EP0689189A1 - Sprachkodierer - Google Patents

Sprachkodierer Download PDF

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Publication number
EP0689189A1
EP0689189A1 EP95108870A EP95108870A EP0689189A1 EP 0689189 A1 EP0689189 A1 EP 0689189A1 EP 95108870 A EP95108870 A EP 95108870A EP 95108870 A EP95108870 A EP 95108870A EP 0689189 A1 EP0689189 A1 EP 0689189A1
Authority
EP
European Patent Office
Prior art keywords
excitation
signal
synthetic
function
objective function
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EP95108870A
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English (en)
French (fr)
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EP0689189B1 (de
Inventor
Silvio Cucchi
Marco Fratti
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Alcatel Lucent SAS
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Alcatel Italia SpA
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    • 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
    • G10L19/083—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being an excitation gain
    • 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
    • G10L19/12—Determination 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

Definitions

  • Speech coding is of application in several communication fields: from transmission via satellite to radiomobile, store-forward systems, automatic responders, etc.
  • the present specification discloses some techniques for improving the features of speech coders based on the above-mentioned techniques.
  • the voice coders based on the Linear Prediction (LP) are parametric coders; typically Analysis-by-Synthesis (A-b-S) techniques are used for the correct determination of the parameters of the system.
  • Such coders synthesize the voice through the use of a suitable input excitation to a synthesis LP filter.
  • the excitation should have the characteristics of the "physical" excitation wave which, coming from the glottis, is then spectrally modified in function of the characteristics of the system that simulates the voice segment (LP filter).
  • A-b-S coders make use of an excitation structure which is composed of an Adaptive Codebook and of a Fixed Codebook (eventually structured). Without prejudicing the generality, it can be assumed that the Fixed Codebook is composed of independent vectors of random numbers, as in the case of CELP coders (M.R. Schroeder, B.S. Atal, "Code Excited Linear Prediction (CELP): high-quality speech at very low bit rates", Proc. ICASSP '85, pages 26-29).
  • CELP Code Excited Linear Prediction
  • Fig. 1 there is represented a block diagram of a typical CELP voice synthesizer; block LPC-IIR denotes the synthesis filter for reconstructing the voice waveform; e a (n) is the adaptive codebook vector (and Ga is the corresponding scaling factor) and e s (n) is the fixed codebook vector (and Gs is the corresponding scaling factor); e(n) is the composite excitation vector.
  • e a (n) is the adaptive codebook vector (and Ga is the corresponding scaling factor)
  • e s (n) is the fixed codebook vector (and Gs is the corresponding scaling factor)
  • e(n) is the composite excitation vector.
  • e a (n) and e s (n) are selected from a suitable set of vectors and are determined simultaneously with respective Ga and Gs. The determination occurs in a time interval of about 5 to 10 ms (analysis frame) and is based on the minimization of the objective function according to the well-known criterion of the perceptively weighted minimum-squared error (see M.R. Schroeder, B.S. Atal, "Code Excited Linear Prediction (CELP): high-quality speech at very low bit-rates", Proc.
  • CELP Code Excited Linear Prediction
  • ICASSP '85 pages 26 to 29), according to the following expression : where N is the length of the time interval for minimization; u i (n) is the zero-state synthesis filter response at the the i-th input of the Codebook (either adaptive or fixed) and G is the corresponding gain; lastly, r s (n) is the reference signal or "objective" signal (i.e. the original voice segment from which the contribution of the reconstruction filter memory deriving from previous synthesis has been subtracted).
  • a first approach consists in using a signal r s el (n) longer than N samples as a reference signal of the objective function (i.e. signal r s (n) of eq. (1) ).
  • the voice can always be considered as obtained from an ideal excitation that constitutes the input of an all-pole synthesis filter (the filter denoted by LPC-IIR in Fig. 1).
  • Such ideal excitation is nothing else than the prediction residue, obtained by filtering the voice through the "inverse filter", i.e. the all-zero filter derived from LPC-IIR.
  • the synthetic excitation has spectral and time location (e.g. the pitch pulse) characteristics similar to those of the ideal excitation. Therefore, it is evident that by including, in the objective function, the contributions of the free evolutions due to both the ideal excitation and to the synthetic excitation, it is possible to carry out a more correct choice of the latter. In fact, depending on the spectral/time characteristics of the signal, the difference between the ideal free evolution and the synthetic one may have a preponderant weight in the modified objective function.
  • the above-mentioned concepts may be expressed according to the revised objective function: in which where u i (n) is the (zero state) synthesis filter response at the i-th input and el i (n) is the corresponding "synthetic" free evolution.
  • the excitation parameters i.e. the i-th index and the corresponding gain G are then choosen in such a way as to minimize the modified objective function (4).
  • the synthetic excitation has spectral and time location (e.g. pitch pulse) characteristics, similar to the ones present in the ideal excitation. From this it derives that it may be important to obtain not only a good similarity between the original voice and the synthetic voice, but also a good similarity between ideal excitation and synthetic excitation.
  • the parameters of the reconstructed excitation allow the achievement of a synthetic voice which "averagely" is similar to the original voice.
  • e s (n) is the prediction residue obtained from the reference signal r s (n)
  • e i (n) is the codebook excitation generating the synthetic signal u i (n).
  • the prediction residue e s (n) must be calculated starting from r s (n) through inverse filtering (with all-zero filter) with null initial state.
  • the reference has been obtained from the voice signal by subtracting its reconstruction filter memory deriving from the previous synthesis. The reference signal is then "free" from every contribution due to the filter memory and can be considered as obtained from a suitable ideal excitation e s (n) coming into the synthesis filter with null initial state.
  • excitation parameters i.e. the i-th index and the corresponding gain G
  • Parameter ⁇ can be either fixed or even made adaptive (i.e. varying with time), for instance in function of certain characteristics of the signal that can be estimated a priori (e.g.: estimate of voiced/unvoiced, estimate of transients, estimate of the pitch period or of the synthesis filter, etc.).

