KR960027374A - LSP frequency quantization method and quantizer - Google Patents

LSP frequency quantization method and quantizer Download PDF

Info

Publication number
KR960027374A
KR960027374A KR1019940035158A KR19940035158A KR960027374A KR 960027374 A KR960027374 A KR 960027374A KR 1019940035158 A KR1019940035158 A KR 1019940035158A KR 19940035158 A KR19940035158 A KR 19940035158A KR 960027374 A KR960027374 A KR 960027374A
Authority
KR
South Korea
Prior art keywords
lsp
frequency
value
quantizer
quantization
Prior art date
Application number
KR1019940035158A
Other languages
Korean (ko)
Other versions
KR0138868B1 (en
Inventor
윤병식
이인성
김재원
Original Assignee
양승택
재단법인 한국전자통신연구소
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by 양승택, 재단법인 한국전자통신연구소 filed Critical 양승택
Priority to KR1019940035158A priority Critical patent/KR0138868B1/en
Publication of KR960027374A publication Critical patent/KR960027374A/en
Application granted granted Critical
Publication of KR0138868B1 publication Critical patent/KR0138868B1/en

Links

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • G10L19/07—Line spectrum pair [LSP] vocoders
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
    • H03M7/00—Conversion of a code where information is represented by a given sequence or number of digits to a code where the same, similar or subset of information is represented by a different sequence or number of digits
    • H03M7/30—Compression; Expansion; Suppression of unnecessary data, e.g. redundancy reduction
    • H03M7/3002—Conversion to or from differential modulation
    • H03M7/3044—Conversion to or from differential modulation with several bits only, i.e. the difference between successive samples being coded by more than one bit, e.g. differential pulse code modulation [DPCM]
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
    • G10H2250/541—Details of musical waveform synthesis, i.e. audio waveshape processing from individual wavetable samples, independently of their origin or of the sound they represent
    • G10H2250/571—Waveform compression, adapted for music synthesisers, sound banks or wavetables
    • G10H2250/601—Compressed representations of spectral envelopes, e.g. LPC [linear predictive coding], LAR [log area ratios], LSP [line spectral pairs], reflection coefficients

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Theoretical Computer Science (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

본 발명은 엘에스피(LSP) 주파수 양자화 방법 및 양자화기에 관한 것으로, LSP를 이용한 음성 부호화 시스팀에서 DPCM AQBW 양자화 방법을 이용하여 비교적 간단한 구조와 적은 계산량, 적은 메모리 공간을 사용한 효율적인 LSP의 양자화 방법을 제시하고 이를 통하여 더욱 향상된 음성 품질을 제공하는 LSP 주파수 양자화 방법 및 양자화기를 제공하는데 그 목적이 있다.The present invention relates to an LSP frequency quantization method and a quantizer, and proposes an efficient LSP quantization method using a relatively simple structure, a small amount of computation, and a small memory space using a DPCM AQBW quantization method in an LSP speech coding system. The purpose of the present invention is to provide an LSP frequency quantization method and a quantizer that provide improved speech quality.

Description

엘에스피(LSP) 주파수 양자화 방법 및 양자화기LSP frequency quantization method and quantizer

본 내용은 요부공개 건이므로 전문내용을 수록하지 않았음Since this is an open matter, no full text was included.

제1도는 본 발명이 적용되는 QCELP 음성 부호화기의 인코딩 구조를 도시한 일예시도, 제2도는 본 발명이 적용되는 QCELP 음성 부호화기의 디코딩 구조를 도시한 일예시도.1 is a diagram showing an encoding structure of a QCELP speech coder to which the present invention is applied, and FIG. 2 is a diagram showing a decoding structure of a QCELP speech coder to which the present invention is applied.

Claims (3)

