ETSI EN 301 704-2000 Digital Cellular Telecommunications System (Phase 2+) Adaptive Multi-Rate (AMR) Speech Transcoding (GSM 06 90 Version 7 2 1 Release 1998)《数字蜂窝通信系统(第2+阶段) 适配多速率.pdf

《ETSI EN 301 704-2000 Digital Cellular Telecommunications System (Phase 2+) Adaptive Multi-Rate (AMR) Speech Transcoding (GSM 06 90 Version 7 2 1 Release 1998)《数字蜂窝通信系统(第2+阶段) 适配多速率.pdf》由会员分享，可在线阅读，更多相关《ETSI EN 301 704-2000 Digital Cellular Telecommunications System (Phase 2+) Adaptive Multi-Rate (AMR) Speech Transcoding (GSM 06 90 Version 7 2 1 Release 1998)《数字蜂窝通信系统(第2+阶段) 适配多速率.pdf（58页珍藏版）》请在麦多课文档分享上搜索。

1、ETSI EN 301 704 V7.2.1 (2000-04) European Standard (Telecomm unica tions series) Digital cellular telecommunications system (Phase 2+); Adaptive Multi-Rate (AMR) speech transcoding (GSM 06.90 version 7.2.1 Release 1998) (GSM 06.90 version 7.2.1 Release 1998) 2 ETSI EN 301 704 V7.2.1 (2000-04) Refere

2、nce RENEMG-1 10690Q7R1 Keywords Digital cellular telecommunications system, Global System for Mobile communication (GSM), speech, AMR ETSI 650 Route des Lucioles F-O6921 Sophia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 O0 Fax: +33 4 93 65 47 16 Siret No 348 623 562 0001 7 - NAF 742 C Association

3、 but non lucratif enregistre la Sous-prfecture de Grasse (06) No 7803/88 Important notice The pre ent d Individual copies of the present document can be downloaded from: http:/w.etsi.aq cument mav be made available in more than one electronic version or in orint. In anv case f existing or perceived

4、difference in coitents between such versions, the reference version is the Poiable Docuhent Format (PDF). In case of dispute, the reference shall be the printing on ETSI printers of the PDF version kept on a specific network drive within ETSI Secretariat. Users of the present document should be awar

5、e that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at ht?p:/.etsi.orn/lb/status/ If you find errors in the present document, send your comment to: editor et.si.fr Coovriaht Notification No part may be re

6、produced except as authorized by written permission. The copyright and the foregoing restriction extend to reproduction in all media. O European Telecommunications Standards Institute 2000. All rights reserved. ETSI (GSM 06.90 version 7.2.1 Release 1998) 3 ETSI EN 301 704 V7.2.1 (2000-04) Contents I

7、ntellectual Property Rights 5 Foreword 5 1 2 3 3.1 3.2 3.3 4 4.1 4.2 4.2.1 4.3 4.4 4.5 5 5.1 5.2 5.2.1 5.2.2 5.2.3 5.2.4 5.2.5 5.2.6 5.2.7 5.3 5.4 5.5 5.6 5.6.1 5.6.2 5.7 5.7.1 5.7.2 5.8 5.8.1 5.8.2 5.8.3 5.9 6 6.1 6.2 6.2.1 6.2.2 7 8 8.1 8.2 8.3 8.4 Scope 6 References 6 Definitions, symbols and abb

8、reviations . 7 Definitions 7 Symbols 9 Abbreviations . 12 Outline description . 13 Functional description of audio parts . 13 Preparation of speech samples 14 Principles of the GSM adaptive multi-rate speech encoder 14 Principles of the GSM adaptive multi-rate speech decoder 16 Sequence and subjecti

9、ve importance of encoded parameters . 17 Functional description of the encoder 17 Linear prediction analysis and quantization . 17 PCM format conversion 14 Pre-processing (all modes) . 17 Windowing and auto-correlation computation 18 Levinson-Durbin algorithm (all modes) 19 LP to LSP conversion (all

10、 modes) 20 LSP to LP conversion (all modes) 21 Quantization of the LSP coefficients 22 Interpolation of the LSPs 23 Open-loop pitch analysis 25 Monitoring resonance in the LPC spectrum (all modes) . 24 Impulse response computation (all modes) . 28 Target signal computation (all modes) . 28 Adaptive

