive relation: for down to end k x f k C x x f k k k = − + − = − + = + + 4 1 2 5 5 2 1 2 1 2 λ λ λ λ λ ( ) ( ) ( ) / , with initial values λ5 1 = and λ6 0 = . The details of the Chebyshev polynomial evaluation method are found in P. Kabal and R.P. Ramachandran [4]. 5.2.4 LSP to LP conversion (all modes) Once the LSPs are quantified and interpolated, they are converted back to the LP coefficient domain { } ak . The conversion to the LP domain is done as follows. The coefficients of ( ) F z 1 or ( ) F z 2 are found by expanding equations (14) and (15) knowing the quantified and interpolated LSPs q i i , = , , 1 10  . The following recursive relation is used to compute ( ) f i 1 : ( ) ( ) ( ) ( ) ( ) ( ) ( ) for to for down to end end i f i q f i f i j i f j f j q f j f j i i = = − − + − = − = − − + − − − 1 5 2 1 2 2 1 1 2 1 2 1 2 1 1 1 1 1 2 1 1 1 with initial values ( ) f1 0 1 = and ( ) f1 1 0 − = . The coefficients ( ) f i 2 are computed similarly by replacing q i2 1 − by q i2 . Once the coefficients ( ) f i 1 and ( ) f i 2 are found, ( ) F z 1 and ( ) F z 2 are multiplied by 1 1 + − z and 1 1 − − z , respectively, to obtain ( ) ′ F z 1 and ( ) ′ F z 2 ; that is: ( ) ( ) ( ) ( ) ( ) ( ) ′ = + − = ′ = − − = f i f i f i i f i f i f i i 1 1 1 2 2 2 1 1 5 1 1 5 , , , , , ,   . (18) Finally the LP coefficients are found by: ( ) ( ) ( ) ( ) a f i f i i f i f i i i = ′ + ′ = ′ − − ′ − =    05 05 1 5 05 11 05 11 6 10 1 2 1 2 . . , , , . . , , ,   . (19) ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 24 TS 26.090 : December 1999 This is directly derived from the relation ( ) ( ) ( ) ( ) A z F z F z = ′ + ′ 1 2 2 , and considering the fact that ( ) ′ F z 1 and ( ) ′ F z 2 are symmetric and anti-symmetric polynomials, respectively. 5.2.5 Quantization of the LSP coefficients 12.2 kbit/s mode The two sets of LP filter coefficients per frame are quantified using the LSP representation in the frequency domain; that is: ( ) f f q i i s i = = 2 1 10 π arccos , , , ,  (20) where fi are the line spectral frequencies (LSF) in Hz [0,4000] and fs =8000 is the sampling frequency. The LSF vector is given by [ ] f t f f f = 1 2 10  , with t denoting transpose. A 1st order MA prediction is applied, and the two residual LSF vectors are jointly quantified using split matrix quantization (SMQ). The prediction and quantization are performed as follows. Let ( ) z(1) n and ( ) z(2) n denote the mean-removed LSF vectors at frame n . The prediction residual vectors ( ) r(1) n and ( ) r(2) n are given by: ( ) ( ) ( ) ( ) ( ) ( ) r z p r z p ( ) ( ) ( ) ( ) , , 1 1 2 2 n n n n n n = − = − and (21) where p( ) n is the predicted LSF vector at frame n . First order moving-average (MA) prediction is used where: ( ) ( ) p r n n = − 0 65 1 2 .  ( ) , (22) where ( )  ( ) r 2 1 n − is the quantified second residual vector at the past frame. The two LSF residual vectors r(1) and r(2) are jointly quantified using split matrix quantization (SMQ). The matrix ( ) r r (1) (2) is split into 5 submatrices of dimension 2 x 2 (two elements from each vector). For example, the first submatrix consists of the elements r1 1 ( ) , r2 1 ( ) , r1 2 ( ) , and r2 2 ( ) . The 5 submatrices are quantified with 7, 8, 8+1, 8, and 6 bits, respectively. The third submatrix uses a 256-entry signed codebook (8-bit index plus 1-bit sign). A weighted LSP distortion measure is used in the quantization process. In general, for an input LSP vector f and a quantified vector at index k , f k , the quantization is performed by finding the index k which minimizes: [ ] E f w f w LSP i i i k i i = − =∑  . 1 10 2 (23) The weighting factors w i i, , , =1 10  , are given by ( ) otherwise, 450 1050 0.8 - 1.8 = , 450 for 450 547 .1 347 .3 − < − = i i i i d d d w (24) ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 25 TS 26.090 : December 1999 where d f f i i i = − + − 1 1 with f0 0 = and f11 4000 = . Here, two sets of weighting coefficients are computed for the two LSF vectors. In the quantization of each submatrix, two weighting coefficients from each set are used with their corresponding LSFs. 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes The set of LP filter coefficients per frame is quantified using the LSP representation in the frequency domain using equation (20). A 1st order MA prediction is applied, and the residual LSF vector is quantified using split vector quantization. The prediction and quantization are performed as follows. Let ) (n z denote the mean-removed LSF vectors at frame n . The prediction residual vectors ) (n r is given by: ) ( ) ( ) ( n n n p z r − = (25) where p( ) n is the predicted LSF vector at frame n . First order moving-average (MA) prediction is used where: ( ) 10 , ,1 1 ˆ ) (  = − = j n r n p j j j α , (26) where )1 (ˆ − n r is the quantified residual vector at the past frame and j α is the prediction factor for the jth LSF. The LSF residual vectors r is quantified using split vector quantization. The vector r is split into 3 subvectors of dimension 3, 3, and 4. The 3 subvectors are quantified with 7-9 bits according to table 2. Table 2. Bit allocation split vector quantization of LSF residual vector. Mode Subvector 1 Subvector 2 Subvector 3 10.2 kbit/s 8 9 9 7.95 kbit/s 9 9 9 7.40 kbit/s 8 9 9 6.70 kbit/s 8 9 9 5.90 kbit/s 8 9 9 5.15 kbit/s 8 8 7 4.75 kbit/s 8 8 7 The weighted LSP distortion measure of equation (23) with the weighting of equation (24) is used in the quantization process. 5.2.6 Interpolation of the LSPs 12.2 kbit/s mode The two sets of quantified (and unquantized) LP parameters are used for the second and fourth subframes whereas the first and third subframes use a linear interpolation of the parameters in the adjacent subframes. The interpolation is performed on the LSPs in the q domain. Let  ( ) q4 n be the LSP vector at the 4th subframe of the present frame n ,  ( ) q2 n be the LSP vector at the 2nd subframe of the present frame n , and  ( ) q4 1 n− the LSP vector at the 4th subframe of the past frame n−1. The interpolated LSP vectors at the 1st and 3rd subframes are given by:  .  .  ,  .  .  . ( ) ( ) ( ) ( ) ( ) ( ) q q q q q q 1 4 1 2 3 2 4 05 05 05 05 n n n n n n = + = + − (27) The interpolated LSP vectors are used to compute a different LP filter at each subframe (both quantified and unquantized coefficients) using the LSP to LP conversion method described in subclause 5.2.4. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 26 TS 26.090 : December 1999 10.2, 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes The set of quantified (and unquantized) LP parameters is used for the fourth subframe whereas the first, second, and third subframes use a linear interpolation of the parameters in the adjacent subframes. The interpolation is performed on the LSPs in the q domain. The interpolated LSP vectors at the 1st, 2nd, and 3rd subframes are given by:  .  .  ,  .  .  ,  .  .  . ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) ( ) q q q q q q q q q 1 4 1 4 2 4 1 4 2 4 1 4 0 75 0 25 05 05 0 25 0 75 n n n n n n n n n = + = + = + − − − (28) The interpolated LSP vectors are used to compute a different LP filter at each subframe (both quantified and unquantized coefficients) using the LSP to LP conversion method described in subclause 5.2.4. 5.2.7 Monitoring resonance in the LPC spectrum (all modes) Resonances in the LPC filter are monitored to detect possible problem areas where divergence between the adaptive codebook memories in the encoder and the decoder could cause unstable filters in areas with highly correlated continuos signals. Typically, this divergence is due to channel errors. The monitoring of resonance signals is performed using unquantized LSPs q i i, ,..., = 1 10 . The LSPs are available after the LP to LSP conversion in section 5.2.3. The algorithm utilises the fact that LSPs are closely located at a peak in the spectrum. First, two distances, dist1 and dist 2 , are calculated in two different regions, defined as dist q q i i i 1 1 4 8 = − = + min( ), ,..., , and dist q q i i i 2 1 2 3 = − = + min( ), , . Either of these two minimum distance conditions must be fulfilled to classify the frame as a resonance frame and increase the resonance counter. if dist TH OR if dist TH counter counter else counter ( ) ( ) 1 1 2 2 1 0 < < = + = TH 1 0 046 = . is a fixed threshold while the second one is depending on q 2 according to: TH q q otherwise 2 2 2 0 018 0 98 0 024 0 93 0 98 0 034 = > < ≤    . , . . , . . . , 12 consecutive resonance frames are needed to indicate possible problem conditions, otherwise the LSP_flag is cleared. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 27 TS 26.090 : December 1999 if counter counter LSP flag else LSP flag ( ) _ _ ≥ = = = 12 12 1 0 5.3 Open-loop pitch analysis Open-loop pitch analysis is performed in order to simplify the pitch analysis and confine the closed-loop pitch search to a small number of lags around the open-loop estimated lags. Open-loop pitch estimation is based on the weighted speech signal ( ) s n w which is obtained by filtering the input speech signal through the weighting filter ( ) ( ) ( ) W z A z A z = γ γ 1 2 . That is, in a subframe of size L , the weighted speech is given by: ( ) ( ) ( ) ( ) s n s n a s n i a s n i n L w i i i i i w i = + − − − = − = = ∑ ∑ γ γ 1 1 10 2 1 10 0 1 , , ,  (29) 12.2 kbit/s mode Open-loop pitch analysis is performed twice per frame (each 10 ms) to find two estimates of the pitch lag in each frame. Open-loop pitch analysis is performed as follows. In the first step, 3 maxima of the correlation: O s n s n k k w w n = − =∑ ( ) ( ) 0 79 (30) are found in the three ranges: i i i = = = 3 2 1 : : : 18 35 36 71 72 143 , , , , , , , , .    The retained maxima O i ti , , , =1 3  , are normalized by dividing by s n t i w i n 2 ( ), − = ∑ 1, ,3  , respectively. The normalized maxima and corresponding delays are denoted by ( ) M t i i i , , , , =1 3  . The winner, Top , among the three normalized correlations is selected by favouring the delays with the values in the lower range. This is performed by weighting the normalized correlations corresponding to the longer delays. The best open-loop delay Top is determined as follows: ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 28 TS 26.090 : December 1999 ( ) ( ) ( ) ( ) ( ) T t M T M if M M T M T M T t end if M M T M T M T t end op op op op op op op op = = > = = > = = 1 1 2 2 2 3 3 3 0 85 0 85 . . This procedure of dividing the delay range into 3 clauses and favouring the lower clauses is used to avoid choosing pitch multiples. 10.2 kbit/s mode Open-loop pitch analysis is performed twice per frame (every 10 ms) to find two estimates of the pitch lag in each frame. The open-loop pitch analysis is performed as follows. First, the correlation of weighted speech is determined for each pitch lag value d by: ( ) ( ) ( ) ( ) C d s n s n d w d d w w n = − = =∑ 0 79 20 143 , , ,  , (31) where ( ) w d is a weighting function. The estimated pitch-lag is the delay that maximises the weighted correlation function ( ) C d . The weighting emphasises lower pitch lag values reducing the likelihood of selecting a multiple of the correct delay. The weighting function consists of two parts: a low pitch lag emphasis function, ( ) w d l , and a previous frame lag neighbouring emphasis function, ( ) w d n : ( ) ( ) ( ) w d w d w d l n = . (32) The low pitch lag emphasis function is a given by: ( ) ( ) w d cw d l = (33) where ( ) cw d is defined by a table in the fixed point computational computational description (ANSI-C code) in [4]. The previous frame lag neighbouring emphasis function depends on the pitch lag of previous speech frames: ( ) ( ) w d cw T d d v n old L = − + >    , . , . , 0 3 10 otherwise, (34) where d L = 20 , Told is the median filtered pitch lag of 5 previous voiced speech half-frames, and v is an adaptive parameter. If the frame is classified as voiced by having the open-loop gain g > 04. , the v-value is set to 1.0 for the next frame. Otherwise, the v-value is updated by v v = 0 9. . The open loop gain is given by: ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 29 TS 26.090 : December 1999 ( ) ( ) ( ) g s n s n d s n w w n w n = − = = ∑ ∑ max 0 79 2 0 79 (35) where dmax is the pitch delay that maximizes ( ) C d . The median filter is updated only during voiced speech frames. The weighting depends on the reliability of the old pitch lags. If previous frames have contained unvoiced speech or silence, the weighting is attenuated through the parameter v. 7.95, 7.40, 6.70, 5.90 kbit/s modes Open-loop pitch analysis is performed twice per frame (each 10 ms) to find two estimates of the pitch lag in each frame. Open-loop pitch analysis is performed as follows. In the first step, 3 maxima of the correlation in equation (30) are found in the three ranges: i i i = = = 3 2 1 : : : . 143 , , 80 , 79 , , 40 , 39 , , 20    The retained maxima O i ti , , , =1 3  , are normalized by dividing by s n t i w i n 2 ( ), − = ∑ 1, ,3  , respectively. The normalized maxima and corresponding delays are denoted by ( ) M t i i i , , , , =1 3  . The winner, Top , among the three normalized correlations is selected by favouring the delays with the values in the lower range. This is performed by weighting the normalized correlations corresponding to the longer delays. The best open-loop delay Top is determined as follows: ( ) ( ) ( ) ( ) ( ) T t M T M if M M T M T M T t end if M M T M T M T t end op op op op op op op op = = > = = > = = 1 1 2 2 2 3 3 3 0 85 0 85 . . This procedure of dividing the delay range into 3 clauses and favouring the lower clauses is used to avoid choosing pitch multiples. 5.15, 4.75 kbit/s modes Open-loop pitch analysis is performed once per frame (each 20 ms) to find an estimate of the pitch lag in each frame. Open-loop pitch analysis is performed as follows. In the first step, 3 maxima of the correlation in equation (30) are found in the three ranges: ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 30 TS 26.090 : December 1999 i i i = = = 3 2 1 : : : . 143 , , 79 , 79 , , 40 , 39 , , 20    The retained maxima O i ti , , , =1 3  , are normalized by dividing by s n t i w i n 2 ( ), − = ∑ 1, ,3  , respectively. The normalized maxima and corresponding delays are denoted by ( ) M t i i i , , , , =1 3  . The winner, Top , among the three normalized correlations is selected by favouring the delays with the values in the lower range. This is performed by weighting the normalized correlations corresponding to the longer delays. The best open-loop delay Top is determined as follows: ( ) ( ) ( ) ( ) ( ) T t M T M if M M T M T M T t end if M M T M T M T t end op op op op op op op op = = > = = > = = 1 1 2 2 2 3 3 3 0 85 0 85 . . This procedure of dividing the delay range into 3 clauses and favouring the lower clauses is used to avoid choosing pitch multiples. 5.4 Impulse response computation (all modes) The impulse response, ( ) h n , of the weighted synthesis filter ( ) ( ) ( ) ( ) ( ) [ ] H z W z A z A z A z = γ γ 1 2  is computed each subframe. This impulse response is needed for the search of adaptive and fixed codebooks. The impulse response ( ) h n is computed by filtering the vector of coefficients of the filter ( ) A z γ 1 extended by zeros through the two filters ( ) 1 A z and ( ) 1 2 A z γ . 5.5 Target signal computation (all modes) The target signal for adaptive codebook search is usually computed by subtracting the zero input response of the weighted synthesis filter ( ) ( ) ( ) ( ) ( ) [ ] H z W z A z A z A z = γ γ 1 2  from the weighted speech signal ( ) s n w . This is performed on a subframe basis. An equivalent procedure for computing the target signal, which is used in this standard, is the filtering of the LP residual signal res n LP( ) through the combination of synthesis filter ( ) 1 A z and the weighting filter ( ) ( ) A z A z γ γ 1 2 . After determining the excitation for the subframe, the initial states of these filters are updated by filtering the difference between the LP residual and excitation. The memory update of these filters is explained in subclause 5.9. The residual signal res n LP( ) which is needed for finding the target vector is also used in the adaptive codebook search to extend the past excitation buffer. This simplifies the adaptive codebook search procedure for delays less than the subframe size of 40 as will be explained in the next clause. The LP residual is given by: ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 31 TS 26.090 : December 1999 res n s n a s n i LP i i ( ) ( )  ( ). = + − =∑ 1 10 (36) 5.6 Adaptive codebook 5.6.1 Adaptive codebook search Adaptive codebook search is performed on a subframe basis. It consists of performing closed-loop pitch search, and then computing the adaptive codevector by interpolating the past excitation at the selected fractional pitch lag. The adaptive codebook parameters (or pitch parameters) are the delay and gain of the pitch filter. In the adaptive codebook approach for implementing the pitch filter, the excitation is repeated for delays less than the subframe length. In the search stage, the excitation is extended by the LP residual to simplify the closed-loop search. 12.2 kbit/s mode In the first and third subframes, a fractional pitch delay is used with resolutions: 1/6 in the range [ ] 6 3 94 , 6 3 17 and integers only in the range [95, 143]. For the second and fourth subframes, a pitch resolution of 1/6 is always used in the range [ ] 6 3 4 , 6 3 5 1 1 + − T T , where T1 is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by 18...143. Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range Top ±3, bounded by 18...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 9 bits in the first and third subframes and the relative delay of the other subframes is encoded with 6 bits. The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term: ( ) R k x n y n y n y n k n k k n = = = ∑ ∑ ( ) ( ) ( ) ( ) , 0 39 0 39 (37) where ( ) x n is the target signal and ( ) y n k is the past filtered excitation at delay k (past excitation convolved with ( ) h n ). Note that the search range is limited around the open-loop pitch as explained earlier. The convolution ( ) y n k is computed for the first delay tmin in the searched range, and for the other delays in the search range k t t = + min max , , 1  , it is updated using the recursive relation: ( ) ( ) ( ) ( ) y n y n u k h n k k = − + − −1 1 , (38) where ( ) ( ) u n n , , , = − + 143 11 39  , is the excitation buffer. Note that in search stage, the samples ( ) u n n , , , = 0 39  , are not known, and they are needed for pitch delays less than 40. To simplify the search, the LP residual is copied to ( ) u n in order to make the relation in equation (38) valid for all delays. Once the optimum integer pitch delay is determined, the fractions from –3/6 to 3/6 with a step of 1/6 around that integer are tested. The fractional pitch search is performed by interpolating the normalized correlation in equation (37) and searching for its maximum. The interpolation is performed using an FIR filter b24 based on a Hamming windowed ( ) sin x x function truncated at ± 23 and padded with zeros at ± 24 ( ( ) b24 24 0 = ). The filter has its ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 32 TS 26.090 : December 1999 cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. The interpolated values of ( ) R k for the fractions –3/6 to 3/6 are obtained using the interpolation formula: ( ) ( ) ( ) ( ) ( ) R k R k i b t i R k i b t i t t i i = − + ⋅ + + + −+ ⋅ = = = ∑ ∑ 24 0 3 0 3 24 6 1 6 6 0 5 , , , ,  (39) where t = 0 5 , ,  corresponds to the fractions 0, 1/6, 2/6, 3/6, -2/6, and –1/6, respectively. Note that it is necessary to compute the correlation terms in equation (37) using a range t t min max , , − + 4 4 to allow for the proper interpolation. Once the fractional pitch lag is determined, the adaptive codebook vector ( ) v n is computed by interpolating the past excitation signal ( ) u n at the given integer delay k and phase (fraction) t : ( ) ( ) ( ) ( ) ( ) v n u n k i b t i u n k i b t i n t i i = − − + ⋅ + − + + −+ ⋅ = = = = ∑ ∑ 60 0 9 0 9 60 6 1 6 6 0 39 0 5 , , , , , , .   (40) The interpolation filter b60 is based on a Hamming windowed ( ) sin x x function truncated at ± 59 and padded with zeros at ± 60 ( ( ) b60 60 0 = ). The filter has a cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. The adaptive codebook gain is then found by: g x n y n y n y n g p n n p = ≤ ≤ = = ∑ ∑ ( ) ( ) ( ) ( ) , . 0 39 0 39 0 12 bounded by (41) where ( ) ( ) ( ) y n v n h n = ∗ is the filtered adaptive codebook vector (zero state response of ( ) ( ) H z W z to ( ) v n ). The computed adaptive codebook gain is quantified using 4-bit non-uniform scalar quantization in the range [0.0,1.2]. 7.95 kbit/s mode In the first and third subframes, a fractional pitch delay is used with resolutions: 1/3 in the range [ ] 3 2 84 , 3 1 19 and integers only in the range [85, 143]. For the second and fourth subframes, a pitch resolution of 1/3 is always used in the range [ ] T T 1 1 10 2 3 9 2 3 − + , , where T1 is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by 20...143. Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range Top ± 3, bounded by 20...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 8 bits in the first and third subframes and the relative delay of the other subframes is encoded with 6 bits. The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. The convolution ( ) y n k is computed for the first delay tmin in the searched range, and for the other delays in the search range k t t = + min max , , 1  , it is updated using the recursive relation of equation (38). Once the optimum integer pitch delay is determined, the fractions from –2/3 to 2/3 with a step of 1/3 around that integer are tested. The fractional pitch search is performed by interpolatingthe normalized correlation in equation ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 33 TS 26.090 : December 1999 (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector ( ) v n is computed by interpolating the past excitation signal ( ) u n at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at ± 11 and the other for interpolating the past excitation with the sinc truncated at ± 29. The filters have their cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. The adaptive codebook gain is then found as in equation (41). The computed adaptive codebook gain is quantified using 4-bit non-uniform scalar quantization as described in section 5.8. 10.2, 7.40 kbit/s mode In the first and third subframes, a fractional pitch delay is used with resolutions: 1/3 in the range [ ] 191 3 84 2 3 , and integers only in the range [85, 143]. For the second and fourth subframes, a pitch resolution of 1/3 is always used in the range [ ] T T 1 1 52 3 4 2 3 − + , , where T1 is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by 20...143. Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range Top ± 3, bounded by 20...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 8 bits in the first and third subframes and the relative delay of the other subframes is encoded with 5 bits. The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. The convolution ( ) y n k is computed for the first delay tmin in the searched range, and for the other delays in the search range k t t = + min max , , 1  , it is updated using the recursive relation of equation (38). Once the optimum integer pitch delay is determined, the fractions from –2/3 to 2/3 with a step of 1/3 around that integer are tested. The fractional pitch search is performed by interpolatingthe normalized correlation in equation (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector ( ) v n is computed by interpolating the past excitation signal ( ) u n at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at ± 11 and the other for interpolating the past excitation with the sinc truncated at ± 29. The filters have their cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. The adaptive codebook gain is then found as in equation (41). The computed adaptive codebook gain (and the fixed codebook gain) is quantified using 7-bit non-uniform vector quantization as described in section 5.8. 6.70, 5.90 kbit/s modes In the first and third subframes, a fractional pitch delay is used with resolutions: 1/3 in the range [ ] 191 3 84 2 3 , and integers only in the range [85, 143]. For the second and fourth subframes, integer pitch resolution is used in the range [ ] T T 1 1 5 4 − + , , where T1 is nearest integer to the fractional pitch lag of the previous (1st or 3rd) subframe, bounded by 20...143. Additionally, a fractional resolution of 1/3 is used in the range [ ] T T 1 1 12 3 2 3 − + , . Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first (and third) subframe the range Top ± 3, bounded by 20...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 34 TS 26.090 : December 1999 is encoded with 8 bits in the first and third subframes and the relative delay of the other subframes is encoded with 4 bits. The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. The convolution ( ) y n k is computed for the first delay tmin in the searched range, and for the other delays in the search range k t t = + min max , , 1  , it is updated using the recursive relation of equation (38). Once the optimum integer pitch delay is determined, the fractions from –2/3 to 2/3 with a step of 1/3 around that integer are tested. The fractional pitch search is performed by interpolatingthe normalized correlation in equation (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector ( ) v n is computed by interpolating the past excitation signal ( ) u n at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at ± 11 and the other for interpolating the past excitation with the sinc truncated at ± 29. The filters have their cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. The adaptive codebook gain is then found as in equation (41). The computed adaptive codebook gain (and the fixed codebook gain) is quantified using vector quantization as described in section 5.8. 5.15, 4.75 kbit/s modes In the first subframe, a fractional pitch delay is used with resolutions: 1/3 in the range [ ] 191 3 84 2 3 , and integers only in the range [85, 143]. For the second, third, and fourth subframes, integer pitch resolution is used in the range [ ] T T 1 1 5 4 − + , , where T1 is nearest integer to the fractional pitch lag of the previous subframe, bounded by 20...143. Additionally, a fractional resolution of 1/3 is used in the range [ ] T T 1 1 12 3 2 3 − + , . Closed-loop pitch analysis is performed around the open-loop pitch estimates on a subframe basis. In the first subframe the range Top ± 5, bounded by 20...143, is searched. For the other subframes, closed-loop pitch analysis is performed around the integer pitch selected in the previous subframe, as described above. The pitch delay is encoded with 8 bits in the first subframe and the relative delay of the other subframes is encoded with 4 bits. The closed-loop pitch search is performed by minimizing the mean-square weighted error between the original and synthesized speech. This is achieved by maximizing the term of equation (37). Note that the search range is limited around the open-loop pitch as explained earlier. The convolution ( ) y n k is computed for the first delay tmin in the searched range, and for the other delays in the search range k t t = + min max , , 1  , it is updated using the recursive relation of equation (38). Once the optimum integer pitch delay is determined, the fractions from –2/3 to 2/3 with a step of 1/3 around that integer are tested. The fractional pitch search is performed by interpolatingthe normalized correlation in equation (37) and searching for its maximum. Once the fractional pitch lag is determined, the adaptive codebook vector ( ) v n is computed by interpolating the past excitation signal ( ) u n at the given integer delay and phase (fraction). The interpolation is performed using two FIR filters (Hamming windowed sinc functions); one for interpolating the term in equation (37) with the sinc truncated at ± 11 and the other for interpolating the past excitation with the sinc truncated at ± 29. The filters have their cut-off frequency (-3 dB) at 3 600 Hz in the over-sampled domain. The adaptive codebook gain is then found as in equation (41). The computed adaptive codebook gain (and the fixed codebook gain) is quantified using vector quantization as described in section 5.8. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 35 TS 26.090 : December 1999 5.6.2 Adaptive codebook gain control (all modes) The average adaptive codebook gain is calculated if the LSP_flag is set and the unquantized adaptive codebook gain exceeds the gain threshold GPth= 0 95 . . The average gain is calculated from the present unquantized gain and the quantized gains of the seven previous subframes. That is, { } GP mean g n g n g n g n ave p p p p = − − − ( ),  ( ),  ( ),...,  ( ) 1 2 7 , where n is the current subframe. If the average adaptive codebook gain exceeds the GP th , the unquantized gain is limited to the threshold value and the GpC_flag is set to indicate the limitation. if GP GP g GP GpC flag else GpC flag ave th p th ( ) _ _ > = = = 1 0 The GpC_flag is used in the gain quantization in section 5.8. 5.7 Algebraic codebook 5.7.1 Algebraic codebook structure The algebraic codebook structure is based on interleaved single-pulse permutation (ISPP) design. 12.2 kbit/s mode In this codebook, the innovation vector contains 10 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 5 tracks, where each track contains two pulses, as shown in table 3. Table 3: Potential positions of individual pulses in the algebraic codebook, 12.2 kbit/s. Track Pulse Positions 1 i0, i5 0, 5, 10, 15, 20, 25, 30, 35 2 i1, i6 1, 6, 11, 16, 21, 26, 31, 36 3 i2, i7 2, 7, 12, 17, 22, 27, 32, 37 4 i3, i8 3, 8, 13, 18, 23, 28, 33, 38 5 i4, i9 4, 9, 14, 19, 24, 29, 34, 39 Each two pulse positions in one track are encoded with 6 bits (total of 30 bits, 3 bits for the position of every pulse), and the sign of the first pulse in the track is encoded with 1 bit (total of 5 bits). For two pulses located in the same track, only one sign bit is needed. This sign bit indicates the sign of the first pulse. The sign of the second pulse depends on its position relative to the first pulse. If the position of the second pulse is smaller, then it has opposite sign, otherwise it has the same sign than in the first pulse. All the 3-bit pulse positions are Gray coded in order to improve robustness against channel errors. This gives a total of 35 bits for the algebraic code. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 36 TS 26.090 : December 1999 10.2 kbit/s mode In this codebook, the innovation vector contains 8 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 4 tracks, where each track contains two pulses, as shown in table 4. Table 4: Potential positions of individual pulses in the algebraic codebook, 10.2 kbit/s. Track Pulse Positions 1 i0, i4 0, 4, 8, 12, 16, 20, 24, 28, 32, 36 2 i1, i5 1, 5, 9, 13, 17, 21, 25, 29, 33, 37 3 i2, i6 2, 6, 10, 14, 18, 22, 26, 30, 34, 38 4 i3, i7 3, 7, 11, 15, 19, 23, 27, 31, 35, 39 The pulses are grouped into 3, 3, and 2 pulses and their positions are encoded with 10, 10, and 7 bits, respectively (total of 27 bits). The sign of the first pulse in each track is encoded with 1 bit (total of 4 bits). For two pulses located in the same track, only one sign bit is needed. This sign bit indicates the sign of the first pulse. The sign of the second pulse depends on its position relative to the first pulse. If the position of the second pulse is smaller, then it has opposite sign, otherwise it has the same sign than in the first pulse. This gives a total of 31 bits for the algebraic code. 7.95, 7.40 kbit/s modes In this codebook, the innovation vector contains 4 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 4 tracks, where each track contains one pulse, as shown in table 5. Table 5: Potential positions of individual pulses in the algebraic codebook, 7.95, 7.40 kbit/s. Track Pulse Positions 1 i0 0, 5, 10, 15, 20, 25, 30, 35 2 i1 1, 6, 11, 16, 21, 26, 31, 36 3 i2 2, 7, 12, 17, 22, 27, 32, 37 4 i3 3, 8, 13, 18, 23, 28, 33, 38, 4, 9, 14, 19, 24, 29, 34, 39 The pulse positions are encoded with 3, 3, 3, and 4 bits (total of 13 bits), and the sign of the each pulse is encoded with 1 bit (total of 4 bits). This gives a total of 17 bits for the algebraic code. 6.70 kbit/s mode In this codebook, the innovation vector contains 3 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 3 tracks, where each track contains one pulse, as shown in table 6. Table 6: Potential positions of individual pulses in the algebraic codebook, 6.70 kbit/s. Track Pulse Positions 1 i0 0, 5, 10, 15, 20, 25, 30, 35 2 i1 1, 6, 11, 16, 21, 26, 31, 36, 3, 8, 13, 18, 23, 28, 33, 38 3 i2 2, 7, 12, 17, 22, 27, 32, 37, 4, 9, 14, 19, 24, 29, 34, 39 The pulse positions are encoded with 3, 4, and 4 bits (total of 11 bits), and the sign of the each pulse is encoded with 1 bit (total of 3 bits). This gives a total of 14 bits for the algebraic code. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 37 TS 26.090 : December 1999 5.90 kbit/s mode In this codebook, the innovation vector contains 2 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 2 tracks, where each track contains one pulse, as shown in table 7. Table 7: Potential positions of individual pulses in the algebraic codebook, 5.90 kbit/s. Track Pulse Positions 1 i0 1, 6, 11, 16, 21, 26, 31, 36, 3, 8, 13, 18, 23, 28, 33, 38 2 i1 0, 5, 10, 15, 20, 25, 30, 35, 1, 6, 11, 16, 21, 26, 31, 36, 2, 7, 12, 17, 22, 27, 32, 37, 4, 9, 14, 19, 24, 29, 34, 39 The pulse positions are encoded with 4 and 5 bits (total of 9 bits), and the sign of the each pulse is encoded with 1 bit (total of 2 bits). This gives a total of 11 bits for the algebraic code. 5.15, 4.75 kbit/s modes In this codebook, the innovation vector contains 2 non-zero pulses. All pulses can have the amplitudes +1 or -1. The 40 positions in a subframe are divided into 5 tracks. Two subsets of 2 tracks each are used for each subframe with one pulse in each track. Different subsets of tracks are used for each subframe. The pulse positions used in each subframe are shown in table 8. Table 8: Potential positions of individual pulses in the algebraic codebook, 5.15, 4.75 kbit/s. Subframe Subset Pulse Positions 1 i0 0, 5, 10, 15, 20, 25, 30, 35 1 i1 2, 7, 12, 17, 22, 27, 32, 37 2 i0 1, 6, 11, 16, 21, 26, 31, 36 i1 3, 8, 13, 18, 23, 28, 33, 38 1 i0 0, 5, 10, 15, 20, 25, 30, 35 2 i1 3, 8, 13, 18, 23, 28, 33, 38 2 i0 2, 7, 12, 17, 22, 27, 32, 37 i1 4, 9, 14, 19, 24, 29, 34, 39 1 i0 0, 5, 10, 15, 20, 25, 30, 35 3 i1 2, 7, 12, 17, 22, 27, 32, 37 2 i0 1, 6, 11, 16, 21, 26, 31, 36 i1 4, 9, 14, 19, 24, 29, 34, 39 1 i0 0, 5, 10, 15, 20, 25, 30, 35 4 i1 3, 8, 13, 18, 23, 28, 33, 38 2 i0 1, 6, 11, 16, 21, 26, 31, 36 i1 4, 9, 14, 19, 24, 29, 34, 39 One bit is needed to encoded the subset used. The two pulse positions are encoded with 3 bits each (total of 6 bits), and the sign of the each pulse is encoded with 1 bit (total of 2 bits). This gives a total of 9 bits for the algebraic code. 5.7.2 Algebraic codebook search The algebraic codebook is searched by minimizing the mean square error between the weighted input speech and the weighted synthesized speech. The target signal used in the closed-loop pitch search is updated by subtracting the adaptive codebook contribution. That is: ( ) ( ) ( ) x n x n g y n n p 2 0 39 = − =  , , ,  (42) ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 38 TS 26.090 : December 1999 where ( ) ( ) ( ) y n v n h n = ∗ is the filtered adaptive codebook vector and gp is the quantified adaptive codebook gain. If ck is the algebraic codevector at index k , then the algebraic codebook is searched by maximizing the term: ( ) ( ) A C E k k Dk t k k t k = = 2 2 d c c c Φ , (43) where d H x = t 2 is the correlation between the target signal ( ) x n 2 and the impulse response ( ) h n , H is a the lower triangular Toepliz convolution matrix with diagonal ( ) h 0 and lower diagonals ( ) ( ) h h 1 39 , ,  , and Φ = H H t is the matrix of correlations of ( ) h n . The vector d (backward filtered target) and the matrix Φ are computed prior to the codebook search. The elements of the vector d are computed by ( ) ( ) ( ) d n x n h i n n i n = − = =∑ 2 39 0 39 , , ,  , (44) and the elements of the symmetric matrix Φ are computed by: ( ) ( ) ( ) ( ) φ i j h n i h n j j i n j , , = − − ≥ =∑ 39 . (45) The algebraic structure of the codebooks allows for very fast search procedures since the innovation vector ck contains only a few nonzero pulses. The correlation in the numerator of Equation (43) is given by: C d m i i i N p = = − ∑ϑ ( ) 0 1 , (46) where mi is the position of the i th pulse, ϑ i is its amplitude, and N p is the number of pulses ( N p = 10 ). The energy in the denominator of equation (43) is given by: E m m m m D i i i j i j j i N i N i N p p p = + = + − = − = − ∑ ∑ ∑φ ϑ ϑ φ ( , ) ( , ). 2 1 1 0 2 0 1 (47) To simplify the search procedure, the pulse amplitudes are preset by the mere quantization of an appropriate signal ( ) b n . This is simply done by setting the amplitude of a pulse at a certain position equal to the sign of ( ) b n at that position. The simplification proceeds as follows (prior to the codebook search). First, the sign signal [ ] s n b n b( ) ( ) =sign and the signal d n d n s n b ' ( ) ( ) ( ) = are computed. Second, the matrix Φ is modified by including the sign information; that is, φ φ ' ( , ) ( ) ( ) ( , ) i j s i s j i j b b = . The correlation in equation (46) is now given by: ( ) C d mi i N p = ′ = − ∑ 0 1 (48) and the energy in equation (47) is given by: E m m m m D i i i j j i N i N i N p p p = + = + − = − = − ∑ ∑ ∑φ φ ' ' ( , ) ( , ). 