EP0616315A1 - Vorrichtung zur digitalen Sprachkodierung und -dekodierung, Verfahren zum Durchsuchen eines pseudologarithmischen LTP-Verzögerungskodebuchs und Verfahren zur LTP-Analyse - Google Patents

Vorrichtung zur digitalen Sprachkodierung und -dekodierung, Verfahren zum Durchsuchen eines pseudologarithmischen LTP-Verzögerungskodebuchs und Verfahren zur LTP-Analyse Download PDF

Info

Publication number
EP0616315A1
EP0616315A1 EP94400525A EP94400525A EP0616315A1 EP 0616315 A1 EP0616315 A1 EP 0616315A1 EP 94400525 A EP94400525 A EP 94400525A EP 94400525 A EP94400525 A EP 94400525A EP 0616315 A1 EP0616315 A1 EP 0616315A1
Authority
EP
European Patent Office
Prior art keywords
delays
dictionary
ltp
segment
module
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.)
Withdrawn
Application number
EP94400525A
Other languages
English (en)
French (fr)
Inventor
Dominique Massaloux
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.)
Orange SA
Original Assignee
France Telecom SA
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by France Telecom SA filed Critical France Telecom SA
Publication of EP0616315A1 publication Critical patent/EP0616315A1/de
Withdrawn legal-status Critical Current

Links

Images

Classifications

    • 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
    • G10L19/04—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
    • 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
    • G10L2019/0001—Codebooks
    • G10L2019/0007—Codebook element generation
    • G10L2019/0008—Algebraic codebooks
    • 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
    • G10L2019/0001—Codebooks
    • G10L2019/0011—Long term prediction filters, i.e. pitch estimation

