EP1103953B1 - Procédé de dissimulation de pertes de trames de parole - Google Patents

Procédé de dissimulation de pertes de trames de parole

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Publication number
EP1103953B1
EP1103953B1 EP00310360A EP00310360A EP1103953B1 EP 1103953 B1 EP1103953 B1 EP 1103953B1 EP 00310360 A EP00310360 A EP 00310360A EP 00310360 A EP00310360 A EP 00310360A EP 1103953 B1 EP1103953 B1 EP 1103953B1
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EP
European Patent Office
Prior art keywords
frame
excitation
muting
gain
synthesis filter
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.)
Expired - Lifetime
Application number
EP00310360A
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German (de)
English (en)
Other versions
EP1103953A2 (fr
EP1103953A3 (fr
Inventor
Takahiro Unno
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.)
Texas Instruments Inc
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Texas Instruments Inc
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Publication of EP1103953A3 publication Critical patent/EP1103953A3/fr
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; 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
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm

Definitions

  • the invention relates to electronic devices, and more particularly, but not exclusively, to speech coding, transmission, storage, and decoding/synthesis methods and circuitry.
  • the performance of digital speech systems using low bit rates has become increasingly important with current and foreseeable digital communications.
  • Both dedicated channel and packetized-over-network (e.g., Voice over IP or Voice over Packet) transmissions benefit from compression of speech signals.
  • the widely-used linear prediction (LP) digital speech coding compression method models the vocal tract as a time-varying filter and a time-varying excitation of the filter to mimic human speech.
  • M the order of the linear prediction filter, is taken to be about 10-12; the sampling rate to form the samples s(n) is typically taken to be 8 kHz (the same as the public switched telephone network sampling for digital transmission); and the number of samples ⁇ s(n) ⁇ in a frame is typically 80 or 160 (10 or 20 ms frames).
  • a frame of samples may be generated by various windowing operations applied to the input speech samples.
  • ⁇ a i ⁇ which furnish the best linear prediction for the frame.
  • the coefficients ⁇ a i ⁇ may be converted to line spectral frequencies (LSFs) for quantization and transmission or storage and converted to line spectral pairs (LSPs) for interpolation between subframes.
  • LSFs line spectral frequencies
  • LSPs line spectral pairs
  • the ⁇ r(n) ⁇ is the LP residual for the frame, and ideally the LP residual would be the excitation for the synthesis filter 1/A(z) where A(z) is the transfer function of equation (1).
  • the LP residual is not available at the decoder; thus the task of the encoder is to represent the LP residual so that the decoder can generate an excitation which emulates the LP residual from the encoded parameters.
  • the LP compression approach basically only transmits/stores updates for the (quantized) filter coefficients, the (quantized) residual (waveform or parameters such as pitch), and (quantized) gain(s).
  • a receiver decodes the transmitted/stored items and regenerates the input speech with the same perceptual characteristics.
  • Figures 5-6 illustrate high level blocks of an LP system. Periodic updating of the quantized items requires fewer bits than direct representation of the speech signal, so a reasonable LP coder can operate at bits rates as low as 2-3 kb/s (kilobits per second).
  • the decoder typically has methods to conceal such frame erasures, and such methods may be categorized as either interpolation-based or repetition-based.
  • An interpolation-based concealment method exploits both future and past frame parameters to interpolate missing parameters.
  • interpolation-based methods provide better approximation of speech signals in missing frames than repetition-based methods which exploit only past frame parameters.
  • the interpolation-based method has a cost of an additional delay to acquire the future frame.
  • future frames are available from a playout buffer which compensates for arrival jitter of packets, and interpolation-based methods mainly increase the size of the playout buffer.
  • Repetition-based concealment which simply repeats or modifies the past frame parameters, finds use in several CELP-based speech coders including G.729, G.723.1 and GSM-EFR.
