1、 ETSI TS 126 090 V15.0.0 (2018-07) Digital cellular telecommunications system (Phase 2+) (GSM); Universal Mobile Telecommunications System (UMTS); LTE; Mandatory Speech Codec speech processing functions; Adaptive Multi-Rate (AMR) speech codec; Transcoding functions (3GPP TS 26.090 version 15.0.0 Rel
2、ease 15) TECHNICAL SPECIFICATION ETSI ETSI TS 126 090 V15.0.0 (2018-07)13GPP TS 26.090 version 15.0.0 Release 15Reference RTS/TSGS-0426090vf00 Keywords GSM,LTE,UMTS ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 00 Fax: +33 4 93 65 47 16 Siret N 348 623 562
3、00017 - NAF 742 C Association but non lucratif enregistre la Sous-Prfecture de Grasse (06) N 7803/88 Important notice The present document can be downloaded from: http:/www.etsi.org/standards-search The present document may be made available in electronic versions and/or in print. The content of any
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9、logo are trademarks registered and owned by the GSM Association. ETSI ETSI TS 126 090 V15.0.0 (2018-07)23GPP TS 26.090 version 15.0.0 Release 15Intellectual Property Rights Essential patents IPRs essential or potentially essential to normative deliverables may have been declared to ETSI. The informa
10、tion pertaining to these essential IPRs, if any, is publicly available for ETSI members and non-members, and can be found in ETSI SR 000 314: “Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in respect of ETSI standards“, which is available from the ET
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13、s being the property of ETSI, and conveys no right to use or reproduce any trademark and/or tradename. Mention of those trademarks in the present document does not constitute an endorsement by ETSI of products, services or organizations associated with those trademarks. Foreword This Technical Speci
14、fication (TS) has been produced by ETSI 3rd Generation Partnership Project (3GPP). The present document may refer to technical specifications or reports using their 3GPP identities, UMTS identities or GSM identities. These should be interpreted as being references to the corresponding ETSI deliverab
15、les. The cross reference between GSM, UMTS, 3GPP and ETSI identities can be found under http:/webapp.etsi.org/key/queryform.asp. Modal verbs terminology In the present document “shall“, “shall not“, “should“, “should not“, “may“, “need not“, “will“, “will not“, “can“ and “cannot“ are to be interpret
16、ed as described in clause 3.2 of the ETSI Drafting Rules (Verbal forms for the expression of provisions). “must“ and “must not“ are NOT allowed in ETSI deliverables except when used in direct citation. ETSI ETSI TS 126 090 V15.0.0 (2018-07)33GPP TS 26.090 version 15.0.0 Release 15Contents Intellectu
17、al Property Rights 2g3Foreword . 2g3Modal verbs terminology 2g3Foreword . 5g31 Scope 6g32 References 6g33 Definitions, symbols and abbreviations . 6g33.1 Definitions 6g33.2 Symbols 8g33.3 Abbreviations . 11g34 Outline description . 12g34.1 Functional description of audio parts . 12g34.2 Preparation
18、of speech samples 12g34.2.1 PCM format conversion 13g34.3 Principles of the adaptive multi-rate speech encoder . 13g34.4 Principles of the adaptive multi-rate speech decoder . 15g34.5 Sequence and subjective importance of encoded parameters . 16g35 Functional description of the encoder 16g35.1 Pre-p
19、rocessing (all modes) . 16g35.2 Linear prediction analysis and quantization . 16g35.2.1 Windowing and auto-correlation computation 17g35.2.2 Levinson-Durbin algorithm (all modes) . 18g35.2.3 LP to LSP conversion (all modes) 18g35.2.4 LSP to LP conversion (all modes) 20g35.2.5 Quantization of the LSP
20、 coefficients 20g35.2.6 Interpolation of the LSPs 22g35.2.7 Monitoring resonance in the LPC spectrum (all modes). 22g35.3 Open-loop pitch analysis 23g35.4 Impulse response computation (all modes) 26g35.5 Target signal computation (all modes) . 26g35.6 Adaptive codebook . 27g35.6.1 Adaptive codebook
21、search 27g35.6.2 Adaptive codebook gain control (all modes) 30g35.7 Algebraic codebook 31g35.7.1 Algebraic codebook structure . 31g35.7.2 Algebraic codebook search . 33g35.8 Quantization of the adaptive and fixed codebook gains . 37g35.8.1 Adaptive codebook gain limitation in quantization 37g35.8.2
22、Quantization of codebook gains 37g35.8.3 Update past quantized adaptive codebook gain buffer (all modes). 39g35.9 Memory update (all modes) 39g36 Functional description of the decoder 40g36.1 Decoding and speech synthesis 40g36.2 Post-processing . 43g36.2.1 Adaptive post-filtering (all modes) . 43g3