2 1 1 0 2 0 1 (49) ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 39 TS 26.090 : December 1999 12.2 kbit/s mode In this case the signal ( ) b n , used for presetting the amplitudes, is a sum of the normalized ( ) d n vector and normalized long-term prediction residual ( ) res n LTP : ( ) ( ) ( ) ( ) ( ) ( ) ( ) b n res n res i res i d n d i d i n LTP LTP LTP i i = + = = = ∑ ∑ 0 39 0 39 0 39 , , , ,  (50) is used. Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. First, for each of the five tracks the pulse positions with maximum absolute values of ( ) b n are searched. From these the global maximum value for all the pulse positions is selected. The first pulse i0 is always set into the position corresponding to the global maximum value. Next, four iterations are carried out. During each iteration the position of pulse i1 is set to the local maximum of one track. The rest of the pulses are searched in pairs by sequentially searching each of the pulse pairs {i2,i3}, {i4,i5}, {i6,i7} and {i8,i9} in nested loops. Every pulse has 8 possible positions, i.e., there are four 8x8-loops, resulting in 256 different combinations of pulse positions for each iteration. In each iteration all the 9 pulse starting positions are cyclically shifted, so that the pulse pairs are changed and the pulse i1 is placed in a local maximum of a different track. The rest of the pulses are searched also for the other positions in the tracks. At least one pulse is located in a position corresponding to the global maximum and one pulse is located in a position corresponding to one of the 4 local maxima. A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter F z E( ) which enhances special spectral components in order to improve the synthesized speech quality. Here the filter F z z E T ( ) ( ) = − − 1 1 β is used, where T is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and β is a pitch gain. In this standard, β is given by the quantified pitch gain bounded by [0.0,1.0]. Note that prior to the codebook search, the impulse response ( ) h n must include the pre-filter F z E( ) . That is, ( ) ( ) ( ) h n h n h n T n T = − − = β , , ,  39 . The fixed codebook gain is then found by: gc t t = x z z z 2 (51) where x2 is the target vector for fixed codebook search and z is the fixed codebook vector convolved with ( ) h n , ( ) ( ) ( ) z n c i h n i n i n = − = =∑ 0 0 39 , , , .  (52) 10.2 kbit/s mode In this case the signal ( ) b n , used for presetting the amplitudes, is given by eq. (50). Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter F z E( ) which enhances special spectral components in order to improve the synthesized speech quality. Here the filter F z z E T ( ) ( ) = − − 1 1 β is used, where T is the nearest integer pitch lag to the closed-loop fractional ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 40 TS 26.090 : December 1999 pitch lag of the subframe, and β is a pitch gain. In this standard, β is given by the quantified pitch gain bounded by [0.0,0.8]. Note that prior to the codebook search, the impulse response ( ) h n must include the pre-filter F z E( ) . That is, ( ) ( ) ( ) h n h n h n T n T = − − = β , , ,  39 . The fixed codebook gain is then found by equation (51). 7.95, 7.40 kbit/s modes In this case the signal ( ) b n , used for presetting the amplitudes, is equal to the signal ( ) d n . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter F z E( ) which enhances special spectral components in order to improve the synthesized speech quality. Here the filter F z z E T ( ) ( ) = − − 1 1 β is used, where T is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and β is a pitch gain. In this standard, β is given by the quantified pitch gain bounded by [0.0,0.8]. Note that prior to the codebook search, the impulse response ( ) h n must include the pre-filter F z E( ) . That is, ( ) ( ) ( ) h n h n h n T n T = − − = β , , ,  39 . The fixed codebook gain is then found by equation (51). 6.70 kbit/s mode In this case the signal ( ) b n , used for presetting the amplitudes, is equal to the signal ( ) d n . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an efficient non-exhaustive analysis-by-synthesis search technique. In this technique, the term in equation (43) is tested for a small percentage of position combinations. A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter F z E( ) which enhances special spectral components in order to improve the synthesized speech quality. Here the filter F z z E T ( ) ( ) = − − 1 1 β is used, where T is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and β is a pitch gain. In this standard, β is given by the quantified pitch gain bounded by [0.0,0.8]. Note that prior to the codebook search, the impulse response ( ) h n must include the pre-filter F z E( ) . That is, ( ) ( ) ( ) h n h n h n T n T = − − = β , , ,  39 . The fixed codebook gain is then found by equation (51). 5.90 kbit/s mode In this case the signal ( ) b n , used for presetting the amplitudes, is equal to the signal ( ) d n . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an exhaustive analysis-by-synthesis search technique. A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter F z E( ) which enhances special spectral components in order to improve the synthesized speech quality. Here the filter F z z E T ( ) ( ) = − − 1 1 β is used, where T is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and β is a pitch gain. In this standard, β is given by the quantified pitch gain bounded by [0.0,0.8]. Note that prior to the codebook search, the impulse response ( ) h n must include the pre-filter F z E( ) . That is, ( ) ( ) ( ) h n h n h n T n T = − − = β , , ,  39 . The fixed codebook gain is then found by equation (51). ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 41 TS 26.090 : December 1999 5.15, 4.75 kbit/s modes In this case the signal ( ) b n , used for presetting the amplitudes, is equal to the signal ( ) d n . Having preset the pulse amplitudes, as explained above, the optimal pulse positions are determined using an exhaustive analysis-by-synthesis search technique. Note that both subsets are searched. A special feature incorporated in the codebook is that the selected codevector is filtered through an adaptive pre-filter F z E( ) which enhances special spectral components in order to improve the synthesized speech quality. Here the filter F z z E T ( ) ( ) = − − 1 1 β is used, where T is the nearest integer pitch lag to the closed-loop fractional pitch lag of the subframe, and β is a pitch gain. In this standard, β is given by the quantified pitch gain bounded by [0.0,0.8]. Note that prior to the codebook search, the impulse response ( ) h n must include the pre-filter F z E( ) . That is, ( ) ( ) ( ) h n h n h n T n T = − − = β , , ,  39 . The fixed codebook gain is then found by equation (51). 5.8 Quantization of the adaptive and fixed codebook gains 5.8.1 Adaptive codebook gain limitation in quantization If the GpC_flag is set, the limited adaptive codebook gain is used in the gain quantization in section 5.8.2. The quantization codebook search range is limited to only include adaptive codebook gain values less than GP th . This is performed in the quantization search for all modes. 5.8.2 Quantization of codebook gains Prediction of the fixed codebook gain (all modes) The fixed codebook gain quantization is performed using MA prediction with fixed coefficients. The 4th order MA prediction is performed on the innovation energy as follows. Let ( ) E n be the mean-removed innovation energy (in dB) at subframe n , and given by: ( ) ( ) E n nN g c i E c i N =      − = − ∑ 10 1 2 2 0 1 log , (53) where N =40 is the subframe size, ( ) c i is the fixed codebook excitation, and E (in dB) is the mean of the innovation energy. The predicted energy is given by: ( ) ( ) ~  E n b R n i i i = − =∑ 1 4 , (54) where [ ] [ ] b b b b 1 2 3 4 0 68 058 034 019 = . . . . are the MA prediction coefficients, and ( ) R k is the quantified prediction error at subframe k . The predicted energy is used to compute a predicted fixed-codebook gain ′ gc as in equation (53) (by substituting ( ) E n by ( ) ~E n and gc by ′ gc ). This is done as follows. First, the mean innovation energy is found by: E N c j I j N =       = − ∑ 10 1 2 0 1 log ( ) (55) ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 42 TS 26.090 : December 1999 and then the predicted gain ′ gc is found by: ( ) ( ) ′ = + − gc E n E EI 100 05 . ~ . (56) A correction factor between the gain gc and the estimated one ′ gc is given by: γ gc c c g g = ′ . (57) Note that the prediction error is given by: R n E n E n gc ( ) ( ) ~( ) ( ). = − = 20 log γ (58) 12.2 kbit/s mode The correction factor γ gc is computed using a mean energy value, E = 36 dB. The correction factor γ gc is quantified using a 5-bit codebook. The quantization table search is performed by minimizing the error: ( ) E g g Q c gc c = − ′ γ 2 . (59) Once the optimum value γ gc is chosen, the quantified fixed codebook gain is given by   g g c gc c = ′ γ . 10.2 kbit/s mode The correction factor γ gc is computed using a mean energy value, E = 33 dB. The adaptive codebook gain g p and the correction factor γ gc are jointly vector quantized using a 7-bit codebook. The gain codebook search is performed by minimizing equation (63). 7.95 kbit/s mode The correction factor γ gc is computed using a mean energy value, E = 36 dB. The same scalar codebooks as for the 12.2 kbit/s mode is used for quantization of the adaptive codebook gain g p and the correction factor γ gc . The search of the codebooks starts with finding 3 candidates for the adaptive codebook gain. These candidates are the best codebook value in scalar quantization and the two adjacent codebook values. These 3 candidates are searched together with the correction factor codebook minimizing the term of equation (63). An adaptor based on the coding gain in the adaptive codebook decides if the coding gain is low. If this is the case, the correction factor codebook is searched once more minimizing a modified criterion in order to find a new quantized fixed codebook gain. The modified criterion is given by: ( ) ( ) E g g E E c gc c res exc mod ( )  = − ⋅ ⋅ − ⋅ ′ + ⋅ − 1 2 2 2 α γ α c (60) where Eres and Eexc are the energy (the squared norm) of the LP residual and the total exictation, respectively. The criterion is searched with the already quantized adaptive codebook gain and the correction factor γ gc that minimizes (60) is selected. The balance factor α decides the amount of energy matching in the modified criterion. This factor is adaptively decided based on the coding gain in the adaptive codebook as computed by: ag LP LP = ⋅ − 10 10 2 2 log res res v . (61) ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 43 TS 26.090 : December 1999 If the coding gain ag is less than 1 dB, the modified criterion is employed, except when an onset is detected. An onset is said to be detected if the fixed codebook gain in the current subframe is more than twice the value of the fixed codebook gain in the previous subframe. A hangover of 8 subframes is used in the onset detection so that the modified criterion is not used for the next 7 subframes either if an onset is detected. The balance factor α is computed from the median filtered adaptive coding gain. The current and the ag-values for the previous 4 subframes are median filtered to get agm . The α -factor is computed by: ( ) α = ⋅ − ⋅ > < < <    0 05 1 05 05 2 0 2 0 . . . ag ag ag ag m m m m . (62) 7.40 kbit/s mode The correction factor γ gc is computed using a mean energy value, E = 30 dB. The adaptive codebook gain g p and the correction factor γ gc are jointly vector quantized using a 7-bit codebook. The gain codebook search is performed by minimizing the square of the weighted error between original and reconstructed speech which is given by E g g g g g g g g p c t p t c t p t c t p c t = − − = + + − − + x y z x x y y z z x y x z y z 2 2 2 2 2 (63) where x is the target vector, y is the filtered adaptive codebook vector, and z is the filtered fixed codebook vector. 6.70 kbit/s mode The correction factor γ gc is computed using a mean energy value, E = 2875 . dB. The adaptive codebook gain g p and the correction factor γ gc are jointly vector quantized using a 7-bit codebook. The gain codebook search is performed by minimizing equation (63). 5.90, 5.15 kbit/s modes The correction factor γ gc is computed using a mean energy value, E = 33 dB. The adaptive codebook gain g p and the correction factor γ gc are jointly vector quantized using a 6-bit codebook. The gain codebook search is performed by minimizing equation (63). 4.75 kbit/s mode The correction factors γ gc are computed using a mean energy value, E = 33 dB. The adaptive codebook gains g p and the correction factors γ gc are jointly vector quantized every 10 ms. This is done by minimizing a weighted sum of the error criterion (63) for each of the two subframes. The default values on the weighing factors are 1. If the energy of the second subframe is more than two times the energy of the first subframe, the weight of the first subrame is set to 2. If the energy of the first subframe is more than four times the energy of the first subframe, the weight of the second subrame is set to 2. 5.8.3 Update past quantized adaptive codebook gain buffer (all modes) After the gain quantization, the buffer with past adaptive codebook gains is updated, regardless of the value of the GpC_flag. That is,  ( )  ( ), ,..., g n i g n i i p p − = −+ = 1 7 1 . ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 44 TS 26.090 : December 1999 5.9 Memory update (all modes) An update of the states of the synthesis and weighting filters is needed in order to compute the target signal in the next subframe. After the two gains are quantified, the excitation signal, ( ) u n , in the present subframe is found by: ( ) ( ) ( ) u n g v n g c n n p c = + =   , , , 0 39  , (64) where g p and gc are the quantified adaptive and fixed codebook gains, respectively, ( ) v n the adaptive codebook vector (interpolated past excitation), and ( ) c n is the fixed codebook vector (algebraic code including pitch sharpening). The states of the filters can be updated by filtering the signal res n u n LP( ) ( ) − (difference between residual and excitation) through the filters ( ) 1 A z and ( ) ( ) A z A z γ γ 1 2 for the 40-sample subframe and saving the states of the filters. This would require 3 filterings. A simpler approach which requires only one filtering is as follows. The local synthesized speech, ( ) s n , is computed by filtering the excitation signal through ( ) 1 A z . The output of the filter due to the input res n u n LP( ) ( ) − is equivalent to ( ) ( ) ( ) e n s n s n = − . So the states of the synthesis filter ( ) 1 A z are given by ( ) e n n , , , = 30 39  . Updating the states of the filter ( ) ( ) ( ) e n s n s n = − can be done by filtering the error signal ( ) e n through this filter to find the perceptually weighted error ( ) e n w . However, the signal ( ) e n w can be equivalently found by: ( ) ( ) ( ) ( ) e n x n g y n g z n w p c = − −   , (65) Since the signals ( ) x n , ( ) y n , and ( ) z n are available, the states of the weighting filter are updated by computing ( ) e n w as in equation (65) for n = 30 39 , ,  . This saves two filterings. 4.75 kbit/s mode The memory update in the first and third subframes use the unquantized gains in equation (64). After the second and fourth subframes respectively, when the gains are quantized, the state is recalculated using the quantized gains. 6 Functional description of the decoder The function of the decoder consists of decoding the transmitted parameters (LP parameters, adaptive codebook vector, adaptive codebook gain, fixed codebook vector, fixed codebook gain) and performing synthesis to obtain the reconstructed speech. The reconstructed speech is then post-filtered and upscaled. The signal flow at the decoder is shown in figure 4. 6.1 Decoding and speech synthesis The decoding process is performed in the following order: Decoding of LP filter parameters: The received indices of LSP quantization are used to reconstruct the quantified LSP vectors. The interpolation described in subclause 5.2.6 is performed to obtain 4 interpolated LSP vectors (corresponding to 4 subframes). For each subframe, the interpolated LSP vector is converted to LP filter coefficient domain ak , which is used for synthesizing the reconstructed speech in the subframe. The following steps are repeated for each subframe: 1) Decoding of the adaptive codebook vector: The received pitch index (adaptive codebook index) is used to find the integer and fractional parts of the pitch lag. The adaptive codebook vector ( ) v n is found by interpolating the past excitation ( ) u n (at the pitch delay) using the FIR filter described in subclause 5.6. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 45 TS 26.090 : December 1999 2) Decoding of the innovative codebook vector: The received algebraic codebook index is used to extract the positions and amplitudes (signs) of the excitation pulses and to find the algebraic codevector ( ) c n . If the integer part of the pitch lag, T, is less than the subframe size 40, the pitch sharpening procedure is applied which translates into modifying ( ) c n by ( ) ( ) ( ) c n c n c n T = + − β , where β is the decoded pitch gain, g p , bounded by [0.0,1.0] or [0.0,0.8], depending on mode. 3) Decoding of the adaptive and fixed codebook gains: In case of scalar quantization of the gains (12.2 kbit/s and 7.95 kbit/s modes) the received indices are used to readily find the quantified adaptive codebook gain, g p , and the quantified fixed codebook gain correction factor, γ gc , from the corresponding quantization tables. In case of vector quantization of the gains (all other modes), the received index gives both the quantified adaptive codebook gain, g p , and the quantified fixed codebook gain correction factor, γ gc . The estimated fixed codebook gain ′ gc is found as described in subclause 5.7. First, the predicted energy is found by: ( ) ( ) ~  E n b R n i i i = − =∑ 1 4 (66) and then the mean innovation energy is found by: E N c j I j N =       = − ∑ 10 1 2 0 1 log ( ) . (67) The predicted gain ′ gc is found by: ( ) ( ) ′ = + − gc E n E EI 100 05 . ~ . (68) The quantified fixed codebook gain is given by: g g c gc c = ′ γ . (69) 4) Smoothing of the fixed codebook gain (10.2, 6.70, 5.90, 5.15, 4.75 kbit/s modes): An adaptive smoothing of the fixed codebook gain is performed to avoid unnatural fluctuations in the energy contour. The smoothing is based on a measure of the stationarity of the short-term spectrum in the q domain. The smoothing strength is computed from this measure. An averaged q-value is computed for each frame n by: ( ) ( ) ( ) q q q n n n = ⋅ − + ⋅ 084 1 016 4 . .  . (70) For each subframe m, a difference measure between the averaged vector and the quantized and interpolated vector is computed by: ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 46 TS 26.090 : December 1999 ( ) ( ) ( ) diff q n q n q n m j m j j m j = − ∑ ∑ ( ) ( ) ( )  , (71) where j runs over the 10 LSPs. Furthermore, a smoothing factor, km , is computed by: ( ) ( ) k K diff K K m m = − min ,max , 2 1 2 0 , (72) where the constants are set to K1 0 4 = . and K2 0 25 = . . A hangover period of 40 subframes is used where the km -value is set 1.0 if the diffm has been above 0.65 for 10 consecutive frames. A value of 1.0 corresponds to no smoothing. An averaged fixed codebook gain value is computed for each subframe by: ( ) ( ) g m g m i c i = − =∑ 1 5 0 4  . (73) The fixed codebook gain used for synthesis is now replaced by a smoothed value given by: ( )   g g k g k c c m c m = ⋅ + ⋅ − 1 . (74) 5) Anti-sparseness processing (7.95, 6.70, 5.90, 5.15, 4.75 kbit/s modes): An adaptive anti-sparseness post- processing procedure is applied to the fixed codebook vector ( ) c n in order to reduce perceptual artifacts arising from the sparseness of the algebraic fixed codebook vectors with only a few non-zero samples per subframe. The anti-sparseness processing consists of circular convolution of the fixed codebook vector with an impulse response. Three pre-stored impulse responses are used and a number impNr = 0 1 2 , , is set to select one of them. A value of 2 corresponds to no modification, a value of 1 corresponds to medium modification, while a value of 0 corresponds to strong modification. The selection of the impulse response is performed adaptively from the adaptive and fixed codebook gains. The following procedure is employed: if then else if then else  . ;  . ; ; g impNr g impNr impNr p p < = < = = 0 6 0 0 9 1 2 Detect onset by comparing the fixed codebook gain to the previous fixed codebook gain. If the current value is more than twice the previous value an onset is detected. If not onset and impNr = 0 , the median filtered value of the current and the previous 4 adaptive codebook gains are computed. If this value is less than 0.6, impNr = 0 . ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 47 TS 26.090 : December 1999 If not onset, the impNr -value is restricted to increase by one step from the previous subframe. If an onset is declared, the impNr -value is increased by one if it is less than 2. 6) Computing the reconstructed speech: The excitation at the input of the synthesis filter is given by: ( ) ( ) ( ) u n g v n g c n p c = +   . (75) Before the speech synthesis, a post-processing of excitation elements is performed. This means that the total excitation is modified by emphasizing the contribution of the adaptive codebook vector: ( )    ≤ > + > + = 5.0 ˆ ) ( modes other all ,5.0 ˆ ), ( ˆ 5.0 ) ( mode kbit/s 12.2 ,5.0 ˆ ), ( ˆ 25 .0 ) ( ˆ p p p p p g n u g n v g n u g n v g n u n u β β (76) Adaptive gain control (AGC) is used to compensate for the gain difference between the non-emphasized excitation ( ) u n and emphasized excitation ( ) u n The gain scaling factor η for the emphasized excitation is computed by: ( ) ( ) η = > ≤       = = ∑ ∑ u n u n g g n n p p 2 0 39 2 0 39 05 10 05  ,  . , . ,  . . (77) The gain-scaled emphasized excitation signal ( ) ′ u n is given by: ( ) ( )   ′ = u n u n η . (78) The reconstructed speech for the subframe of size 40 is given by: ( ) ( ) ( )     , , , s n u n a s n i n i i = ′ − − = =∑ 1 10 0 39  . (79) where ai are the interpolated LP filter coefficients. 7) Additional instability protection: An additional instability protection is implemented in the speech decoder which is monitoring overflows in the synthesis filter. If an overflow has occurred in the synthesis part, the whole adaptive codebook memory, v n n ( ), ( ),..., = − + 143 11 39 is scaled down by a factor of 4, and the synthesis filtering is repeated using this down-scaled memory. I.e. in this case step 6) is repeated, except that the post-processing in (76) - (78) of the excitation signal is by-passed. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 48 TS 26.090 : December 1999 The synthesized speech ( ) s n is then passed through an adaptive postfilter which is described in the following clause. 6.2 Post-processing 6.2.1 Adaptive post-filtering (all modes) The adaptive postfilter is the cascade of two filters: a formant postfilter, and a tilt compensation filter. The postfilter is updated every subframe of 5 ms. The formant postfilter is given by: ( ) ( ) ( ) H z A z A z f n d =   γ γ (80) where ( ) A z is the received quantified (and interpolated) LP inverse filter (LP analysis is not performed at the decoder), and the factors γ n and γ d control the amount of the formant post-filtering. Finally, the filter ( ) H z t compensates for the tilt in the formant postfilter ( ) H z f and is given by: ( ) H z z t = − − 1 1 µ (81) where µ γ = ′ t k1 is a tilt factor, with ′ k1 being the first reflection coefficient calculated on the truncated ( Lh = 22 ) impulse response, ( ) h n f , of the filter ( ) ( )   A z A z n d γ γ . ′ k1 is given by: ( ) ( ) ( ) ( ) ( ) ′ = = + = −− ∑ k r r r i h j h j i h h h f f j L i h 1 0 1 1 0 ; . (82) The post-filtering process is performed as follows. First, the synthesized speech ( ) s n is inverse filtered through ( ) A z n γ to produce the residual signal ( ) r n . The signal ( ) r n is filtered by the synthesis filter ( ) 1 A z d γ . Finally, the signal at the output of the synthesis filter ( ) 1 A z d γ is passed to the tilt compensation filter ( ) H z t resulting in the post-filtered speech signal ( ) s n f . Adaptive gain control (AGC) is used to compensate for the gain difference between the synthesized speech signal ( ) s n and the post-filtered signal ( ) s n f . The gain scaling factor γ sc for the present subframe is computed by: γ sc n f n s n s n = = = ∑ ∑  ( )  ( ) 2 0 39 2 0 39 . (83) The gain-scaled post-filtered signal ( ) ′s n is given by: ( ) ( ) ( )   ′ = s n n s n sc f β (84) where β sc n ( ) is updated in sample-by-sample basis and given by: ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 49 TS 26.090 : December 1999 β αβ α γ sc sc sc n n ( ) ( ) ( ) = − + − 1 1 (85) where α is a AGC factor with value of 0.9. 12.2, 10.2 kbit/s modes The adaptive post-filtering factors are given by: γ n = 0 7. , γ d = 0 75 . and γ t k = ′ >    08 0 0 1 . , , , otherwise.. (86) 7.95, 7.40, 6.70, 5.90, 5.15, 4.75 kbit/s modes The adaptive post-filtering factors are given by: γ n = 055 . , γ d = 0 7. and γ t = 08. . 6.2.2 High-pass filtering and up-scaling (all modes) The high-pass filter serves as a precaution against undesired low frequency components. A filter cut-off frequency of 60 Hz is used, and the filter is given by H z z z z z h2 1 2 1 2 0 939819335 1879638672 0939819335 1 1933105469 0935913085 ( ) . . . . . = − + − + − − − − . (87) Up-scaling consists of multiplying the post-filtered speech by a factor of 2 to compensate for the down-scaling by 2 which is applied to the input signal. 7 Detailed bit allocation of the adaptive multi-rate codec The detailed allocation of the bits in the adaptive multi-rate speech encoder is shown for each mode in table 9a-9h. These tables show the order of the bits produced by the speech encoder. Note that the most significant bit (MSB) of each codec parameter is always sent first. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 50 TS 26.090 : December 1999 Table 9a: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 244 bits/20 ms, 12.2 kbit/s mode. Bits (MSB-LSB) Description s1 - s7 index of 1st LSF submatrix s8 - s15 index of 2nd LSF submatrix s16 - s23 index of 3rd LSF submatrix s24 sign of 3rd LSF submatrix s25 - s32 index of 4th LSF submatrix s33 - s38 index of 5th LSF submatrix subframe 1 s39 - s47 adaptive codebook index s48 - s51 adaptive codebook gain s52 sign information for 1st and 6th pulses s53 - s55 position of 1st pulse s56 sign information for 2nd and 7th pulses s57 - s59 position of 2nd pulse s60 sign information for 3rd and 8th pulses s61 - s63 position of 3rd pulse s64 sign information for 4th and 9th pulses s65 - s67 position of 4th pulse s68 sign information for 5th and 10th pulses s69 - s71 position of 5th pulse s72 - s74 position of 6th pulse s75 - s77 position of 7th pulse s78 - s80 position of 8th pulse s81 - s83 position of 9th pulse s84 - s86 position of 10th pulse s87 - s91 fixed codebook gain subframe 2 s92 - s97 adaptive codebook index (relative) s98 - s141 same description as s48 - s91 subframe 3 s142 - s194 same description as s39 - s91 subframe 4 s195 - s244 same description as s92 - s141 ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 51 TS 26.090 : December 1999 Table 9b: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 204 bits/20 ms, 10.2 kbit/s mode. Bits (MSB-LSB) Description s1 – s8 index of 1st LSF subvector s9 - s17 index of 2nd LSF subvector s18 – s26 index of 3rd LSF subvector subframe 1 s27 – s34 adaptive codebook index s35 sign information for 1st and 5th pulses s36 sign information for 2nd and 6th pulses s37 sign information for 3rd and 7th pulses s38 sign information for 4th and 8th pulses s39-s48 position for 1st, 2nd, and 5th pulses s49-s58 position for 3rd, 6th, and 7th pulses s59-s65 position for 4th and 8th pulses s66 – s72 codebook gains subframe 2 s73 – s77 adaptive codebook index (relative) s78 – s115 same description as s35 – s72 subframe 3 s116 – s161 same description as s27 – s72 subframe 4 s162 – s204 same description as s73 – s115 Table 9c: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 159 bits/20 ms, 7.95 kbit/s mode. Bits (MSB-LSB) Description s1 – s9 index of 1st LSF subvector s10 - s18 index of 2nd LSF subvector s19 – s27 index of 3rd LSF subvector subframe 1 s28 – s35 adaptive codebook index s36 – s39 position of 4th pulse s40 – s42 position of 3rd pulse s43 – s45 position of 2nd pulse s46 – s48 position of 1st pulse s49 sign information for 4th pulse s50 sign information for 3rd pulse s51 sign information for 2nd pulse s52 sign information for 1st pulse s53 – s56 adaptive codebook gain s57 – s61 fixed codebook gain subframe 2 s62 – s67 adaptive codebook index (relative) s68 – s93 same description as s36 – s61 subframe 3 s94 – s127 same description as s28 – s61 subframe 4 s128 – s159 same description as s62 – s93 ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 52 TS 26.090 : December 1999 Table 9d: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 148 bits/20 ms, 7.40 kbit/s mode. Bits (MSB-LSB) Description s1 – s8 index of 1st LSF subvector s9 - s17 index of 2nd LSF subvector s18 – s26 index of 3rd LSF subvector subframe 1 s27 – s34 adaptive codebook index s35 – s38 position of 4th pulse s39 – s41 position of 3rd pulse s42 - s44 position of 2nd pulse s45 – s47 position of 1st pulse s48 sign information for 4th pulse s49 sign information for 3rd pulse s50 sign information for 2ndd pulse s51 sign information for 1st pulse s52 – s58 codebook gains subframe 2 s59 – s63 adaptive codebook index (relative) s64 – s87 same description as s35 – s58 subframe 3 s88 – s119 same description as s27 – s58 subframe 4 s120 – s148 same description as s59 – s87 Table 9e: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 134 bits/20 ms, 6.70 kbit/s mode. Bits (MSB-LSB) Description s1 – s8 index of 1st LSF subvector s9 - s17 index of 2nd LSF subvector s18 – s26 index of 3rd LSF subvector subframe 1 s27 – s34 adaptive codebook index s35 – s38 position of 3rd pulse s39 – s42 position of 2nd pulse s43 – s45 position of 1st pulse s46 sign information for 3rd pulse s47 sign information for 2nd pulse s48 sign information for 1st pulse s49 – s55 codebook gains subframe 2 s56 – s59 adaptive codebook index (relative) s60 – s80 same description as s35 – s55 subframe 3 s81 – s109 same description as s27 – s55 subframe 4 s110 – s134 same description as s56 – s80 ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 53 TS 26.090 : December 1999 Table 9f: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 118 bits/20 ms, 5.90 kbit/s mode. Bits (MSB-LSB) Description s1 – s8 index of 1st LSF subvector s9 - s17 index of 2nd LSF subvector s18 – s26 index of 3rd LSF subvector subframe 1 s27 – s34 adaptive codebook index s35 – s39 position of 2nd pulse s40 – s43 position of 1st pulse s44 sign information for 2nd pulse s45 sign information for 1st pulse s46 – s51 codebook gains subframe 2 s52 – s55 adaptive codebook index (relative) s56 – s72 same description as s35 – s51 subframe 3 s73 – s97 same description as s27 – s51 subframe 4 s98 – s118 same description as s52 – s72 Table 9g: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 103 bits/20 ms, 5.15 kbit/s mode. Bits (MSB-LSB) Description s1 – s8 index of 1st LSF subvector s9 - s16 index of 2nd LSF subvector s17 – s23 index of 3rd LSF subvector subframe 1 s24 – s31 adaptive codebook index s32 position subset s33 – s35 position of 2nd pulse s36 – s38 position of 1st pulse s39 sign information for 2nd pulse s40 sign information for 1st pulse s41 – s46 codebook gains subframe 2 s47 – s50 adaptive codebook index (relative) s51 – s65 same description as s32 – s46 subframe 3 s66 – s84 same description as s47 – s65 subframe 4 s85 – s103 same description as s47 – s65 ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 54 TS 26.090 : December 1999 Table 9h: Source encoder output parameters in order of occurrence and bit allocation within the speech frame of 95 bits/20 ms, 4.75 kbit/s mode. Bits (MSB-LSB) Description s1 – s8 index of 1st LSF subvector s9 - s16 index of 2nd LSF subvector s17 – s23 index of 3rd LSF subvector subframe 1 s24 – s31 adaptive codebook index s32 position subset s33 – s35 position of 2nd pulse s36 – s38 position of 1st pulse s39 sign information for 2nd pulse s40 sign information for 1st pulse s41 – s48 codebook gains subframe 2 s49 – s52 adaptive codebook index (relative) s53 – s61 same description as s32 – s40 subframe 3 s62 - s65 same description as s49 – s52 s66 – s82 same description as s32– s48 subframe 4 s83 – s95 same description as s49 – s61 8 Homing sequences 8.1 Functional description The adaptive multi-rate speech codec is described in a bit-exact arithmetic to allow for easy type approval as well as general testing purposes of the adaptive multi-rate speech codec. The response of the codec to a predefined input sequence can only be foreseen if the internal state variables of the codec are in a predefined state at the beginning of the experiment. Therefore, the codec has to be put in a so called home state before a bit-exact test can be performed. This is usually done by a reset (a procedure in which the internal state variables of the codec are set to their defined initial values). The codec mode of the speech encoder and speech decoder shall be set to the tested codec mode by external means at reset. To allow a reset of the codec in remote locations, special homing frames have been defined for the encoder and the decoder, thus enabling a codec homing by inband signalling. The codec homing procedure is defined in such a way, that in either direction (encoder or decoder) the homing functions are called after processing the homing frame that is input. The output corresponding to the first homing frame is therefore dependent on the used codec mode and the codec state when receiving that frame and hence usually not known. The response of the encoder to any further homing frame is by definition the corresponding decoder homing frame for the used codec mode. The response of the decoder to any further homing frame is by definition the encoder homing frame. This procedure allows homing of both, the encoder and decoder from either side, if a loop back configuration is implemented, taking proper framing into account. 8.2 Definitions Encoder homing frame: The encoder homing frame consists of 160 identical samples, each 13 bits long, with the least significant bit set to "one" and all other bits set to "zero". When written to 16-bit words with left justification, the samples have a value of 0008 hex. The speech decoder has to produce this frame as a response to the second and any further decoder homing frame if at least two decoder homing frames were input to the decoder consecutively. The encoder homing frame is identical for all codec modes. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 55 TS 26.090 : December 1999 Decoder homing frame: There exist eight different decoder homing frames, which correspond to the eight AMR codec modes. Using one of these codec modes, the corresponding decoder homing frame is the natural response of the speech encoder to the second and any further encoder homing frame if at least two encoder homing frames were input to the encoder consecutively. In [4], for each decoder homing frame the parameter values are given. 8.3 Encoder homing Whenever the adaptive multi-rate speech encoder receives at its input an encoder homing frame exactly aligned with its internal speech frame segmentation, the following events take place: Step 1: The speech encoder performs its normal operation including VAD and SCR and produces in accordance with the used codec mode a speech parameter frame at its output which is in general unknown. But if the speech encoder was in its home state at the beginning of that frame, then the resulting speech parameter frame is identical to that decoder homing frame, which corresponds to the used codec mode (this is the way how the decoder homing frames were constructed). Step 2: After successful termination of that operation the speech encoder provokes the homing functions for all sub-modules including VAD and SCR and sets all state variables into their home state. On the reception of the next input frame, the speech encoder will start from its home state. NOTE: Applying a sequence of N encoder homing frames will cause at least N-1 decoder homing frames at the output of the speech encoder. 8.4 Decoder homing Whenever the speech decoder receives at its input a decoder homing frame, which corresponds to the used codec mode, then the following events take place: Step 1: The speech decoder performs its normal operation and produces a speech frame at its output which is in general unknown. But if the speech decoder was in its home state at the beginning of that frame, then the resulting speech frame is replaced by the encoder homing frame. This would not naturally be the case but is forced by this definition here. Step 2: After successful termination of that operation the speech decoder provokes the homing functions for all sub-modules including the comfort noise generator and sets all state variables into their home state. On the reception of the next input frame, the speech decoder will start from its home state. NOTE 1: Applying a sequence of N decoder homing frames will cause at least N-1 encoder homing frames at the output of the speech decoder. NOTE 2: By definition (!) the first frame of each decoder test sequence must differ from the decoder homing frame at least in one bit position within the parameters for LPC and first subframe. Therefore, if the decoder is in its home state, it is sufficient to check only these parameters to detect a subsequent decoder homing frame. This definition is made to support a delay-optimized implementation in the TRAU uplink direction. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) Page 56 TS 26.090 : December 1999 A(z) 1 s(n) ^ + v(n) c(n) u(n) g c fixed codebook adaptive codebook g p LP synthesis post-filtering s'(n) ^ Figure 2: Simplified block diagram of the CELP synthesis model (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) ETSI Page 57 TS 26.090 : December 1999 w indow ing and autocorrelation R [ ] Levinson- D urbin R[ ] A(z) A(z) LSP quantization com pute target for innovation update filter m em ories for next subfram e O pen -loop p itc h search Adaptive codebook search Innovative codebook search Filter memory update interpolation subfram es L SP A(z) LSP com pute w eighted speech (4 subfram es) find o pen-loop pitch find best innovation fixed codebook gain quantization A(z) ^ x(n) pitch index code index fram e subframe s(n) com pute target for adaptive codebook To find best delay and gain x(n) com pute im pulse response A(z) ^ A(z) h(n) h(n) A(z) LPC analysis (twice per fram e) A(z) (twice per fram e) x (n) 2 quantize LTP-gain compute adaptive codebook contribution LSP indices L TP g ain ind ex g ain in dex fixed codebook interpolation for the 4 su bfram es L SP A(z) ^ for the 4 Pre-pro cessing Pre-processing com pute excitation Figure 3: Simplified block diagram of the adaptive multi-rate encoder (3G TS 26.090 version 3.1.0 Release 1999) ETSI ETSI TS 126 090 V3.1.0 (2000-01) Page 58 TS 26.090 : December 1999 L SP indices decode L SP interpolation of LS P for the 4 subframes LS P decode adaptive codebook decode innovative codebook pitch index code index decode gains A(z) ^ construct excitation fram e subfram e post-processing s'(n) ^ s(n) ^ post filter gains indices synthesis filter Figure 4: Simplified block diagram of the adaptive multi-rate decoder (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) ETSI 3GPP 3G TS 26.090 V3.1.0 (1999-12) 59 3G TS 26.090 version 3.1.0 1) M.R. Schroeder and B.S. Atal, "Code-Excited Linear Prediction (CELP): High quality speech at very low bit rates," in Proc. ICASSP'85, pp. 937-940, 1985. 2) L.R. Rabiner and R.W. Schaefer. Digital processing of speech signals. Prentice-Hall Int., 1978. 3) F. Itakura, "Line spectral representation of linear predictive coefficients of speech signals," J. Acoust. Soc. Amer., vol. 57, Supplement no. 1, S35, 1975. 4) F.K. Soong and B.H. Juang, "Line spectrum pair (LSP) and speech data compression", in Proc. ICASSP'84, pp. 1.10.1-1.10.4. 5) K.K Paliwal and B.S. Atal, "Efficient vector quantization of LPC parameters at 24 bits/frame", IEEE Trans. Speech and Audio Processing, vol. 1, no 1, pp. 3-14, 1993. 6) P. Kabal and R.P. Ramachandran, "The computation of line spectral frequencies using Chebyshev polynomials", IEEE Trans. on ASSP, vol. 34, no. 6, pp. 1419-1426, Dec. 1986. 7) K. Järvinen, J. Vainio, P. Kapanen, T. Honkanen, P. Haavisto, R. Salami, C. Laflamme, and J.-P. Adoul, “GSM enhanced full rate speech codec”, in Proc. ICASSP’97, pp. 771-774. 8) T. Honkanen, J. Vainio, K. Järvinen, P. Haavisto, R. Salami, C. Laflamme, and J.-P. Adoul, “Enhanced full rate speech codec for IS-136 digital cellular system”, in Proc. ICASSP’97, pp. 731-734. 9) R. Hagen, E. Ekudden, B. Johansson, and W.B. Kleijn, “Removal of sparse-excitation artifacts in CELP”, in Proc. ICASSP’98, pp. I-145-I-148. ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) 3GPP 3G TS 26.090 V3.1.0 (1999-12) 60 3G TS 26.090 version 3.1.0 Annex A: Change history Tdoc SPEC CR RE VER SUBJECT CAT NEW SP-99570 26.090 A001 3.0.1 Bit allocation of the adaptive multi-rate codec F 3.1.0 ETSI ETSI (3G TS 26.090 version 3.1.0 Release 1999) ETSI TS 126 090 V3.1.0 (2000-01) 61 ETSI ETSI TS 126 090 V3.1.0 (2000-01) (3G TS 26.090 version 3.1.0 Release 1999) History Document history V3.1.0 January 2000 Publication
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1 Scope
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2 Normative references
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3 Definitions and abbreviations
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3.1 Abbreviations
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4 General
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5 Adaptive Multi-Rate speech codec transcoding functions
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6 Adaptive Multi-Rate speech codec ANSI C-code
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7 Adaptive Multi-Rate speech codec test vectors
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8 Adaptive Multi-Rate speech codec source controlled rate operation
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9 Adaptive Multi-Rate speech codec voice activity detection
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10 Adaptive Multi-Rate speech codec comfort noise insertion
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11 Adaptive Multi-Rate speech codec error concealment of lost frames
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12 Adaptive Multi-Rate speech codec frame structure
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13 Adaptive Multi-Rate speech codec interface to RAN
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14 Adaptive Multi-Rate speech codec performance characterisation
.........................................................11 Annex A: Change history......................................................................................................................12 History..............................................................................................................................................................13 (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 4 3G TS 26.071 version 3.0.1 Foreword This Technical Specification has been produced by the 3GPP. The present document is an introduction to the speech processing parts of the narrowband telephony speech service employing the Adaptive Multi-Rate (AMR) speech coder within the 3GPP system. The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of this TS, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: Version 3.y.z where: x the first digit: 1 presented to TSG for information; 2 presented to TSG for approval; 3 Indicates TSG approved document under change control. y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. z the third digit is incremented when editorial only changes have been incorporated in the specification; (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 5 3G TS 26.071 version 3.0.1 1 Scope The present document is an introduction to the speech processing parts of the narrowband telephony speech service employing the Adaptive Multi-Rate (AMR) speech coder. A general overview of the speech processing functions is given, with reference to the documents where each function is specified in detail. 2 Normative references This TS 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 TS only when incorporated in it by amendment or revision. For undated references, the latest edition of the publication referred to applies. [1] GSM 03.50 : "Digital cellular telecommunications system (Phase 2); Transmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) system". [2] 3G TS 26.090 : “AMR Speech Codec; Transcoding functions". [3] 3G TS 26.073 : “AMR Speech Codec; ANSI-C code". [4] 3G TS 26.074 : “AMR Speech Codec; Test sequences". [5] 3G TS 26.093 : “AMR Speech Codec; Source Controlled Rate operation". [6] 3G TS 26.094 : “AMR Speech Codec; Voice Activity Detection (VAD)". [7] 3G TS 26.092 : “AMR Speech Codec; Comfort Noise Aspects". [8] 3G TS 26.091 : “AMR Speech Codec; Error Concealment of Lost Frames. [9] 3G TS 26.101 : “AMR Speech Codec; Frame Structure". [10] 3G TS 26.102 : “AMR Speech Codec; Interface to RAN". [11] TS 26.901: “AMR Speech Codec; Performance characterisation". 3 Definitions and abbreviations 3.1 Abbreviations For the purposes of this TS, the following abbreviations apply: ACELP Algebraic Code Excited Linear Prediction AMR Adaptive Multi-Rate BFI Bad Frame Indication CHD Channel Decoder CHE Channel Encoder GSM Global System for Mobile communications ITU-T International Telecommunication Union – Telecommunication standardisation sector (former CCITT) PCM Pulse Code Modulation PLMN Public Land Mobile Network PSTN Public Switched Telephone Network RX Receive SCR Source Controlled Rate (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 6 3G TS 26.071 version 3.0.1 SPD SPeech Decoder SPE SPeech Encoder TC Transcoder TX Transmit UE User Equipment (terminal) 4 General The AMR speech coder consists of the multi-rate speech coder, a source controlled rate scheme including a voice activity detector and a comfort noise generation system, and an error concealment mechansim to combat the effects of transmission errors and lost packets. The multi-rate speech coder is a single integrated speech codec with eight source rates from 4.75 kbit/s to 12.2 kbit/s, and a low rate background noise encoding mode. The speech coder is capable of switching its bit-rate every 20 ms speech frame upon command. A reference configuration where the various speech processing functions are identified is given in Figure 1. In this figure, the relevant specifications for each function are also indicated. In Figure 1, the audio parts including analogue to digital and digital to analogue conversion are included, to show the complete speech path between the audio input/output in the User Equipment (UE) and the digital interface of the network. The detailed specification of the audio parts is not within the scope of this document. These aspects are only considered to the extent that the performance of the audio parts affect the performance of the speech transcoder. (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 7 3G TS 26.071 version 3.0.1 8bit / A-law to 13-bit uniform LPF A/D 1 2 MS side only BSS side only GSM 06.60.AMR GSM 03.50 TRANSMIT SIDE Speech Encoder Comfort Noise TX Functions Voice Activity Detector DTX Control and Operation 3 6 4 5 6 7 GSM 06.82.AMR GSM 06.81.AMR GSM 06.60.AMR GSM 06.62.AMR SID frame Speech frame VAD 13-bit uniform to 8bit / A-law LPF D/A 1 8 MS side only BSS side only GSM 06.60.AMR GSM 03.50 RECEIVE SIDE Speech Decoder Speech frame substitution DTX Control and Operation 4 5 9 10 GSM 06.61.AMR GSM 06.81.AMR GSM 06.60.AMR GSM 06.62.AMR SID frame Speech frame Comfort Noise R X Functions 11 2 SP flag Info. bits BFI Info. bits SID TAF Figure 1: Overview of audio processing functions. 1) 8-bit A-law or µ -law PCM (ITU-T recommendation G.711), 8 000 samples/s 2) 13-bit uniform PCM, 8 000 samples/s 3) Voice Activity Detector (VAD) flag 4) Encoded speech frame, 50 frames/s, number of bits/frame depending on the AMR codec mode 5) SIlence Descriptor (SID) frame. 6) TX_TYPE, 2 bits, indicates whether information bits are available and if they are speech or SID information 7) Information bits delivered to the 3G AN 8) Information bits received from the 3G AN 9) RX_TYPE, the type of frame received quantized into three bits (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 8 3G TS 26.071 version 3.0.1 5 Adaptive Multi-Rate speech codec transcoding functions The adaptive multi-rate speech codec is described in [2]. The technical content is identical to that of GSM 06.90. As shown in Figure 1, the speech encoder takes its input as a 13-bit uniform Pulse Code Modulated (PCM) signal either from the audio part of the UE or on the network side, from the Public Switched Telephone Network (PSTN) via an 8-bit A-law or µ -law to 13-bit uniform PCM conversion. The encoded speech at the output of the speech encoder is packetized and delivered to the network interface. In the receive direction, the inverse operations take place. The detailed mapping between input blocks of 160 speech samples in 13-bit uniform PCM format to encoded blocks (in which the number of bits depends on the presently used codec mode) and from these to output blocks of 160 reconstructed speech samples is described in [2]. The coding scheme is Multi-Rate Algebraic Code Excited Linear Prediction. The bit-rates of the source codec are listed in Table 1. An AMR speech codec capable UE shall support all source rates listed in Table 1. Table 1: Source codec bit-rates for the AMR codec. Codec mode Source codec bit-rate AMR_12.20 12.20 kbit/s (GSM EFR) AMR_10.20 10.20 kbit/s AMR_7.95 7.95 kbit/s AMR_7.40 7.40 kbit/s (IS-641) AMR_6.70 6.70 kbit/s (PDC-EFR) AMR_5.90 5.90 kbit/s AMR_5.15 5.15 kbit/s AMR_4.75 4.75 kbit/s AMR_SID 1.80 kbit/s * (*) Assuming SID frames are continously transmitted NOTE 1: GSM-EFR is the ETSI GSM 06.90 Enhanced Full Rate Speech Codec (also identical to the TIA TDMA-US1 Enhanced speech codec) NOTE 2: IS-641is the TIA/EIA IS-641 TDMA Enhanced Full Rate Speech Codec NOTE 3: PDC-EFR is the ARIB 6.7 kbit/s Enhanced Full Rate Speech Codec 6 Adaptive Multi-Rate speech codec ANSI C-code The ANSI-C code of the speech codec, VAD and CNG system are described in [3]. The ANSI C-code is mandatory. The ANSI C-code is identical to that of GSM 06.73. 7 Adaptive Multi-Rate speech codec test vectors A set of digital test sequences is specified in [4], thus enabling the verification of compliance, i.e. bit- exactness, to a high degree of confidence. The test vectors are identical to those of GSM 06.74. The test sequences are defined separately for: - The speech codec described in [2], (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 9 3G TS 26.071 version 3.0.1 - The VAD described in [6] , - The CN generation described in [7] The adaptive multi-rate speech transcoder, VAD, SCR system and comfort noise parts of the audio processing functions (see Figure 1) are defined in bit exact arithmetic. Consequently, they shall react on a given input sequence always with the corresponding bit exact output sequence, provided that the internal state variables are also always exactly in the same state at the beginning of the test. The input test sequences provided shall force the corresponding output test sequences, provided that the tested modules are in their home-state when starting. The modules may be set into their home states by provoking the appropriate homing-functions. NOTE: This is normally done during reset (initialisation of the codec). Special inband signalling frames (encoder-homing-frame and decoder-homing-frame) described in [2]have been defined to provoke these homing-functions also in remotely placed modules. At the end of the first received homing frame, the audio functions that are defined in a bit exact way shall go into their predefined home states. The output corresponding to the first homing frame is dependent on the codec state when the frame was received. Any consecutive homing frames shall produce corresponding homing frames at the output. 8 Adaptive Multi-Rate speech codec source controlled rate operation The source controlled rate operation of the adaptive multi-rate speech codec is defined in [5]. During a normal telephone conversation, the participants alternate so that, on the average, each direction of transmission is occupied about 50 % of the time. Source controlled rate (SCR) is a mode of operation where the speech encoder encodes speech frames containing only background noise with a lower bit-rate than normally used for encoding speech. A network may adapt its transmission scheme to take advantage of the varying bit-rate. This may be done for the following two purposes: 1) In the UE, battery life will be prolonged or a smaller battery could be used for a given operational duration. 2) The average required bit-rate is reduced, leading to a more efficient transmission with decreased load and hence increased capacity. The following functions are required for the source controlled rate operation: - a Voice Activity Detector (VAD) on the TX side; - evaluation of the background acoustic noise on the TX side, in order to transmit characteristic parameters to the RX side; - generation of comfort noise on the RX side during periods when no normal speech frames are received. The transmission of comfort noise information to the RX side is achieved by means of a Silence Descriptor (SID) frame, which is sent at regular intervals. (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 10 3G TS 26.071 version 3.0.1 9 Adaptive Multi-Rate speech codec voice activity detection The adaptive multi-rate VAD function is described in [6]. The input to the VAD is the input speech itself together with a set of parameters computed by the adaptive multi-rate speech encoder. The VAD uses this information to decide whether each 20 ms speech coder frame contains speech or not. The VAD algorithm is described in [6], and the corresponding C code is defined in [3]. The verification of compliance to [6]. is achieved by use of digital test sequences applied to the same interface as the test sequences for the speech codec. 10 Adaptive Multi-Rate speech codec comfort noise insertion The adaptive multi-rate comfort noise insertion function is described in [7]. When speech is absent, the synthesis in the speech decoder is different from the case when normal speech frames are received. The synthesis of an artificial noise based on the received non-speech parameters is termed comfort noise generation. The comfort noise generation process is as follows: - the evaluation of the acoustic background noise in the transmitter; - the noise parameter encoding (SID frames) and decoding, and - the generation of comfort noise in the receiver. The comfort noise processes and the algorithm for updating the noise parameters during speech pauses are defined in detail in [7], and the corresponding C code is defined in [3]. The comfort noise mechanism is based on the adaptive multi-rate speech codec defined in [2]. 11 Adaptive Multi-Rate speech codec error concealment of lost frames The adaptive multi-rate speech codec error concealment of lost frames is described in [8]. Frames may be lost due to transmission errors or frame stealing in a wireless environment. Actions which shall be taken in these cases, both for lost speech frames and for lost SID frames are described in [8]. Error concealment actions shall be used also in the case of lost speech packets in the transport network. The methods described in [8] may with some modifications be used as a basis for such actions. In order to mask the effect of isolated lost frames, the speech decoder shall be informed and the error concealment actions shall be initiated, whereby a set of predicted parameters are used in the speech synthesis. Insertion of speech signal independent silence frames is not allowed. For several subsequent lost frames, a muting technique shall be used to indicate to the listener that transmission has been interrupted. (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 11 3G TS 26.071 version 3.0.1 12 Adaptive Multi-Rate speech codec frame structure The adaptive multi-rate speech frame structure is described in [9]. The output interface format from the encoder and input interface format to the decoder is divided into two parts; the core speech data part, which is the speech coded bits, and the other part is an additional data part with mode information. The interface format described in [9] is termed AMR interface format 1 (AMR IF1). Annex A of [9] describes an octet aligned frame format which shall be used in applications requiring octet alignment, such as for 3G H.324. This format is termed AMR interface format 2 (AMR IF2). 13 Adaptive Multi-Rate speech codec interface to RAN The adaptive multi-rate speech service interface to RAN is described in [10]. [F.F.S] 14 Adaptive Multi-Rate speech codec performance characterisation The adaptive multi-rate speech channel performance characterisation is described in [11]. [F.F.S.] (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 3GPP 3G TS 26.071 V3.0.1 (1999-08) 12 3G TS 26.071 version 3.0.1 Annex A: Change history Document history V. 0.1.0 March 1999 First Draft V. 0.1.1 April 1999 References changed V. 1.0.0 April 22, 1999 Editorial changes V 2.0.0 June 15, 1999 Minor Editorial changes V 3.0.0 June 22, 1999 Approved at 3GPP TSG SA#4 Plenary meeting V 3.0.1 August 22, 1999 Reformatted (3G TS 26.071 version 3.0.1 Release 1999) ETSI TS 126 071 V3.0.1 (2000-01) ETSI 13 ETSI ETSI TS 126 071 V3.0.1 (2000-01) (3G TS 26.071 version 3.0.1 Release 1999) History Document history V3.0.1 January 2000 Publication