  • the repetition-based concealment method in these coders does not introduce any additional delay or playout buffer size, but the performance of reconstructed speech with erased frames is poorer than that of the interpolation-based approach, especially in a high erased-frame ratio or bursty frame erasure environment.
  • the ITU standard G.729 uses frames of 10 ms length (80 samples) divided into two 5-ms 40-sample subframes for better tracking of pitch and gain parameters plus reduced codebook search complexity.
  • Each subframe has an excitation represented by an adaptive-codebook contribution and a fixed (algebraic) codebook contribution.
  • the adaptive-codebook contribution provides periodicity in the excitation and is the product of v(n), the prior frame's excitation translated by the current frame's pitch lag in time and interpolated, multiplied by a gain, g P .
  • the algebraic codebook contribution approximates the difference between the actual residual and the adaptive codebook contribution with a four-pulse vector, c(n), multiplied by a gain, g C .
  • G.729 handles frame erasures by reconstruction based on previously received information; that is, repetition-based concealment. Namely, replace the missing excitation signal with one of similar characteristics, while gradually decaying its energy by using a voicing classifier based on the long-term prediction gain (which is computed as part of the long-term postfilter analysis).
  • the long-term postfilter finds the long-term predictor for which the prediction gain is more than 3 dB by using a normalized correlation greater than 0.5 in the optimal delay determination.
  • a 10 ms frame is declared periodic if at least one 5 ms subframe has a long-term prediction gain of more than 3 dB. Otherwise the frame is declared nonperiodic.
  • An erased frame inherits its class from the preceding (reconstructed) speech frame. Note that the voicing classification is continuously updated based on this reconstructed speech signal. The specific steps taken for an erased frame are as follows:
  • US-A-5 717 822 describes a speech coding system robust to frame erasure (or packet loss).
  • Illustrative embodiments are directed to a modified version of CCIIT standard G.728.
  • frame erasure vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. This synthesis differs for voiced and non-voiced speech.
  • linear prediction filter coefficients are synthesized as a weighted extrapolation of a set of linear prediction filter coefficients determined during non-erased frames. The weighting factor is a number less than 1. This weighting accomplishes a bandwidth-expansion of peaks in the frequency response of a linear predictive filter.
  • Computational complexity during erased frames is reduced through the elimination of certain computations needed during non-erased frames only. This reduction in computational complexity offsets additional computation required for excitation signal synthesis and linear prediction filter coefficient generation during erased frames.
  • An embodiments of the present invention provides concealment of erased frames by frame repetition together with one or more of: excitation signal muting, LP coefficient bandwidth expansion with cutoff frequency, and pitch delay jittering.
  • Preferred embodiment decoders and methods for concealment of frame erasures in CELP-encoded speech or other signal transmissions have one or more of three features: (1) muting the excitation outside of the feedback loop, this replaces the attenuation of the adaptive and fixed codebook gains; (2) expanding the bandwidth of the LP synthesis filter with a threshold frequency for differing expansion factors; and (3) jittering the pitch delay to avoid overly periodic repetition frames.
  • Features (2) and (3) especially apply to bursty noise leading to frame erasures.
  • Figure 1 illustrates a preferred embodiment decoder using all three concealment features; this contrasts with the G.729 standard decoder concealment illustrated in Figure 2.
  • Preferred embodiment systems e.g., Voice over IP or Voice over Packet
  • Preferred embodiment concealment methods in decoders.
  • Figure 3 illustrates a speech encoder using LP encoding with excitation contributions from both adaptive and algebraic codebook, and preferred embodiment concealment features affect the pitch delay, the codebook gains, and the LP synthesis filter. Encoding proceeds as follows:
  • the final codeword encoding the (sub)frame would include bits for: the quantized LSF coefficients, adaptive codebook pitch delay, algebraic codebook vector, and the quantized adaptive codebook and algebraic codebook gains.
  • Figure 1 illustrates preferred embodiment decoders and decoding methods which essentially reverse the encoding steps of the foregoing encoding method plus provide repetition-based concealment features for erased frame reconstructions as described in the next section.
