EP1112625A1 - Procede et appareil de codage d'un signal d'informations - Google Patents

Procede et appareil de codage d'un signal d'informations

Info

Publication number
EP1112625A1
EP1112625A1 EP99943854A EP99943854A EP1112625A1 EP 1112625 A1 EP1112625 A1 EP 1112625A1 EP 99943854 A EP99943854 A EP 99943854A EP 99943854 A EP99943854 A EP 99943854A EP 1112625 A1 EP1112625 A1 EP 1112625A1
Authority
EP
European Patent Office
Prior art keywords
positions
pulse
pulses
signal
combinations
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.)
Granted
Application number
EP99943854A
Other languages
German (de)
English (en)
Other versions
EP1112625A4 (fr
EP1112625B1 (fr
Inventor
James P. Ashley
Weimin Peng
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.)
Motorola Solutions Inc
Original Assignee
Motorola Inc
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 Motorola Inc filed Critical Motorola Inc
Publication of EP1112625A1 publication Critical patent/EP1112625A1/fr
Publication of EP1112625A4 publication Critical patent/EP1112625A4/fr
Application granted granted Critical
Publication of EP1112625B1 publication Critical patent/EP1112625B1/fr
Anticipated expiration legal-status Critical
Expired - Lifetime legal-status Critical Current

Links

Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/10Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/10Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation
    • G10L19/107Sparse pulse excitation, e.g. by using algebraic codebook

