1、 ETSI TS 1Digital cellular telecoUniversal Mobile TelSpeech codec sAdaptive Multi-Rate - Tran(3GPP TS 26.1TECHNICAL SPECIFICATION126 190 V13.0.0 (2016communications system (Phaelecommunications System (LTE; c speech processing functionWideband (AMR-WB) speecanscoding functions .190 version 13.0.0 Re
2、lease 1316-01) hase 2+); (UMTS); ons; ech codec; 13) ETSI ETSI TS 126 190 V13.0.0 (2016-01)13GPP TS 26.190 version 13.0.0 Release 13Reference RTS/TSGS-0426190vd00 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
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9、go are Trade Marks registered and owned by the GSM Association. ETSI ETSI TS 126 190 V13.0.0 (2016-01)23GPP TS 26.190 version 13.0.0 Release 13Intellectual Property Rights IPRs essential or potentially essential to the present document may have been declared to ETSI. The information pertaining to th
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12、b server) which are, or may be, or may become, essential to the present document. Foreword This Technical Specification (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 id
13、entities or GSM identities. These should be interpreted as being references to the corresponding ETSI deliverables. 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“, “sh
14、all not“, “should“, “should not“, “may“, “need not“, “will“, “will not“, “can“ and “cannot“ are to be interpreted 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 dire
15、ct citation. ETSI ETSI TS 126 190 V13.0.0 (2016-01)33GPP TS 26.190 version 13.0.0 Release 13Contents Intellectual Property Rights 2g3Foreword . 2g3Modal verbs terminology 2g3Foreword . 5g31 Scope 6g32 Normative references . 6g33 Definitions, symbols and abbreviations . 6g33.1 Definitions 6g33.2 Symb
16、ols 7g33.3 Abbreviations . 12g34 Outline description . 12g34.1 Functional description of audio parts . 12g34.2 Preparation of speech samples 13g34.3 Principles of the adaptive multi-rate wideband speech encoder . 13g34.4 Principles of the adaptive multi-rate speech decoder . 16g34.5 Sequence and sub
17、jective importance of encoded parameters . 16g35 Functional description of the encoder 16g35.1 Pre-processing 16g35.2 Linear prediction analysis and quantization . 17g35.2.1 Windowing and auto-correlation computation 17g35.2.2 Levinson-Durbin algorithm 17g35.2.3 LP to ISP conversion 18g35.2.4 ISP to
18、 LP conversion 19g35.2.5 Quantization of the ISP coefficients . 20g35.2.6 Interpolation of the ISPs . 21g35.3 Perceptual weighting 21g35.4 Open-loop pitch analysis 22g35.4.1 6.60 kbit/s mode 22g35.4.2 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 and 23.85 kbit/s modes 23g35.5 Impulse response comp
19、utation . 24g35.6 Target signal computation 24g35.7 Adaptive codebook . 24g35.8 Algebraic codebook 26g35.8.1 Codebook structure . 26g35.8.1.1 23.85 and 23.05 kbit/s mode . 26g35.8.1.2 19.85 kbit/s mode 26g35.8.1.3 18.25 kbit/s mode 27g35.8.1.4 15.85 kbit/s mode 27g35.8.1.5 14.25 kbit/s mode 27g35.8.
20、1.6 12.65 kbit/s mode 28g35.8.1.7 8.85 kbit/s mode 28g35.8.1.8 6.60 kbit/s mode 28g35.8.2 Pulse indexing . 29g35.8.3 Codebook search . 32g35.9 Quantization of the adaptive and fixed codebook gains . 35g35.10 Memory update 36g35.11 High-band gain generation . 36g36 Functional description of the decod
21、er 36g36.1 Decoding and speech synthesis 37g36.2 High-pass filtering, up-scaling and interpolation . 39g36.3 High frequency band 40g3ETSI ETSI TS 126 190 V13.0.0 (2016-01)43GPP TS 26.190 version 13.0.0 Release 136.3.1 Generation of high-band excitation 40g36.3.2 LP filter for the high frequency band
22、 40g36.3.2.1 6.60 kbit/s mode 40g36.3.2.2 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 or 23.85 kbit/s modes . 41g36.3.3 High band synthesis 41g37 Detailed bit allocation of the adaptive multi-rate wideband codec 41g38 Homing sequences 47g38.1 Functional description 47g38.2 Definitions 48g38.3 Enc
23、oder homing . 48g38.4 Decoder homing . 48g39 Bibliography . 50g3Annex A (informative): Change history . 51g3History 52g3ETSI ETSI TS 126 190 V13.0.0 (2016-01)53GPP TS 26.190 version 13.0.0 Release 13Foreword The present document describes the detailed mapping of the wideband telephony speech service
24、 employing the Adaptive Multi-Rate (AMR-WB) 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 i
25、dentifying 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 Indicates TSG approved document under change control. y the second digit is incremented for all changes of su
26、bstance, 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 TS 126 190 V13.0.0 (2016-01)63GPP TS 26.190 version 13.0.0 Release 131 Scope This Telecommunication Standard (TS) descr
27、ibes the detailed mapping from input blocks of 320 speech samples in 16-bit uniform PCM format to encoded blocks of 132, 177, 253, 285, 317, 365, 397, 461 and 477 bits and from encoded blocks of 132, 177, 253, 285, 317, 365, 397, 461 and 477 bits to output blocks of 320 reconstructed speech samples.
