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EP1383113A1 - Verfahren und Vorrichtung für Breitbandsprachkodierung geeignet zur Kontrolle von Kurzzeit- und Langzeitverzerrungen - Google Patents

Verfahren und Vorrichtung für Breitbandsprachkodierung geeignet zur Kontrolle von Kurzzeit- und Langzeitverzerrungen Download PDF

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Publication number
EP1383113A1
EP1383113A1 EP02015919A EP02015919A EP1383113A1 EP 1383113 A1 EP1383113 A1 EP 1383113A1 EP 02015919 A EP02015919 A EP 02015919A EP 02015919 A EP02015919 A EP 02015919A EP 1383113 A1 EP1383113 A1 EP 1383113A1
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EP
European Patent Office
Prior art keywords
weighting filter
term
speech
filter
formant
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EP02015919A
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English (en)
French (fr)
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désignation de l'inventeur n'a pas encore été déposée La
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STMicroelectronics NV
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STMicroelectronics NV
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Application filed by STMicroelectronics NV filed Critical STMicroelectronics NV
Priority to EP02015919A priority Critical patent/EP1383113A1/de
Priority to EP03291749A priority patent/EP1388846A3/de
Priority to US10/622,019 priority patent/US20040073421A1/en
Publication of EP1383113A1 publication Critical patent/EP1383113A1/de
Withdrawn legal-status Critical Current

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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12Determination 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
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters

Definitions

  • the invention relates to speech encoding / decoding extended band, in particular but not limited to telephony mobile.
  • the bandwidth of the speech signal is between 50 and 7000 Hz.
  • Successive speech sequences sampled at one predetermined sampling frequency are processed in a coding device using a prediction linear excitation by coded sequences (ACELP: “algebraic-code-excited linear-prediction ”), well known to those skilled in the art, and described in particular in recommendation ITU-TG 729, version 3/96, titled “speech coding at 8 kbit / s by prediction linear with excitation by coded sequences with algebraic structure conjugate ”.
  • ACELP “algebraic-code-excited linear-prediction ”
  • the prediction coder CD of the ACELP type, is based on the linear predictive coding model with code excitation.
  • the coder operates on vocal superframes equivalent for example to 20 ms of signal and each comprising 320 samples.
  • the extraction of the linear prediction parameters, ie the coefficients of the linear prediction filter also called short-term synthesis filter 1 / A (z), is carried out for each speech superframe.
  • each superframe is subdivided into 5 ms frames comprising 80 samples.
  • the speech signal is analyzed to extract the parameters of the CELP prediction model (that is to say, in particular, a long-term digital excitation word v i extracted from an adaptive coded DLT directory, also called “adaptive long term dictionary", an associated long term gain Ga, a short term excitation word c j , extracted from a DCT algebraic coded repertoire, also called “fixed coded repertoire” or “short term dictionary algebraic ", and an associated short-term gain Gc).
  • a long-term digital excitation word v i extracted from an adaptive coded DLT directory, also called “adaptive long term dictionary", an associated long term gain Ga
  • a short term excitation word c j extracted from a DCT algebraic coded repertoire, also called “fixed coded repertoire” or “short term dictionary algebraic ", and an associated short-term gain Gc).
  • these parameters are used, in a decoder, to retrieve the excitation and predictive filter parameters. We then reconstitutes speech by filtering this excitation flow in a short-term synthesis filter.
  • the short-term dictionary DCT is founded on an algebraic structure using a permutation model intertwined with Dirac pulses.
  • this coded directory which contains innovative excitations, also called excitations algebraic or short-term, each vector contains a certain number of non-zero pulses, for example four, each of which can have amplitude +1 or -1 with predetermined positions.
  • the CD encoder processing means include functionally of the first MEXT1 extraction means intended to extract the word long-term excitement, and second MEXT2 extraction means intended to extract the word short-term excitement. Functionally, these means are made for example in software within a processor.
  • These extraction means include a predictive filter FP having a transfer function equal to 1 / A (z), as well as a filter FPP perceptual weighting with a transfer function W (z).
  • the perceptual weighting filter is applied to the signal to model the perception of the ear.
  • the extraction means include means MECM intended to perform a minimization of a square error average.
  • the linear prediction FP synthesis filter models the spectral envelope of the signal. Linear predictive analysis is performed all superframes, so as to determine the linear predictive filter coefficients. These are converted to spectral line pairs (LSP: “Line Spectrum Pairs”) and digitized by predictive vector quantization in two stages.
  • LSP Line Spectrum Pairs
  • Each 20 ms speech superframe is divided into four frames of 5 ms each containing 80 samples.
  • the settings Quantized LSPs are transmitted to the decoder once per superframe while long term and short term parameters are passed at each frame.
  • the coefficients of the linear prediction filter, quantified and not quantified, are used for the most recent frame of a super-frame, while the other three frames of the same super-frame use an interpolation of these coefficients.
  • Tonal delay open loop is estimated every two frames based on the perceptually weighted voice signal. Then, the following operations are repeated at each frame:
  • the long-term target signal X LT is calculated by filtering the sampled speech signal s (n) by the perceptual weighting filter FPP.
  • the impulse response of the weighted synthesis filter is calculated.
  • a closed loop tonal analysis using a minimization of the mean square error is then carried out in order to determine the long-term excitation word v i and the associated gain Ga, by means of the target signal and the impulse response, by searches around the value of the tone delay in open loop.
  • the long-term target signal is then updated by subtracting the filtered contribution y from the adaptive coded directory DLT and this new short-term target signal X ST is used when exploring the fixed coded directory DCT in order to determine the password.
  • short term excitation c j and the associated gain G c is used when exploring the fixed coded directory DCT in order to determine the password.
  • the object of the invention is to independently control the short-term and long-term distortions.
  • the invention therefore provides a speech encoding method with wide band, in which the speech is sampled so as to obtain successive voice frames each comprising a predetermined number of samples, and for each voice frame, we determines parameters of a linear prediction model at excitation by code, these parameters comprising a numeric word of long-term excitement extracted from an adaptive coded repertoire, as well that a word of short-term excitement extracted from a coded repertoire associated algebraic.
  • long term excitation word extraction using a prime perceptual weighting filter comprising a first filter formantic weighting
  • the denominator of the transfer function of the first formantic weighting filter is equal to the numerator of the second formantic weighting filter.
  • the use of two filters weighting different formant allows to control regardless of short-term and long-term distortions.
  • the short-term weighting filter is cascaded to the filter of long-term weighting.
  • tying the denominator of the long-term weighting filter in the numerator of the short-term weighting allows these two to be controlled separately filters and also allows a clear simplification when these two filters are cascaded.
  • the first extraction means include a first filter perceptual weighting including a first weighting filter formantic, by the fact that the second means of extraction include the first perceptual weighting filter and a second perceptual weighting filter including a second formantic weighting filter, and the denominator of the function of transfer of the first formantic weighting filter is equal to numerator of the second formantic weighting filter.
  • the invention also relates to a terminal of a system wireless communication, such as a mobile phone cell, incorporating a device as defined above.
  • the FPP perceptual weighting filter uses the masking properties of the human ear compared to the spectral envelope of the speech signal, whose shape is a function resonances of the vocal tract. This filter allows you to assign more importance of the error appearing in the spectral valleys by compared to formic peaks.
  • W (z) AT ( z / ⁇ 1 ) AT ( z / ⁇ 2 ) in which 1 / A (z) is the transfer function of the predictive filter FP and ⁇ 1 and ⁇ 2 are the perceptual weighting coefficients, the two coefficients being positive or zero and less than or equal to 1 with the coefficient ⁇ 2 less than or equal to the coefficient ⁇ 1.
  • the perceptual weighting filter consists of a formantic weighting filter and a weighting of the slope of the spectral envelope of the signal (tilt).
  • FIG. 2 Such an embodiment according to the invention is illustrated in the Figure 2, in which, compared to Figure 1, the FPP single filter has been replaced by a first formantic weighting filter FPP1 for long-term research, cascaded with a second FPP2 formantic weighting filter for short search term.
  • FPP1 formantic weighting filter
  • FPP2 formantic weighting filter
  • the filters appearing in the long-term research loop should also appear in the short-term research loop.
  • the transfer function W 1 (z) of the formantic weighting filter FPP1 is given by formula (II) below.
  • W 1 ( z ) AT ( z / ⁇ 11 ) AT ( z / ⁇ 12 ) while the transfer function W 2 (z) of the formantic weighting filter FPP2 is given by formula (III) below.
  • W 2 ( z ) AT ( z / ⁇ 21 ) AT ( z / ⁇ 22 )
  • the coefficient ⁇ 12 is equal to the coefficient ⁇ 21 . This allows a clear simplification when cascading these two filters.
  • the filter equivalent to the cascade of these two filters has a transfer function given by formula (IV) below.
  • the synthesis filter FP (having the transfer function 1 / A (z)) followed by the long-term weighting filter FPP1 and the weighting filter FPP2 is then equivalent to the filter whose transfer function is given by formula (V) below. 1 AT ( z / ⁇ 22 )
  • the invention advantageously applies to telephony mobile, and in particular to all remote terminals belonging to a wireless communication system.
  • Such a terminal for example a TP mobile telephone, such as that illustrated in FIG. 3, conventionally comprises a antenna connected via a DUP duplexer to a chain reception CHR and a CHT transmission chain.
  • a baseband processor BB is connected to the chain respectively of reception CHR and to the chain of transmission CHT by via analog digital ADCs and analog digital DACs.
  • the processor BB performs processing in baseband, including DCN channel decoding, followed by DCS source decoding.
  • the processor For transmission, the processor performs source coding CCS followed by CCN channel coding.
  • the mobile phone incorporates an encoder according to the invention, it is incorporated within the coding means of CCS source, while the decoder is incorporated within the means DCS source decoding.