Definitions

  • the present invention relates, in general, to communication systems and, more particularly, to coding information signals in such communication systems.
  • CDMA communication systems are well known.
  • One exemplary CDMA communication system is the so-called IS-95 which is defined for use in North America by the Telecommunications Industry Association (TLA).
  • TLA Telecommunications Industry Association
  • TIA/EIA/IS-95 Mobile Station-Base-station Compatibility Standard for Dual Mode Wideband Spread Spectrum Cellular System, January 1997, published by the Electronic Industries Association (EIA), 2001 Eye Street, N.W., Washington, D.C. 20006.
  • a variable rate speech codec, and specifically Code Excited Linear Prediction (CELP) codec, for use in communication systems compatible with IS-95 is defined in the document known as IS- 127 and titled Enhanced Variable Rate Codec, Speech Service Option 3 for Wideband Spread Spectrum Digital Systems, September 1996. IS-127 is also published by the Electronic Industries Association (EIA), 2001 Eye Street, N. W., Washington, D.C. 20006.
  • EIA Electronic Industries Association
  • FIG. 1 generally depicts a CELP decoder as is known in the prior art.
  • FIG. 2 generally depicts a Code Excited Linear Prediction (CELP) encoder as is known in the prior art.
  • CELP Code Excited Linear Prediction
  • FIG. 3 generally depicts a joint interleaved pulse permutation matrix in accordance with the invention.
  • FIG. 4 generally depicts a flow chart describing how the codebook is generated in accordance with the invention.
  • FIG. 5 generally depicts a joint interleaved pulse permutation matrix for pulses 3 and 4 in accordance with the present invention.
  • a method for coding an information signal comprises the steps of dividing the information signal into blocks and deriving a target signal based on a block of the information signal.
  • the method further includes the steps of coding the target signal using pulse positioning techniques based on an error criteria, wherein the allowable positions of a given pulse are dependent on the positions of one or more other pulses, to produce coded pulse positions and transmitting the coded pulse positions to a destination.
  • the information signal further comprises a speech signal or an audio signal and a block of the information signals further comprise a frame or a subframe of the information signals.
  • the error criteria further comprises a perceptually weighted squared error criteria and the allowable pulse positions are determined using an arbitrary closed-form expression E( ⁇ ), in which at least one of the conditions within the expression pertain to at least two of the elements within ⁇ .
  • FIG. 1 generally depicts a Code Excited Linear Prediction (CELP) decoder 100 as is known in the art.
  • CELP Code Excited Linear Prediction
  • This signal is scaled using the FCB gain factor / and combined with a signal E(n) output from an adaptive codebook 104 (ACB) and scaled by a factor ⁇ , which is used to model the long term (or periodic) component of a speech signal (with period r).
  • the signal E t (n) which represents the total excitation, is used as the input to the LPC synthesis filter 106, which models the coarse short term spectral shape, commonly referred to as "formants”.
  • the output of the synthesis filter 106 is then perceptually postfiltered by perceptual postfilter 108 in which the coding distortions are effectively "masked” by amplifying the signal spectra at frequencies that contain high speech energy, and attenuating those frequencies that contain less speech energy. Additionally, the total excitation signal E,(n) is used as the adaptive codebook for the next block of synthesized speech.
  • FIG. 2 generally depicts a CELP encoder 200.
  • the goal is to code the perceptually weighted target signal x w (n), which can be represented in general terms by the z-transform:
  • W(z) is the transfer function of the perceptual weighting filter 208, and is of the form:
  • H(z) is the transfer function of the perceptually weighted synthesis filters 206 and 210, and is of the form:
  • H zs ( ) is the "zero state" response of H(z) from filter 206, in which the initial state of H(z) is all zeroes
  • H m (z) is the "zero input response" of H(z) from filter 210, in which the previous state of H(z) is allowed to evolve with no input excitation.
  • the initial state used for generation of H zlR (z) is derived from the total excitation E t (n) from the previous subframe.
  • FCB perceptually weighted target signal x w (n) and the perceptually weighted excitation signal x w (n). This can be expressed in time domain form as:
  • c k (ri) is the codevector corresponding to FCB codebook index k
  • ⁇ k is the optimal FCB gain associated with codevector c k ( ⁇ )
  • h( ⁇ ) is the impulse response of the perceptually weighted synthesis filter H(z)
  • M is the codebook size
  • L is the subframe length
  • x w (n) ⁇ k c k ( ⁇ ) * h(ri) .
  • speech is coded every 20 milliseconds (ms) and each frame includes three subframes of length L.
  • Eq.4 can also be expressed in vector-matrix form as:
  • H is the L x L zero-state convolution matrix
  • the FCB utilizes a multipulse configuration in which the excitation vector c k contains very few non-zero, unit magnitude values. This configuration is known in the art as Algebraic CELP, or ACELP.
  • Table 1 generally depicts pulse positions defined for IS-127 Rate 1/2.
  • the excitation codevector c k can contain " holes" in which certain positions are not represented by the vector space. That is, an optimal match to the target vector may require a pulse at position 12, but the definitions of the pulse positions in Table 1 does not allow a pulse to be located at that position.
  • the constraints on positions may cause the pulse to be placed either at locations close to the optimal position, or worse, the energy of the target signal may be completely missed at that position. This can cause distortion, and possibly audible artifacts in the synthesized speech signal.
  • the bit allocation of 16 bits would be divided between the four tracks equally so that each track would receive four bits.
  • the four bits per track would further be composed of three bits for position (comprising 8 different positions) and one sign bit to indicate the polarity of the pulse.
  • the pulse positions can then be extracted at the decoder by:
  • the respective positions of pulse 0 are shown along the horizontal axis, and the positions of pulse 1 are shown along the vertical axis.
  • the "forbidden" pulse combinations are designated by the shaded regions while the allowable combinations are unshaded.
  • FIG. 4 generally depicts a flow chart describing how the codebook is generated in accordance with the invention.
  • the flowchart shows a basic nested loop structure in which all permutations of 0 ⁇ / ⁇ M and 0 ⁇ j ⁇ N are generated.
  • N and M are the total number of allowable positions for each pulse.
  • the decision in the innermost loop simply checks for forbidden combinations [i,j] according to function F(i,j) at step 402, which in the example of FIG. 3 is described as:
  • This function returns a value of 1 for cases when the absolute value of the difference of / andy is an element of the given set; otherwise, a zero is returned. This is shown in step 403.
  • the elements of the given set correspond to the distances between the diagonal shaded elements of FIG. 3, and the expression is therefore sufficient in describing all necessary shaded regions.
  • the respective positions are calculated using the following expression:
  • is the decimated track position
  • N lr ⁇ ck is the number of tracks
  • n is the track number.
  • FIG. 5 generally depicts a joint interleaved pulse permutation matrix for pulses p 2 and p 3 in accordance with the present invention. As shown in FIG.
  • n is the number of pulses.

