EP0570362B1 - Decodeur de parole numerisee utilisant un postfiltre a distorsion spectrale reduite - Google Patents

Decodeur de parole numerisee utilisant un postfiltre a distorsion spectrale reduite Download PDF

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Publication number
EP0570362B1
EP0570362B1 EP90913916A EP90913916A EP0570362B1 EP 0570362 B1 EP0570362 B1 EP 0570362B1 EP 90913916 A EP90913916 A EP 90913916A EP 90913916 A EP90913916 A EP 90913916A EP 0570362 B1 EP0570362 B1 EP 0570362B1
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Prior art keywords
component
postfilter
speech
signal
filter
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EP90913916A
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German (de)
English (en)
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EP0570362A4 (fr
EP0570362A1 (fr
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Ira Alan Gerson
Mark Antoni Jasiuk
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Motorola Solutions Inc
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Motorola Inc
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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
    • G10L19/26—Pre-filtering or post-filtering

Definitions

  • This invention relates generally to speech coders, and more particularly to digital speech coders that use postfilters to enhance the speech quality.
  • Speech coders and decoders are known in the art. Some speech coders convert analog voice samples into digitized representations, and subsequently represent the spectral speech information through use of linear predictive coding (see, for example, document EP-A-0 294 020). Other speech coders improve upon ordinary linear predictive coding (LPC) techniques by providing an excitation signal that is related to the original voice signal.
  • LPC linear predictive coding
  • U.S. Patent No. 4,817,157 describes a digital speech coder and decoder having an improved vector excitation source wherein a codebook of codebook excitation vectors is accessed to select a codebook excitation signal that best fits the available information, and is used to provide a synthesized speech signal from an LPC filter that closely represents the original.
  • various post-LPC filters are often used to further condition the signal.
  • One such filter is an adaptive spectral postfilter (which is typically intended to enhance the perceptual quality of the synthetic speech), and another is a post emphasis filter (which contributes brightness to the synthetic speech result).
  • the numerator term attempts to cancel the general spectral shape introduced by the denominator. In prior art applications, ⁇ is often set to about 0.8, and ⁇ to about 0.5.
  • the numerator polynomial is only partially successful in tracking the spectral shape of the denominator (in effect, the spectral characteristic of the filter tilts with time), and that discrepancy typically manifests itself as a time varying modulation of the postfiltered speech brightness.
  • a method for producing a synthesized speech signal comprising the steps of: providing an excitation signal to a linear predictive coding (LPC) filter; providing from the LPC filter a synthesized speech signal; providing a speech synthesis postfilter that requires a first component and a second component; providing the first component including a first set of coefficients; the method characterised by the steps of: transforming at least some of the first set of coefficients into an autocorrelation domain set of parameters; spectrally smoothing the autocorrelation domain set of parameters to provide a modified first set of coefficients; using the modified first set of coefficients to provide the second component for use by the speech synthesis postfilter; filtering the synthesized speech signal in the speech synthesis postfilter using the first component and the second component to provide a filtered synthesized speech signal, wherein the second component adaptively tracks and cancels out a general spectral shape of the first component; and rendering the filtered synthesized speech signal audible.
  • LPC linear predictive coding
  • Z transform (filter) coefficients that represent the first component are converted to the autocorrelation domain.
  • a spectral smoothing technique that makes use of a bandwidth expansion function is then applied to the autocorrelation sequence, and the second component polynomial coefficients are calculated from the modified autocorrelation sequence via the Levinson recursion.
  • the first component is then used as the denominator, and the second component as the numerator, in the above noted filter characteristic.
  • the numerator polynomial is replaced by a spectrally smoothed version of the A(z/ ⁇ ) polynomial.
  • Formant bandwidth expansion does not change the smoothed spectral envelope.
  • the spectrally smoothed bandwidth expanded version of the A(z/ ⁇ ) polynomial effectively minimizes time varying spectral tilt and allows the numerator to adaptively track the general spectral shape of the denominator and cancel it out.
