EP0361432A2 - Verfahren und Einrichtung zur Codierung und Decodierung von Sprachsignalen unter Anwendung von Multipuls-Anregung - Google Patents

Verfahren und Einrichtung zur Codierung und Decodierung von Sprachsignalen unter Anwendung von Multipuls-Anregung Download PDF

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Publication number
EP0361432A2
EP0361432A2 EP89117837A EP89117837A EP0361432A2 EP 0361432 A2 EP0361432 A2 EP 0361432A2 EP 89117837 A EP89117837 A EP 89117837A EP 89117837 A EP89117837 A EP 89117837A EP 0361432 A2 EP0361432 A2 EP 0361432A2
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European Patent Office
Prior art keywords
signal
gain
long
term
excitation
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French (fr)
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EP0361432A3 (en
EP0361432B1 (de
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Maurizio Omologo
Daniele Sereno
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SIP SAS
Italtel SpA
Telecom Italia SpA
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SIP SAS
Italtel SpA
Italtel Societa Italiana Telecomunicazioni SpA
SIP Societa Italiana per lEsercizio delle Telecomunicazioni SpA
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Publication of EP0361432A3 publication Critical patent/EP0361432A3/en
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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
    • G10L19/10—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation

Definitions

  • the present invention concerns medium-low bit-rate speech signal coding systems, and more particularly it relates to a coding-­decoding method and device using a multipulse analysis-by-synthesis excitation technique.
  • Multipulse linear prediction coding is one of the most promising techniques for obtaining high quality synthetic speech at bit rates below 16 kbit/s. This technique has been originally proposed by B. S. Atal and J. R. Remde in the paper entitled “A new method of LPC excitation for producing natural-sounding speech at low bit rates", International Conference on Acoustic, Speech, Signal Processing (ICASSP), pages 614-617, Paris, 1982.
  • IICASSP International Conference on Acoustic, Speech, Signal Processing
  • the excitation signal for the synthesis filter consists of a train of pulses whose amplitudes and time positions are determined so as to minimize a perceptually-meaningful distorsion measurement; such a measurement is obtained by comparing the samples at the synthesis filter output with the original speech samples and simultaneous weighting the difference by a function which takes into account how the human perception evaluates the distorsion introduced (analysis-by-synthesis procedure)
  • the synthesizer comprises the cascade of a long-term and a short-term synthesis filter are of particular interest: in fact they provide signals whose quality gradually decreases as the bit rate decreases and do not present a dramatic performance deterioration below a threshold rate.
  • the invention provides on the contrary a method and a device allowing quality to be increased leaving bit rate unchanged or a given quality to be maintained even at lower bit rate.
  • This can be achieved by using a combined optimization technique, of sequential type, of the parameters of the long-term synthesis filter and of the excitation within the analysis-by-synthesis procedure; the sequential procedure is sub-optimum with respect to the original optimum one, but it is easier to be implemented.
  • the method according to the invention comprises a coding phase including the following operations: - speech signal conversion into frames of digital samples; - short-term analysis of the speech signal, to determine a group of linear prediction coefficients relevant to a current frame and a representation of said coefficients as line spectrum pairs; - coding of said representation of the linear prediction coefficients; - spectral shaping of the speech signal, by weighting the digital samples in a frame by a first and a second weighting functions, the weighting according to the first weighting function generating a residual signal which is then weighted by the second function to generate the spectrally-shaped speech signal; - long-term analysis of the speech signal, by using said residual signal and said spectrally-shaped signal, to determine the lag separating a current sample from a preceding sample used to process said current sample, and the gain by which said preceding sample is weighted for the processing; - determination of the locations and amplitudes of the excitation pulses, by exploiting the results of the short-term and long-term analysis;
  • the excitation is reconstructed starting from the coded values of the amplitudes, the locations and the r.m.s. values of the pulses, and, by using the reconstructed excitation, a synthesized filtering followed by a short-term synthesis filtering, using long-term analysis parameters and respectively the linear prediction coefficients reconstructed starting from said coded line spectrum pairs or line pair differences, which method is characterized in that the long-term analysis and excitation pulse generation are performed in successive steps, in the first of which the long-term analysis gain and lag are determined by minimizing a mean squared error between the spectrally-shaped speech signal and a further signal obtained by weighting by said second weighting function the signal resulting from a long-term synthesis filtering in which the input signal used for the synthesis is a null signal, while in the second step the amplitudes and positions of the excitation pulses are actually determined, and in that said coding of the representation of
  • the invention provides also a device for implementing the method, comprising, for speech signal coding: - means for converting the speech signal into digital sample frames; - means for the short-term analysis of the speech signal, which means receive a group of samples from said converting means, compute a set of linear prediction coefficients relevant to a current frame, and emit a representation of said linear prediction coefficients as line spectrum pairs; - means for coding said representation of the linear prediction coefficients; - means for obtaining quantized linear prediction coefficients from said coded representation; - a circuit for the spectral shaping of the speech signal, connected to the converting means and to the means obraining the quantized coefficients, and comprising a pair of cascaded weighting digital filters, which weight the digital samples according to a first and a second weighting function, respectively, said first filter supplying a residual signal; - means for the long-term analysis of the speech signal, connected to the output of said first filter and of the spectral shaping circuit, to determine the lag which separates a current sample