Definitions

  • the present invention relates to a device for digital coding and decoding of speech, a method of exploring a pseudo-logarithmic dictionary of LTP delays, and a method of LTP analysis.
  • a digital speech coding device consists, after sampling the analog signal, in performing the compression of the binary data of the digitized speech signal.
  • the decoding device performs the opposite operation and reproduces an analog signal different from the original signal, but as close as possible from a perceptual point of view.
  • a digital coding-decoding device for speech is characterized by the digital bit rate of the data to be transmitted between the coder and the decoder, the quality of the signal reconstituted at the decoder, and the complexity of the compression technique used.
  • Predictive coders are used for fairly low bit rates (4 to 16 kbit / s for a sampling frequency of 8 kHz) and good coding quality.
  • a predictive coder is composed of a short-term prediction module, a long-term prediction module, then a module performing the coding of the residual wave using a method of analysis by synthesis, as described in the article by P. Kroon and BS Atal entitled “Predictive Coding of Speech Using Analysis by Synthesis Techniques” (Advances in Speech Signal Processing, Ed. Furui S., Sondhi MM, pages 141-164, 1991 ).
  • This type of coding device is widely used, mainly in terrestrial or satellite transmission systems, or in storage applications.
  • long-term prediction module or LTP module Various embodiments of the long-term prediction module or LTP module, known to those skilled in the art, will now be reviewed.
  • the parameters p and ⁇ are determined by minimizing the energy of an error signal e (n) on a block of N samples of the signal x (n): x (n) represents the input signal itself s (n) or the LPC residue r (n).
  • x (n) represents the input signal itself s (n) or the LPC residue r (n).
  • This type of analysis can advantageously be replaced by a closed-loop analysis, anticipating the operation performed at the decoder to produce the synthesis signal (n).
  • the signal t (n) ("target") is expressed from the LPC residue r (n) and the signal p (n) obtained by prolonging the past excitation (n) by null samples:
  • closed loop analyzes use the signal (n) which is known, at the start of the analyzed block, only for n ⁇ 0, which means that the LTP analysis must be limited to the values ⁇ ⁇ N. This restriction decreases the effectiveness of a long-term predictor on high fundamental frequency voices (voices of women and children). This can be remedied by extrapolating the signal (n) for n ⁇ 0.
  • the subject of the invention is a digital speech coding and decoding device in which the operation of the long-term prediction module as defined in these various documents of the prior art is improved.
  • the invention proposes a device for digital coding and decoding of speech comprising, when coding: an LPC analysis module (short-term prediction), an LTP analysis module (long-term prediction) , a residual wave coding module using a synthesis analysis method; during decoding: a residual wave decoding module, an LTP synthesis module and an LPC synthesis module; characterized in that the LTP analysis module uses a delay dictionary with a pseudo-logarithmic structure in which the delays are arranged in ascending order; this dictionary being made up of Q adjacent segments, each one of a given resolution, the resolutions of the successive segments decreasing geometrically in a rational ratio k such that k> 1 while the number of elements L of each segment remains constant.
  • the interest of these nested precisions is to keep the relative precision over the delay almost constant, and hence the error in the periodicity of the signal due to sampling.
  • the invention also makes it possible to obtain a simple and efficient coding of the delay.
  • ⁇ i the last delay of the segment S i
  • ⁇ i the first delay of the segment S i
  • the size of the segments L is a multiple of K i L - 1 , the choice for ⁇ (0) of L / K l L-1 or of a sub-multiple of L / K l L -1, introducing a spacing regular delays explored in the first pass.
  • the present invention thus makes it possible to define a structure over the set of delays explored in the long-term prediction module, the set of delays thus structured being called in the invention "pseudo-logarithmic dictionary of LTP delays". It is known that maintaining great precision in LTP delays, when these delays increase, is useless from a perceptual point of view.
  • the pseudo-logarithmic dictionary of the invention exploits this idea and makes it possible to maintain the performance of uniform dictionaries for a lower bit rate: for example, it has been observed that the performance of the dictionary D, composed of 256 elements, was similar to those of all 960 delays obtained by uniformly sampling the same range of delays with a precision of 1/8, which represents a gain of more than 20% in throughput.
  • the pseudo-logarithmic structure in addition to organizing the previously stated concept, also makes it possible to establish a simple correspondence between the index of each delay of the pseudo-logarithmic dictionary and its value, facilitating the coding and decoding operations of the delay. No storage is necessary to find the delays in the dictionary.
  • the processing of the LTP module using the technique proposed in the invention is three times faster than that of the module using an optimized version of the reference technique.
  • This optimized version makes maximum use of the methods making it possible to reduce the complexity of the reference technique: if we compare the calculation times of the non-optimized version of the reference technique with those of the proposed technique, we obtain a higher gain at 11.
  • the present invention relates to a digital coding device for speech of the predictive coder type using a short-term prediction of the signal allowing the modeling of the formants, a long-term prediction intended to restore the fine structure of the spectrum, then a coding of the residual wave using a synthetic analysis method; a general description of this type of coder being provided in the article by Messrs Kroon and Atal cited above.
  • Short-term and long-term predictors are calculated by linear prediction methods known as LPC analysis ("Linear Prediction Coding") and LTP analysis (“Long Term Prediction”)
  • This coding device operates as follows:
  • the analog signal after conversion to digital, is segmented into frames of N o samples s (n). These samples are analyzed in the LPC module 13 by a conventional method of linear prediction. This module 13 produces the output of the PLPC parameters transmitted to the decoder and N o residual signal samples r (n).
  • the LTP module 15 accepts as input N samples of a signal x (n) which can come from a sub-segmentation of the signal s (n) itself or else from r (n).
  • a signal x (n) which can come from a sub-segmentation of the signal s (n) itself or else from r (n).
  • the LTP module can optionally also use the PLPC parameters (adaptive dictionary, perceptual filter). This module 15 produces the PLTP output parameters (quantized gain ⁇ and delay index i d ) and develops a long-term prediction signal p (n).
  • the residue coding module 14 performs the coding of the residual excitation.
  • the coding parameters of this excitation are transmitted to the decoder.
  • this module 14 includes a local decoder allowing the calculation of the synthesis (or reconstructed residual) excitation (n).
  • the residue decoding module 21 decodes the parameters P CODRES and calculates N samples of a signal u (n). This signal enters the module 22 together with the P LTP parameters which will be decoded there. After filtering u (n) by 1 / P (z), we obtain (n).
  • This signal then enters the module 23 which performs the decoding of the P LPC parameters and the filtering of (n) by 1 / A (z).
  • This module 23 outputs the N o samples of the synthesis signal (n), for a frame, which are converted to analog.
  • the present invention is located at the LTP module, the operation of which will now be described.
  • the LTP analysis module of the invention is based on the exploration of a dictionary of pseudo-logarithmic type delays.
  • An LTP analysis module of order 1 calculates the delay of the predictor P (z) which minimizes a certain error criterion.
  • the present invention combines all the time periods explored into a dictionary having a pseudo-logarithmic structure. These delays ⁇ are rational numbers, arranged in ascending order in the dictionary.
  • Each segment Si corresponds to a resolution R i , and if we call ⁇ i the last delay of the segment Si, the segment Si is formed as follows, as shown in FIGS. 3A and 3B:
  • the delay ⁇ i can possibly be fractional but the delays ⁇ j , must verify ⁇ j , R i integer i , j ' therefore for each segment S i , it is necessary and sufficient that ⁇ i .R i is integer.
  • D dictionary with 256 delays (8 bits) such as:
  • the signal x "(n) resulting from this filtering is then sub-sampled by a factor q, in a sub-sampler 32 to give y (n).
  • H (z) a windowed cardinal sine sampled by a factor Max (p, q).
  • finding the optimal delay means minimizing a criterion:
  • the second pass uses the complete criterion E '( ⁇ ) and must also be performed on all the segments: even for the segments i ⁇ i L tq ⁇ (i) ⁇ L, because E' ( ⁇ ) must be evaluated on the local extrema of N ( ⁇ ) selected in the first pass.
  • LTP analysis by adaptive dictionary very efficient, is also very complex, due to the presence of the closed loop on the one hand, and the perceptual filter on the other hand.
  • a variant of this analysis, reducing the intrinsic complexity of the process without degrading the subjective performance is proposed here: it is based on a modification of the expression (3) of the error signal whose energy is minimized (criterion E ( ⁇ ) to minimize).
  • the commutability of the linear filters is used and the interpolation filter is applied to the pre-filtered samples w (n) (this is however not applicable to samples using an extrapolated signal (n)).
  • the LTP module thus designed is integrated by way of example into the coding device presented above.
  • ETWO, ETW1, ETW2, ETW3 represented in FIGS. 8A, 8B, 8C and 8D, we have:
  • the search is carried out in two passes according to the principle described above.
  • the first pass, performed only on the digitizers N ( ⁇ 0 ), is very fast because it does not involve any interpolation operation.
  • the LTP module given here by way of example is integrated into the device presented above as a particularly advantageous embodiment of the present invention.
  • H (z) is an FIR (finite impulse response filter) of length 33.
  • P2S modules i , i 0 to 3 referenced respectively 50, 51, 52 and 53.
  • the P2S modules i , i 0 to 3 referenced respectively 50, 51, 52 and 53.
  • the outputs of the corresponding P1 Si modules in addition to the signals resw (n), w (n) and é ( n), we find the outputs of the corresponding P1 Si modules.
  • Each P2Si module maximizes criterion E (A) and outputs the delay A associated with the maximum criterion.
  • the delay value A from the second pass is the delay selected by the search module in the dictionary D.