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  • 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)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
EP95108870A 1994-06-20 1995-06-08 Methode zur Verbesserung der Eigenschaften von Sprachkodierern Expired - Lifetime EP0689189B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
ITMI941283A IT1271182B (it) 1994-06-20 1994-06-20 Metodo per migliorare le prestazioni dei codificatori vocali
ITMI941283 1994-06-20

Publications (2)

Publication Number Publication Date
EP0689189A1 true EP0689189A1 (de) 1995-12-27
EP0689189B1 EP0689189B1 (de) 1999-09-22

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EP95108870A Expired - Lifetime EP0689189B1 (de) 1994-06-20 1995-06-08 Methode zur Verbesserung der Eigenschaften von Sprachkodierern

Country Status (4)

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EP (1) EP0689189B1 (de)
AU (1) AU698340B2 (de)
DE (1) DE69512323T2 (de)
IT (1) IT1271182B (de)

Families Citing this family (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP3273455B2 (ja) * 1994-10-07 2002-04-08 日本電信電話株式会社 ベクトル量子化方法及びその復号化器
DE10047172C1 (de) * 2000-09-22 2001-11-29 Siemens Ag Verfahren zur Sprachverarbeitung

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0465057A1 (de) * 1990-06-29 1992-01-08 AT&T Corp. 32 Kb/s codeangeregte prädiktive Codierung mit niedrigen Verzögerung für Breitband-Sprachsignal
EP0515138A2 (de) * 1991-05-20 1992-11-25 Nokia Mobile Phones Ltd. Digitaler Sprachkodierer
EP0516439A2 (de) * 1991-05-31 1992-12-02 Motorola, Inc. Wirksamer CELP-Vocoder und Verfahren

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0465057A1 (de) * 1990-06-29 1992-01-08 AT&T Corp. 32 Kb/s codeangeregte prädiktive Codierung mit niedrigen Verzögerung für Breitband-Sprachsignal
EP0515138A2 (de) * 1991-05-20 1992-11-25 Nokia Mobile Phones Ltd. Digitaler Sprachkodierer
EP0516439A2 (de) * 1991-05-31 1992-12-02 Motorola, Inc. Wirksamer CELP-Vocoder und Verfahren

Non-Patent Citations (3)

* Cited by examiner, † Cited by third party
Title
"Proc. ICASSP '85 (International Conference on Acoustics, Speech and Signal Processing)", article M.R. SCHROEDER, B.S. ATAL: "Code Excited Linear Prediction (CELP): high-quality speech at very low bit rates", pages: 26 - 29
"Proc. ICASSP '88 (International Conferences on Acoustics, Speech and Signal Processing)", article W.B. KLEIJN, D.J. KRASINSKI, R.H. KETCHUM: "Improved Speech Quality and Efficient Vector Quantization in SELP", pages: 155 - 158
P. KROON, E.F. DEPRETTERE: "A class of analysis-by-synthesis predictive coders for high quality speech coding at rates between 4.8 and 16 Kbit/s", IEEE JOURNAL ON SELECTED AREAS IN COMMUNICATIONS, vol. 6, no. 2, February 1988 (1988-02-01), pages 353 - 363

Also Published As

Publication number Publication date
DE69512323T2 (de) 2000-07-06
ITMI941283A1 (it) 1995-12-20
AU2175395A (en) 1996-01-04
EP0689189B1 (de) 1999-09-22
DE69512323D1 (de) 1999-10-28
IT1271182B (it) 1997-05-27
ITMI941283A0 (it) 1994-06-20
AU698340B2 (en) 1998-10-29

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