EP0076234A1 - Procédé et dispositif pour traitement digital de la parole réduisant la redondance - Google Patents

Procédé et dispositif pour traitement digital de la parole réduisant la redondance Download PDF

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Publication number
EP0076234A1
EP0076234A1 EP82810391A EP82810391A EP0076234A1 EP 0076234 A1 EP0076234 A1 EP 0076234A1 EP 82810391 A EP82810391 A EP 82810391A EP 82810391 A EP82810391 A EP 82810391A EP 0076234 A1 EP0076234 A1 EP 0076234A1
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EP
European Patent Office
Prior art keywords
speech
parameters
section
pitch
sections
Prior art date
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.)
Granted
Application number
EP82810391A
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German (de)
English (en)
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EP0076234B1 (fr
Inventor
Stephan Dr. Horvath
Carlo Bernasconi
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Omnisec AG Te Regensdorf Zwitserland
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Gretag AG
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Priority to AT82810391T priority Critical patent/ATE15415T1/de
Publication of EP0076234A1 publication Critical patent/EP0076234A1/fr
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Publication of EP0076234B1 publication Critical patent/EP0076234B1/fr
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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

Definitions

  • the invention relates to a method operating according to the linear prediction method and to a corresponding device for redundancy-reducing digital speech processing according to the preamble of patent claim 1 and patent claim 13.
  • the LPC vocoders known and available today do not yet work fully satisfactorily. Although the language that was re-synthesized after the analysis is usually still relatively understandable, it is distorted and sounds artificial. One of the reasons for this is with difficulty in making the decision as to whether there is a voiced or an unvoiced speech section with sufficient certainty. Other causes include poor determination of the pitch period and inaccurate determination of the sound formation filter parameters.
  • the data rate must in many cases be limited to a relatively low value. It is e.g. in the case of telephone networks, preferably only 2.4 kbit / sec.
  • the data rate is determined by the number of speech parameters analyzed in each speech section, by the number of bits required for these parameters and by the so-called frame rate, i.e. given the number of speech sections per second.
  • frame rate i.e. given the number of speech sections per second.
  • at least slightly more than 50 bits are required per speech section. This automatically sets the maximum frame rate, e.g. in a 2.4 kbit / sec system to around 45 / sec.
  • the voice quality at these relatively low frame rates is also correspondingly poor. It is not possible to increase the frame rate, which would in itself improve the voice quality, as this would exceed the specified data rate. To reduce the number of bits required per frame, on the other hand, a reduction in the number of parameters used or a coarsening of their quantization would be necessary, but this would automatically result in a deterioration in the quality of the speech reproduction.
  • the present invention now deals primarily with these difficulties caused by predetermined data rates and, in particular, has the aim of improving a method or a device of the type defined at the outset with regard to the quality of the speech reproduction, without increasing the data rates.
  • the basic idea of the invention is therefore to save bits by improved coding of the speech parameters, so that the frame rate can be increased.
  • the frame rate can be increased.
  • there is also an interrelation between the coding of the parameters and the frame rate since less bit-intensive, redundancy-reducing coding is only possible or makes sense at higher frame rates.
  • the coding of the parameters according to the invention is based on the use of the correlation between adjacent voiced speech sections (interframe correlation), which of course becomes increasingly stronger with increasing frame rate.
  • FIG. 1 The general structure and mode of operation of the speech processing device according to the invention are shown in FIG. 1. That from any source, e.g. Analog voice signal originating from a microphone 1 is band-limited in a filter 2 and then sampled and digitized in an A / D converter 3. The sampling rate is about 6 to 16 kHz, preferably about 8 kHz.
  • the resolution is about 8 to 12 bit.
  • the pass band of the filter 2 usually extends from approximately 80 Hz to approximately 3.1-3.4 kHz in the case of so-called broadband speech, and from approximately 300 Hz to 3.1-3.4 kHz in the telephone language.
  • the speech section length is approximately 10 to 30 msec, preferably approximately 20 msec.
  • the frame rate i.e. the number of frames per second is approximately 30-100, preferably approximately 50 to 70.
  • sections which are as short as possible and correspondingly high frame rates are desirable, but this is on the one hand in real-time processing limited performance of the computer used and, on the other hand, the demand for the lowest possible bit rates during the transmission.
