EP0782128A1 - Verfahren zur Analyse eines Audiofrequenzsignals durch lineare Prädiktion, und Anwendung auf ein Verfahren zur Kodierung und Dekodierung eines Audiofrequenzsignals - Google Patents

Verfahren zur Analyse eines Audiofrequenzsignals durch lineare Prädiktion, und Anwendung auf ein Verfahren zur Kodierung und Dekodierung eines Audiofrequenzsignals Download PDF

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EP0782128A1
EP0782128A1 EP96402715A EP96402715A EP0782128A1 EP 0782128 A1 EP0782128 A1 EP 0782128A1 EP 96402715 A EP96402715 A EP 96402715A EP 96402715 A EP96402715 A EP 96402715A EP 0782128 A1 EP0782128 A1 EP 0782128A1
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signal
stage
coefficients
transfer function
filter
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French (fr)
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EP0782128B1 (de
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Catherine Quinquis
Alain Le Guyader
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Orange SA
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France Telecom SA
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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

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  • the present invention relates to a method for linear prediction analysis of an audio frequency signal.
  • This process finds a particular, but not exclusive, application in predictive audio coders, in particular in coders with analysis by synthesis, of which the most widespread type is the coder CELP ("Code-Excited Linear Prediction").
  • the predictive coders used for bit rate compression perform a modeling of the spectral envelope of the signal. This modeling results from an analysis by linear prediction of order M (M ⁇ 10 typically in narrow band), consisting in determining M coefficients a i of linear prediction of the input signal. These coefficients characterize a synthesis filter used at the decoder, whose transfer function is of the form 1 / A (z) with
  • Linear prediction analysis has a broader field of application than that of speech coding.
  • the order M of the prediction constitutes one of the variables that the analysis by linear prediction aims to obtain, this variable being influenced by the number of peaks present in the spectrum of the analyzed signal (see US-A- 5,142,581).
  • the filter calculated by the linear prediction analysis can have various structures, leading to different choices of parameters for the representation of the coefficients (the coefficients a i themselves, the parameters LAR, LSF, LSP, the reflection coefficients or PARCOR. ..).
  • DSP digital signal processors
  • it was common to use recursive structures for the calculated filter for example structures using PARCOR coefficients of the type described in the article by F. ITAKURA and S. SAITO "Digital Filtering Techniques for Speech Analysis and Synthesis", Proc. of the 7th International Congress on Acoustics, Budapest 1971, pages 261-264 (see FR-A-2 284 946 or US-A-3 975 587).
  • the coefficients a i are also used to construct a perceptual weighting filter used by the coder to determine the excitation signal to be applied to the short-term synthesis filter to obtain a synthetic signal representative of the speech signal.
  • This perceptual weighting accentuates the portions of the spectrum where the coding errors are the most perceptible, that is to say the interformant areas.
  • the linear prediction coefficients a i are also used to define a post-filter used to attenuate the frequency zones between the formants and the harmonics of the speech signal, without modifying the slope of the spectrum of the signal.
  • Modeling the spectral envelope of the signal by the coefficients a i therefore constitutes an essential element of the coding and decoding process, in the sense that it must represent the spectral content of the signal to be reconstructed at the decoder and that it also controls masking quantization noise as well as post-filtering at the decoder.
  • An object of the present invention is to improve the modeling of the spectrum of an audiofrequency signal in a system using a method of analysis by linear prediction. Another aim is to make the performance of such a system more homogeneous for different input signals (speech, music, sinusoids, DTMF signals %), different bandwidths (telephone band, extended band, stereo band %), different registration conditions (directive microphone, acoustic antenna %) and filtering.
  • the invention thus proposes a method of analysis by linear prediction of an audiofrequency signal, to determine spectral parameters dependent on a short-term spectrum of the audiofrequency signal, comprising q successive prediction stages, q being an integer greater than 1
  • parameters representing a predefined number M p of coefficients a 1 p , ... a M p p of linear prediction of an input signal of said stage are determined, the analyzed audio signal constituting the input signal of the first stage, and the input signal of a stage p + 1 being constituted by the input signal of the stage p filtered by a transfer function filter
  • the number Mp of linear prediction coefficients can notably increase from one stage to the next.
