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EP0075311B1 - Dispositif pour la transmission de la parole utilisant le principe des vocodeurs à canaux - Google Patents

Dispositif pour la transmission de la parole utilisant le principe des vocodeurs à canaux Download PDF

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Publication number
EP0075311B1
EP0075311B1 EP82108691A EP82108691A EP0075311B1 EP 0075311 B1 EP0075311 B1 EP 0075311B1 EP 82108691 A EP82108691 A EP 82108691A EP 82108691 A EP82108691 A EP 82108691A EP 0075311 B1 EP0075311 B1 EP 0075311B1
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EP
European Patent Office
Prior art keywords
filter
fir
speech
filters
sum
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Expired
Application number
EP82108691A
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German (de)
English (en)
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EP0075311A2 (fr
EP0075311A3 (en
Inventor
Rüdiger Dr.-Ing. Reiss
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Siemens AG
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Siemens AG
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Publication of EP0075311A2 publication Critical patent/EP0075311A2/fr
Publication of EP0075311A3 publication Critical patent/EP0075311A3/de
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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
    • G10L19/02—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 an arrangement for the transmission of speech according to the channel vocoder principle, in which, in a vocoder analyzer on the transmission side, envelope values of the speech signal to be transmitted for a number of M spectral channels predetermined by a filter bank, as well as additional speech-specific parameters, such as basic speech frequency and width, differ by frequency position and width Vocal characteristics derived and combined in a suitable manner to form a digital sum signal in the rhythm of successive frames each comprising an analysis interval are transmitted to the reception side, in which furthermore the digital sum signal on the reception side is again divided into the individual M spectral channels and, if appropriate, the channels assigned to the additional language-specific parameters and these channel signals are then fed to the receiver-side vocoder synthesizer, which in turn corresponds to the transmitter-side vocoder analyzer has a computing filter bank and a pulse generator and outputs the generated synthetic speech signal on the output side.
  • Channel vocoder transmission methods are known, for example, from the literature IEEE Transactions on Audio and Electroacoustics, Vol. AU-15, No. 4, Dec. 1967, pages 148 to 161.
  • the same filter bank is used for such channel vocoders in the analysis section and in the synthesis section. In the case of voice transmission in half duplex, this results in an inexpensive outlay because the filter bank can be switched over and thus only needs to be present once per device.
  • So-called “switched capacitor filters” can also be used as digital filters, which are available in an integrated embodiment and manage with a small installation volume.
  • such filters have the disadvantage that their transfer function is repeated periodically in the frequency domain.
  • the input signal for such a filter must not have any spectral components above half the sampling frequency.
  • the upper band limit of the input signal must be further away from half the sampling frequency if interference is to be avoided.
  • the analysis part there are no problems with this, since this distance can be reliably maintained with simple means.
  • this requirement cannot be guaranteed without considerable additional expenditure due to the pulse-shaped excitation function.
  • considerable analog screening means have to be used here in order to sufficiently suppress disruptive effects in the form of a chirping background noise.
  • the invention has for its object to provide a solution for a digital filter bank for the synthesis part of a channel vocoder implemented in digital technology, which can ensure optimal speech quality of the generated synthetic speech with relatively little technical effort.
  • the filter bank designed in digital technology consists of non-recursive time-variant filters with a finite impulse response (FIR filter) and that the filter bank the excitation variable of the pulse generator is supplied on the input side with a constant pulse amplitude and that the time variance of the gain factor of the filter bank is brought about in multipliers by multiplying the filter bank coefficients by the envelope values transmitted in rhythm with successive frames.
  • FIR filter finite impulse response
  • the invention is based on the essential finding that the impulse response of a FIR filter amounts to the same whether the pulse train supplied to its input is weighted with the transmitted envelope values or the weighting is carried out by multiplying the filter coefficients by the transmitted envelope values .
  • the FIR filters to which the excitation variable is fed on the input side, each have a chain connection of N-1 identical delay stages with a maximum of N taps.
  • Each FIR filter has a number of switches corresponding to the taps, the control inputs of which are connected to the taps.
  • Each switch connects the output of a multiplier to the filter output.
  • the inputs of the multipliers together form the Input for the assigned transmitted envelope value and at each of the other inputs of the multipliers there is a filter coefficient, which in their entirety determine the filter response.
  • the filter bank is an FIR sum filter, which has a chain connection of N-1 equal delay stages with a maximum of N taps and has a number of switches corresponding to the taps, the control inputs of which are connected to the taps.
  • Each switch connects the output of a multiplier summing circuit to the filter output, which carries out the product formation of the individual filter coefficients with the associated transmitted envelope values and forms the product sum.
