TIA EIA-136-420-1999 TDMA Celular PCS VSELP (ANSI APPROVAL WITHDRAWN JUNE 12 2015)《TDMA蜂窝 PCS VSELP》.pdf

1、 ANSI/TIA/EIA-136-420-1999 APPROVED: MARCH 3, 1999 REAFFIRMED: JUNE 19, 2003 REAFFIRMED: MAY 28, 2013 WITHDRAWN: JUNE 12, 2015 TIA/EIA-136-420 March 1999TDMA Cellular/ PCS VSELP NOTICE TIA Engineering Standards and Publications are designed to serve the public interest through eliminating misunderst

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19、TION OF DAMAGES IS A FUNDAMENTAL ELEMENT OF THE USE OF THE CONTENTS HEREOF, AND THESE CONTENTS WOULD NOT BE PUBLISHED BY TIA WITHOUT SUCH LIMITATIONS. TINEIA-1 36-420 Contents 1 . Speech Coding (Full-Rate) . Mobile Station 1 1.1 Definitions and Basic Codec Parameters 1 1.2 Audio Interface . 3 1.2.1

20、Transmit Level Adjustment 3 1.2.2 Bandpass Filter . 3 1.2.3 Analog to Digital Converter . 4 1.2.4 Echo Return Loss . 4 1.3 Pre-Processing . 4 1.4 Short-Term Predictor Coefficients 5 1.4.1 Solution for Reflection Coefficients . 5 1.4.2 Bandwidth Expansion . 7 1.4.3 Quantization and Encoding of Coeffi

21、cients 8 1.4.4 Conversion to Direct Form Coefficients . 12 1.4.5 Interpolation of Coefficients . 12 1.5 Frame Energy 13 Computation of Frame Energy . 13 Quantization and Encoding of Frame Energy . 13 1.5.3 Interpolation of Frame Energy 14 1.6 Subframe Processing . 14 Weighting of Input Speech . 14 1

22、.6.2 Subtraction of Zero Input Response . 15 1.6.3 Long Term Predictor Lag . 15 1.6.3.1 Computation of Lag . 17 1.6.3.2 Encoding of Lag . 19 1.6.4 Codebook Excitation 19 1.5.1 1.5.2 1.6.1 25 1.6.4.1 1.6.4.2 Orthogonalization of Filtered Basis Vectors 25 1.6.4.2.1 Orthogonalization for Codebook 1 . 2

23、5 1.6.4.2.2 Orthogonalization for Codebook 2 . 26 1.6.4.3 VSELP Codebook Search . 27 1.6.4.4 Encoding of Excitation Codewords . 28 Quantization of Gains . 28 Filtering of Basis Vectors . 1.6.5 1.6.5.1 1.6.5.2 Transformation of Gains to GS. PO and P1 Vector Quantization and Encoding of GS. PO and P1

24、30 31 1.6.6 Update Filter States 41 2 . Channel Coding . Mobile Station . 42 Definition of Terms. Nomenclature. and Assumptions . 42 2.1 2.2 Speech Data Classes 43 2.3 Cyclic Redundancy Check (CRC) . 45 2.4 Convolutional Encoding 46 i TINEIA-1 36-420 3 . Interleaving . Mobile Station 50 4 . De-Inter

25、leaving . 53 5 . Channel Decoding 54 5.1 Convolutional Decoding 54 5.1.1 Cyclic Redundancy Check (CRC) 54 5.1.2 Bad Frame Masking 55 5.2 Speech Decoding . 56 5.2.1 Definitions and Basic Definitions and Basic Decoder Parameters . 56 5.2.2 Short Term Predictor Coefficients 57 57 5.2.2.1 Decoding of Co

26、efficients 5.2.2.2 Conversion to Direct Form Coefficients . 57 5.2.2.3 Interpolation of Coefficients 57 5.2.3 Frame Energy 57 5.2.4 Subframe Processing . 58 5.2.4.1 Decoding of Lag . 58 5.2.4.2 5.2.4.3 5.2.4.4 Transformation of GS. PO and P1 to Gains 5.2.4.5 Generation of Combined Excitation Decodin

