DE3478065D1 - Method and apparatus for coding digital signals - Google Patents

Method and apparatus for coding digital signals

Info

Publication number
DE3478065D1
DE3478065D1 DE8484112041T DE3478065T DE3478065D1 DE 3478065 D1 DE3478065 D1 DE 3478065D1 DE 8484112041 T DE8484112041 T DE 8484112041T DE 3478065 T DE3478065 T DE 3478065T DE 3478065 D1 DE3478065 D1 DE 3478065D1
Authority
DE
Germany
Prior art keywords
block
signal
excitation signal
lpc coefficients
excitation
Prior art date
Legal status (The legal status 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 status listed.)
Expired
Application number
DE8484112041T
Other languages
German (de)
Inventor
Michael George Beyrouti
Paul Mermelstein
Haim Garten
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Nortel Networks Ltd
Original Assignee
Northern Telecom Ltd
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 Northern Telecom Ltd filed Critical Northern Telecom Ltd
Application granted granted Critical
Publication of DE3478065D1 publication Critical patent/DE3478065D1/en
Expired legal-status Critical Current

Links

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; 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
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/10—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; 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

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Health & Medical Sciences (AREA)
  • Signal Processing (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Computational Linguistics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Analogue/Digital Conversion (AREA)

Abstract

In a method and apparatus for coding of digital signals, especially so-called "waveform" coding of voice frequency signals to reduce bit rates for reduced transmission or storage requirements, in which, for each block, an excitation signal is derived which, when applied to be a synthesis filter having suitable LPC coefficients, will generate an approximation to the input signal, low signal delay is achieved by deriving such excitation signal using the LPC coefficients corresponding to the last sample period of the corresponding block. Thus the excitation signal calculations in the encoder need only be delayed by the duration of the block. The LPC coefficients can be derived on a continuous or sample-by-sample basis using an adaptive lattice. Simplified computation of the excitation signal, which comprises a set of pulses fewer in number than the number of samples in the original block, may be achieved by cross-correlating the impulse response of the modified synthesis filter with the output of the filter and using the result, together with the covariance of the impulse response, to derive the parameters of the excitation pulses. Further computation simplifications may be achieved by using an auto-correlation as the covariance signal.
DE8484112041T 1983-11-30 1984-10-08 Method and apparatus for coding digital signals Expired DE3478065D1 (en)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
CA000442281A CA1236922A (en) 1983-11-30 1983-11-30 Method and apparatus for coding digital signals

Publications (1)

Publication Number Publication Date
DE3478065D1 true DE3478065D1 (en) 1989-06-08

Family

ID=4126639

Family Applications (1)

Application Number Title Priority Date Filing Date
DE8484112041T Expired DE3478065D1 (en) 1983-11-30 1984-10-08 Method and apparatus for coding digital signals

Country Status (5)

Country Link
EP (1) EP0149724B1 (en)
JP (1) JPS60185432A (en)
AT (1) ATE42853T1 (en)
CA (1) CA1236922A (en)
DE (1) DE3478065D1 (en)

Families Citing this family (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CA1328509C (en) * 1988-03-28 1994-04-12 Tetsu Taguchi Linear predictive speech analysis-synthesis apparatus
US5701392A (en) * 1990-02-23 1997-12-23 Universite De Sherbrooke Depth-first algebraic-codebook search for fast coding of speech
US5754976A (en) * 1990-02-23 1998-05-19 Universite De Sherbrooke Algebraic codebook with signal-selected pulse amplitude/position combinations for fast coding of speech
WO1995001673A1 (en) * 1993-06-30 1995-01-12 Royal Melbourne Institute Of Technology Filter windows for fourier transform signal compression
KR100510399B1 (en) * 1998-02-17 2005-08-30 모토로라 인코포레이티드 Method and Apparatus for High Speed Determination of an Optimum Vector in a Fixed Codebook
KR100587099B1 (en) 2003-05-10 2006-06-07 엘지전자 주식회사 Dust removing unit of cyclone cleaner

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4472832A (en) * 1981-12-01 1984-09-18 At&T Bell Laboratories Digital speech coder

Also Published As

Publication number Publication date
EP0149724B1 (en) 1989-05-03
CA1236922A (en) 1988-05-17
ATE42853T1 (en) 1989-05-15
JPS60185432A (en) 1985-09-20
EP0149724A1 (en) 1985-07-31

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Legal Events

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8364 No opposition during term of opposition
8339 Ceased/non-payment of the annual fee