  • the analysis is essentially divided into two main procedures, firstly in the calculation of the amplification factor or volume parameter and the coefficients or filter parameters of the underlying vocal tract model filter and secondly in the voiced-unvoiced decision and in determining the pitch -Period in voiced case.
  • the filter coefficients are obtained in a parameter calculator 4 by solving the system of equations which is obtained when the energy of the prediction error, ie the energy of the difference between the actual samples and the samples estimated on the basis of the model assumption in the interval under consideration (speech section) is minimized as a function of the coefficients becomes.
  • the system of equations is preferably solved using the autocorrelation method using an algorithm according to Durbin (cf., for example, BLB Rabiner and RW Schafer, "Digital Processing of Speech Signals", Prentice-Hall Inc., Englewood Cliffs, NJ, 1978, pages 411- 413).
  • reflection coefficients (k.) are less sensitive transforms of the filter coefficients (a.) To quantization.
  • the reflection coefficients are always smaller than 1 and, in addition, their amount decreases with an increasing atomic number. Because of these advantages, these reflection coefficients (k.) Are preferably transmitted instead of the filter coefficients (a j ).
  • the volume parameter G results from the algorithm as a by-product.
  • the digital voice signal s in a buffer 5 until at first are calculated until the Filterparamet he (a j).
  • Inverse filtering is a prediction error signal e which is similar to the excitation signal x n multiplied by the gain factor G.
  • This prediction error signal en is now supplied in the case of telephone speech directly or in the case of broadband speech via a low-pass filter 7 to an autocorrelation stage 8, which forms the autocorrelation function AKF standardized to the zero-order autocorrelation maximum, on the basis of which the pitch period p is determined in a pitch extraction stage 9, and although in a known manner as the distance between the second autocorrelation maximum RXX and the first maximum (zero order), an adaptive search method is preferably used.
  • the language section under consideration is classified as voiced or unvoiced in a decision stage 11 according to certain criteria, which include also include the energy of the speech signal and the number of zero crossings in the section under consideration. These two values are determined in an energy determination stage 12 and a zero crossing determination stage 13.
  • the parameter calculator described above determines a set of filter parameters for each speech section (frame).
  • the filter parameters could also be determined differently, for example continuously by means of adaptive inverse filtering or another known method, the filter parameters being readjusted continuously with each sampling cycle, but only at the times determined by the frame rate for further processing or Transmission will be provided.
  • the invention is in no way restricted in this regard. It is only essential that there is a set of filter parameters for each language section.
  • the parameters (kj), G and p obtained according to the method just described are then fed to a coding stage 14, where they are brought (formatted) and made available in a particularly bit-efficient form suitable for the transmission in a manner to be described in more detail below.
  • the speech signal is recovered or synthesized from the parameters in a known manner in that the parameters initially decoded in a decoder 15 are fed to a pulse-noise generator 16, an amplifier 17 and a vocal tract model filter 18 and the output signal of the model filter 18 by means of a D / A converter 19 is brought into analog form and then made audible after the usual filtering 20 by a playback device, for example a loudspeaker 21.
  • the volume parameter G controls the amplification factor of the amplifier 17, the filter parameters (k . ) Define the transfer function of the sound formation or vocal tract model filter 18.
  • Fig. 2 An example of such a system is shown in Fig. 2 as a block diagram.
  • the multi-processor system shown essentially comprises four functional blocks, namely a main processor 50, two secondary processors 60 and 70 and an input / output unit 80. It implements both analysis and synthesis.
  • the input / output unit 80 contains the stages designated 81 for analog signal processing, such as amplifiers, filters and automatic gain control, as well as the A / D converter and the D / A converter.
  • the main processor 50 carries out the actual speech analysis or synthesis, for which purpose the determination of the filter parameters and the volume parameters (parameter calculator 4), the determination of energy and zero crossings of the speech signal (stages 13 and 12), the voiced-unvoiced decision (stage 11 ) and the determination of the pitch period (stage 9) or synthesis-side the generation of the output signal (stage 16), its volume variation (stage 17) and its filtering in the speech model filter (filter 18).
  • the main processor 50 is supported by the secondary processor 60, which carries out the intermediate storage (buffer 5), inverse filtering (stage 6), optionally the low-pass filtering (stage 7) and the autocorrelation (stage 8).
  • the secondary processor 70 deals exclusively with the coding or decoding of the speech parameters and with the data traffic with e.g. a modem 90 or the like via an interface designated 71.