  • the excitation quantity supplied to the FIR filter or the FIR sum filter can be a pulse sequence that is generated periodically by a pulse generator and can be controlled in its repetition rate by the transmitted basic speech frequency value.
  • the filter coefficients are dimensioned such that an optimal speech quality of the synthesized speech is guaranteed.
  • a first such preferred dimensioning can consist in that the filter coefficients of the FIR filter or the filter areas of the FIR sum filter with center frequencies ⁇ 2 kHz for impulse responses and those of the FIR filters or the filter areas of the sum filter with center frequencies> 2 kHz for noise responses are measured.
  • a further preferred dimensioning can consist in dimensioning all filter coefficients of the FIR filter or the FIR sum filter for noise responses.
  • the envelope values Ai combined within an analysis interval on the transmission side to form a time-division multiplex frame, as well as the fundamental frequency information No and the voiced-unvoiced criterion Sc are first of all returned to the individual channels, namely the channels, on the receiving side in the demultiplexer DEMUX, to which the sum signal ss is fed on the input side M spectral channels for the M envelope values A1, A2 ... AM as well as the channel for the voiced-unvoiced criterion Sc and the channel for the fundamental frequency information No are divided.
  • the actual synthesis part ST has the filter bank with the bandpasses B1, B2 ... BM, to which the envelope values are fed via a modulator M1, M2 ... MM.
  • the excitation variable x (n) in the form of the periodic pulse sequence generated by the pulse generator PB or in the form of the pseudo-random pulse sequence generated by a random pulse generator PNG is fed to the second input of the modulators.
  • the pulses of the pulse generator PG or of the random pulse generator PNG which are output at the modulator outputs and weighted by the envelope values A1, A2 ... AM are passed over the bandpass filters B1, B2 ... BM.
  • the filter responses Y1 (n), Y2 (n) ... YM (n) are then combined via the summer SU to the synthesized speech signal y (n) at the output of the synthesis part ST.
  • the output signal y (n) is obtained taking into account the filter coefficients h1 (k), h2 (k) ... hM (k ) the relationship for these bandpass filters
  • Equation (1) the same result is obtained for the output signal y (n) if, instead of a weighting, that of the pulse generator DG or of the random pulse generator PNG supplied excitation variable x (n) with the envelope values Ai, the filter coefficients hi (k) are multiplied with the envelope values Ai.
  • the equation (1) for the output signal of the synthesis part y (n) can therefore be rewritten as follows.
  • the synthesis part ST which realizes equations (3) to (5) and is constructed with FIR filters is shown in FIG. 2.
  • the bandpasses B1 ', B2', ... BM 'representing time-variant FIR filters are now used.
  • the excitation variable x (n) is fed directly to the inputs of these bandpass filters in the form of a pulse train with a constant pulse amplitude.
  • the time variance of the filters is controlled via the envelope values A1, A2 ... AM by weighting the filter coefficients hi (k) by forming the products gi (k). This weighting must be done once for each incoming frame of the sum signal ss for each filter.
  • FIG. 3 shows a bandpass filter Bi 'in the form of such a time-variant FIR filter. It has N-1 identical delay stages Z- 1 and N taps connected in chain. The N taps are fed to control inputs of N switches S, which each connect the output of a multiplier MU to the sum line 1 supplying the filter response yi (n). Each multiplier MU assigned to a switch S is supplied with the associated envelope value Ai at its one input.
  • the filter coefficients hi (0), hi (1), ... hi (N-1) are present at the N second inputs of the N multipliers MU.
  • the excitation variable x (n) supplied to the input of the chain of delay elements has only the function of controlling the switches S "on or” within the filter, depending on whether no pulse or a pulse occurs. This has become possible because the excitation variable itself is no longer weighted by the envelope values.
  • the transition from time-invariant to time-variant FIR filters thus saves M multipliers. As has already been mentioned, these multipliers can be designed in an extremely simple manner as sliding mechanisms, since the transmitted envelope values can only have values which are either zero or potencies of two for reasons of redundancy reduction.
  • FIG. 4 shows an embodiment of the synthesis part ST according to FIG. 2, in which the bandpasses B1 ', B2' ... BM 'are combined to form a sum filter SB.
  • a multiplier-summing arrangement MS is provided here, each of which has M multipliers. The outputs of the M multipliers are connected to the associated switch S via a summer SU.
  • the filter coefficients hi (k) of all equivalent taps of the individual filters are supplied to one input of the M multipliers MU of a multiplier-summing arrangement MS, while the envelope values A1, A2 ... AM are present at their second inputs.
  • the respective sum function connected to the sum line 1 and in this way the output signal y (n) obtained at the output of the synthesis part ST '.
  • the transmission of the voiced-unvoiced criterion Sc and the random generator PNG can be dispensed with in an extremely advantageous manner in such a synthesis part with time-variant FIR filters with a suitable measurement of the filter coefficients defining the impulse response or the noise response will.
  • This is shown schematically in Fig. 5.
  • the filter coefficients hi '(k) are expediently determined here experimentally so that the synthesized speech signal has an optimal speech quality at the output. As has already been explained, this can be done in such a way that the filter coefficients hi '(n) of the band-pass filters B1', B2 '...