  • a generic speech signal coding-­deconding system can be schematized by a coder COD, a transmission channel CH and a decoder DEC.
  • coder COD receives digital samples s(n) of the original speech signal, organized into frames comprising each a predetermined number of samples, and sends onto channel CH, for each sample frame, the coding of a suitable representation ⁇ (k) of a group of linear prediction coefficients a(k) obtained by a short-term analysis of the speech signal, the coded amplitudes and positions A(i), Cp of the pulses forming the excitation signal, the coded r.m.s. values ⁇ (i) of the excitation pulses, and the codings of two parameters (gain B and lag M) determined by the long-term analysis.
  • Decoder DEC reconstructs the excitation and generates a synthesized speech signal on the basis of the reconstructed excitation, the linear prediction coefficients reconstructed starting from the transmitted representation thereof, and long-term analysis parameters.
  • the digital sample frames, present on connection 1 are supplied to a spectral shaping circuit SW and to a short-term analysis circuit STA.
  • Spectral shaping circuit SW performs a frequency-shaping of the speech signal in order to render the differences between the original and the reconstructed speech signals less perceptible in correspondence with the formants of the original speech signal.
  • Such a circuit consists of a pair of cascaded digital filters F1, F2, whose transfer functions, in z transform, are given in a non-limiting example respectively by relations and where z represents a sampling interval delay;
  • â(k) is a quantized linear prediction coefficient vector (1 ⁇ k ⁇ p, where p is the filter order) reconstructed from the coded representation of the linear prediction coefficients short-term analysis result;
  • ⁇ is an experimentally determined constant correcting factor, determining the bandwidth increase around the formants.
  • a signal r(n) hereinafter referred to as "residual signal”
  • spectrally weighted speech signal s w (n) is obtained on output connection 3 of F2: both signals are used in long-­term analysis.
  • Short-term analysis circuit STA is to determine linear prediction coefficients a(k), which depend on short-term correlations deriving from a non-flat spectral envelope of speech signal. Circuit STA calculates coefficients a(k) according to the classical autocorrelation method, as described in "Digital Signal Processing of Speech Signals" by L.R. Rabiner and R.W. Schafer (Prentice-Hall, Englewood Cliffs, N.J., USA, 1978), page 401, and uses to this aim a set of digital samples s h (n) which can comprise, besides the samples of the current frame, a certain number of samples of both the preceding and the following frames.
  • Block STA also comprises circuits for transforming the coefficients into a group of parameters ⁇ (k) in the frequency domain, known as "line spectrum pairs", which are presented on output 5 of STA.
  • line spectrum pairs denote the resonant frequencies at which the acoustic tube, the vocal tract can be assimilated to, exhibits a line spectrum structure under extreme boundary conditions corresponding to complete opening and closure at the glottis.
  • the conversion of linear prediction coefficients into line spectrum pairs is described e.g. by N. Sugamura and F.Itakura in the paper "Speech analysis and synthesis method developed at ECL in NTT - From LPC to LSP", Speech Communication, Vol.5, No.2, June 1986, pages 199-215.
  • Line spectrum pairs ⁇ (k) or the differences ⁇ between adjacent line pairs are then vectorially quantized in a vector quantization circuit VQ exploiting techniques of the type descrbed in published European Patent application EP-A-186763 (CSELT), applied to a set of codebooks.
  • CSELT published European Patent application
  • That vector instead of being coded by a single word with that number of bits, is quantized by a group of words of smaller size chosen out of suitable sub-codebooks.
  • the modality of quantization of the above patent application are applied to obtain each of said words.
  • vector quantizer VQ is one of the characteristics of the present invention and allows a reduction in the number of bits necessary to code the results of the short-term analysis, while maintaining the same quality of the coded signal, from about 36-34 bits (scalar quantization) to 24 (vector quantization).
  • differences ⁇ organized into three vectors of 3, 3 and 4 components respectively, may be quantized with 24 bits organized into three groups of 256 words, each group corresponding to one of said vectors.
  • the indices of the vectors are sent by VQ on a connection 6 which belongs to channel CH.
  • a circuit DCO obtains from said indices quantized linear prediction coefficients â(k) which are supplied, through connection 4, to filters F1, F2 of circuit SW, to an excitation generator EG and to a long-term analysis circuit LTA.
  • LTA supplies information dependent on the fine spectral structure of the signal, which information is used to make the synthesized signal more natural-sounding.
  • the samples relevant to M preceding sampling instants weighted by a weighting factor (gain) B, are used.
  • LTA is just to determine both M and B.
  • Lag M in case of a voiced sound, corresponds to the pitch period.
  • the lag can range from 20 to 83 samples and it is updated every frame. The gain is on the contrary updated every half frame.
  • Values M and B are emitted on a connection 7 and are supplied to excitation generator EG which also receives, through a connection 8, a signal s we (n), obtained from s w (n) in a manner which will be described hereinafter. Values M and B are also sent to a coder LTC, which transfers the coded signals onto a connection 9 belonging to channel CH.
  • LTC liquid crystal display
  • Long-term analysis circuit LTA performs a closed-loop analysis as a part of the procedure for determining the pulse positions, with modalities allowing a good coder performance to be maintained even if a sub-optimum procedure is used, as will be better described hereinafter.