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...................................................................................................................................................... 7 Annex A (informative): Change History................................................................................................8 History................................................................................................................................................................9 3G TS 25.442 version 3.1.0 Release 1999 ETSI 4 3G TS 25.442 V3.1.0 ETSI TS 125 442 V3.1.0 (2000-03) Foreword This Technical Specification (TS) has been produced by the 3rd Generation Partnership Project (3GPP). The contents of this informal TS may be subject to continuing work within the 3GPP and may change following formal TSG-S4 approval. Should TSG-S4 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 x.y.z where: x the first digit: 1 presented to TSG for information; 2 presented to TSG for approval; 3 or greater indicates TSG approved document under change control. y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. z the third digit is incremented when editorial only changes have been incorporated in the document. 3G TS 25.442 version 3.1.0 Release 1999 ETSI 5 3G TS 25.442 V3.1.0 ETSI TS 125 442 V3.1.0 (2000-03) 1 Scope The present document specifies the transport of implementation specific O&M signalling between Node B and the Management Platform in case that the transport is routed via the RNC. 2 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. • For this Release 1999 document, references to 3G documents are for Release 1999 versions (version 3.x.y). [1] 3GPP TS 25.431: "UTRAN Iub interface Layer 1". [2] 3GPP TS 25.401: "UTRAN Overall Description". [3] ITU-T Recommendation I.363.5 (8/1996): "B-ISDN ATM Adaptation Layer Type 5 Specification". [4] IETF RFC 2225 (4/1998): "Classical IP and ARP over ATM". [5] IETF RFC 2684 (9/1999): "Multiprotocol Encapsulation over ATM Adaptation Layer 5". [6] IETF RFC 791 (9/1981): "Internet Protocol". 3 Definitions and abbreviations 3.1 Definitions For the purposes of the present document, the following terms and definitions apply: Logical O&M: Logical O&M is the signalling associated with the control of logical resources owned by the RNC but physically implemented in Node B. Implementation Specific O&M: Implementation Specific O&M functions depend on the implementation of the Node B, both for it’s hardware and software components. 3.2 Abbreviations For the purposes of the present document, the following abbreviations apply: AAL5 ATM Adaptation Layer type 5 ATM Asynchronous Transfer Mode ARP Address Resolution Protocol RFC Request For Comment IP Internet Protocol O&M Operation and Maintenance RNC Radio Network Controller 3G TS 25.442 version 3.1.0 Release 1999 ETSI 6 3G TS 25.442 V3.1.0 ETSI TS 125 442 V3.1.0 (2000-03) 4 Implementation Specific O&M Transport 4.1 Requirements While this specification only addresses the transport of Node B Implementation Specific O&M signalling, many of the following requirements are derived from generic requirements for O&M of UMTS network elements: • Common O&M infrastructure for all network elements. • Independence from various data link protocols. • Support of various higher layer protocols and applications. • Secure transmission. • No Impact of O&M transport on traffic transport and signalling. • Re-use of existing transport facilities, i.e. co-existence of Iub and Implementation Specific O&M on the same bearer. 4.2 Routing It is the responsibility of the RNC to route Implementation Specific O&M signalling traffic. The traffic exchanged over this signalling link is completely transparent to the RNC. Both RNC and Node B have to support the routing of Implementation specific O&M via the RNC. Management Platform(s) Node B RNC Management Model Implementation Specific O&M Iub Logical O&M Node B Management Model Node B Implementation Specific O&M Logical O&M RNC O&M Node B Logical O&M Node B Management Model RNC Iub Implementation Specific O&M transport Implementation Specific O&M Transport Physical bearer Physical bearer Figure 1: Implementation Specific O&M Transport via RNC 3G TS 25.442 version 3.1.0 Release 1999 ETSI 7 3G TS 25.442 V3.1.0 ETSI TS 125 442 V3.1.0 (2000-03) An appropriate transport bearer for Implementation Specific O&M should consider the requirements listed in subclause 4.1. IP [6] should be the transport mechanism in order to allow a data link independent support of a variety of O&M applications and protocols for the Implementation Specific O&M of the Node B. IP datagrams containing O&M signalling have to be carried over the same bearer as Iub. Since ATM will be used on Iub, IP over ATM should be the bearer for O&M signalling. The following figure shows the protocol stack for Implementation Specific O&M transport between Node B and RNC: IP AAL5 ATM PHY Data Link Layer PHY IP AAL5 ATM PHY Implementation Specific O&M RNC Node B Figure 2: Protocol Stack for Implementation Specific O&M Transport AAL5 shall be used according to ITU-T Recommendation I.363.5. AAL5 virtual circuits are used to transport the IP packets containing Implementation Specific O&M signalling data between Node B and RNC. Multiple VCs can be used over the interface. An association shall be made between a VC and the IP addresses that are related to this VC in the peer node side. This association can be made using O&M or using ATM Inverse ARP according to Classical IP over ATM. Classical IP over ATM protocols are used to carry the IP packets over the ATM transport network. Classical IP over ATM is specified in IETF RFC 2225. Multiprotocol Encapsulation over AAL5 is specified in IETF RFC 2684. 3G TS 25.442 version 3.1.0 Release 1999 ETSI 8 3G TS 25.442 V3.1.0 ETSI TS 125 442 V3.1.0 (2000-03) Annex A (informative): Change History Change history TSG RAN# Version CR Tdoc RAN New Version Subject/Comment RAN_05 - - 3.0.0 A Approved at TSG RAN #5 by correspondence and placed under Change Control RAN_07 3.0.0 - - 3.1.0 Approved at TSG RAN #7 3G TS 25.442 version 3.1.0 Release 1999 9 ETSI ETSI TS 125 442 V3.1.0 (2000-03) 3G TS 25.442 version 3.1.0 Release 1999 History Document history V3.1.0 March 2000 Publication
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............................................................................................................................................... 6 History................................................................................................................................................................8 (3G TS 25.432 version 3.1.0 Release 1999) ETSI TS 125 432 V3.1.0 (2000-01) ETSI 3GPP TS 25.432 V3.1.0 (1999-10) 4 Foreword This Technical Specification has been produced by the 3GPP. The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of this TS, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: Version 3.y.z where: x the first digit: 1999.´ presented to TSG for information; 1999.´ presented to TSG for approval; 1999.´ Indicates TSG approved document under change control. Y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. z the third digit is incremented when editorial only changes have been incorporated in the specification. ETSI TS 125 432 V3.1.0 (2000-01) (3G TS 25.432 version 3.1.0 Release 1999) ETSI 3GPP TS 25.432 V3.1.0 (1999-10) 5 1 Scope The present document specifies the signalling transport related to NBAP signalling to be used across the Iub Interface. The Iub interface is a logical interface for the interconnection of NodeB and Radio Network Controller (RNC) components of the UMTS Terrestrial Radio Access Network (UTRAN) for the UMTS system. The radio network control signalling between these nodes is based on the NodeB application part (NBAP). 2 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. [1] ITU-T Recommendation Q.2100 (07/94). “B-ISDN signalling ATM adaptation layer (SAAL) overview description”. [2] ITU-T Recommendation Q.2130 (07/94). “B-ISDN signalling ATM adaptation layer – Service specific coordination function for support of signalling at the user network interface (SSCF–UNI)”. [3] ITU-T Recommendation Q.2110 (07/94). “B-ISDN ATM adaptation layer – Service specific connection oriented protocol (SSCOP)”. [4] ITU-T Recommendation I.363.5 (08/96). “B-ISDN ATM Adaptation Layer Type 5 Specification”. [5] ITU-T Recommendation I.361 B-ISDN ATM Layer Specification (11/95) [6] ITU-T Rec. I.630 (2/99) ATM Protection Switching 3 Definitions, symbols and abbreviations 3.1 Definitions For the purposes of the present document, the [following] terms and definitions [given in … and the following] apply. ETSI TS 125 432 V3.1.0 (2000-01) (3G TS 25.432 version 3.1.0 Release 1999) ETSI 3GPP TS 25.432 V3.1.0 (1999-10) 6 3.2 Symbols For the purposes of the present document, the following symbols apply: 3.3 Abbreviations For the purposes of the present document, the following abbreviations apply: AAL ATM Adaptation Layer ATM Asynchronous Transfer Mode NBAP NodeB Application Part RNC Radio Network Controller SAAL Signalling ATM Adaptation Layer SSCF Service Specific Coordination Function SSCOP Service Specific Connection Oriented Protocol UNI User-Network Interface 4 ATM Layer 4.1 General ATM shall be used in the radio network control plane according to I.361 [5] 4.2 Protection Switching at ATM Layer If redundancy of pathways at ATM layer between RNC and Node B is supported, it shall be implemented using ATM Protection Switching according to I.630 [6]. 5 NBAP Signalling Bearer 5.1 Introduction The Signalling Bearer for NBAP is a point-to-point protocol. There may be multiple point-to-point links between an RNC and a NodeB. 5.2 Signalling Bearer The signalling bearer in the Radio Network Control Plane is SAAL-UNI [1] over ATM. The figure below shows the protocols to be used to support NBAP signalling. These are SSCF-UNI [2] on top of SSCOP [3] and AAL Type 5 [4]. ETSI TS 125 432 V3.1.0 (2000-01) (3G TS 25.432 version 3.1.0 Release 1999) ETSI 3GPP TS 25.432 V3.1.0 (1999-10) 7 NodeB Application Part (NBAP) ATM AAL Type 5 SSCF-UNI SSCOP Figure 1: Iub NBAP Signalling Transport ETSI TS 125 432 V3.1.0 (2000-01) (3G TS 25.432 version 3.1.0 Release 1999) ETSI 8 ETSI ETSI TS 125 432 V3.1.0 (2000-01) (3G TS 25.432 version 3.1.0 Release 1999) History Document history V3.1.0 January 2000 Publication
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.............................................................................................. 7 Annex A: Change history........................................................................................................................8 History................................................................................................................................................................9 ETSI TS 124 096 V3.0.0 (2000-01) (3G TS 24.096 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.096 V3.0.0 (1999-05) 4 3G TS 24.096 version 3.0.0 Foreword This Technical Specification has been produced by the 3GPP. This TS specifies the procedures used at the radio interface for normal operation, registration, erasure, activation, deactivation, invocation and interrogation of line identification supplementary services within the 3GPP system. The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of this TS, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: Version 3.y.z where: x the first digit: 1 presented to TSG for information; 2 presented to TSG for approval; 3 Indicates TSG approved document under change control. y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. z the third digit is incremented when editorial only changes have been incorporated in the specification; ETSI TS 124 096 V3.0.0 (2000-01) (3G TS 24.096 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.096 V3.0.0 (1999-05) 5 3G TS 24.096 version 3.0.0 1 Scope This Technical Specification (TS ) specifies the procedures used at the radio interface for normal operation, registration, erasure, activation, deactivation, invocation and interrogation of name identification supplementary services. Provision and withdrawal of supplementary services is an administrative matter between the mobile subscriber and the service provider and cause no signalling on the radio interface. In GSM 04.10 the general aspects of the specification of supplementary services at the layer 3 radio interface are given. GSM 04.80 specifies the formats and coding for the supplementary services. Definitions and descriptions of supplementary services are given in GSM 02.04 and GSM 02.8x and 02.9x-series. GSM 02.96 is related specially to name identification supplementary services. Technical realization of supplementary services is described in GSM 03.11 and GSM 03.8x and 03.9x-series. Technical specification GSM 03.96 is related specially to name identification supplementary services. The procedures for Call Control, Mobility Management and Radio Resource management at the layer 3 radio interface are defined in GSM 04.07 and GSM 04.08. The following supplementary services belong to the name identification supplementary services and are described in this TS: - Calling name presentation (CNAP); 2 Normative 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. [1] GSM 01.04: "Digital cellular telecommunication system (Phase 2+); Abbreviations and acronyms". [2] GSM 02.04: "Digital cellular telecommunications system (Phase 2+); General on supplementary services". [3] GSM 02.96: "Digital cellular telecommunications system (Phase 2+); Name identification supplementary services - Stage 1". [4] GSM 03.11: "Digital cellular telecommunications system (Phase 2+); Technical realization of supplementary services". [5] GSM 03.96: "Digital cellular telecommunications system(Phase 2+); Name identification supplementary services - Stage 2". [6] GSM 04.07: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface signalling layer 3; General aspects". [7] GSM 04.08: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface layer 3 specification". ETSI TS 124 096 V3.0.0 (2000-01) (3G TS 24.096 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.096 V3.0.0 (1999-05) 6 3G TS 24.096 version 3.0.0 [8] GSM 04.10: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface layer 3; Supplementary services specification; General aspects". [9] GSM 04.80: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface layer 3 supplementary services specification; Formats and coding". 3 Abbreviations Abbreviations used in the TS are listed in GSM 01.04. 4 Calling Name Presentation (CNAP) 4.1 Normal operation The calling name identity is made up of calling party's name - up to 80 characters. The MS shall be given one of the following in the name indicator: - calling name identity; - presentation indicator of presentation restricted; - presentation indicator of name unavailable; or - calling name identity and presentation restricted (for the case where the CNAP override category is provisioned). If the network has received a non-zero SS screening indicator from the MS, then calling name information indicated above shall be sent to the MS as defined in figure 1. If the network did not receive a non-zero SS screening indicator form the MS, then calling name information shall not be sent to the MS. MS Network SETUP <------------------------------------------------------------------------------------------------------- Facility (Invoke = NotifySS (CNAP, nameIndicator )) or MS Network SETUP <------------------------------------------------------------------------------------------------------- : : FACILITY <------------------------------------------------------------------------------------------------------- Facility (Invoke = NotifySS (CNAP, nameIndicator)) Figure 1: Notification by the network to the called mobile subscriber ETSI TS 124 096 V3.0.0 (2000-01) (3G TS 24.096 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.096 V3.0.0 (1999-05) 7 3G TS 24.096 version 3.0.0 4.2 Interrogation Status Check The mobile subscriber can request the status of the supplementary service and be informed if the service is provided to him/her. MS Network REGISTER -------------------------------------------------------------------------------------------------------> Facility (Invoke = InterrogateSS (CNAP)) RELEASE COMPLETE <------------------------------------------------------------------------------------------------------- Facility (Return Result = InterrogateSS (SS-Status)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_problem)) Figure 2: Interrogation of calling name identification presentation Activation, deactivation, registration and erasure of the supplementary service Calling Name presentation (CNAP) are not applicable. ETSI TS 124 096 V3.0.0 (2000-01) (3G TS 24.096 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.096 V3.0.0 (1999-05) 8 3G TS 24.096 version 3.0.0 Annex A: Change history Change history TSG CN# Spec Version CR <Phase> New Version Subject/Comment Apr 1999 GSM 04.96 6.0.1 Transferred to 3GPP CN1 CN#03 24.096 3.0.0 Approved at CN#03 ETSI TS 124 096 V3.0.0 (2000-01) (3G TS 24.096 version 3.0.0 Release 1999) ETSI 9 ETSI ETSI TS 124 096 V3.0.0 (2000-01) (3G TS 24.096 version 3.0.0 Release 1999) History Document history V3.0.0 January 2000 Publication
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1 Scope
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2 References
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3 Definitions and abbreviations
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3.1 Definitions
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4 General