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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)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
  • Electrophonic Musical Instruments (AREA)
  • Analogue/Digital Conversion (AREA)

Claims (7)

  1. . 1. Dispositif pour la transmission de la parole conformément au principe des vocodeurs à canaux, dans lequel du côté émission dans un analyseur à vocodeur, des valeurs de la courbe enveloppe pour un nombre, prédéterminé par un ensemble de filtres, de M canaux spectraux, qui diffèrent entre eux en ce qui concerne la position de fréquence et la largeur de la bande des fréquences, ainsi que, éventuellement, des paramètres supplémentaires spécifiques à la parole, comme par exemple la fréquence fondamentale de la voix et la caractéristique d'accord, sont dérivés du signal vocal devant être transmis et sont transmis de façon appropriée au côté réception, en étant réunis pour former un signal somme numérique, au rythme de trames successives englobant chacune un intervalle d'analyse, et dans lequel en outre le signal somme numérique est à nouveau réparti, du côté réception, entre les M différents canaux spectraux et éventuellement entre les canaux associés aux paramètres supplémentaires spécifiques de la parole, et ces signaux répartis dans les canaux sont ensuite envoyés au synthétiseur à vocodeur situé du côté réception et qui, pour sa part, comporte un ensemble de filtrage correspondant à l'analyseur à vocodeur situé du côté émission, ainsi qu'un générateur d'impulsions et délivre, sur sa sortie, le signal vocal synthétique produit, caractérisé par le fait que l'ensemble de filtrage réalisé conformément à la technique numérique est constitué par des filtres non récursifs variables dans le temps (B1' ... BM') présentant une réponse impulsionnelle infinie (filtres FIR), 'qu'en outre la grandeur d'excitation (x(n)) du générateur d'impulsions (PG, PNG) possédant une amplitude d'impulsion constante est envoyée, sur le côté entrée, à l'ensemble de filtrage et que la variabilité dans le temps du facteur d'amplification de l'ensemble de filtrage est réalisée par multiplication des coefficients de l'ensemble de filtrage (hi(n), hi'(n)) avec les valeurs (Al...AM) de la courbe enveloppe transmises au rythme de trames successives, dans des multiplicateurs (MU).
  2. 2. Dispositif suivant la revendication 1, caractérisé par le fait que les filtres (FIR), à l'entrée desquels la grandeur d'excitation (x(n)) est envoyée, possèdent chacun un circuit itératif formé de N-1 étages de retardement identiques (Z-1) possédant au maximum N prises, que chaque filtre (FIR) possède en outre un nombre, correspondant aux prises de commutateurs (S) dont les entrées de commande sont reliées aux prises, et qu'en outre chaque commutateur relie la sortie d'un multiplicateur (MU) à la sortie d'un filtre et que des premières entrées du multiplicateur forment en commun l'entrée pour la valeur de la courbe enveloppe transmise associée et qu'aux autres entrées respectives des multiplicateurs sont envoyés des coefficients respectifs de filtre (hi(n)), qui fixent dans sa totalité la réponse des filtres.
  3. 3. Dispositif suivant la revendication 1, caractérisé par le fait que l'ensemble de filtrage est un filtre de sommation (FIR), qui comporte un circuit itératif formé de N-1 étages de retardement (Z-1) identiques comportant au maximum N prises, qu'en outre il est prévu un nombre, correspondant aux prises, de commutateurs (S) dont les entrées de commande sont reliées aux prises, et que chaque commutateur relie à la sortie des filtres la sortie d'un circuit de multiplication et de sommation (MS) qui réalise la formation du produit des différents coefficients (Hi(n)) des filtres par les valeurs transmises, et associées (Ai) de la courbe enveloppe et forment ta somme des produits.
  4. 4. Dispositif suivant l'une des revendications précédentes, caractérisé par le fait que les multiplicateurs sont réalisés par des dispositifs à décalage.
  5. 5. Dispositif suivant l'une des revendications précédentes, caractérisé par le fait que la grandeur d'excitation (x(n)), envoyée aux filtres FIR (B1' ... BM', Bi') ou au filtre de sommation FIR (SB), et, en l'absence d'une commutation « non accordée-accordée est une suite d'impulsions produites périodiquement par un générateur d'impulsions (PG) et dont la fréquence de récurrence peut être commandée par la valeur transmise (No) de la fréquence fondamentale de la voix, et que les coefficients (hi(n)) des filtres sont dimensionnés de telle sorte qu'une qualité optimale de parole de la voix synthétisée est de ce fait garantie.
  6. 6. Dispositif suivant la revendication 5, caractérisé par le fait que les coefficients (hi'(n)) des filtres FIR (B1' ... BM', Bi) ou des gammes du filtre de sommation FIR (SB) possédant des fréquences centrales < 2 kHz sont dimensionnées pour des réponses impulsionnelles, et que les coefficients des filtres FIR ou des gammes du filtre de sommation FIR possédant les fréquences centrales > 2 kHz sont dimensionnés pour des réponses de bruits parasites.
  7. 7. Dispositif suivant la revendication 5, caractérisé par le fait que tous les coefficients (Hi'(n)) des filtres FIR (B1' ... BM', Bi') ou des filtres de sommation FIR (SB) sont dimensionnés pour des réponses de bruits parasites.
EP82108691A 1981-09-22 1982-09-20 Dispositif pour la transmission de la parole utilisant le principe des vocodeurs à canaux Expired EP0075311B1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE19813137679 DE3137679A1 (de) 1981-09-22 1981-09-22 Anordnung zur uebertragung von sprache nach dem kanalvocoderprinzip
DE3137679 1981-09-22