  • Excitation generator EG is to supply the sequence of Ns pulses (e.g. 6), distributed within a time period Ls (more particularly corresponding to half a frame), forming the excitation signal; such a signal is computed so as to minimize a mean squared error, frequency-weighted as mentioned, between the original signal and the reconstructed one.
  • Ns pulses e.g. 6
  • Ls more particularly corresponding to half a frame
  • Excitation generator EG supplies, through a connection 10, the pulses it has generated to a circuit PAC coding the amplitudes and the positions of such pulses, which circuits calculate and code also the r.m.s. values of said pulses.
  • the coded values ⁇ (i), A(i) (1 ⁇ i ⁇ Ns) and Cp are emitted on a connection 11, also belonging to channel CH.
  • circuit PAC The structure of circuit PAC is known to the skilled in the art.
  • an excitation decoder ED reconstructs the excitation starting from the coded values ⁇ (i), A(i), Cp.
  • reconstructed excitation pulses ê are supplied by ED to a long-term synthesis filter LTPl which, together with a short-term synthesis filter STP, forms synthesizer SYN.
  • Reconstructed residual signal r ⁇ is present at the output of LTP1 and is sent via a connection 14 to short-term synthesis filter STP.
  • This is a filter whose transfer function in z transform is 1/A(z), where A(z) is the function already examined for filter F1 of spectral weighting circuit SW.
  • Coefficients â(k) for filter STP are supplied through a connection 15 from a circuit STC, which reconstructs them by decoding the information relevant to line spectrum pairs.
  • Filter STP emits on connection 16 the reconstructed or synthesized speech signal ⁇ .
  • the optimum solution would be determining, for each pair of possible values m, b of the lag and gain used to determine the optimum values M, B to be exploited in the synthesis, the combination of excitation pulses, gain and lag minimizing the mean squared error between the original signal and the reconstructed signal.
  • the optimum solution is too complex and hence, according to the invention, the determination of M and B is separated from that of the excitation pulses. There are hence two successive operation phases.
  • M, B of m and b are to be found which minimize mean squared error between frequency-shaped speech signal s w (n) and a signal s w0 (n) obtained by weighting, in the same way as the residual signal, a signal r0 obtained as a response from a long-term synthesis filter (similar to the one of the synthesizer) when at the filter input a zero has been forced (long-term synthesis filter memory).
  • a predetermined value b is allotted to the gain and the error is minimized for each value m of lag: once found optimum lag M, the successive step is that of determining the optimum gain B.
  • a second and simpler solution is that of computing M by using a signal 0 which consists of the signal r0, when the lag is greater than the frame length (or, more generally, when a sample of the current frame is processed by using a sample of the preceding frame), while in the opposite case it is equal to residual signal r(n), and minimising the error Under said conditions the previous constraint for the lag is eliminated, since signals r0 are always available, and hence M can be determined as the number m of samples which maximizes the function
  • gain B can be determined either in exhaustive manner or by the following procedure, which reduces the necessary amount of computations.
  • B is computed every half frame, and hence also the excitation pulses will be computed every half frame.
  • Fig. 3 shows a block diagram of the devices of LTP and EG in case signal 0 is used to determine M and B.
  • a synthesis filter LTP2 having a transfer function similar to that of LTP1 (Fig. 1), is fed with a null signal.
  • filter LTP2 successively uses the different values m and, for each of them, an optimum value b(ott) which is implicitly obtained in the above-mentioned derivative operation.
  • B LTP2 uses value M of the lag determined in the preceding step and different values b.
  • Values m and b are supplied to LTP2 by a processing unit CMB, carrying out the computation and comparisons mentioned above.
  • Signal 0 is present on output 20 of LTP2.
  • Output 20 is connected to a first input of a multiplexer MX1 receiving at a second input the residual signal r(n) present on connection 3, and letting through signal r0 or signal r depending on the relative value of m and n.
  • signal r0 is present on output connection 21 of MX1, and that signal is delayed by a time equal to m samples in a delay element DL1 before being sent to CMB.
  • the latter receives also signal r(n) and, for each frame and for all values m, calculates function R′(m) and determines the value M of m which maximizes such function.
  • the value is stored into a register RM and made available on wires 7a of connection 7.
  • Output 20 of LTP2 is also connected to a weighting filter F3, which is enabled only while B is being computed and has the same transfer function 1/ A(z, ⁇ ) as filter F2 in SW (Fig. 1).
  • Filter F3 weights signal r0 (or r′0, when the gain used in LTP2 is 1) giving at output 22 signal s w0 (s′ w0 ).
  • the latter is supplied at an input of an adder SM1 where it is subtracted from signal s w coming from spectral weighting filter SW (Fig. 1) via connection 3.
  • SM1 supplies on output 8 signal s we .
  • device CMB determines, every half frame, value B of b which minimizes E and stores it into register RB which keeps it available, for the whole half frame, on a group of wires 7b of connection 7.
  • Values B, M computed by CMB are supplied to LTC (Fig.1) and to a long-term synthesis filter LTP3 which is part of the excitation generator EG and is followed by a weighting filter F4.
  • Filters LTP3, F4 have transfer functions similar to those of LTP1 and F2, respectively;
  • LTP3 is fed, during the analysis-by-synthesis procedure, with the excitation pulses e(i) supplied via connection 10 by a processing unit CE which sequentially determines the positions and the amplitudes of the various pulses.
  • F4 emits on output 24 signal ⁇ we which is supplied to a first input of an adder SM2 receiving at a second input signal s we outgoing from SM1. The difference between the two signals is then supplied via connection 25 to CE, which determines pulses e(i) by minimizing mean squared error dw.

