EP0308817A2 - Procédé pour transformer les paramètres d'un vocodeur à canaux en paramètres d'un vocodeur à prédiction linéaire - Google Patents

Procédé pour transformer les paramètres d'un vocodeur à canaux en paramètres d'un vocodeur à prédiction linéaire Download PDF

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Publication number
EP0308817A2
EP0308817A2 EP88115139A EP88115139A EP0308817A2 EP 0308817 A2 EP0308817 A2 EP 0308817A2 EP 88115139 A EP88115139 A EP 88115139A EP 88115139 A EP88115139 A EP 88115139A EP 0308817 A2 EP0308817 A2 EP 0308817A2
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EP
European Patent Office
Prior art keywords
parameters
vocoder
channel
lpc
vocoder parameters
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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.)
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EP88115139A
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German (de)
English (en)
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EP0308817A3 (fr
Inventor
Hans Dipl.-Ing. Brandl
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Siemens AG
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Siemens AG
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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
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • 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
    • G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders

Definitions

  • the invention relates to a method according to the preamble of patent claim 1.
  • Digital narrowband communication networks with low data transmission rates (1-2 kbit / s) are currently being planned.
  • the coding methods used are based either on the principle of the channel vocoder or the linear prediction (LPC vocoder). Communication between the vocoders is only possible if a suitable data transcoding takes place at their interface.
  • the converter required for this should be designed to be as inexpensive as possible and should not deteriorate the speech quality if possible.
  • One way to build a converter is to transform the speech data back into the speech signal and re-encode it.
  • This method is very complex since two analysis units and two synthesis units are required.
  • the analysis quality also deteriorates the speech quality.
  • the deterioration of the speech quality can be avoided by directly re-encoding the data of the different vocoders. This possibility results from the very similar synthesis principle, that of the channel vocoder and the LPC vocoder is applied.
  • the speech signal is generated by an excitation signal that is filtered by a variable filter.
  • the excitation signal consists of a pulse train for voiced sounds and white noise for unvoiced sounds. With the excitation parameters, the pulse frequency and the excitation mode - voiced or unvoiced - are determined.
  • the variable transmission behavior of the filter corresponds to the variable resonance behavior of the human vocal tract. This changes slowly and is reset by filter parameters every 10 to 20 ms.
  • the task of the speech signal analysis of a vocoder is to obtain the excitation parameters and the filter parameters from a speech signal.
  • the LPC vocoder and the channel vocoder differ essentially in the structure of the filter. LPC assumes an all-pole filter and the channel vocoder assumes a filter bank.
  • the analysis methods for determining the corresponding filter parameters differ and there are other filter parameters that are transmitted in the different networks. In contrast, the excitation parameters are basically the same.
  • a recoding process is therefore sought which converts the filter parameters of a filter bank of a channel vocoder into the filter parameters of an all-pole filter of an LPC vocoder.
  • the channel vocoder parameters usually represent a non-equidistantly scanned spectrum in terms of message theory.
  • the power spectrum is now calculated from the amplitude spectrum and transformed into the autocorrelation function (AKF) using the Fourier transformation.
  • the corresponding LPC vocoder parameter set can now be calculated from the AKF in a known manner using the usual methods (eg Levinson recursion) (see H. Hermansky, B. Hanson, H. Witka; "Perceptually based Predictive Analysis of Speech" on ICASSP 85, p. 13.10 conference proceedings).
  • the direct transformation is associated with high technical expenditure. Powerful real-time processors are required to calculate spectra and correlation functions.
  • the invention is based on the object of specifying a method for transcoding channel vocoder parameters into LPC vocoder parameters which requires relatively few arithmetic operations with high accuracy.
  • the starting point are the channel vocoder parameters, which are available, for example, as a power spectrum (see FIG. 1). This range of services is only available in a channel vocoder in a section-wise constant form b k with jumps at the transition points from b k to b k + 1 .
  • b k energy values e j shown where the value e j is the energy in the channel with the number j corresponds.
  • the channel energy corresponds to the power in a 20 ms interval (this is the interval after which new filter parameters are set in each case). This interval is also the transformation interval.
  • a smoothed spectrum a k (see FIG. 2) is formed by folding with a smoothing function g (i, s).
  • Gaussian bell curves or similar functions are suitable for this smoothing function g, for example.
  • the following function is given as an example for the Gaussian bell curve:
  • smoothing functions g are the low-pass functions known from filter theory and digital signal processing.
  • the spread s defines the corner frequencies of the respective low-pass filter.
  • the scatter s can be a function of the current spectral line.
  • a larger scatter s is selected for the smoothing function g (i, s) at higher frequencies and thus wider channels in b k than at lower frequencies. This makes it possible to adjust the smoothing to the sensation of tonality (Bark scale) of the human ear.
  • the "harmony" in speech synthesis can be empirically selected by the choice of the scatter (s).
  • the LPC coefficients are generally calculated from the short-term autocorrelation function (approx. 20 ms), AKF for short, of the speech signal. These AKF, ie their correlation coefficients r i , can also be determined from the power spectrum of the speech signal by the inverse, discrete Fourier transformation.
  • the N spectral lines b l of the raw spectrum can be derived from the channel energy values e j (see FIG. 1)
  • the number of channels and thus also the number of channel energy values e j is around 16-18.
  • the elements of matrix C are calculated only once for a certain vocoder combination in the method according to the invention. Subsequently, only matrix multiplications between the energy vectors E (which contains the parameters) and the matrix C have to be carried out in order to recode the respective speech parameters.
  • the smoothed channel vocoder parameters a p are present at an input 1 of a first memory 2. For example, it will A set of these parameters, in the case of 18 channels, ie 18 values, is written into the first memory 2.
  • the transformation coefficients c ip of the matrix C are calculated and stored in a coefficient memory 3.
  • the channel vocoder parameters a p in the first memory 2 are addressed in succession by a first counter 4.
  • the coefficients c ip are addressed in the coefficient memory 3 according to their index p.
  • the addressed channel vocoder parameters a p and the addressed coefficients c ip are multiplied in a multiplier 5 and added up in a downstream adder 6.
  • the index i of the coefficients c ip is kept constant until the index i has reached its greatest value, for example 17 in formula 8.
  • the sum formed is written into a second memory 7 as LPC parameter l i .
  • the index i is then increased by one by a second counter 8 and the next LPC parameter l i + 1 is calculated.
  • the second counter 8 addresses the coefficients c ip in the coefficient memory 3 on the one hand according to their index i, and on the other hand the LPC vocoder parameters in the second memory 7.
  • the two counters 4 and 8 are clocked by a clock controller 9.
  • a transformed or recoded set of LPC vocoder parameters can then be removed at an output 10 of the second memory 7.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Carbon Steel Or Casting Steel Manufacturing (AREA)
EP88115139A 1987-09-23 1988-09-15 Procédé pour transformer les paramètres d'un vocodeur à canaux en paramètres d'un vocodeur à prédiction linéaire Withdrawn EP0308817A3 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE19873732047 DE3732047A1 (de) 1987-09-23 1987-09-23 Verfahren zur umcodierung von kanalvocoder-parameter in lpc-vocoder-parameter
DE3732047 1987-09-23