  • the data rate in an LPC vocoder system is determined by the so-called frame rate, i.e. the number of speech segments per second, the number of language parameters used and the number of bits required to encode the language parameters.
  • the basic principle of the invention consists in the consideration that if the speech signal is analyzed more often, that is to say the frame rate is increased, a better tracking of the transientities of the speech signal is possible. It is thus achieved in stationaryTransabitese- "a greater correlation between the parameters of successive speech sections, which in turn can be utilized in a more efficient, ie bitêtden encoding so that the total data rate is not increased despite an increase in frame rate, the voice quality is, however, considerably improved.
  • This Special coding of the speech parameters according to the invention is explained in more detail below.
  • the basic idea of the parameter coding according to the invention is the so-called block coding principle, that is to say that the speech parameters are not coded independently of one another for each individual speech section, but rather two or three speech sections are combined to form a block and the parameters of all two or are coded within this block three language sections according to uniform rules and in such a way that in each case only the parameters of the first section are coded in full form, while the parameters of the other language section (s) are coded in differential form or possibly omitted or substituted entirely.
  • the coding within each block is also carried out differently, taking into account the typical properties of human speech, depending on whether it is a voiced or an unvoiced block, the first language section in each case determining the voiced character of the block.
  • Complete coding is understood to mean the customary coding of the parameters, in which e.g. reserved for the pitch parameter 6 bit, for the volume parameter 5 bit and (for a ten-pole filter, for example) for the first four filter coefficients each 5 bits, for the next four each 4 bits and for the last two 3 or 2 bits.
  • the decreasing number of bits for the higher filter coefficients is explained by the fact that the reflection coefficients usually used decrease in magnitude with increasing atomic number and essentially only determine the fine structure of the short-term speech spectrum.
  • the coding according to the invention is different for the individual parameter types (filter coefficients, volume, pitch). It is explained below using the example of blocks consisting of three language sections each.
  • the filter coefficients of the second speech section can also be adopted immediately with those of the first section and therefore do not need to be coded or transmitted at all.
  • the bits released in this way can be used to encode the difference between the filter parameters of the third section and those of the first section with greater resolution.
  • the coding is done in a different way.
  • the filter parameters of the first section are full again, i.e. encoded in full form or full bit length, the filter parameters of the other two sections are not coded differentially, but also in full form.
  • bit reduction use is made of the fact that in the unvoiced case the higher filter coefficients make little contribution to the sound image, and accordingly the higher filter coefficients, e.g. from the seventh, not encoded or transmitted at all. On the synthesis side, they are then interpreted as zero.
  • a change is indicated by a special code word, in that the difference to the pitch parameter of the first speech section, which in any case exceeds the representable difference range, is replaced by this code word.
  • the code word of course has the same format as the pitch parameter differences.
  • the decoded pitch parameter is preferably compared on the synthesis side with a running average of the pitch parameters of a number, for example 2 to 7, previous speech sections and, for example, when a predetermined maximum deviation is exceeded about ⁇ 30% to ⁇ 60%, replaced by the running average.
  • the coding is basically the same as for the blocks with three sections. All parameters of the first section are encoded in their entirety.
  • the filter parameters of the second speech section are either coded in differential form in voiced blocks or assumed to be the same as in the first section and accordingly not coded at all.
  • the filter coefficients of the second speech section are also encoded in their entirety, but the higher coefficients are omitted.
  • the pitch parameter of the second speech section is coded the same again in the voiced and in the unvoiced case, namely in the form of its difference to the pitch parameter of the first section.
  • a code word is used again.
  • the volume parameter of the second speech section is coded in the same way as in the case of blocks with three sections, that is to say in differential form or not at all.
  • the coding and decoding is preferably carried out by software using the computer system which is already available for the remaining speech processing.
  • the creation of a suitable program is within the skill of the average professional.
  • the coding rules contained in Fig. 3 A 1 , A 2 and A 3 and B 1 , B 2 and B 3 are in Fi g . 4 shown in more detail and each indicate the format (bit assignments) of the parameters to be encoded.
  • the programs for decoding are of course analog.