28、 The sampling rate is 16 000 samples/s leading to a bit rate for the encoded bit stream of 6.60, 8.85, 12.65, 14.25, 15.85, 18.25, 19.85, 23.05 or 23.85 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
29、 ACELP. The multi-rate wideband ACELP coder is referred to as MRWB-ACELP. 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 hereaft
30、er. 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 s
31、ystem (Phase 2); Transmission planning aspects of the speech service in the GSM Public Land Mobile Network (PLMN) system“ 2 3GPP TS 26.201 : “AMR wideband speech codec; Frame structure“. 3 3GPP TS 26.194: “AMR wideband speech codec; Voice Activity Detection (VAD)“. 4 3GPP TS 26.173: “AMR wideband sp
32、eech codec; ANSI-C code“. 5 3GPP TS 26.174: “AMR wideband 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“. 3 Definitions, symbols and abbreviations 3.1 Definitions For the purpo
33、ses 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. algebraic c
34、odebook: 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 potential positions. The amplitudes and positions of the pulses of the kthexcitation codevector
35、can be derived from its index k through a rule requiring no or minimal physical storage, in contrast with stochastic codebooks whereby the path from the index to the associated codevector involves look-up tables. anti-sparseness processing: An adaptive post-processing procedure applied to the fixed
36、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,
37、 the lag is searched using error minimization loop (analysis-by-synthesis). In the adaptive multi-rate wideband 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 wideb
38、and codec, all filters which are used to modify speech samples use direct form coefficients. ETSI ETSI TS 126 190 V13.0.0 (2016-01)73GPP TS 26.190 version 13.0.0 Release 13fixed codebook: The fixed codebook contains excitation vectors for speech synthesis filters. The contents of the codebook are no
39、n-adaptive (i.e., fixed). In the adaptive multi-rate wideband 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 wideband codec a sub-sample resolution of 1/4th or 1/2nd of a sample is us
40、ed. frame: A time interval equal to 20 ms (320 samples at an 16 kHz sampling rate). Immittance Spectral Frequencies: (see Immittance Spectral Pair) Immittance Spectral Pair: Transformation of LPC parameters. Immittance Spectral Pairs are obtained by decomposing the inverse filter transfer function A
41、(z) to a set of two transfer functions, one having even symmetry and the other having odd symmetry. The Immittance Spectral Pairs (also called as Immittance Spectral Frequencies) are the roots of these polynomials on the z-unit circle. integer lags: A set of lag values having whole sample resolution
42、. interpolating filter: An FIR filter used to produce an estimate of sub-sample resolution samples, given an input sampled with integer sample resolution. In this implementation, the interpolating filter has low pass filter characteristics. Thus the adaptive codebook consists of the low-pass filtere
43、d interpolated past excitation. 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. LP ana
44、lysis 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 wideband codec, the length of the analysis window is always 384 samples. For all the modes, a single asymmetric window is u
45、sed to generate a single set of LP coefficients. The 5 ms look-ahead is used in the analysis. 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 alon
46、e, refers to the source codec mode, i.e., to one of the source codecs employed in the AMR-WB 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
47、small number of lags around the open-loop estimated lags. In the adaptive multi-rate wideband 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 th
48、e 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 we
49、ighting the error less in regions near the formant frequencies and more in regions away from them. subframe: A time interval equal to 5 ms (80 samples at 16 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