23、6.2.2 High-pass filtering and up-scaling (all modes) . 44g37 Detailed bit allocation of the adaptive multi-rate codec . 45g38 Homing sequences 49g38.1 Functional description 49g38.2 Definitions 50g38.3 Encoder homing . 50g38.4 Decoder homing . 50g3ETSI ETSI TS 126 090 V15.0.0 (2018-07)43GPP TS 26.09
24、0 version 15.0.0 Release 159 Bibliography . 54g3Annex A (informative): Change history . 55g3History 56g3ETSI ETSI TS 126 090 V15.0.0 (2018-07)53GPP TS 26.090 version 15.0.0 Release 15Foreword This Technical Specification has been produced by the 3rdGeneration Partnership Project (3GPP). The contents
25、 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 the present document, it will be re-released by the TSG with an identifying change of release date and an increase in version number as follows:
26、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 th
27、e third digit is incremented when editorial only changes have been incorporated in the document. ETSI ETSI TS 126 090 V15.0.0 (2018-07)63GPP TS 26.090 version 15.0.0 Release 151 Scope The present document describes the detailed mapping from input blocks of 160 speech samples in 13-bit uniform PCM fo
28、rmat 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
29、, 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
30、 described in the present document 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 the present document, see 4 for a description of the ANSI-C code. The transcoding procedure spec
31、ified in the present document is mandatory for systems using the AMR speech codec. 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, editio
32、n 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. In the case of a reference to a 3GPP document (including a GSM document), a non-specific reference implicitly refers to the late
33、st version of that document in the same Release as the present document. 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 3GPP TS 26.101 : “Frame Structure“. 3 3GPP TS 26.
34、094: “AMR Speech Codec; Voice Activity Detector“. 4 3GPP TS 26.073: “Adaptive Multi-Rate (AMR); ANSI C source code“. 5 3GPP TS 26.074: “Adaptive Multi-Rate (AMR); Test sequences“. 6 ITU-T Recommendation G.711 (1988): “Pulse code modulation (PCM) of voice frequencies“. 7 ITU-T Recommendation G.726: “
35、40, 32, 24, 16 kbit/s Adaptive Differential Pulse Code Modulation (ADPCM)“. 8 ITU-T Recommendation G.712 3 Definitions, symbols and abbreviations 3.1 Definitions For the purposes of the present document, the following terms and definitions apply: adaptive codebook: contains excitation vectors that a
36、re 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 re
37、constructed 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: fixed codebook where algebraic code is used to populate the excitation vectors (innovation vectors). The excitation conta
38、ins a small number of nonzero pulses with predefined interlaced sets of positions ETSI ETSI TS 126 090 V15.0.0 (2018-07)73GPP TS 26.090 version 15.0.0 Release 15anti-sparseness processing: adaptive post-processing procedure applied to the fixed codebook vector in order to reduce perceptual artefacts
39、 from a sparse fixed codebook vector closed-loop pitch analysis: 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-synt
40、hesis). 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 coeffici
41、ents. 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
42、 sub-sample resolution. In the adaptive multi-rate codec a sub-sample resolution of 1/6thor 1/3rdof a sample is used. frame: time interval equal to 20 ms (160 samples at an 8 kHz sampling rate) integer lags: set of lag values having whole sample resolution interpolating filter: FIR filter used to pr
43、oduce 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: long term filter delay. This is typi
44、cally 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 hav
45、ing 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 LP analysis window: for each frame, the short term filter coefficients are computed using the high pass filtered speech samp
46、les 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 g
47、enerate 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
48、 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: process of estimating the near optimal lag directly from the weighted speech input. This is done to simplify the pi
49、tch 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 filte