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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)
  • Exchange Systems With Centralized Control (AREA)
EP82810391A 1981-09-24 1982-09-20 Procédé et dispositif pour traitement digital de la parole réduisant la redondance Expired EP0076234B1 (fr)

Priority Applications (1)

Application Number Priority Date Filing Date Title
AT82810391T ATE15415T1 (de) 1981-09-24 1982-09-20 Verfahren und vorrichtung zur redundanzvermindernden digitalen sprachverarbeitung.

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
CH616881 1981-09-24
CH6168/81 1981-09-24

Publications (2)

Publication Number Publication Date
EP0076234A1 true EP0076234A1 (fr) 1983-04-06
EP0076234B1 EP0076234B1 (fr) 1985-09-04

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EP82810391A Expired EP0076234B1 (fr) 1981-09-24 1982-09-20 Procédé et dispositif pour traitement digital de la parole réduisant la redondance

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US (1) US4618982A (fr)
EP (1) EP0076234B1 (fr)
JP (1) JPS5870300A (fr)
AT (1) ATE15415T1 (fr)
CA (1) CA1184656A (fr)
DE (1) DE3266042D1 (fr)

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0360265A2 (fr) * 1988-09-21 1990-03-28 Nec Corporation Système de transmission capable de modifier la qualité de la parole par classement des signaux de paroles
EP0676744A1 (fr) * 1994-04-04 1995-10-11 Digital Voice Systems, Inc. Evaluation des paramètres d'excitation
DE4033350B4 (de) * 1989-10-20 2004-04-08 Canon K.K. Verfahren und Vorrichtung für die Sprachverarbeitung