DPCM(Differential Pulse Code Modulation)과 AQBW(Adaptive Quantization Backward)의 혼합 양자화를 사용한 LSP(Line Spectrum Pair) 주파수 양자화 방법에 있어서, 현재 프레임에 대한 선형 예측계수(LPC:Linear Prediction Coefficient)를 계산하는 제1단계; 상기 선형 예측 계수를 LSP(Line Spectrum Pair) 주파수로 변환시키는 제2단계; 및 상기 LSP 주파수를 전송을 위해 양자화하는 제3단계를 포함하여 이루어지는 것을 특징으로 하는 LSP 주파수 양자화 방법.In a method of linear spectral pair (LSP) frequency quantization using mixed quantization of differential pulse code modulation (DPCM) and adaptive quantization backward (AQBW), a first method for calculating a linear prediction coefficient (LPC) for a current frame step; A second step of converting the linear prediction coefficients into a line spectrum pair (LSP) frequency; And a third step of quantizing the LSP frequency for transmission. DPCM(Differential Pulse Code Modulation)과 AQBW(Adaptive Quantization Backward)의 혼합 양자화를 사용한 LSP(Line Spectrum Pair) 주파수 양자기에 있어서, LSP 주파수와 각각의 LSP 주파수에 해당하는 주파수값에 바이어스값을 빼는 제1가산수단(31); 상기 제1가산수단(31)의 출력값에 다음 프레임에 대한 LSP 주파수를 예측하기 위한 제2가산수단(32); 상기 제2가산수단(32)의 가산값을 전송 속도에 따라 변화 영역과 스텝 크기를 변화시켜 양자화된 전송 LSP 파라미터를 출력하는 선형 양자화기(33); 상기 선형 양자화기(33)의 LSP 파라미터를 역양자화하는 역양자화기(34); 상기 예측값에 상기 역양자화기(34)의 출력값을 더하는 제3가산수단(35); 및 상기 제3가산수단(35)의 가산 결과를 통해 상기 제2 및 제3가산기(32,35)에 입력되는 예측값을 예측하는 예측수단(36)을 포함하여 이루어지는 것을 특징으로 하는 양자화기.In a Line Spectrum Pair (LSP) frequency quantizer using a mixed quantization of DPCM (Differential Pulse Code Modulation) and AQBW (Adaptive Quantization Backward), a first bias is obtained by subtracting a bias value from an LSP frequency and a frequency value corresponding to each LSP frequency. Adding means 31; Second adding means (32) for predicting the LSP frequency for the next frame to the output value of the first adding means (31); A linear quantizer (33) for outputting a quantized transmission LSP parameter by varying the change area and the step size according to the transmission speed by adding the value of the second adding means (32); An inverse quantizer (34) for inversely quantizing the LSP parameter of the linear quantizer (33); Third adding means (35) for adding the output value of the inverse quantizer (34) to the predicted value; And predicting means (36) for predicting prediction values input to the second and third adders (32, 35) through the addition result of the third adding means (35). DPCM(Differential Pulse Code Modulation)과 AQBW(Adaptive Quantization Backward)의 혼합 양자화를 사용한 LSP(Line Spectrum Pair) 주파수 양자기에 있어서, 양자화된 전송 LSP 파라미터를 역양자화 하는 역양자화수단(41); 상기 역양자화수단(41)의 출력값에 예측값을 더하는 제1가산수단(42); 상기 제1가산수단(42)의 가산된 값을 통해 상기 예측값을 예측하는 예측수단(43); 상기 제1가산수단(42)의 가산된 값에 바이어스 값을 더하여 역양자화된 LSP 주파수를 출력하는 제2가산수단(44)을 포함하여 이루어지는 것을 특징으로 하는 역양자화기.A line spectrum pair (LSP) frequency quantizer using mixed quantization of differential pulse code modulation (DPCM) and adaptive quantization backward (AQBW), comprising: dequantization means (41) for dequantizing a quantized transmission LSP parameter; First adding means (42) for adding a predicted value to an output value of said inverse quantization means (41); Prediction means (43) for predicting the prediction value through the added value of the first addition means (42); And a second adding means (44) for outputting the dequantized LSP frequency by adding a bias value to the added value of the first adding means (42). ※ 참고사항 : 최초출원 내용에 의하여 공개하는 것임.※ Note: The disclosure is based on the initial application.
KR1019940035158A 1994-12-19 1994-12-19 Lsp frequency quantizer KR0138868B1 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
KR1019940035158A KR0138868B1 (en) 1994-12-19 1994-12-19 Lsp frequency quantizer

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
KR1019940035158A KR0138868B1 (en) 1994-12-19 1994-12-19 Lsp frequency quantizer

Publications (2)

Publication Number Publication Date
KR960027374A true KR960027374A (en) 1996-07-22
KR0138868B1 KR0138868B1 (en) 1998-06-15

Family

ID=19402281

Family Applications (1)

Application Number Title Priority Date Filing Date
KR1019940035158A KR0138868B1 (en) 1994-12-19 1994-12-19 Lsp frequency quantizer

Country Status (1)

Country Link
KR (1) KR0138868B1 (en)

Also Published As

Publication number Publication date
KR0138868B1 (en) 1998-06-15

Similar Documents

Publication Publication Date Title
FI114248B (en) Method and apparatus for audio coding and audio decoding
US6502069B1 (en) Method and a device for coding audio signals and a method and a device for decoding a bit stream
US7756350B2 (en) Lossless encoding and decoding of digital data
RU2509379C2 (en) Device and method for quantising and inverse quantising lpc filters in super-frame
KR910008975A (en) Audio signal processing method
KR960019069A (en) Code Excited Linear Predictor and Decoder
US8463615B2 (en) Low-delay audio coder
WO2002093551A3 (en) Method and system for line spectral frequency vector quantization in speech codec
JP2010170142A (en) Method and device for generating bit rate scalable audio data stream
KR960006301A (en) Sound signal encoding / decoding method
EP0375551B1 (en) A speech coding/decoding system
JP2000508091A (en) Methods and apparatus for encoding discrete signals and decoding encoded discrete signals
JP2003015694A (en) Device and method for converting bit rate
US7349842B2 (en) Rate-distortion control scheme in audio encoding
JP2007510170A (en) Fast codebook selection method in audio coding
KR960027374A (en) LSP frequency quantization method and quantizer
US7050967B2 (en) Speech coding system
JP3496618B2 (en) Apparatus and method for speech encoding / decoding including speechless encoding operating at multiple rates
KR101341188B1 (en) Apparatus and method for coding of voice signal using non negative factorization algorithm
KR100703325B1 (en) Apparatus and method for converting rate of speech packet
KR960027371A (en) Vector-Scalar Coupled Quantizer with Line Spectrum Pair Frequency
KR970014385A (en) Coder Using Adaptive Quantization
KR970057832A (en) Coder Using Partial Search Quantization
KR970057922A (en) Image Compression Device Using Two-Step Quantization
JPH086598A (en) Lsp quantization device

Legal Events

Date Code Title Description
A201 Request for examination
N231 Notification of change of applicant
E902 Notification of reason for refusal
E701 Decision to grant or registration of patent right
GRNT Written decision to grant
FPAY Annual fee payment

Payment date: 20080214

Year of fee payment: 11

LAPS Lapse due to unpaid annual fee