  • the first stage will be able to give a fairly faithful account of the general slope of the spectrum or of the signal, while the following stages will refine the representation of the signal formants.
  • the transfer function A (z) thus obtained can also be used to define according to formula (2) the transfer function of the perceptual weighting filter when the coder is a coder for analysis by synthesis with closed loop determination of the signal of excitation.
  • Another interesting possibility is to adopt coefficients of spectral expansion ⁇ 1 and ⁇ 2 which can vary from one stage to the next, that is to say to give the perceptual weighting filter a function of transfer of the form.
  • ⁇ 1 p , ⁇ 2 p denote pairs of spectral expansion coefficients such that 0 ⁇ 2 p ⁇ 1 p ⁇ 1 for 1 ⁇ p ⁇ q.
  • This transfer function A (z) can also be used to define a post-filter whose transfer function comprises, as in formula (3) above, a term of the form A (z / ⁇ 1 ) / A (z / ⁇ 2 ), where ⁇ 1 and ⁇ 2 denote coefficients such as 0 ⁇ 1 ⁇ 2 ⁇ 1.
  • the method of analysis by linear prediction with multiple stages proposed according to the invention comprises many other applications in the processing of audio signals, for example in transform predictive coders, in speech recognition systems, in speech enhancement systems ...
  • the audiofrequency signal to be analyzed in the method illustrated in FIG. 1 is denoted s 0 (n). It is assumed to be available in the form of digital samples, the integer n denoting the successive sampling instants.
  • the linear prediction analysis method comprises q successive stages 5 1 , ..., 5 p , ..., 5 q . At each prediction stage 5 p (1 p p q q), a linear order Mp prediction of an input signal s p-1 (n) is carried out.
  • the input signal of the first stage 5 1 is constituted by the audio frequency signal to be analyzed s 0 (n), while the input signal of a stage 5 p + 1 (1 ⁇ p ⁇ q) is constituted by the signal s p (n), obtained in a step denoted 6 p by applying to the input signal s p-1 (n) of the p-th stage 5 p a filtering by means of a transfer function filter where the coefficients a i p (1 ⁇ i ⁇ Mp) are the coefficients of linear prediction obtained on the floor 5 p .
  • the quantity E (Mp) is the energy of the residual prediction error of stage p.
  • the quantification can relate to the normalized frequencies ⁇ i p or to their cosines.
  • the analysis can be performed at each 5 p prediction stage according to the classic Levinson-Durbin algorithm mentioned above.
  • Other algorithms providing the same results, developed more recently, can be used advantageously, in particular the exploded Levinson algorithm (see “A new Efficient Algorithm to Compute the LSP Parameters for Speech Coding", by S. Saoudi, JM Boucher and A. Le Guyader, Signal Processing, Vol.28, 1992, pages 201-212), or the use of Chebyshev polynomials (see “The Computation of Line Spectrum Frequencies Using Chebyshev Polynomials, by P. Kabal and RP Ramachandran, IEEE Trans. On Acoustics, Speech, and Signal Processing, Vol. ASSP-34, n ° 6, pages 1419-1426, December 1986).
  • the orders Mp of the linear predictions carried out preferably increase from one stage to the following: M1 ⁇ M2 ⁇ ... ⁇ Mq.
  • M1 2 for example
  • M1 2 for example
  • the signal sampling frequency Fe was 16 kHz.
  • the signal spectrum (modulus of its Fourier transform) is represented by curve I. This spectrum is representative of audio frequency signals which have, on average, more energy at low frequencies than at high frequencies. The spectral dynamics are sometimes higher than that of Figure 2 (60 dB).
  • Curves (II) and (III) correspond to the modeled spectral envelopes
  • the invention is described below in its application to a CELP type speech coder.
  • FIG. 3 The speech synthesis process implemented in a CELP coder and decoder is illustrated in FIG. 3.
  • An excitation generator 10 delivers an excitation code c k belonging to a predetermined repertoire in response to an index k.
  • An amplifier 12 multiplies this excitation code by an excitation gain ⁇ , and the resulting signal is subjected to a long-term synthesis filter 14.
  • the output signal u of the filter 14 is in turn subjected to a short-term synthesis filter 16, the output of which constitutes what is considered here as the synthetic speech signal.
  • This synthetic signal is applied to a post-filter 17 intended to improve the subjective quality of the reconstructed speech.
  • Post-filtering techniques are well known in the field of speech coding (see JH Chen and A.
  • the coefficients of the post-filter 17 are obtained from the LPC parameters characterizing the short-term synthesis filter 16. It will be understood that, as in certain current CELP decoders, the post-filter 17 could also include a long-term post-filtering component.
  • the aforementioned signals are digital signals represented for example by words of 16 bits at a sampling rate Fe equal for example to 16 kHz for an encoder in wide band (50-7000 Hz).
  • the synthesis filters 14, 16 are generally purely recursive filters.
  • the delay T and the gain G constitute long-term prediction parameters (LTP) which are determined adaptively by the coder.