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CA1229681A (fr) * 1984-03-06 1987-11-24 Kazunori Ozawa Methode et appareil de codage de signaux dans la bande de frequences vocales
CA1255802A (fr) * 1984-07-05 1989-06-13 Kazunori Ozawa Codage et decodage de signaux a faible debit binaire utilisant un nombre restreint d'impulsions d'excitation
CA1252568A (fr) * 1984-12-24 1989-04-11 Kazunori Ozawa Codeur et decodeur de signaux a faible debit binaire pouvant reduire la vitesse de transmission de l'information
US4912764A (en) * 1985-08-28 1990-03-27 American Telephone And Telegraph Company, At&T Bell Laboratories Digital speech coder with different excitation types
US4890328A (en) * 1985-08-28 1989-12-26 American Telephone And Telegraph Company Voice synthesis utilizing multi-level filter excitation
EP0245531A1 (fr) * 1986-05-14 1987-11-19 Deutsche ITT Industries GmbH Application d'une mémoire morte semi-conductrice
US4972474A (en) * 1989-05-01 1990-11-20 Cylink Corporation Integer encryptor
US6006174A (en) 1990-10-03 1999-12-21 Interdigital Technology Coporation Multiple impulse excitation speech encoder and decoder
JP2810252B2 (ja) * 1991-05-22 1998-10-15 シャープ株式会社 音声再生装置
US5317567A (en) * 1991-09-12 1994-05-31 The United States Of America As Represented By The Secretary Of The Air Force Multi-speaker conferencing over narrowband channels
US5272698A (en) * 1991-09-12 1993-12-21 The United States Of America As Represented By The Secretary Of The Air Force Multi-speaker conferencing over narrowband channels
FI95086C (fi) * 1992-11-26 1995-12-11 Nokia Mobile Phones Ltd Menetelmä puhesignaalin tehokkaaksi koodaamiseksi
US5517511A (en) * 1992-11-30 1996-05-14 Digital Voice Systems, Inc. Digital transmission of acoustic signals over a noisy communication channel
FI96248C (fi) * 1993-05-06 1996-05-27 Nokia Mobile Phones Ltd Menetelmä pitkän aikavälin synteesisuodattimen toteuttamiseksi sekä synteesisuodatin puhekoodereihin
US5457685A (en) * 1993-11-05 1995-10-10 The United States Of America As Represented By The Secretary Of The Air Force Multi-speaker conferencing over narrowband channels
PL174216B1 (pl) * 1993-11-30 1998-06-30 At And T Corp Sposób redukcji w czasie rzeczywistym szumu transmisji mowy
AU696092B2 (en) * 1995-01-12 1998-09-03 Digital Voice Systems, Inc. Estimation of excitation parameters
US5754974A (en) * 1995-02-22 1998-05-19 Digital Voice Systems, Inc Spectral magnitude representation for multi-band excitation speech coders
US5701390A (en) * 1995-02-22 1997-12-23 Digital Voice Systems, Inc. Synthesis of MBE-based coded speech using regenerated phase information
US6240384B1 (en) * 1995-12-04 2001-05-29 Kabushiki Kaisha Toshiba Speech synthesis method
SE506034C2 (sv) * 1996-02-01 1997-11-03 Ericsson Telefon Ab L M Förfarande och anordning för förbättring av parametrar representerande brusigt tal
US6131084A (en) * 1997-03-14 2000-10-10 Digital Voice Systems, Inc. Dual subframe quantization of spectral magnitudes
US6161089A (en) * 1997-03-14 2000-12-12 Digital Voice Systems, Inc. Multi-subframe quantization of spectral parameters
US6199037B1 (en) 1997-12-04 2001-03-06 Digital Voice Systems, Inc. Joint quantization of speech subframe voicing metrics and fundamental frequencies
US6377916B1 (en) 1999-11-29 2002-04-23 Digital Voice Systems, Inc. Multiband harmonic transform coder
US7080009B2 (en) * 2000-05-01 2006-07-18 Motorola, Inc. Method and apparatus for reducing rate determination errors and their artifacts
DE102004001293A1 (de) * 2004-01-07 2005-08-11 Deutsche Thomson-Brandt Gmbh Vorrichtung und Verfahren zur Datenübertragung mit reduzierter Datenmenge

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Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0360265A2 (fr) * 1988-09-21 1990-03-28 Nec Corporation Système de transmission capable de modifier la qualité de la parole par classement des signaux de paroles
EP0360265A3 (en) * 1988-09-21 1990-09-26 Nec Corporation Communication system capable of improving a speech quality by classifying speech signals
DE4033350B4 (de) * 1989-10-20 2004-04-08 Canon K.K. Verfahren und Vorrichtung für die Sprachverarbeitung
EP0676744A1 (fr) * 1994-04-04 1995-10-11 Digital Voice Systems, Inc. Evaluation des paramètres d'excitation
CN1113333C (zh) * 1994-04-04 2003-07-02 数字语音***公司 激励参数判定方法及其语言编码***

Also Published As

Publication number Publication date
DE3266042D1 (en) 1985-10-10
EP0076234B1 (fr) 1985-09-04
US4618982A (en) 1986-10-21
CA1184656A (fr) 1985-03-26
JPS5870300A (ja) 1983-04-26
ATE15415T1 (de) 1985-09-15

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