27、g of Excitation Codewords . Decoding of GS. PO and P1 . 5.2.4.6 Update Long Term Filter State . 59 5.2.4.7 Synthesis Filter . 59 5.2.4.8 Adaptive Spectral Postfilter . . 59 5.2.5 Audio Interface . 61 5.2.5.1 5.2.5.2 Reconstruction Filter 5.2.5.3 Receive Level Adjustment Digital to Analog Converter 5

28、.3 Delay Interval Requirements . 62 6 . Speech Coding . Base Station 63 6.1 Audio Interface 63 6.1.1 Transmit Audio Level Adjustment . 63 6.1.2 p-law to 14-Bit Uniform Format Conversion 63 6.1.3 Echo Control 63 7 . Channel Coding . Base Station 64 8 . Interleaving . Base Station 65 9 . Speech Decodi

29、ng . Base Station 66 9.1 Audio Interface 66 Receive Audio Level Adjustment 66 Uniform to S-BitIp-law Code Conversion . 66 10 . Change History for TlA/ElA-136-420 . 67 9.1.1 9.1.2 ii TINEIA-1 36-420 List of Tables Table 1 Table 2 Table 3 Table 4 Table 5 Table 6 Table 7 Codebooks for Quantization of C

30、oefficients 9 Format in Which the Basis Vector Samples Are Given: 20 GS. PO. P1 Codebook 33 Speech Codec Parameter Class Bit Assignments . 44 Input . Output Relationship of Convolutional Coder . 47 Bit Ordering Into Convolutional Coder 48 Bit Ordering Into Class 2 Array . 52 iii TINEIA-1 36-420 List

31、 of Figures Figure 1 Figure 2 Figure 3 Figure 4 Figure 5 Figure 6 Figure 7 Speech Decoder 1 Analysis by Synthesis Procedure for Long Term Predictor Lag and Code Search . 16 Weighted Synthesizer . 41 Error Correction For Speech Codec . 45 Interleaving Array 50 Class 1 and Class 2 Interleaving . 51 Sp

32、eech Decoder 58 iv TINEIA-1 36-420 1 1. Speech Coding (Full-Rate) - Mobile 2 Station 3 A 12 1.1 13 14 1s See TIMEIA- 136-0 10. The speech coding algorithms for alternate codecs are defined in IS-641. The speech coding algorithm described in this standard (see Sections 1.1 to 1.6) is a member of a cl

33、ass of speech codecs known as Code Excited Linear Predictive Coding (CELP), Stochastic Coding or Vector Excited Speech Coding. These techniques use codebooks to vector quantize the excitation (residual) signal. The speech coding algorithm is a variation on CELP called Vector-Sum Excited Linear Predi

34、ctive Coding (VSELP). VSELP uses a codebook which has a predefined structure such that the computations required for the Codebook Search process can be significantly reduced. Definitions and Basic Codec Parameters Figure 1 shows a block diagram of the speech decoder. This figure indicates the variou

35、s parameters which must be determined and encoded by the speech codec. Figure 1 Speech Decoder I r- 1 I I I I I filter speech I synthesis output spectral t t ex(n) Codeboo k 1 a I q I yl Codeboo k 2 H 1 TINEIA-1 36-420 5 6 I 8 The speech decoder utilizes two VSELP excitation codebooks. The two codeb

36、ooks each have their own gain. The two codebook excitations are each multiplied by their corresponding gains and summed to create a combined codebook excitation. The following are the basic parameters for the 7950 bps speech codec and decoder. NF N NP M1 M2 Sampling, Rate frame length subframe lengt

37、h short term predictor order # of taps for long term predictor # of bits in codeword 1 (# of basis vectors) # of bits in codeword 2 (# of basis vectors) 8 kHz 160 samples (20 msec) 40 samples (5 msec) 10 1 7 7 The basic data rate of the speech codec is 7950 bps. There are 159 bits per speech frame (