  • the LPC parameters defining the short-term synthesis filter 16 are determined at the coder by a method of analysis by linear prediction of the speech signal.
  • the transfer function of the filter 16 is generally of the form 1 / A (z) with A (z) of the form (1).
  • the present invention proposes to adopt a similar form of the transfer function, in which A (z) is broken down according to (7) as indicated above.
  • excitation signal is used here to denote the signal u (n) applied to the short-term synthesis filter 14.
  • This excitation signal comprises an LTP Gu (nT) component and a residual component, or innovation sequence, ⁇ c k (n).
  • the parameters characterizing the residual component and, optionally, the LTP component are evaluated in a closed loop, using a perceptual weighting filter.
  • FIG 4 shows the diagram of a CELP coder.
  • the speech signal s (n) is a digital signal, for example supplied by an analog-digital converter 20 processing the amplified and filtered output signal from a microphone 22.
  • the LPC, LTP and EXC parameters are obtained at the coder by three respective analysis modules 24, 26, 28. These parameters are then quantified in a known manner for transmission effective digital, then subjected to a multiplexer 30 which forms the output signal of the encoder. These parameters are also supplied to a module 32 for calculating the initial states of certain coder filters.
  • This module 32 essentially comprises a decoding chain such as that shown in FIG. 3. Like the decoder, the module 32 operates on the basis of the quantized LPC, LTP and EXC parameters. If an interpolation of the LPC parameters is carried out at the decoder, as is common, the same interpolation is carried out by the module 32.
  • the module 32 makes it possible to know at the level of the coder the previous states of the synthesis filters 14, 16 of the decoder, determined according to the synthesis and excitation parameters prior to the subframe considered.
  • the next step in coding is to determine the LTP parameters for long-term prediction. These are for example determined once per subframe of L samples.
  • the output signal from the subtractor 34 is subjected to a perceptual weighting filter 38 whose role is to accentuate the portions of the spectrum where the errors are most perceptible, that is to say the inter-forming zones.
  • the respective coefficients b i and c i (1 i i M M) of the functions AN (z) and AP (z) are calculated for each frame by a module 39 for evaluating the perceptual weighting which supplies them to the filter 38.
  • AN (z) A (z / ⁇ 1 )
  • AP (z) A (z / ⁇ 2 ) with 0 ⁇ 2 ⁇ 1 ⁇ 1, which comes back to the usual form (2 ) with A (z) of the form (7).
  • the invention however allows, with a very low computational overload, to have greater flexibility as regards the shaping of the quantization noise, by adopting the form (6) for W (z), that is:
  • the closed loop LTP analysis performed by the module 26 consists, in a conventional manner, in selecting for each subframe the delay T which maximizes the normalized correlation: where x '(n) denotes the output signal of the filter 38 during the sub-frame considered, and y T (n) denotes the convolution product u (nT) * h' (n).
  • h '(0), h' (1) ..., h '(L-1) denotes the impulse response of the weighted synthesis filter, with transfer function W (z) / A (z).
  • This impulse response h ′ is obtained by a module 40 for calculating impulse responses, as a function of the coefficients b i and c i provided by the module 39 and of the LPC parameters which have been determined for the sub-frame, if appropriate after quantification. and interpolation.
  • the samples u (nT) are the previous states of the synthesis filter 14 at long term, supplied by module 32.
  • the missing samples u (nT) are obtained by interpolation on the basis of the previous samples, or from the speech signal.
  • the delays T, whole or fractional, are selected in a specific window.
  • the signal Gy T (n) which has been calculated by the module 26 for the optimal delay T, is first subtracted from the signal x '(n) by the subtractor 42.
  • the resulting signal x (n) is subjected to a reverse filter 44 which provides a signal D (n) given by: where h (0), h (1), ..., h (L-1) denotes the impulse response of the filter composed of the synthesis filters and of the perceptual weighting filter, calculated by the module 40.
  • the compound filter has the transfer function W (z) / [A (z) .B (z)] .
  • the vector D constitutes a target vector for the module 28 for searching for the excitation.
  • the CELP decoder comprises a demultiplexer 8 receiving the bit stream from the coder.
  • the quantized values of the excitation parameters EXC and of the synthesis parameters LTP and LPC are supplied to the generator 10, to the amplifier 12 and to the filters 14, 16 to reconstruct the synthetic signal ⁇ , which is subjected to the post-filter 17 then converted to analog by the converter 18 before being amplified and then applied to a loudspeaker 19 to restore the original speech.
  • the LPC parameters are for example constituted by quantization indexes of the reflection coefficients r i p (also called partial correlation coefficients or PARCOR) relating to the different stages of linear prediction.