  • an additional post emphasis filter can be used to afford more control over postfiltered speech brightness.
  • This filter is a first order filter of the form where typically 0.2 ⁇ u ⁇ 0.5.
  • a radio (100) embodying the invention includes an antenna (102) for receiving a speech coded radio frequency (RF) signal (101).
  • An RF unit (103) processes the received signal to recover the speech coded information.
  • This information is provided to a parameter decoder (105) that develops control parameters for various subsequent processes.
  • An excitation source (104) as described above utilizes the parameters provided to it to create an excitation signal.
  • This resultant excitation signal from the excitation source (104) is provided to an LPC filter (106) that yields a synthesized speech signal in accordance with the coded information.
  • the synthesized speech signal is then pitch postfiltered (107) and spectrally postfiltered (108) to enhance the quality of the reconstructed speech.
  • a post emphasis filter (109) can also be included to further enhance the resultant speech signal. (Additional details regarding the spectral postfilter (108) and the post emphasis filter (109) will be provided below.)
  • the speech signal is then processed in an audio processing unit (111) and rendered audible by an audio transducer (112).
  • the excitation source (104), LPC filter (106), pitch postfilter (107), adaptive spectral postfilter (108), and post emphasis filter (109) can all be provided through appropriate programming of a DSP (113).
  • the adaptive spectral postfilter (108) is characterized by a first component (a denominator that is related to the filter characteristics of the LPC filter (106)) and a second component (a numerator that adaptively tracks the general spectral shape of the denominator to thereby cancel it out).
  • a first component a denominator that is related to the filter characteristics of the LPC filter (106)
  • a second component a numerator that adaptively tracks the general spectral shape of the denominator to thereby cancel it out.
  • the general form of such a filter can be found described in an article entitled "Real-Time Vector APC Speech Coding at 4800 bps With Adaptive Postfiltering," by Chen and Gersho, which appeared in the April, 1987 edition of the Proceedings of The International Conference on Acoustics, Speech, and Signal Processing, at pages 2185-2188.
  • the numerator is developed by applying spectral smoothing techniques to the denominator polynomial.
  • spectral smoothing techniques are described in an article entitled "Spectral Smoothing Technique in PARCOR Speech Analysis - Synthesis," by Tohkura, Itakura, and Hashimoto, which appeared in the December, 1978 edition of the I.E.E.E. Transactions on Acoustics, Speech, and Signal Processing.
  • Z transform coefficients that represent the denominator are converted to the autocorrelation domain.
  • Examples of such conversions can be found in Markel, J.D. Gray, A.H., Jr.; Linear Prediction of Speech (Springer-Verlag, Berlin, Heidelberg, New York, 1976.)
  • the spectral smoothing technique bandwidth expansion function is then applied to the autocorrelation sequence, with the numerator polynomial coefficients being calculated from the modified autocorrelation sequence via the Levinson recursion.
  • the autocorrelation coefficients are multiplied by the following factors to provide the resultant numerator coefficients: Autocorrelation Lag Spectral Smoothing Factor 0 1.0000000 1 0.9230769 2 0.7252747 3 0.4835164 4 0.2719780 5 0.1279896 6 4.9773753E-02 7 1.5718028E-02 8 3.9295070E-03 9 7.4847753E-04 10 1.0206513E-04
  • the denominator and numerator are then used to characterize the adaptive spectral postfilter (108).
  • the numerator polynomial is provided by a spectrally smoothed version of the denominator polynomial.
  • the spectrally smoothed bandwidth expanded version of the denominator polynomial effectively minimizes time varying spectral tilt and allows the numerator to adaptively track the general spectral shape of the denominator and cancel it out.
  • a bandwidth expansion factor (which specifies the degree of smoothing that is performed on the denominator) of about 1,200 Hz was used.
  • the adaptive spectral postfilter is characterized by a first component, or denominator, and a second component, or numerator.
  • the first component which can be expressed as: is provided in block 202.
  • the z-transform coefficients are converted to the autocorrelation domain.
  • a spectral smoothing bandwidth expansion function is applied to the autocorrelation sequence, and, in the subsequent block (205), the numerator (second component) polynomial coefficients are calculated from the autocorrelation sequence modified in the previous step (204), through the use of the Levinson recursion.
  • the numerator, or second component can be expressed as: 1-B(z).
  • the first and second components are used to characterize the adaptive spectral postfilter, which can be represented as:
  • the post emphasis filter (109) may be provided to afford more control over postfiltered speech brightness.
  • This filter is a first order filter of the form