38、20 msec) for the speech codec. These 159 bits are allocated as follows: short-term filter coefficients, CtiS frame energy, R(0) 5 bitdframe lag, L 7 bits/subframe 28 bitdframe codewords, I, H 7+7 bitdsubframe 5 6 bitdframe gainsf? lr 2 8 bits/subframe 32 bitdframe 38 bitdframe The following is a lis

39、t of all the parameter codes transmitted for each 20 msec. speech frame. The codes are: RO LPC 1 LPC2 LPC3 LPC4 LPC5 LPC6 LPC7 LPC8 LPC9 LPC10 LAG-1 LAG-2 LAG-3 LAG-4 CODE1-I 5 bits 6 bits 5 bits 5 bits 4 bits 4 bits 3 bits 3 bits 3 bits 3 bits 2 bits 7 bits 7 bits 7 bits 7 bits 7 bits frame energy

40、Ist reflection coefficient 2nd reflection coefficient 3rd reflection coefficient 4th reflection coefficient 5th reflection coefficient 6th reflection coefficient 7th reflection coefficient 8th reflection coefficient 9th reflection coefficient 10th reflection coefficient lag for first subframe lag fo

41、r second subframe lag for third subframe lag for fourth subframe Ist codebook code, I, for first subframe 2 TINEIA-1 36-420 1 1.2 2 3 11 13 14 15 16 17 CODE 1-2 CODE 1-3 CODE 14 CODE2-1 CODE2-2 CODE2-3 CODE24 GSPO-1 GSPO-2 GSPO-3 GSPO-4 7 bits 7 bits 7 bits 7 bits 7 bits 7 bits 7 bits 8 bits 8 bits

42、8 bits 8 bits Audio Interface 12 1.2.1 Due to the delays inherent in the air interface specification, which may exceed 100 msec, the implementer is cautioned that echo control measures are necessary. The function of the audio interface at the mobile station transmitter is to convert the analog speec

43、h signal to uniform PCM format with a minimum resolution of 13 bits for further processing by the speech codec. The speech codec is preceded by the following voice processing stages: Level adjustment Bandpass Filter Analog to Digital Converter. The characteristics of these stages are described in th

44、e following sections. Trans m it Leve I Ad i u s t m e n t Pending the generation of a complete speech transmission plan for these systems, the following requirements shall be met to ensure compatibility with the transmission plan for fixed digital speech networks. The transmit audio sensitivity of

45、the mobile station shall meet the requirements stated in IS-1 37 under “Digital Transmitter Audio Sensitivity“. 18 1.2.2 BandDass Filter 19 20 1st codebook code, I, for second subframe 1st codebook code, I, for third subframe 1st codebook code, I, for fourth subframe 2nd codebook code, H, for first

46、subframe 2nd codebook code, H, for second subframe 2nd codebook code, H, for third subframe 2nd codebook code, H, for fourth subframe GS, PO, Pi code for first subframe GS, PO, Pi code for second subframe GS, PO, Pi code for third subframe GS, PO, Pi code for fourth subframe The function of the filt

47、er is to avoid aliasing distortion of the input signal. The attenuation of the filter shall comply with ITU-T (formerly CCITT) Red Book G.714 sending filter. 3 TINEIA-1 36-420 1 2 3 A 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 1.2.3 Analog to Digital Converter The A/D function shall be p

48、erformed according to either of the following: by direct conversion analog to a uniform PCM format with a minimum resolution of 13 bits, or by converting analog to an 8-bitlplaw format followed by a plawhniform code conversion. The AID conversion is based on the standard 8-bitlplaw codec specified i

49、n ITU-T (formerly CCITT) Red Book G.711. The plawhniform code conversion is performed according to definition in ITU-T (formerly CCITT) Red Book G.721 Section 4.2.1 sub-block EXPAND. The parameter LAW shall be set to LAW = O. 1.2.4 Echo Return Loss The echo return loss of the mobile station shall have a minimum value of 45 dB, measured in accordance with the procedure given in IEEE 269-1990, Section 7.10.2, condition 3, using the continuous-spectrum signal method; and calculated using the technique described in TIA SP-l920A, Section 4.4.1.1. This requirement must be met by