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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)
EP02015919A 2002-07-17 2002-07-17 Verfahren und Vorrichtung für Breitbandsprachkodierung geeignet zur Kontrolle von Kurzzeit- und Langzeitverzerrungen Withdrawn EP1383113A1 (de)

Priority Applications (3)

Application Number Priority Date Filing Date Title
EP02015919A EP1383113A1 (de) 2002-07-17 2002-07-17 Verfahren und Vorrichtung für Breitbandsprachkodierung geeignet zur Kontrolle von Kurzzeit- und Langzeitverzerrungen
EP03291749A EP1388846A3 (de) 2002-07-17 2003-07-15 Verfahren und Vorrichtung zur Breitbandkodierung von Sprachsignalen geeignet zurunabhängigen Steuerung lang- und kurzzeitiger Verzerrungen
US10/622,019 US20040073421A1 (en) 2002-07-17 2003-07-17 Method and device for encoding wideband speech capable of independently controlling the short-term and long-term distortions

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
EP02015919A EP1383113A1 (de) 2002-07-17 2002-07-17 Verfahren und Vorrichtung für Breitbandsprachkodierung geeignet zur Kontrolle von Kurzzeit- und Langzeitverzerrungen

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EP1383113A1 true EP1383113A1 (de) 2004-01-21

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EP02015919A Withdrawn EP1383113A1 (de) 2002-07-17 2002-07-17 Verfahren und Vorrichtung für Breitbandsprachkodierung geeignet zur Kontrolle von Kurzzeit- und Langzeitverzerrungen

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Publication number Priority date Publication date Assignee Title
CN105976830B (zh) 2013-01-11 2019-09-20 华为技术有限公司 音频信号编码和解码方法、音频信号编码和解码装置

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5926785A (en) * 1996-08-16 1999-07-20 Kabushiki Kaisha Toshiba Speech encoding method and apparatus including a codebook storing a plurality of code vectors for encoding a speech signal
US6173257B1 (en) * 1998-08-24 2001-01-09 Conexant Systems, Inc Completed fixed codebook for speech encoder

Family Cites Families (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6073092A (en) * 1997-06-26 2000-06-06 Telogy Networks, Inc. Method for speech coding based on a code excited linear prediction (CELP) model
EP0932141B1 (de) * 1998-01-22 2005-08-24 Deutsche Telekom AG Verfahren zur signalgesteuerten Schaltung zwischen verschiedenen Audiokodierungssystemen
US7072832B1 (en) * 1998-08-24 2006-07-04 Mindspeed Technologies, Inc. System for speech encoding having an adaptive encoding arrangement
US6330533B2 (en) * 1998-08-24 2001-12-11 Conexant Systems, Inc. Speech encoder adaptively applying pitch preprocessing with warping of target signal

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5926785A (en) * 1996-08-16 1999-07-20 Kabushiki Kaisha Toshiba Speech encoding method and apparatus including a codebook storing a plurality of code vectors for encoding a speech signal
US6173257B1 (en) * 1998-08-24 2001-01-09 Conexant Systems, Inc Completed fixed codebook for speech encoder

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
CHEN J-H ET AL: "Improving the performance of the 16 kb/s LD-CELP speech coder", DIGITAL SIGNAL PROCESSING 2, ESTIMATION, VLSI. SAN FRANCISCO, MAR. 23 - 26, 1992, PROCEEDINGS OF THE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (ICASSP), NEW YORK, IEEE, US, vol. 5 CONF. 17, 23 March 1992 (1992-03-23), pages 69 - 72, XP010058714, ISBN: 0-7803-0532-9 *

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