Landscapes

  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
EP94400525A 1993-03-12 1994-03-10 Vorrichtung zur digitalen Sprachkodierung und -dekodierung, Verfahren zum Durchsuchen eines pseudologarithmischen LTP-Verzögerungskodebuchs und Verfahren zur LTP-Analyse Withdrawn EP0616315A1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
FR9302881A FR2702590B1 (fr) 1993-03-12 1993-03-12 Dispositif de codage et de décodage numériques de la parole, procédé d'exploration d'un dictionnaire pseudo-logarithmique de délais LTP, et procédé d'analyse LTP.
FR9302881 1993-03-12

Publications (1)

Publication Number Publication Date
EP0616315A1 true EP0616315A1 (de) 1994-09-21

Family

ID=9444907

Family Applications (1)

Application Number Title Priority Date Filing Date
EP94400525A Withdrawn EP0616315A1 (de) 1993-03-12 1994-03-10 Vorrichtung zur digitalen Sprachkodierung und -dekodierung, Verfahren zum Durchsuchen eines pseudologarithmischen LTP-Verzögerungskodebuchs und Verfahren zur LTP-Analyse

Country Status (3)

Country Link
US (1) US5704002A (de)
EP (1) EP0616315A1 (de)
FR (1) FR2702590B1 (de)