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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)
  • Filters That Use Time-Delay Elements (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Claims (6)

  1. Procédé de production d'un signal vocal synthétisé, comprenant les étapes suivantes :
    A) fourniture d'un signal d'excitation à un filtre de codage prédictif linéaire ;
    B) fourniture d'un signal vocal synthétisé par un filtre de codage prédictif linéaire ;
    C) fourniture d'un postfiltre de synthèse vocale qui requiert une première composante et une seconde composante ;
    fourniture de la première composante comportant un premier jeu de coefficients;
    E) transformation d'au moins certains coefficients du premier jeu de coefficients en un jeu de paramètres du domaine d'autocorrélation ;
    F) lissage spectral du jeu de paramètres du domaine d'autocorrélation pour fournir un premier jeu de coefficients modifié ;
    G) utilisation du premier jeu de coefficients modifié pour fournir la seconde composante qui doit être utilisée par le postfiltre de synthèse vocale ;
    H) filtrage du signal vocal synthétisé dans le postfiltre de synthèse vocale en utilisant la première composante et la seconde composante pour fournir un signal vocal synthétisé filtré ; et
    I) transformation du signal vocal synthétisé filtré en un signal audible.
  2. Procédé de production d'un signal vocal synthétisé selon la revendication 1, comprenant en outre les étapes suivantes :
    A) réception d'un signal à haute fréquence (HF) qui comporte des informations vocales codées ;
    B) récupération d'un signal d'excitation dans les informations vocales codées ; et
    C) fourniture du signal d'excitation à un filtre de codage prédictif linéaire (CPL) ;
       et dans lequel une première composante doit être utilisée par le postfiltre de synthèse vocale.
  3. Procédé selon la revendication 1 ou 2, dans lequel le filtre CPL est au moins partiellement défini par l'expression suivante : 11- A(z)
  4. Procédé selon la revendication 1 ou 2, dans lequel la première composante du postfiltre de synthèse vocale a la forme suivante :
    Figure 00150001
    telle que représentée dans la notation de la transformée en Z.
  5. Procédé selon la revendication 1 ou 2, comportant en outre l'étape suivante :
    A) filtrage du signal vocal synthétisé dans un filtre de désaccentuation sensiblement défini, dans la transformée en Z, comme étant : _(z) = 1-uz-1    où 0,2 _ u _ 0,5.
  6. Procédé selon l'une quelconque des revendications précédentes, dans lequel l'étape de fonctionnement comporte l'étape de multiplication.
EP90913916A 1989-10-17 1990-09-17 Decodeur de parole numerisee utilisant un postfiltre a distorsion spectrale reduite Expired - Lifetime EP0570362B1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
US42292689A 1989-10-17 1989-10-17
US422926 1989-10-17
PCT/US1990/005190 WO1991006093A1 (fr) 1989-10-17 1990-09-17 Decodeur de parole numerisee utilisant un postfiltre a distorsion spectrale reduite

Publications (3)

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EP0570362A4 EP0570362A4 (fr) 1993-07-01
EP0570362A1 EP0570362A1 (fr) 1993-11-24
EP0570362B1 true EP0570362B1 (fr) 1999-03-17

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EP (1) EP0570362B1 (fr)
JP (1) JP3158434B2 (fr)
CN (1) CN1078371C (fr)
AT (1) ATE177867T1 (fr)
AU (1) AU635342B2 (fr)
DE (1) DE69033011T2 (fr)
ES (1) ES2131498T3 (fr)
WO (1) WO1991006093A1 (fr)

Families Citing this family (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2729246A1 (fr) * 1995-01-06 1996-07-12 Matra Communication Procede de codage de parole a analyse par synthese
FR2729244B1 (fr) * 1995-01-06 1997-03-28 Matra Communication Procede de codage de parole a analyse par synthese
FR2729247A1 (fr) * 1995-01-06 1996-07-12 Matra Communication Procede de codage de parole a analyse par synthese
JP2993396B2 (ja) * 1995-05-12 1999-12-20 三菱電機株式会社 音声加工フィルタ及び音声合成装置
DE19643900C1 (de) * 1996-10-30 1998-02-12 Ericsson Telefon Ab L M Nachfiltern von Hörsignalen, speziell von Sprachsignalen
US6137844A (en) * 1998-02-02 2000-10-24 Oki Telecom, Inc. Digital filter for noise and error removal in transmitted analog signals

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* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4301329A (en) * 1978-01-09 1981-11-17 Nippon Electric Co., Ltd. Speech analysis and synthesis apparatus
US4617676A (en) * 1984-09-04 1986-10-14 At&T Bell Laboratories Predictive communication system filtering arrangement
JP2535833B2 (ja) * 1986-07-03 1996-09-18 日本電気株式会社 集積回路
US4852169A (en) * 1986-12-16 1989-07-25 GTE Laboratories, Incorporation Method for enhancing the quality of coded speech
US4969192A (en) * 1987-04-06 1990-11-06 Voicecraft, Inc. Vector adaptive predictive coder for speech and audio
US4817157A (en) * 1988-01-07 1989-03-28 Motorola, Inc. Digital speech coder having improved vector excitation source

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Publication number Publication date
ES2131498T3 (es) 1999-08-01
JPH05500573A (ja) 1993-02-04
DE69033011T2 (de) 2001-10-04
EP0570362A4 (fr) 1993-07-01
AU6411490A (en) 1991-05-16
DE69033011D1 (de) 1999-04-22
ATE177867T1 (de) 1999-04-15
CN1078371C (zh) 2002-01-23
AU635342B2 (en) 1993-03-18
JP3158434B2 (ja) 2001-04-23
EP0570362A1 (fr) 1993-11-24
CN1051101A (zh) 1991-05-01
WO1991006093A1 (fr) 1991-05-02

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