Publications (2)

Publication Number Publication Date
EP0308817A2 true EP0308817A2 (fr) 1989-03-29
EP0308817A3 EP0308817A3 (fr) 1990-04-18

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EP88115139A Withdrawn EP0308817A3 (fr) 1987-09-23 1988-09-15 Procédé pour transformer les paramètres d'un vocodeur à canaux en paramètres d'un vocodeur à prédiction linéaire

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EP (1) EP0308817A3 (fr)
DE (1) DE3732047A1 (fr)

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0626675A1 (fr) * 1993-05-28 1994-11-30 Motorola Inc. Excitation synchrone du temps d'un vocodeur et méthode
WO1995022819A1 (fr) * 1994-02-16 1995-08-24 Qualcomm Incorporated Vocodeur asic
WO1996031873A1 (fr) * 1995-04-03 1996-10-10 Universite De Sherbrooke Quantification des parametres spectraux pour un codage efficace de la parole, utilisant une matrice de prediction scindee
AU725711B2 (en) * 1994-02-16 2000-10-19 Qualcomm Incorporated Block normalisation processor

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0138073A1 (fr) * 1983-09-29 1985-04-24 Siemens Aktiengesellschaft Convertisseur pour le transfert de données entre vocodeurs à canaux et à prédiction linéaire pour la transmission de signaux digitaux de parole au moyen de systèmes de transmission à bande réduite

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0138073A1 (fr) * 1983-09-29 1985-04-24 Siemens Aktiengesellschaft Convertisseur pour le transfert de données entre vocodeurs à canaux et à prédiction linéaire pour la transmission de signaux digitaux de parole au moyen de systèmes de transmission à bande réduite

Cited By (13)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0626675A1 (fr) * 1993-05-28 1994-11-30 Motorola Inc. Excitation synchrone du temps d'un vocodeur et méthode
US5784532A (en) * 1994-02-16 1998-07-21 Qualcomm Incorporated Application specific integrated circuit (ASIC) for performing rapid speech compression in a mobile telephone system
EP0758123A3 (fr) * 1994-02-16 1997-03-12 Qualcomm Incorporated Circuit bouchon pour la normalisation
US5727123A (en) * 1994-02-16 1998-03-10 Qualcomm Incorporated Block normalization processor
WO1995022819A1 (fr) * 1994-02-16 1995-08-24 Qualcomm Incorporated Vocodeur asic
AU697822B2 (en) * 1994-02-16 1998-10-15 Qualcomm Incorporated Vocoder asic
US5926786A (en) * 1994-02-16 1999-07-20 Qualcomm Incorporated Application specific integrated circuit (ASIC) for performing rapid speech compression in a mobile telephone system
AU725711B2 (en) * 1994-02-16 2000-10-19 Qualcomm Incorporated Block normalisation processor
SG87819A1 (en) * 1994-02-16 2002-04-16 John G Mcdonough Vocoder asic
CN100397484C (zh) * 1994-02-16 2008-06-25 高通股份有限公司 数字信号处理器
WO1996031873A1 (fr) * 1995-04-03 1996-10-10 Universite De Sherbrooke Quantification des parametres spectraux pour un codage efficace de la parole, utilisant une matrice de prediction scindee
US5664053A (en) * 1995-04-03 1997-09-02 Universite De Sherbrooke Predictive split-matrix quantization of spectral parameters for efficient coding of speech
CN1112674C (zh) * 1995-04-03 2003-06-25 舍布鲁克大学 用于语音有效编码的谱参数预测分解矩阵量化

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Publication number Publication date
DE3732047C2 (fr) 1992-10-29
DE3732047A1 (de) 1989-04-06
EP0308817A3 (fr) 1990-04-18

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