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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)
  • Analogue/Digital Conversion (AREA)
  • Dc Digital Transmission (AREA)
EP89117837A 1988-09-28 1989-09-27 Verfahren und Einrichtung zur Codierung und Decodierung von Sprachsignalen unter Anwendung von Multipuls-Anregung Expired - Lifetime EP0361432B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
IT6786888 1988-09-28
IT67868/88A IT1224453B (it) 1988-09-28 1988-09-28 Procedimento e dispositivo per la codifica decodifica di segnali vocali con l'impiego di un eccitazione a impulsi multipli

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EP0361432A2 true EP0361432A2 (de) 1990-04-04
EP0361432A3 EP0361432A3 (en) 1990-09-26
EP0361432B1 EP0361432B1 (de) 1994-08-17

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EP (1) EP0361432B1 (de)
AT (1) ATE110180T1 (de)
DE (2) DE361432T1 (de)
ES (1) ES2017906T3 (de)
GR (1) GR900300170T1 (de)
IT (1) IT1224453B (de)

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
ES2042410A2 (es) * 1992-04-15 1993-12-01 Control Sys S A Metodo de codificacion y codificador de voz para equipos y sistemas de comunicacion.
EP0599569A2 (de) * 1992-11-26 1994-06-01 Nokia Mobile Phones Ltd. Verfahren zum Kodieren eines Sprachsignales
US5327519A (en) * 1991-05-20 1994-07-05 Nokia Mobile Phones Ltd. Pulse pattern excited linear prediction voice coder
WO1995029480A2 (en) * 1994-04-22 1995-11-02 Philips Electronics N.V. Analogue signal coder
US5761635A (en) * 1993-05-06 1998-06-02 Nokia Mobile Phones Ltd. Method and apparatus for implementing a long-term synthesis filter
EP0910064A2 (de) * 1991-02-26 1999-04-21 Nec Corporation Sprachkodierungsverfahren und -vorrichtung