  • a module 15 recovers the quantized values of the r i p from the quantization indexes, and converts them to provide the q sets of linear prediction coefficients. This conversion is for example carried out by the same recursive method as in the Levinson-Durbin algorithm.
  • the sets of coefficients a i p are supplied to the short-term synthesis filter 16 constituted by a succession of q filters / stages of transfer functions 1 / A 1 (z), ..., 1 / A q (z) given by the relation (4) .
  • the filter 16 could also be in a single stage of transfer function 1 / A (z) given by the relation (1) in which the coefficients a i have been calculated according to the relations (9) to (13).
  • the reflection coefficient r 1 can be that associated with the coefficients a i of the composite synthesis filter, which it is then necessary to calculate.
  • the invention makes it possible to adopt coefficients ⁇ 1 and ⁇ 2 different from one stage to the next (formula (8)), that is:
  • the invention has been described above in its application to a predictive coder with forward adaptation, that is to say in which the audio frequency signal subject to analysis by linear prediction is the signal of encoder input.
  • the invention also applies to predictive coders / decoders with backward adaptation, in which the synthetic signal is the subject of analysis by linear prediction at the coder and at the decoder (see JH Chen et al: "A Low -Delay CELP Coder for the CCITT 16 kbit / s Speech Coding Standard ", IEEE J. SAC, Vol.10, n ° 5, pages 830-848, June 1992).
  • FIGS. 5 and 6 respectively show a CELP decoder and a CELP coder with "backward" adaptation implementing the present invention. Numerical references identical to those of FIGS. 3 and 4 have been used to designate similar elements.
  • the “backward” adaptation decoder receives only the quantization values of the parameters defining the excitation signal u (n) to be applied to the short-term synthesis filter 16.
  • these parameters are the index k and the associated gain ⁇ as well as the LTP parameters.
  • the synthetic signal ⁇ (n) is processed by a module 124 multi-stage linear prediction analysis identical to module 24 of FIG. 3. Module 124 supplies the LPC parameters to filter 16 for one or more subsequent frames of the excitation signal, and to post-filter 17 whose coefficients are obtained as described above.
  • the corresponding coder represented in FIG. 6, performs the analysis by multistage linear prediction on the locally generated synthetic signal and not on the audio signal s (n). It thus comprises a local decoder 132 essentially consisting of the elements denoted 10, 12, 14, 16 and 124 of the decoder of FIG. 5.
  • the local decoder 132 provides the LPC parameters obtained by analysis of the synthetic signal, which are used by the module 39 for evaluating the perceptual weighting and the module 40 for calculating the impulse responses h and h '.
  • the operation of the encoder is identical to that of the encoder described with reference to FIG. 4, except that the LPC analysis module 24 is no longer necessary. Only the EXC and LTP parameters are sent to the decoder.
  • Figures 7 and 8 are block diagrams of a CELP decoder and a CELP coder with mixed adaptation.
  • the linear prediction coefficients of the first stage or stages result from a "forward" analysis of the audio frequency signal carried out by the coder, while the linear prediction coefficients of the first stage or stages result from a "backward” analysis of the synthetic signal carried out by the decoder (and by a local decoder provided in the coder).
  • Numerical references identical to those of FIGS. 3 to 6 have been used to designate similar elements.
  • the mixed decoder illustrated in FIG. 7 receives the quantization values of the parameters EXC, LTP defining the excitation signal u (n) to be applied to the short-term synthesis filter 16, and the quantization values of the determined LPC / F parameters by the "forward" analysis performed by the coder.
  • These LPC / F parameters represent q F sets of linear prediction coefficients a 1 F, p , ..., a MFp F, p for 1 ⁇ p ⁇ q F , and define a first component 1 / A F (z) of the transfer function 1 / A (z) of filter 16:
  • the mixed decoder comprises an inverse filter 200 of transfer function A F (z) which filters the synthetic signal (n) produced by the short-term synthesis filter 16 to produce a filtered synthetic signal 0 (n).
  • the LPC / B coefficients thus obtained are supplied to the synthesis filter 16 to define its second component for the next frame.
  • the local decoder 232 provided in the mixed coder essentially consists of the elements denoted 10, 12, 14, 16, 200 and 224 / B of the decoder of FIG. 7.
  • the local decoder 232 supplies the LPC / B parameters which are used, with the LPC / F parameters supplied by the analysis module 224 / F, by the module 39 for evaluating the perceptual weighting and the module 40 for calculating the impulse responses h and h '.
  • the operation of the mixed encoder is identical to that of the encoder described with reference to FIG. 4. Only the parameters EXC, LTP and LPC / F are sent to the decoder.