Landscapes

  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Paper (AREA)
  • Control Of El Displays (AREA)
  • Control Of Motors That Do Not Use Commutators (AREA)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
EP99943854A 1998-09-11 1999-08-24 Procede de codage d'un signal d'informations Expired - Lifetime EP1112625B1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
US15143098A 1998-09-11 1998-09-11
US151430 1998-09-11
PCT/US1999/019217 WO2000016501A1 (fr) 1998-09-11 1999-08-24 Procede et appareil de codage d'un signal d'informations

Publications (3)

Publication Number Publication Date
EP1112625A1 true EP1112625A1 (fr) 2001-07-04
EP1112625A4 EP1112625A4 (fr) 2004-06-16
EP1112625B1 EP1112625B1 (fr) 2006-05-31

Family

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Family Applications (1)

Application Number Title Priority Date Filing Date
EP99943854A Expired - Lifetime EP1112625B1 (fr) 1998-09-11 1999-08-24 Procede de codage d'un signal d'informations

Country Status (6)

Country Link
EP (1) EP1112625B1 (fr)
JP (1) JP4460165B2 (fr)
KR (1) KR100409167B1 (fr)
AT (1) ATE328407T1 (fr)
DE (1) DE69931641T2 (fr)
WO (1) WO2000016501A1 (fr)

Families Citing this family (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6539349B1 (en) 2000-02-15 2003-03-25 Lucent Technologies Inc. Constraining pulse positions in CELP vocoding
US7889103B2 (en) * 2008-03-13 2011-02-15 Motorola Mobility, Inc. Method and apparatus for low complexity combinatorial coding of signals

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0397628A1 (fr) * 1989-05-11 1990-11-14 Telefonaktiebolaget L M Ericsson Procédé pour le positionnement des impulsions d'excitation dans un codeur à prédiction linéaire pour signal vocal

Family Cites Families (10)

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Publication number Priority date Publication date Assignee Title
FR2579356B1 (fr) * 1985-03-22 1987-05-07 Cit Alcatel Procede de codage a faible debit de la parole a signal multi-impulsionnel d'excitation
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
JP3057907B2 (ja) * 1992-06-16 2000-07-04 松下電器産業株式会社 音声符号化装置
KR950011967B1 (ko) * 1992-07-31 1995-10-12 임홍식 반도체 녹음기용 메모리 정리 장치
JP3196595B2 (ja) * 1995-09-27 2001-08-06 日本電気株式会社 音声符号化装置
JP4063911B2 (ja) * 1996-02-21 2008-03-19 松下電器産業株式会社 音声符号化装置
US5970444A (en) * 1997-03-13 1999-10-19 Nippon Telegraph And Telephone Corporation Speech coding method
US5963897A (en) * 1998-02-27 1999-10-05 Lernout & Hauspie Speech Products N.V. Apparatus and method for hybrid excited linear prediction speech encoding
JP3180762B2 (ja) * 1998-05-11 2001-06-25 日本電気株式会社 音声符号化装置及び音声復号化装置
JP3824810B2 (ja) * 1998-09-01 2006-09-20 富士通株式会社 音声符号化方法、音声符号化装置、及び音声復号装置

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0397628A1 (fr) * 1989-05-11 1990-11-14 Telefonaktiebolaget L M Ericsson Procédé pour le positionnement des impulsions d'excitation dans un codeur à prédiction linéaire pour signal vocal

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
CHENG DEYUAN ED - YUAN B ET AL: "An 8 kb/s low complexity ACELP speech codec" SIGNAL PROCESSING, 1996., 3RD INTERNATIONAL CONFERENCE ON BEIJING, CHINA 14-18 OCT. 1996, NEW YORK, NY, USA,IEEE, US, 14 October 1996 (1996-10-14), pages 671-674, XP010209596 ISBN: 0-7803-2912-0 *
See also references of WO0016501A1 *

Also Published As

Publication number Publication date
JP4460165B2 (ja) 2010-05-12
KR20010073146A (ko) 2001-07-31
DE69931641T2 (de) 2006-10-05
WO2000016501A1 (fr) 2000-03-23
JP2002525667A (ja) 2002-08-13
ATE328407T1 (de) 2006-06-15
DE69931641D1 (de) 2006-07-06
EP1112625A4 (fr) 2004-06-16
EP1112625B1 (fr) 2006-05-31
KR100409167B1 (ko) 2003-12-12

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