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5899966A (en) * 1995-10-26 1999-05-04 Sony Corporation Speech decoding method and apparatus to control the reproduction speed by changing the number of transform coefficients
EP1164578A2 (de) * 1995-10-26 2001-12-19 Sony Corporation Verfahren und Vorrichtung zur Sprachkodierung und -dekodierung

Families Citing this family (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2729246A1 (fr) * 1995-01-06 1996-07-12 Matra Communication Procede de codage de parole a analyse par synthese
EP0788091A3 (de) * 1996-01-31 1999-02-24 Kabushiki Kaisha Toshiba Verfahren und Vorrichtung zur Sprachkodierung und -dekodierung
US6219641B1 (en) * 1997-12-09 2001-04-17 Michael V. Socaciu System and method of transmitting speech at low line rates
US6104994A (en) * 1998-01-13 2000-08-15 Conexant Systems, Inc. Method for speech coding under background noise conditions
JP2001109489A (ja) * 1999-08-03 2001-04-20 Canon Inc 音声情報処理方法、装置および記憶媒体
US6760698B2 (en) * 2000-09-15 2004-07-06 Mindspeed Technologies Inc. System for coding speech information using an adaptive codebook with enhanced variable resolution scheme
CN112863539B (zh) * 2019-11-28 2024-04-16 科大讯飞股份有限公司 一种高采样率语音波形生成方法、装置、设备及存储介质

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1991003790A1 (en) * 1989-09-01 1991-03-21 Motorola, Inc. Digital speech coder having improved sub-sample resolution long-term predictor
EP0443548A2 (de) * 1990-02-22 1991-08-28 Nec Corporation Sprachcodierer
EP0523979A2 (de) * 1991-07-19 1993-01-20 Motorola, Inc. Methode und Mittel zur Sprachcodierung mit niedriger Bitrate

Family Cites Families (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS61134000A (ja) * 1984-12-05 1986-06-21 株式会社日立製作所 音声分析合成方式
JP2903533B2 (ja) * 1989-03-22 1999-06-07 日本電気株式会社 音声符号化方式
GB2235354A (en) * 1989-08-16 1991-02-27 Philips Electronic Associated Speech coding/encoding using celp
US5371853A (en) * 1991-10-28 1994-12-06 University Of Maryland At College Park Method and system for CELP speech coding and codebook for use therewith

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1991003790A1 (en) * 1989-09-01 1991-03-21 Motorola, Inc. Digital speech coder having improved sub-sample resolution long-term predictor
EP0443548A2 (de) * 1990-02-22 1991-08-28 Nec Corporation Sprachcodierer
EP0523979A2 (de) * 1991-07-19 1993-01-20 Motorola, Inc. Methode und Mittel zur Sprachcodierung mit niedriger Bitrate

Non-Patent Citations (7)

* Cited by examiner, † Cited by third party
Title
KEMP ET AL.: "Multi-frame coding of LPC parameters at 600-800 bps", INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING 91, vol. 1, 14 May 1991 (1991-05-14), TORONTO, ONTARIO, CA, pages 609 - 612, XP000245302, DOI: doi:10.1109/ICASSP.1991.150413 *
KLEIJN ET AL.: "Fast methods for the CELP speech coding algorithm", IEEE TRANSACTIONS ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, vol. 38, no. 8, August 1990 (1990-08-01), NEW YORK, US, pages 1330 - 1342, XP000148377, DOI: doi:10.1109/29.57568 *
KROON ET AL.: "Pitch predictors with high temporal resolution", INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING 90, vol. 2, 3 April 1990 (1990-04-03), ALBUQUERQUE, NM, US, pages 661 - 664 *
MARQUES ET AL.: "Pitch prediction with fractional delays in CELP coding", EUROPEAN CONFERENCE ON SPEECH COMMUNICATION AND TECHNOLOGY 89, vol. 2, 26 September 1989 (1989-09-26), PARIS, FR, pages 509 - 510 *
OZAWA: "A hybrid speech coding based on multi-pulse and CELP at 3.2kb/2", INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING 90, vol. 2, 3 April 1990 (1990-04-03), ALBUQUERQUE, NM, US, pages 677 - 680, XP000146860 *
PICARD ET AL.: "PCM code converter for two speech channels", NACHRICHTENTECHNISCHE ZEITSCHRIFT, vol. 33, no. 3, March 1980 (1980-03-01), DE, pages 150 - 155 *
REININGER ET AL.: "Prädiktive Sprachkodierung mit stochastischer Anregung", AEU ARCHIV FÜR ELEKTRONIK UND ÜBERTRAGUNGSTECHNIK, vol. 43, no. 5, September 1989 (1989-09-01), STUTTGART, DE, pages 307 - 312, XP000067790 *