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EP96402715A 1995-12-15 1996-12-12 Verfahren zur Analyse eines Audiofrequenzsignals durch lineare Prädiktion, und Anwendung auf ein Verfahren zur Kodierung und Dekodierung eines Audiofrequenzsignals Expired - Lifetime EP0782128B1 (de)

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FR9514925A FR2742568B1 (fr) 1995-12-15 1995-12-15 Procede d'analyse par prediction lineaire d'un signal audiofrequence, et procedes de codage et de decodage d'un signal audiofrequence en comportant application
FR9514925 1995-12-15

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Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0867862A2 (de) * 1997-03-26 1998-09-30 Nec Corporation Vorrichtung zur Kodierung und Dekodierung von Sprach- und Musiksignalen
US8027242B2 (en) 2005-10-21 2011-09-27 Qualcomm Incorporated Signal coding and decoding based on spectral dynamics
US8392176B2 (en) 2006-04-10 2013-03-05 Qualcomm Incorporated Processing of excitation in audio coding and decoding
US8428957B2 (en) 2007-08-24 2013-04-23 Qualcomm Incorporated Spectral noise shaping in audio coding based on spectral dynamics in frequency sub-bands

Families Citing this family (43)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5621852A (en) 1993-12-14 1997-04-15 Interdigital Technology Corporation Efficient codebook structure for code excited linear prediction coding
FR2729247A1 (fr) * 1995-01-06 1996-07-12 Matra Communication Procede de codage de parole a analyse par synthese
FR2729246A1 (fr) * 1995-01-06 1996-07-12 Matra Communication Procede de codage de parole a analyse par synthese
JPH10124088A (ja) * 1996-10-24 1998-05-15 Sony Corp 音声帯域幅拡張装置及び方法
FI973873A (fi) * 1997-10-02 1999-04-03 Nokia Mobile Phones Ltd Puhekoodaus
FR2774827B1 (fr) 1998-02-06 2000-04-14 France Telecom Procede de decodage d'un flux binaire representatif d'un signal audio
US6223157B1 (en) * 1998-05-07 2001-04-24 Dsc Telecom, L.P. Method for direct recognition of encoded speech data
US6148283A (en) * 1998-09-23 2000-11-14 Qualcomm Inc. Method and apparatus using multi-path multi-stage vector quantizer
US6778953B1 (en) * 2000-06-02 2004-08-17 Agere Systems Inc. Method and apparatus for representing masked thresholds in a perceptual audio coder
CN1216368C (zh) * 2000-11-09 2005-08-24 皇家菲利浦电子有限公司 用于扩展语音信号的频率范围的方法和系统
CN1270291C (zh) * 2000-12-06 2006-08-16 皇家菲利浦电子有限公司 滤波设备和方法
WO2002067246A1 (en) * 2001-02-16 2002-08-29 Centre For Signal Processing, Nanyang Technological University Method for determining optimum linear prediction coefficients
US6590972B1 (en) * 2001-03-15 2003-07-08 3Com Corporation DTMF detection based on LPC coefficients
US7062429B2 (en) * 2001-09-07 2006-06-13 Agere Systems Inc. Distortion-based method and apparatus for buffer control in a communication system
US6934677B2 (en) 2001-12-14 2005-08-23 Microsoft Corporation Quantization matrices based on critical band pattern information for digital audio wherein quantization bands differ from critical bands