Publications (3)

Publication Number Publication Date
EP0075311A2 EP0075311A2 (fr) 1983-03-30
EP0075311A3 EP0075311A3 (en) 1984-02-01
EP0075311B1 true EP0075311B1 (fr) 1986-08-20

Family

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Family Applications (1)

Application Number Title Priority Date Filing Date
EP82108691A Expired EP0075311B1 (fr) 1981-09-22 1982-09-20 Dispositif pour la transmission de la parole utilisant le principe des vocodeurs à canaux

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US (1) US4574392A (fr)
EP (1) EP0075311B1 (fr)
JP (1) JPS594720B2 (fr)
DE (2) DE3137679A1 (fr)

Families Citing this family (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CA1233891A (fr) * 1985-05-27 1988-03-08 Peter Gillingham Filtre a impulsions a condensateurs commutes
DE3521413C2 (de) * 1985-06-14 1995-09-21 Daimler Benz Aerospace Ag Sprechfunkanordnung
GB2182795B (en) * 1985-11-12 1988-10-05 Nat Res Dev Apparatus and methods for speech analysis
JPS6345933A (ja) * 1986-04-15 1988-02-26 Nec Corp 秘話装置
DE3826858A1 (de) * 1988-08-08 1990-02-15 Thomson Brandt Gmbh Zwei-band-filter
RU2224359C1 (ru) * 2002-06-26 2004-02-20 Московский технический университет связи и информатики Способ передачи и приема сигналов, представленных параметрами спектрально-модуляционного разложения, и устройство для его осуществления
RU2584462C2 (ru) * 2014-06-10 2016-05-20 Федеральное государственное образовательное бюджетное учреждение высшего профессионального образования Московский технический университет связи и информатики (ФГОБУ ВПО МТУСИ) Способ передачи и приема сигналов, представленных параметрами ступенчатого модуляционного разложения, и устройство для его осуществления

Family Cites Families (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
AT272413B (de) * 1967-06-29 1969-07-10 Ibm Oesterreich Internationale Einrichtung zur Sprachsynthese für mehrere Sprachkanäle
JPS5853358B2 (ja) * 1980-03-31 1983-11-29 株式会社東芝 音声分析装置
US4377793A (en) * 1981-01-13 1983-03-22 Communications Satellite Corporation Digital adaptive finite impulse response filter with large number of coefficients
JPS5833313A (ja) * 1981-08-21 1983-02-26 Nec Corp トランスバ−サル定利得可変等化器

Also Published As

Publication number Publication date
DE3272737D1 (en) 1986-09-25
JPS5870298A (ja) 1983-04-26
JPS594720B2 (ja) 1984-01-31
US4574392A (en) 1986-03-04
DE3137679A1 (de) 1983-04-07
EP0075311A2 (fr) 1983-03-30
EP0075311A3 (en) 1984-02-01

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