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5899966A (en) * 1995-10-26 1999-05-04 Sony Corporation Speech decoding method and apparatus to control the reproduction speed by changing the number of transform coefficients
EP1164578A2 (de) * 1995-10-26 2001-12-19 Sony Corporation Verfahren und Vorrichtung zur Sprachkodierung und -dekodierung
EP1164578A3 (de) * 1995-10-26 2002-01-02 Sony Corporation Verfahren und Vorrichtung zur Sprachkodierung und -dekodierung

Also Published As

Publication number Publication date
FR2702590A1 (fr) 1994-09-16
US5704002A (en) 1997-12-30
FR2702590B1 (fr) 1995-04-28

Similar Documents

Publication Publication Date Title
EP0782128B1 (de) Verfahren zur Analyse eines Audiofrequenzsignals durch lineare Prädiktion, und Anwendung auf ein Verfahren zur Kodierung und Dekodierung eines Audiofrequenzsignals
EP1692689B1 (de) Optimiertes mehrfach-codierungsverfahren
EP1145228B1 (de) Kodierung periodischer sprache
EP2254110B1 (de) Stereosignalkodiergerät, stereosignaldekodiergerät und verfahren dafür
WO1996021221A1 (fr) Procede de codage de parole a prediction lineaire et excitation par codes algebriques
EP0428445B1 (de) Verfahren und Einrichtung zur Codierung von Prädiktionsfiltern in Vocodern mit sehr niedriger Datenrate
EP0801790A1 (de) Verfahren zur sprachkodierung mittels analyse durch synthese
EP0801788A1 (de) Verfahren zur sprachkodierung mittels analyse durch synthese
EP1836699B1 (de) Verfahren und Vorrichtung zur Ausführung einer optimalizierten Audiokodierung zwischen zwei Langzeitvorhersagemodellen
EP0685833B1 (de) Verfahren zur Sprachkodierung mittels linearer Prädiktion
FR2784218A1 (fr) Procede de codage de la parole a bas debit
EP0616315A1 (de) Vorrichtung zur digitalen Sprachkodierung und -dekodierung, Verfahren zum Durchsuchen eines pseudologarithmischen LTP-Verzögerungskodebuchs und Verfahren zur LTP-Analyse
JP2779325B2 (ja) ボコーダーにおける前処理の相関関係式を用いたピッチ検索時間短縮方法
US5812966A (en) Pitch searching time reducing method for code excited linear prediction vocoder using line spectral pair
WO2023165946A1 (fr) Codage et décodage optimisé d'un signal audio utilisant un auto-encodeur à base de réseau de neurones
JPH09508479A (ja) バースト励起線形予測
FR2783651A1 (fr) Dispositif et procede de filtrage d'un signal de parole, recepteur et systeme de communications telephonique
EP1192619B1 (de) Audio-kodierung, dekodierung zur interpolation
EP0796490B1 (de) Verfahren und vorrichtung zur signalprädiktion für einen sprachkodierer
EP1192618B1 (de) Audiokodierung mit adaptiver lifterung
EP1192621B1 (de) Audiokodierung mit harmonischen komponenten
EP1194923B1 (de) Verfahren und system für audio analyse und synthese
WO2002029786A1 (fr) Procede et dispositif de codage segmental d'un signal audio
EP0454552A2 (de) Verfahren und Einrichtung zur Sprachcodierung mit niedriger Bitrate
EP1192620A1 (de) Audiosignalkodierer und -dekodierer einschliesslich nicht-harmonischen komponenten

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): DE GB

17P Request for examination filed

Effective date: 19950223

17Q First examination report despatched

Effective date: 19980701

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: THE APPLICATION IS DEEMED TO BE WITHDRAWN

18D Application deemed to be withdrawn

Effective date: 19990720