11、codebook . 28 Adaptive codebook search 28 Adaptive codebook gain control (all modes) 32 Algebraic codebook 33 Algebraic codebook structure . 33 Algebraic codebook search . 35 Quantization of the adaptive and fixed codebook gains . 39 Adaptive codebook gain limitation in quantization 39 Quantization

12、of codebook gains 39 Update past quantized adaptive codebook gain buffer (all modes) . 41 Memory update (all modes) 41 Decoding and speech synthesis 42 Post-processing . 45 Functional description of the decoder 42 Adaptive post-filtering (all modes) . 45 High-pass filtering and up-scaling (all modes

13、) . 46 Detailed bit allocation of the adaptive multi-rate codec 46 Homing sequences 51 Functional description 51 Definitions 51 Encoder homing . 52 Decoder homing . 52 ETSI (GSM 06.90 version 7.2.1 Release 1998) 4 ETSI EN 301 704 V7.2.1 (2000-04) 9 Bibliography . 56 Annex A (informative): Document c

14、hange history 57 History 58 ETSI (GSM 06.90 version 7.2.1 Release 1998) 5 ETSI EN 301 704 V7.2.1 (2000-04) Intellectual Property Rights IPRs essential or potentially essential to the present document may have been declared to ETSI. The information pertaining to these essential IPRs, if any, is publi

15、cly available for ETSI members and non-members, and can be found in SR O00 3 14: “Intellectual Propero Rights (IPRs); Essential, or potentially Essential, IPRs notifed to ETSI in respect of ETSI standards“, which is available from the ETSI Secretariat. Latest updates are available on the ETSI Web se

16、rver (h ttix/www.etsi. ocdini). Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee can be given as to the existence of other IPRs not referenced in SR O00 314 (or the updates on the ETSI Web server) which are, or may be, or may becom

17、e, essential to the present document. Foreword This European Standard (Telecommunications series) has been produced by the Special Mobile Group (SMG). The present document describes the detailed mapping from input blocks of 160 speech samples in 13-bit uniform PCM format to encoded blocks of 95, 103

18、, 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 within the digital cellular telecommunications system. The contents of the present document is subject to continuing work within SMG a

19、nd may change following formal SMG approval. Should SMG modify the contents of the present document it will be re-released with an identifying change of release date and an increase in version number as follows: Version 7.x.y where: 7 indicates Release 1998 of GSM Phase 2+. x the second digit is inc

20、remented for all changes of substance, i.e. technical enhancements, corrections, updates, etc. y the third digit is incremented when editorial only changes have been incorporated in the specification. National transposition dates Date of adoption of this EN: Date of latest announcement of this EN (d

21、oa): Date of latest publication of new National Standard or endorsement of this EN (dop/e): Date of withdrawal of any conflicting National Standard (dow): 31 March 2000 30June2000 3 1 December 2000 3 1 December 2000 ETSI (GSM 06.90 version 7.2.1 Release 1998) 6 ETSI EN 301 704 V7.2.1 (2000-04) 1 Sco

22、pe The present document 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 reconstruct

23、ed speech samples. The sampling rate is 8 O00 samplesk 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 kbitls. The coding scheme for the multi-rate coding modes is the so-called Algebraic Code Excited Linear Prediction Coder, hereafter referred to

24、 as ACELP. The multi-rate ACELP coder is referred to as MR-ACELP. In the case of discrepancy between the requirements described in the present document and the fixed point computational description (ANSI-C code) of these requirements contained in GSM 06.73 6, the description in GSM 06.73 6 will prev

25、ail. The ANSI-C code is not described in the present document, see GSM 06.73 6 for a description of the ANSI-C code. The transcoding procedure specified in the present document is applicable for the adaptive multi-rate full rate and half rate speech traffic channels (TCH) in the GSM system. In GSM 0

26、6.71 5, a reference configuration for the speech transmission chain of the GSM adaptive multi-rate (AMR) system is shown. According to this reference configuration, the speech encoder takes its input as a 13-bit uniform PCM signal either from the audio part of the Mobile Station or on the network si

