EP0689189B1 - Methode zur Verbesserung der Eigenschaften von Sprachkodierern - Google Patents

Methode zur Verbesserung der Eigenschaften von Sprachkodierern Download PDF

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EP0689189B1
EP0689189B1 EP95108870A EP95108870A EP0689189B1 EP 0689189 B1 EP0689189 B1 EP 0689189B1 EP 95108870 A EP95108870 A EP 95108870A EP 95108870 A EP95108870 A EP 95108870A EP 0689189 B1 EP0689189 B1 EP 0689189B1
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Prior art keywords
excitation
signal
synthetic
free
objective
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French (fr)
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EP0689189A1 (de
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Silvio Cucchi
Marco Fratti
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Alcatel Lucent SAS
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Alcatel CIT SA
Alcatel SA
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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.
  • an excitation parameters computing method is provided as claimed in claim 1.
  • an audio coder is provided as claimed in claim 6.
  • 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 937-940).
  • 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 mean-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 937 to 940 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 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) and 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)

Claims (6)

  1. Verfahren zum Berechnen der Anregungsparameter in Sprachcodierern basierend auf linearen Vorhersage- und Analyse-durch-Synthese-Techniken, die eine zu minimierende Zielfunktion verwenden, dadurch gekennzeichnet, daß die Zielfunktion gemeinsam oder alternativ a) die freie Entwicklung des Zielsignals und des synthetischen Signals und b) eine Gewichtung im Hinblick auf den Fehler zwischen dem Vorhersage-Rückstand und der synthetischen Anregung umfaßt.
  2. Verfahren nach Anspruch 1 in den Alternativen a) oder a) und b), dadurch gekennzeichnet, daß die Zielfunktion: Ex = αE1 + (1 - α)E3    verwendet wird, wobei die Funktion E1 neben dem Fehler zwischen den Ziel signalen und den synthetischen Signalen auch den Fehler zwischen den relativen freien Entwicklungen berücksichtigt, und die Funktion E3 den Fehler zwischen dem Vorhersage-Rückstand und der synthetischen Anregung berücksichtigt, und 0 < α ≤ 0 ist.
  3. Verfahren nach Anspruch 2, dadurch gekennzeichnet, daß die Funktion E1 gegeben ist durch:
    Figure 00140001
       wobei N die Länge des Zeitintervalls für die Minimierung ist, M die freie Entwicklungslänge ist, rs el(n) das durch eine freie Entwicklung erhaltene erweiterte Referenzsignal ist, ui el(n) die erweiterte Null-Zustands-Synthesefilterantwort an dem i-ten Eingang des Codebuches ist, und G die entsprechende Verstärkung ist.
  4. Verfahren nach Anspruch 2, dadurch gekennzeichnet, daß die Funktion E3 gegeben ist durch:
    Figure 00140002
       wobei es(n) der von dem Referenzsignal erhaltene Vorhersage-Rückstand ist und ei(n) das Codebuch-Anregungssignal ist.
  5. Verfahren nach Anspruch 2, dadurch gekennzeichnet, daß der Gewichtsfaktor zeitlich variierbar ist.
  6. Toncodierer, der umfaßt:
    Mittel zum Ausführen einer linearen Vorhersage,
    Mittel zum Ausführen einer Analyse-durch-Synthese, und
    Mittel zum Berechnen der Anregungsparameter unter Verwendung einer zu minimierenden Zielfunktion,
       dadurch gekennzeichnet, daß die Zielfunktion gemeinsam oder alternativ
    a) die freie Entwicklung des Zielsignals und des synthetischen Signals, und
    b) eine Gewichtung im Hinblick auf den Fehler zwischen dem Vorhersage-Rückstand und der synthetischen Anregung umfaßt.
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
ITMI941283 1994-06-20
ITMI941283A IT1271182B (it) 1994-06-20 1994-06-20 Metodo per migliorare le prestazioni dei codificatori vocali

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

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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

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US5235669A (en) * 1990-06-29 1993-08-10 At&T Laboratories Low-delay code-excited linear-predictive coding of wideband speech at 32 kbits/sec
FI98104C (fi) * 1991-05-20 1997-04-10 Nokia Mobile Phones Ltd Menetelmä herätevektorin generoimiseksi ja digitaalinen puhekooderi
JPH06138896A (ja) * 1991-05-31 1994-05-20 Motorola Inc 音声フレームを符号化するための装置および方法

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EP0689189A1 (de) 1995-12-27
AU2175395A (en) 1996-01-04
ITMI941283A0 (it) 1994-06-20
ITMI941283A1 (it) 1995-12-20
AU698340B2 (en) 1998-10-29
IT1271182B (it) 1997-05-27
DE69512323T2 (de) 2000-07-06

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