US7240001B2 (en) * 2001-12-14 2007-07-03 Microsoft Corporation Quality improvement techniques in an audio encoder
US20030216921A1 (en) * 2002-05-16 2003-11-20 Jianghua Bao Method and system for limited domain text to speech (TTS) processing
EP1383109A1 (de) * 2002-07-17 2004-01-21 STMicroelectronics N.V. Verfahren und Vorrichtung für breitbandige Sprachkodierung
US7502743B2 (en) * 2002-09-04 2009-03-10 Microsoft Corporation Multi-channel audio encoding and decoding with multi-channel transform selection
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US7299190B2 (en) * 2002-09-04 2007-11-20 Microsoft Corporation Quantization and inverse quantization for audio
US7254533B1 (en) * 2002-10-17 2007-08-07 Dilithium Networks Pty Ltd. Method and apparatus for a thin CELP voice codec
US20040260540A1 (en) * 2003-06-20 2004-12-23 Tong Zhang System and method for spectrogram analysis of an audio signal
US7539612B2 (en) * 2005-07-15 2009-05-26 Microsoft Corporation Coding and decoding scale factor information
US8417185B2 (en) * 2005-12-16 2013-04-09 Vocollect, Inc. Wireless headset and method for robust voice data communication
US7885419B2 (en) * 2006-02-06 2011-02-08 Vocollect, Inc. Headset terminal with speech functionality
US7773767B2 (en) 2006-02-06 2010-08-10 Vocollect, Inc. Headset terminal with rear stability strap
CN101114415B (zh) * 2006-07-25 2011-01-12 元太科技工业股份有限公司 双稳态显示器的驱动装置及其方法
JP5061111B2 (ja) * 2006-09-15 2012-10-31 パナソニック株式会社 音声符号化装置および音声符号化方法
CN101536311B (zh) 2007-01-25 2012-09-26 夏普株式会社 脉冲输出电路、使用该脉冲输出电路的显示装置的驱动电路、显示装置及脉冲输出方法
TWI346465B (en) * 2007-09-04 2011-08-01 Univ Nat Central Configurable common filterbank processor applicable for various audio video standards and processing method thereof
USD605629S1 (en) 2008-09-29 2009-12-08 Vocollect, Inc. Headset
FR2938688A1 (fr) 2008-11-18 2010-05-21 France Telecom Codage avec mise en forme du bruit dans un codeur hierarchique
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US8160287B2 (en) 2009-05-22 2012-04-17 Vocollect, Inc. Headset with adjustable headband
US8438659B2 (en) 2009-11-05 2013-05-07 Vocollect, Inc. Portable computing device and headset interface
WO2011118977A2 (ko) * 2010-03-23 2011-09-29 엘지전자 주식회사 오디오 신호 처리 방법 및 장치
KR101257776B1 (ko) * 2011-10-06 2013-04-24 단국대학교 산학협력단 상태-체크 코드를 이용한 부호화 방법 및 부호화 장치
CN102638846B (zh) * 2012-03-28 2015-08-19 浙江大学 一种基于最优量化策略的wsn通信负载降低方法
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ES2863554T3 (es) * 2014-01-24 2021-10-11 Nippon Telegraph & Telephone Aparato de análisis predictivo lineal, método, programa y soporte de registro
US9626983B2 (en) * 2014-06-26 2017-04-18 Qualcomm Incorporated Temporal gain adjustment based on high-band signal characteristic
US10542289B2 (en) * 2015-07-16 2020-01-21 Dolby Laboratories Licensing Corporation Signal reshaping and coding for HDR and wide color gamut signals