27、de, from the PSTN via an 8-bit A-law or ,Li -law to 13-bit uniform PCM conversion. The encoded speech at the output of the speech encoder is delivered to a channel encoder unit which is specified in GSM 05.03 3. In the receive direction, the inverse operations take place. 2 References The following

28、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. Fo

29、r a non-specific reference, the latest version applies. A non-specific reference to an ETS shall also be taken to refer to later versions published as an EN with the same number. For this Release 1998 document, references to GSM documents are for Release 1998 versions (version 7.x.y). il GSM O 1.04:

30、 “Digital cellular telecommunications system (Phase 2+); Abbreviations and acronyms“. 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“. GSM 05.03: “Digital cellular telecommunic

31、ations system (Phase 2+); Channel coding“. Pl 31 41 GSM 06.94: “Digital cellular telecommunications system (Phase 2+); Voice Activity Detection (VAD) for Adaptive Multi-Rate speech traffic channels“. GSM 06.7 1: “Digital cellular telecommunications system (Phase 2+); Adaptive Multi-Rate speech proce

32、ssing functions; General description“. GSM 06.73: “Digital cellular telecommunications system (Phase 2+); ANSI-C code for the Adaptive Multi-Rate speech codec“. GSM 06.74: “Digital cellular telecommunications system (Phase 2+); Test sequences for the GSM Adaptive Multi-Rate speech codec“. 51 61 71 E

33、TSI (GSM 06.90 version 7.2.1 Release 1998) 7 ETSI EN 301 704 V7.2.1 (2000-04) SI ITU-T Recommendation G.711 (1988): “Coding of analogue signals by pulse code modulation Pulse code modulation (PCM) of voice frequencies“. Pl ITU-T Recommendation G.726: “40, 32, 24, 16 kbitls adaptive differential puls

34、e code modulation (ADPCM)“. 3 3.1 Definitions, symbols and abbreviations Definitions For the purposes of the present document, the following terms and definitions apply. adaptive codebook The adaptive codebook contains excitation vectors that are adapted for every subframe. The adaptive codebook is

35、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, t

36、he adaptive postfilter is a cascade of two filters: a formant postfilter and a tilt compensation filter. Adaptive Multi-Rate (AMR) codec: Speech and channel codec capable of operating at gross bit-rates of 11.4 kbitls (“half-rate“) and 22.8 kbitls (“full-rate“). In addition, the codec may operate at

37、 various combinations of speech and channel coding (codec mode) bit-rates for each channel mode. 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 interla

38、ced sets of positions. AMR handover: Handover between the FR and HR channel modes to optimise AMR operation. 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. channel mod

39、e: Half-rate or full-rate operation. channel mode adaptation: The control and selection of the (FR or HR) channel mode. channel repacking: Repacking of HR (and FR) radio channels of a given radio cell to achieve higher capacity within the cell. closed-loop pitch analysis: This is the adaptive codebo

40、ok search, Le., 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 f

41、or every subframe. codec mode: For a given channel mode, the bit partitioning between the speech and channel codecs. codec mode adaptation: The control and selection of the codec mode bit-rates. Normally, implies no change to the channel mode. direct form coefficients: One of the formats for storing

42、 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 (Le., fixe

43、d). 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 U6th or U3rd of a sample is used. full-rate (FR): Full-rate channel or c

44、hannel mode. frame: A time interval equal to 20 ms (160 samples at an 8 kHz sampling rate). ETSI (GSM 06.90 version 7.2.1 Release 1998) 8 ETSI EN 301 704 V7.2.1 (2000-04) gross bit-rate: The bit-rate of the channel mode selected (22.8 kbs or 11.4 kbs). half-rate (HR): Half-rate channel or channel mo

45、de. in-band signalling: Signalling for DTX, Link Control, Channel and codec mode modification, etc. carried within the traffic channel. 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, g

46、iven 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 o

47、r 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 s

48、ymmetry. 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 samples within the analysis window. In the adap

49、tive 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 kbitls mode. For the other modes, only a single asymmetric window is used to generate a single set of LP coefficients. In the 12,2 kbitls 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 th