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4.1 Overview
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4.3 CCBS Recall and CCBS Call Set-up
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4.3.1 CCBS Call Set-up (MS A idle)
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4.3.2 CCBS Call Set-up (MS A not idle)
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4.3.2.1 Existing call released
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4.4 Deactivation
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4.5 Interrogation
.................................................................................................................................................... 16 Annex A (Informative): Operation for Non-Supporting MS’s...........................................................18 A.0 Scope......................................................................................................................................................18 A.1 MSs which do not support CCBS..........................................................................................................18 A.1.1 Activation for non supporting MSs.................................................................................................................. 18 A.2 CCBS Call Set-up for non supporting MSs...........................................................................................20 A.3 Deactivation for non supporting MSs....................................................................................................21 A.4 Interrogation for non supporting MSs....................................................................................................21 Annex B: Change history......................................................................................................................22 History..............................................................................................................................................................23 ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 4 3G TS 24.093 version 3.0.0 Foreword This Technical Specification has been produced by the 3GPP. This TS gives the stage 3 description of the Completion of Calls to Busy Subscriber (CCBS) supplementary service within the 3GPP system. The contents of the present document are subject to continuing work within the TSG and may change following formal TSG approval. Should the TSG modify the contents of this TS, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows: Version 3.y.z where: x the first digit: 1 presented to TSG for information; 2 presented to TSG for approval; 3 Indicates TSG approved document under change control. y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. z the third digit is incremented when editorial only changes have been incorporated in the specification; ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 5 3G TS 24.093 version 3.0.0 1 Scope The present dcoument gives the stage 3 description of the Completion of Calls to Busy Subscriber (CCBS) supplementary service. The present document specifies the procedures used at the radio interface (Reference point Um as defined in GSM 04.02) for normal operation, activation, deactivation, invocation and interrogation of the completion of calls to busy subscriber supplementary services. Provision and withdrawal of supplementary services is an administrative matter between the mobile subscriber and the service provider and cause no signalling on the radio interface. In GSM 04.10 the general aspects of the specification of supplementary services at the layer 3 radio interface are given. GSM 04.80 specifies the formats and coding for the supplementary services. Definitions and descriptions of supplementary services are given in GSM 02.04, GSM 02.8x and GSM 02.9x-series. Technical specification GSM 02.93 is related specifically to the Completion of Calls to Busy Subscriber supplementary service. The technical realization of supplementary services is described in technical specifications GSM 03.11, GSM 03.8x and 03.9x-series. GSM 03.93 is related specifically to Completion of Calls to Busy Subscriber supplementary service. The procedures for Call Control, Mobility Management and Radio Resource management at the layer 3 radio interface are defined in GSM 04.07 and GSM 04.08. The following supplementary services belong to the call completion supplementary services and are described in the present document: - Completion of Calls to Busy Subscriber (CCBS) (see clause 4). 2 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. [1] GSM 01.04: "Digital cellular telecommunications system (Phase 2+); "Abbreviations and acronyms". [2] GSM 02.04: "Digital cellular telecommunications system (Phase 2+); General on supplementary services". [3] GSM 02.07: "Digital cellular telecommunications system (Phase 2+); Mobile Stations (MS) features". [4] GSM 02.30: "Digital cellular telecommunications system (Phase 2+); Man-Machine Interface (MMI) of the Mobile Station (MS)". [5] GSM 02.93: "Digital cellular telecommunications system (Phase 2+); Completion of Calls to Busy Subscriber (CCBS) Service description, Stage 1". [6] GSM 03.11: "Digital cellular telecommunications system (Phase 2+); Technical realization of supplementary services". ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 6 3G TS 24.093 version 3.0.0 [7] GSM 03.93: "Digital cellular telecommunications system (Phase 2+); Technical realization of Completion of Calls to Busy Subscriber (CCBS)". [8] GSM 04.02: "Digital cellular telecommunications system (Phase 2+); GSM Public Land Mobile Network (PLMN) access reference configuration". [9] GSM 04.07: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface signalling layer 3 General aspects". [10] GSM 04.08: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface layer 3 specification". [11] GSM 04.10: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface layer 3 Supplementary services specification General aspects". [12] GSM 04.80: "Digital cellular telecommunications system (Phase 2+); Mobile radio interface layer 3 supplementary services specification Formats and coding". [13] GSM 04.83: "Digital cellular telecommunications system (Phase 2+); Call Waiting (CW) and Call Hold (HOLD) supplementary services - Stage 3". 3 Definitions and abbreviations 3.1 Definitions For the purposes of the present document the following definitions apply. Subscriber A: The user of MS A, requesting CCBS. Destination B: The entity addressed in the original call set up, which is busy when first called by subscriber A. 3.2 Abbreviations For the purposes of the present document the following abbreviations apply: CCBS Completion of Calls to Busy Subscriber MS A Mobile Station of subscriber A NDUB Network Determined User Busy Further related abbreviations are given in GSM 01.04 ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 7 3G TS 24.093 version 3.0.0 4 General 4.1 Overview CCBS allows a calling subscriber A, encountering a Network Determined User Busy (NDUB) called destination B, to be notified when destination B is idle and to have the network reinitiate the call to destination B, if subscriber A desires. All of the radio signalling specific to CCBS is at the subscriber A-side. Each procedure is described in turn. There is no radio signalling specific to CCBS at destination B-side. The radio signalling on the destination B-side uses basic call signalling procedures only. A mobile station that supports CCBS shall support the requirements of the following options used in GSM 04.08: 1) Prolonged Clearing Procedure; 2) Network Initiated Mobile Originated Connection Management (MO CM) Connection Request; 3) Network initiated MO call. A network supporting CCBS shall support the requirements of the following options used in GSM 04.08: 1) CCBS Request activation; and 2) Network initiated MO call. 4.2 Activation When CCBS is allowed the network shall give subscriber A the option of activating a CCBS Request. The network shall send a DISCONNECT message to MS A (cause #17 (User Busy) or cause #34 (no circuit / channel available)) with diagnostic field indicating CCBS is Possible and allowed actions indicating CCBS is Possible. The network starts the retention timer T1 when it sends the DISCONNECT message. The MS shall not release the connection with the network if allowed actions is present. If subscriber A attempts to activate a CCBS Request, MS A shall send a RELEASE message, with the Facility information element indicating CCBSRequest and the network shall stop T1. If the subscriber A does not accept CCBS activation, the MS shall send normal RELEASE message and the network shall stop T1 and continue normal call clearing. If the timer T1 expires before the RELEASE message is received from the MS, the network shall continue normal call clearing. If the network accepts the activation attempt, it shall acknowledge it by sending a RELEASE COMPLETE message containing the Facility information element with the CCBS index and optionally the AddressOfB, SubAddressOfB and the BasicServiceCode. If the network rejects the activation attempt, it shall send a RELEASE COMPLETE message containing the Facility information element with a return error indication. If a TCH has been allocated for the initial call and there are no further need for this channel configuration, the network may reconfigure the ongoing connection from TCH(s) mode to SDCCH only mode while waiting for further user input activity. It is a network option to maintain the ongoing connection in TCH mode while waiting for further user input activity. SS Version Indicator value 3 or above has to be used. ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 8 3G TS 24.093 version 3.0.0 MS A NETWORK SETUP (Note 1) -------------------------------------------------------------------------------------------------------------------------------> (Bearer capability, CC capabilities, Called party BCD number) DISCONNECT (Note 2) <------------------------------------------------------------------------------------------------------------------------------ ((Cause #17 (User Busy) / Cause #34 (no circuit/channel available)), diagnostic = CCBSPossible, allowed actions = CCBS Possible) RELEASE (Note 3) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - RELEASE (Note 4) <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - ((Cause #17 (User Busy) / Cause #34 (no circuit/channel available) ) / Cause #102 (recovery on timer expiry)) RELEASE COMPLETE - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> RELEASE ---------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = AccessRegisterCCEntry) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return Result (CCBS Index, AddressOfB, Sub_AddressOfB, BasicServiceCode)) (Note 5) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error = (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject = (Invoke Problem)) NOTE 1: The original call set-up is shown for completeness. NOTE 2: The CCBS activation is possible only when allowed actions field contains CCBS Possible indication NOTE 3: If Subscriber A rejects the CCBS Possible indication, then the MS shall send RELEASE. NOTE 4: If T1 (Retention timer) expires then the network shall send a RELEASE message to the MS. The Timer expiry cause may be included. NOTE 5: AddressOfB, Sub_AddressOfB and BasicServiceCode are optional parameters. Figure 4.2: Activation of a CCBS Request for supporting MSs ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 9 3G TS 24.093 version 3.0.0 4.3 CCBS Recall and CCBS Call Set-up When destination B becomes free the network shall offer subscriber A the option of recalling destination B. The network shall prompt MS A to allocate a Transaction Identifier (TI) and establish the CC connection by sending a CM SERVICE PROMPT message. MS A establishes the CC connection by sending a START CC message to the network. The network shall then send a CC ESTABLISHMENT message to MS A and shall include the Setup container. The Setup container contains information necessary to set-up the CCBS Call. The MS can modify the Bearer Capability (BC), High Level Compatibility (HLC) and Low Level Compatibility (LLC) information within the Setup container provided that the same Basic Service Group is maintained. If MS A is compatible with the basic service group it sends CC ESTABLISHMENT CONFIRMED message to the network. Once the network has received the CC ESTABLISHMENT CONFIRMED message it shall send a RECALL message to MS A, which contains information to be presented to the subscriber. At this stage, if the MS detects that ACM ≥ ACMmax, the MS shall interrupt the recall procedure, shall not alert the user and shall send a RELEASE COMPLETE message with the appropriate cause value to the network. If subscriber A accepts the CCBS recall, MS A shall establish a new call with the SETUP message. MSC A shall maintain the RR connection with MS A throughout the time when acceptance of the CCBS Recall is possible. Once the SETUP message is received, normal call handling continues. 4.3.1 CCBS Call Set-up (MS A idle) Figure 4.3.1 shows the case where MS A is idle when a CCBS Recall is received by the originating network. The different possibilities for allocating a traffic channel are described in GSM 04.08. ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 10 3G TS 24.093 version 3.0.0 MS A NETWORK RR CONNECTION ESTABLISHED <--------------------------------------------------------------------------------------------------------------------------------> CM SERVICE PROMPT <---------------------------------------------------------------------------------------------------------------------------------- START CC ---------------------------------------------------------------------------------------------------------------------------------> MM_STATUS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #32 (Service option not supported) / Cause #34 (Service option temporarily out of order) / Cause #97 (message type non-existent or not implemented)) CC ESTABLISHMENT <---------------------------------------------------------------------------------------------------------------------------------- (Setup container) CC ESTABLISHMENT CONFIRMED ---------------------------------------------------------------------------------------------------------------------------------> (BC’(s)), (Note 1) RELEASE COMPLETE - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #29 (facility rejected) / Cause #88 (incompatible destination) / Cause #17 ( User Busy)) (Note 2) RECALL <---------------------------------------------------------------------------------------------------------------------------------- Facility (Invoke = NotifySS (SS-Code = CCBS, CCBS index, AddressOfB, Sub_AddressOfB, BasicServiceCode, Alerting Pattern)) (Note 3) RELEASE COMPLETE - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #21 (call rejected) / Cause #17 (User Busy) / Cause #68 (ACM ≥ ACMmax)) (Note 4) RELEASE COMPLETE < - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - (Cause #102 (recovery on timer expiry)) (Note 5) SETUP ----------------------------------------------------------------------------------------------------------------------------------> (Note 6) NOTE 1: The BC may be modified by the MS as long as the same Basic Service Group is maintained. NOTE 2: The MS releases the transaction if the BC,HLC,LLC received in the CC ESTABLISHMENT message are incompatible with the MS, the MS cannot decode the contents of the "advanced recall alignment" Facility information element correctly (see GSM 04.10) or the MS responds by indicating UDUB. NOTE 3: MS shall start CCBS Recall alerting on receiving the RECALL message. The CCBS timer T4 is started when the RECALL message is sent to the MS. Sub_Address information may be included. The Alerting Pattern parameter may be included by the network to give some indication about alerting (category or level). If supported in the MS, this optional parameter is to be used by the MS as specified in GSM 02.07. NOTE 4: Subscriber A explicitly rejects the CCBS Recall or subscriber A responds to the CCBS Recall by indicating UDUB or ACM ≥ ACMmax. NOTE 5: The network releases the transaction if CCBS timer T4 expires. NOTE 6: The information elements within the SETUP message are derived from the Setup container in the CC ESTABLISHMENT Message. The SETUP message must contain the same BC(s) that was (were) sent to the network in the CC ESTABLISHMENT CONFIRMED message. Figure 4.3.1: CCBS Call Set-up for supporting MSs - subscriber A idle when RECALL arrives ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 11 3G TS 24.093 version 3.0.0 4.3.2 CCBS Call Set-up (MS A not idle) If a CCBS Recall is offered to MS A and MS A is not idle, subscriber A may accept the CCBS Recall and either release the existing call or put the existing call on hold. 4.3.2.1 Existing call released MS A NETWORK CM SERVICE PROMPT <---------------------------------------------------------------------------------------------------------------------------------- START CC (TI = CCBS call) (Note 1) ----------------------------------------------------------------------------------------------------------------------------------> MM_STATUS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #32 (Service option not supported) / Cause #34 (Service option temporarily out of order) / Cause #97 (message type non-existent or not implemented)) CC ESTABLISHMENT (TI = CCBS call) <---------------------------------------------------------------------------------------------------------------------------------- (Setup container) CC ESTABLISHMENT CONFIRMED (TI = CCBS call) ---------------------------------------------------------------------------------------------------------------------------------> (BC’(s), Cause #17 ( User Busy)) (Note 2) RELEASE COMPLETE (TI = CCBS call) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #29 (facility rejected) / Cause #88 (incompatible destination) / Cause #17 (User Busy)) (Note 3) RECALL (TI = CCBS call) <------------------------------------------------------------------------------------------------------------------------------------- Facility (Invoke = NotifySS (SS-Code = CCBS, CCBS index, AddressOfB, Sub_AddressOfB, BasicServiceCode, Alerting Pattern)) (Note 4) RELEASE COMPLETE (TI = CCBS call) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #21 (call rejected) / Cause #17 (User Busy) / Cause #68 (ACM ≥ ACMmax)) (Note 5) RELEASE COMPLETE (TI = CCBS call) < - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - (Cause #102 (recovery on timer expiry)) (Note 6) DISCONNECT (TI = existing call) ----------------------------------------------------------------------------------------------------------------------------------> (Cause #16 (normal clearing)) (Note 1, Note 7) SETUP (TI = CCBS call) ----------------------------------------------------------------------------------------------------------------------------------> (Note 8) Figure 4.3.2: CCBS Recall arrives while MS involved in a call, the existing call is released on acceptance of the CCBS Recall ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 12 3G TS 24.093 version 3.0.0 Notes to figure 4.3.2: NOTE 1: A new TI value indicated by "TI = CCBS call" is allocated by the MS for the subsequent CCBS call. The already existing call is referred to by the TI value "TI = existing call". NOTE 2: The BC may be modified by the MS as long as the same Basic Service Group is maintained. The MS shall indicate "User Busy" if it is not idle. NOTE 3: The MS releases the transaction if the BC,HLC,LLC received in the CC ESTABLISHMENT message are incompatible with the MS, the MS cannot decode the contents of the "advanced recall alignment" Facility information element correctly (see GSM 04.10) or the MS responds by indicating UDUB. NOTE 4: MS shall start CCBS Recall alerting on receiving the RECALL message. The CCBS timer T10 is started when the RECALL message is sent to the MS. Sub_Address information may be included. The Alerting Pattern parameter may be included by the network to give some indication about alerting (category or level). If supported in the MS, this optional parameter is to be used by the MS as specified in GSM 02.07. NOTE 5: Subscriber A explicitly rejects the CCBS Recall or subscriber A responds to the CCBS Recall by indicating UDUB or ACM ≥ ACMmax. NOTE 6: The network releases the transaction if CCBS timer T10 expires. NOTE 7: The existing call is released to make resources available for the CCBS call. The existing call is released according to normal call clearing procedures (see GSM 04.08). NOTE 8: The information elements within the SETUP message are derived from the Setup container in the CC ESTABLISHMENT message. The SETUP message must contain the same BC(s) that was (were) sent to the network in the CC ESTABLISHMENT CONFIRMED message. ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 13 3G TS 24.093 version 3.0.0 4.3.2.2 Existing call placed on hold If the existing call is a telephony call, subscriber A may place this call on hold in order to accept the CCBS Recall. MS A NETWORK CM SERVICE PROMPT <---------------------------------------------------------------------------------------------------------------------------------- START CC (TI = CCBS call) (Note 1) ----------------------------------------------------------------------------------------------------------------------------------> MM_STATUS - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #32 (Service option not supported) / Cause #34 (Service option temporarily out of order) / Cause #97 (message type non-existent or not implemented)) CC ESTABLISHMENT(TI = CCBS call) <---------------------------------------------------------------------------------------------------------------------------------- (Setup container) CC ESTABLISHMENT CONFIRMED (TI = CCBS call) ---------------------------------------------------------------------------------------------------------------------------------> (BC’(s), Cause #17 ( User Busy)) (Note 2) RELEASE COMPLETE (TI = CCBS call) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - > (Cause #29 (facility rejected) / Cause #88 (incompatible destination) / Cause #17 (User Busy)) (Note 3) RECALL (TI = CCBS call) <------------------------------------------------------------------------------------------------------------------------------------- Facility (Invoke = NotifySS (SS-Code = CCBS, CCBS index, AddressOfB, Sub_AddressOfB, BasicServiceCode, Alerting Pattern)) (Note 4) RELEASE COMPLETE (TI = CCBS call) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -> (Cause #21 (call rejected) / Cause #17 (UDUB) / Cause #68 (ACM ≥ ACMmax)) (Note 5) RELEASE COMPLETE (TI = CCBS call) < - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -- (Cause #102 (recovery on timer expiry)) (Note 6) HOLD (TI = existing call) (Note 1, Note 7) ----------------------------------------------------------------------------------------------------------------------------------> HOLD ACKNOWLEDGE (TI = existing call) <---------------------------------------------------------------------------------------------------------------------------------- SETUP (TI = CCBS call) (Note 8) ----------------------------------------------------------------------------------------------------------------------------------> Figure 4.3.3: CCBS Recall arrives while MS involved in a call, the existing call is placed on hold on acceptance of the CCBS Recall ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 14 3G TS 24.093 version 3.0.0 Notes to figure 4.3.3: NOTE 1: A new TI value indicated by "TI = CCBS call" is allocated by the MS for the subsequent CCBS call. The already existing call is referred to by the TI value "TI = existing call". NOTE 2: The BC may be modified by the MS as long as the same Basic Service Group is maintained. The MS shall indicate "User Busy" if it is not idle. NOTE 3: The MS releases the transaction if the BC,HLC,LLC received in the CC ESTABLISHMENT message are incompatible with the MS, the MS cannot decode the contents of the "advanced recall alignment" Facility information element correctly, or the MS responds by indicating UDUB. NOTE 4: MS shall start CCBS Recall alerting on receiving the RECALL message. The CCBS timer T10 is started when the RECALL message is sent to the MS. Sub_Address information may be included. The Alerting Pattern parameter may be included by the network to give some indication about alerting (category or level). If supported in the MS, this optional parameter is to be used by the MS as specified in GSM 02.07. NOTE 5: Subscriber A explicitly rejects the CCBS Recall or subscriber A responds to the CCBS Recall by indicating UDUB or ACM ≥ ACMmax. NOTE 6: The network releases the transaction if CCBS timer T10 expires. NOTE 7: The existing call is placed on hold to make resources available for the CCBS call. The existing call is placed on hold according to normal call hold procedures (see GSM 04.83). NOTE 8: The information elements within the SETUP message derived from the Setup container in the CC ESTABLISHMENT Message. The SETUP message must contain the same BC(s) that was (were) sent to the network in the CC ESTABLISHMENT CONFIRMED message. ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 15 3G TS 24.093 version 3.0.0 4.4 Deactivation Subscriber A can perform the following operations: - deactivate all outstanding CCBS requests; - deactivate a specific CCBS request. MS A shall send a REGISTER message, with the Facility information element, indicating EraseCCEntry. SS Version Indicator value 3 or above has to be used. MS A NETWORK REGISTER --------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = EraseCCEntry (CCBS)) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return Result) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) Figure 4.4.1: Deactivation of all CCBS requests MS A NETWORK REGISTER --------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = EraseCCEntry (CCBS, CCBS Index)) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return Result) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) Figure 4.4.2: Deactivation of a specific CCBS request ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 16 3G TS 24.093 version 3.0.0 Subscriber A can perform an interrogation of the CCBS service, with the three possible outcomes: - the CCBS service is not provisioned for the subscriber; - the CCBS service is provisioned for the subscriber, but the queue of requests is empty; - the CCBS service is provisioned for the subscriber and there are requests in the queue. MS A shall send a REGISTER message, with the Facility information element, indicating InterrogateSS. SS Version Indicator value 2 or above has to be used. Depending on the outcome of the interrogation, the network shall return: a) SS-status set to not provisioned when the CCBS service is not provisioned (figure 4.5.1); b) SS status set to provisioned when the CCBS service is provisioned, but there are no outstanding requests (figure 4.5.2); c) SS-status set to provisioned and the list of outstanding CCBS requests in the queue (figure 4.5.3). For each request in the queue, the following data shall be returned: - CCBS index; - Address of B; - Sub-Address of B (optional); - Basic Service Code. MS A NETWORK REGISTER ---------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = InterrogateSS (CCBS)) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return Result = InterrogateSS (SS-status)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) Figure 4.5.1: Interrogation of the CCBS - service not provisioned ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 17 3G TS 24.093 version 3.0.0 MS A NETWORK REGISTER ---------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = InterrogateSS (CCBS)) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return Result = InterrogateSS (SS-status)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) Figure 4.5.2: Interrogation of the CCBS - the request queue is empty MS A NETWORK REGISTER ---------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = InterrogateSS(CCBS)) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return Result = InterrogateSS (SS-status, CCBS index, AddressOfB, Sub-AddressOfB, BasicServiceCode)) (Note) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) NOTE: The information for up to five CCBS Requests can be included. Figure 4.5.3: Interrogation of the CCBS - all existing requests ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 18 3G TS 24.093 version 3.0.0 Annex A (Informative): Operation for Non-Supporting MS’s A.0 Scope This annex is included for information only and is for further study. A.1 MSs which do not support CCBS MSs which do not explicitly support CCBS are not precluded from attempting to activate CCBS or from accepting a CCBS Recall. The mechanisms employed to offer the CCBS service to these MSs shall be a matter for individual networks. A.1.1 Activation for non supporting MSs The network shall send DISCONNECT to MS A (cause #17 or #34) with diagnostic field indicating CCBS is Possible, and a progress indicator indicating inband information is available. This inband information shall be used to indicate CCBS possible. The absence of a progress indicator in the DISCONNECT, prevents subscriber A from successfully activating CCBS. If subscriber A requests CCBS, MS A will send a REGISTER message, containing a ProcessUnstructuredSS-Request invoke component. The receiving network entity shall pass the data received in the request to the application handling USSD operations and shall wait for the response of the application. When the application accepts the request and terminates the dialogue, the network shall clear the transaction by sending a RELEASE COMPLETE message containing a return result component. If the network is unable to process the request received from the MS, it shall clear the call independent transaction by sending a RELEASE COMPLETE message containing a return error component. When the call independent transaction has been cleared, either the MS or the network shall release the call related transaction by sending a RELEASE COMPLETE message. ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 19 3G TS 24.093 version 3.0.0 MS A NETWORK DISCONNECT (PD = CC) <------------------------------------------------------------------------------------------------------------------------------------ (Cause = #17 or 34, Diagnostic = CCBS Possible) Progress Indicator = #8 REGISTER (PD = SS) ------------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = ProcessUnstructuredSS-Request (ussd-DataCodingScheme, ussd-String)) RELEASE COMPLETE (PD = SS) <------------------------------------------------------------------------------------------------------------------------------------ Facility (Return Result = ProcessUnstructuredSS-Request (ussd-DataCodingScheme, ussd-String)) RELEASE COMPLETE (PD = SS) <------------------------------------------------------------------------------------------------------------------------------------ Facility (Return Error = ProcessUnstructuredSS-Request (ussd-DataCodingScheme, ussd-String)) RELEASE COMPLETE (PD = SS) < - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE (PD = SS) < - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) RELEASE (PD = CC) (Note1) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - > RELEASE COMPLETE (PD = CC) < - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - RELEASE (PD = CC) (Note 2) < - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - RELEASE COMPLETE (PD = CC) - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - > NOTE 1: If Subscriber A rejects the CCBS Possible indication or ends the call having received either a successful or unsuccessful indication following an activation attempt, then the MS shall send a RELEASE message. NOTE 2: If the call control timer expires (T306) or if T1 expires, then network shall send a RELEASE message to MS. Figure A.1: Activation of CCBS for non supporting MSs ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 20 3G TS 24.093 version 3.0.0 A.2 CCBS Call Set-up for non supporting MSs The CCBS recall shall be treated as a mobile terminated call set-up. The network shall send a SETUP message to MS A, which causes the MS to ring indicating that destination B is now idle. If subscriber A accepts the CCBS recall, MS A shall establish a new call with the CONNECT message. MS A NETWORK SETUP <---------------------------------------------------------------------------------------------------------------------------------- (Calling Party BCD Number, BearerCapability, CCBS Remote User Free) CALL CONFIRMED ----------------------------------------------------------------------------------------------------------------------------------> ALERTING ----------------------------------------------------------------------------------------------------------------------------------> CONNECT ----------------------------------------------------------------------------------------------------------------------------------> CONNECT ACKNOWLEDGE <---------------------------------------------------------------------------------------------------------------------------------- START DTMF ----------------------------------------------------------------------------------------------------------------------------------> START DTMF ACKNOWLEDGE <---------------------------------------------------------------------------------------------------------------------------------- STOP DTMF ----------------------------------------------------------------------------------------------------------------------------------> STOP DTMF ACKNOWLEDGE <---------------------------------------------------------------------------------------------------------------------------------- ALERT <---------------------------------------------------------------------------------------------------------------------------------- FACILITY <---------------------------------------------------------------------------------------------------------------------------------- NOTE*: It is an operator option to request the user if he/she wishes to continue Figure A.2: CCBS Call Set-up for non supporting MSs Editors Note: The network needs to know that this is a CCBS Call so that the destination network can be informed. How does the originating network know that this is a CCBS Call? Can the network determine this based on the transaction identifier? Editors Note: This area is currently under discussion within SMG1 & SMG3 ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 21 3G TS 24.093 version 3.0.0 A.3 Deactivation for non supporting MSs MS A shall send a REGISTER message to the network, with the Facility information element, indicating ProcessUnstructuredSS-Request. Different MMI is required (as specified in GSM 02.30) for the three different deactivation operations, although each deactivation operation uses the USSD mechanism to transport the information to the network. MS A NETWORK REGISTER --------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = ProcessUnstructuredSS-Request (ussd-DataCodingScheme, ussd-String)) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return Result = ProcessUnstructuredSS-Request (ussd-DataCodingScheme, ussd-String)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) Figure A.3: Deactivation of all CCBS requests, the last or a single CCBS Request for non supporting MSs A.4 Interrogation for non supporting MSs MS A shall send a REGISTER message to the network, with the Facility information element, indicating ProcessUnstructuredSS-Request. Different MMI is required (as specified in GSM 02.30) for the two different interrogation operations, although each interrogation operation uses the USSD mechanism to transport the information to the network. MS A NETWORK REGISTER --------------------------------------------------------------------------------------------------------------------------------> Facility (Invoke = ProcessUnstructuredSS-Request (ussd-DataCodingScheme, ussd-String)) RELEASE COMPLETE <-------------------------------------------------------------------------------------------------------------------------------- Facility (Return result = ProcessUnstructuredSS-Request (ussd-DataCodingScheme, ussd-String)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Return Error (Error)) RELEASE COMPLETE <- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - Facility (Reject (Invoke_Problem)) Figure A.4: Interrogation of all CCBS requests or a single CCBS request for Non supporting MSs ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 3GPP 3G TS 24.093 V3.0.0 (1999-05) 22 3G TS 24.093 version 3.0.0 Annex B: Change history Change history TSG CN# Spec Version CR <Phase> New Version Subject/Comment Apr 1999 GSM 04.93 6.1.1 Transferred to 3GPP CN1 CN#03 24.093 3.0.0 Approved at CN#03 ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) ETSI 23 ETSI ETSI TS 124 093 V3.0.0 (2000-01) (3G TS 24.093 version 3.0.0 Release 1999) History Document history V3.0.0 January 2000 Publication
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1 Scope
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2 References
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3 Abbreviations
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4 Explicit Call Transfer (ECT)
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4.1 Normal operation
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4.2 Explicit Call Transfer invocation
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4.3 Notification to the remote parties
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4.3.1 Notification to the held remote party
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4.3.2 Notification to the active or alerting remote party
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4.4 Activation and deactivation
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