Non-Patent Citations (7)

* Cited by examiner, † Cited by third party
Title
ELECTRONICS LETTERS, vol. 23, no. 24, 19th November 1987, pages 1286-1288, Hitchin, GB; A. KONDOZ et al.: "Vector-quantised transform coder for speech coding at 9.6kbits/s and below" *
ICASSP 84, IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, San Diego, Calfornia, 19th - 21st March 1984, vol. 1, pages 10.4.1 - 10.4.4, IEEE, New York, US; P. KROON et al.: "Experimental evaluation of different approaches to the multi-pulse coder" *
ICASSP 86, IEEE-IECEJ-ASJ INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Tokyo, 7th - 11th April 1986, vol. 4, pages 3067-3070, IEEE, New York, US; G. OHYAMA et al.: "A novel approach to estimating excitation code in code-excited linear prediction coding" *
ICASSP 86, IEEE-IECEJ-ASJ INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Toyko, 7th - 11th April 1986, vol. 3, pages 1689-1692, IEEE, New York, US; K. OZAWA et al.: "High quality multi-pulse speech coder with pitch predicton" *
ICASSP 86, IEEE-IECEJ-ASJ INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, ANDSIGNAL PROCESSING, Tokyo, 7th - 11th April 1986, vol. 4, pages 3067-3070, IEEE,New York, US; G. OHYAMA et al.: "A novel approach to estimating excitation codein code-excited linear prediction coding" *
ICASSP 87, INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Dallas, Texas, 6th - 9th April 1987, vol. 3, pages 1649-1652, IEEE, New York, US; P. KROON et al.: "Quantization procedures for the excitation in celp coders" *
SIGNAL PROCESSING, Toyko, 7th - 11th April 1986, vol. 3, pages 1689-1692,IEEE, New York, US; K. OZAWA et al.: "High quality multi-pulse speech coderwith pitch predicton" *

Cited By (11)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0910064A2 (de) * 1991-02-26 1999-04-21 Nec Corporation Sprachkodierungsverfahren und -vorrichtung
EP0910064A3 (de) * 1991-02-26 1999-06-16 Nec Corporation Sprachkodierungsverfahren und -vorrichtung
US5327519A (en) * 1991-05-20 1994-07-05 Nokia Mobile Phones Ltd. Pulse pattern excited linear prediction voice coder
ES2042410A2 (es) * 1992-04-15 1993-12-01 Control Sys S A Metodo de codificacion y codificador de voz para equipos y sistemas de comunicacion.
EP0599569A2 (de) * 1992-11-26 1994-06-01 Nokia Mobile Phones Ltd. Verfahren zum Kodieren eines Sprachsignales
EP0599569A3 (en) * 1992-11-26 1994-09-07 Nokia Mobile Phones Ltd A method of coding a speech signal.
AU665283B2 (en) * 1992-11-26 1995-12-21 Nokia Mobile Phones Limited A method for the efficient coding of a speech signal
US5596677A (en) * 1992-11-26 1997-01-21 Nokia Mobile Phones Ltd. Methods and apparatus for coding a speech signal using variable order filtering
US5761635A (en) * 1993-05-06 1998-06-02 Nokia Mobile Phones Ltd. Method and apparatus for implementing a long-term synthesis filter
WO1995029480A2 (en) * 1994-04-22 1995-11-02 Philips Electronics N.V. Analogue signal coder
WO1995029480A3 (en) * 1994-04-22 1995-12-07 Philips Electronics Nv Analogue signal coder

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DE68917552D1 (de) 1994-09-22
EP0361432A3 (en) 1990-09-26
ES2017906T3 (es) 1994-10-16
ES2017906A4 (es) 1991-03-16
IT8867868A0 (it) 1988-09-28
ATE110180T1 (de) 1994-09-15
EP0361432B1 (de) 1994-08-17
IT1224453B (it) 1990-10-04
GR900300170T1 (en) 1991-09-27
DE361432T1 (de) 1991-03-21
DE68917552T2 (de) 1995-01-12

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