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2284946A1 (fr) * 1974-09-13 1976-04-09 Int Standard Electric Corp Vocodeur numerique
WO1983002346A1 (en) * 1981-12-22 1983-07-07 Motorola Inc A time multiplexed n-ordered digital filter
US5142581A (en) * 1988-12-09 1992-08-25 Oki Electric Industry Co., Ltd. Multi-stage linear predictive analysis circuit

Family Cites Families (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CA1245363A (en) * 1985-03-20 1988-11-22 Tetsu Taguchi Pattern matching vocoder
US4868867A (en) * 1987-04-06 1989-09-19 Voicecraft Inc. Vector excitation speech or audio coder for transmission or storage
GB2235354A (en) * 1989-08-16 1991-02-27 Philips Electronic Associated Speech coding/encoding using celp
US5307441A (en) * 1989-11-29 1994-04-26 Comsat Corporation Wear-toll quality 4.8 kbps speech codec
FI98104C (fi) * 1991-05-20 1997-04-10 Nokia Mobile Phones Ltd Menetelmä herätevektorin generoimiseksi ja digitaalinen puhekooderi
IT1257065B (it) * 1992-07-31 1996-01-05 Sip Codificatore a basso ritardo per segnali audio, utilizzante tecniche di analisi per sintesi.
US5706395A (en) * 1995-04-19 1998-01-06 Texas Instruments Incorporated Adaptive weiner filtering using a dynamic suppression factor
US5692101A (en) * 1995-11-20 1997-11-25 Motorola, Inc. Speech coding method and apparatus using mean squared error modifier for selected speech coder parameters using VSELP techniques

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
FR2284946A1 (fr) * 1974-09-13 1976-04-09 Int Standard Electric Corp Vocodeur numerique
WO1983002346A1 (en) * 1981-12-22 1983-07-07 Motorola Inc A time multiplexed n-ordered digital filter
US5142581A (en) * 1988-12-09 1992-08-25 Oki Electric Industry Co., Ltd. Multi-stage linear predictive analysis circuit

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
KWOK-WAH LAW ET AL: "A novel split residual vector quantization scheme for low bit rate speech coding", ICASSP-94. IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING (CAT. NO.94CH3387-8), PROCEEDINGS OF ICASSP '94. IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, ADELAIDE, SA, AUSTRALIA, 19-22 APRIL 1994, ISBN 0-7803-1775-0, 1994, NEW YORK, NY, USA, IEEE, USA, pages I/493 - 496 VOL., XP002013349 *
ORDENTLICH E ET AL: "LOW-DELAY CODE-EXCITED LINEAR-PREDICTIVE CODING OF WIDEBAND SPEECH AT 32 KBPS", SPEECH PROCESSING 1, TORONTO, MAY 14 - 17, 1991, vol. 1, 14 May 1991 (1991-05-14), INSTITUTE OF ELECTRICAL AND ELECTRONICS ENGINEERS, pages 9 - 12, XP000245155 *

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0867862A2 (de) * 1997-03-26 1998-09-30 Nec Corporation Vorrichtung zur Kodierung und Dekodierung von Sprach- und Musiksignalen
EP0867862A3 (de) * 1997-03-26 1999-06-09 Nec Corporation Vorrichtung zur Kodierung und Dekodierung von Sprach- und Musiksignalen
US6101464A (en) * 1997-03-26 2000-08-08 Nec Corporation Coding and decoding system for speech and musical sound
US8027242B2 (en) 2005-10-21 2011-09-27 Qualcomm Incorporated Signal coding and decoding based on spectral dynamics
US8392176B2 (en) 2006-04-10 2013-03-05 Qualcomm Incorporated Processing of excitation in audio coding and decoding
US8428957B2 (en) 2007-08-24 2013-04-23 Qualcomm Incorporated Spectral noise shaping in audio coding based on spectral dynamics in frequency sub-bands

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US5787390A (en) 1998-07-28
EP0782128B1 (de) 2000-06-21
FR2742568B1 (fr) 1998-02-13
CN1159691A (zh) 1997-09-17
JP3678519B2 (ja) 2005-08-03
FR2742568A1 (fr) 1997-06-20
KR970050107A (ko) 1997-07-29
JPH09212199A (ja) 1997-08-15
DE69608947T2 (de) 2001-02-01
KR100421226B1 (ko) 2004-07-19

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