  • Figure 4 shows a decoder without concealment features, and for the m th (sub)frame proceed as follows:
  • Figure 1 shows preferred embodiment concealment features in a preferred embodiment decoder and contrasts with Figure 2.
  • the preferred embodiment concealment features construct an (m+j) st frame with one or more of the following modified decoder steps:
  • Another alternative preferred embodiment omits the pitch delay jittering but may use the incrementing as in G.729 for erased frames together with excitation muting and LP coefficient bandwidth expansion.
  • an alternative preferred embodiment omits the excitation muting and uses the G.729 construction together with the pitch delay jittering and synthesis filter coefficient bandwidth expansion.
  • preferred embodiments may use just one of the three features (excitation muting, pitch delay jittering, and synthesis filiter coefficient bandwidth expansion) and follow G.729 in other aspects.
  • FIGS 5-6 show in functional block form preferred embodiment systems which use the preferred embodiment encoding and decoding. This applies to speech and also other signals which can be effectively CELP coded.
  • the encoding and decoding can be performed with digital signal processors (DSPs) or general purpose programmable processors or application specific circuitry or systems on a chip such as both a DSP and RISC processor on the same chip with the RISC processor controlling.
  • DSPs digital signal processors
  • Codebooks would be stored in memory at both the encoder and decoder, and a stored program in an onboard or external ROM, flash EEPROM, or ferroelectric memory for a DSP or programmable processor could perform the signal processing.
  • Analog-to-digital converters and digital-to-analog converters provide coupling to the real world, and modulators and demodulators (plus antennas for air interfaces) provide coupling for transmission waveforms.
  • the encoded speech can be packetized and transmitted over networks such as the Internet.
  • the preferred embodiments may be modified in various ways while retaining one or more of the features of erased frame concealment by synthesis filter coefficient bandwidth expansion, pitch delay jittering, and excitation muting.
  • interval (frame and subframe) size and sampling rate could differ;
  • the bandwidth expansion factor could apply for C B > 0 or C B > 2, the multipliers 0.95 and 1.05 and limits 0.8 and 1.0 could vary, and the 100 Hz threshold could vary;
  • the pitch delay jitter could be with a larger or smaller percentage of the pitch delay and could also apply to the first erased frame, and the jitter size could vary with the number of consecutive erased frames or erasure density;
  • the excitation muting could vary nonlinearly with number of consecutive erased frames or erasure density, and the multipliers 0.95499 and 1.09648 could vary.
  • a software-controlled programmable processing device such as a Digital Signal Processor, microprocessor, other processing devices, data processing apparatus or computer system
  • a computer program for configuring a programmable device, apparatus or system to implement the foregoing described methods is envisaged as an aspect of the present invention.
  • the computer program may be embodied as source code and undergo compilation for implementation on a processing device, apparatus or system, or may be embodied as object code, for example.
  • object code for example.
  • the term computer in its most general sense encompasses programmable devices such as referred to above, and data processing apparatus and computer systems.
  • the computer program is stored on a carrier medium in machine or device readable form, for example in solid-state memory or magnetic memory such as disc or tape and the processing device utilises the program or a part thereof to configure it for operation.
  • the computer program may be supplied from a remote source embodied in a communications medium such as an electronic signal, radio frequency carrier wave or optical carrier wave.
  • a communications medium such as an electronic signal, radio frequency carrier wave or optical carrier wave.
  • carrier media are also envisaged as aspects of the present invention.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Multimedia (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Computational Linguistics (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Two-Way Televisions, Distribution Of Moving Picture Or The Like (AREA)
  • Materials For Medical Uses (AREA)
  • Catalysts (AREA)
  • Laminated Bodies (AREA)

Claims (10)

  1. Un procédé de décodage de paroles numériques, le procédé comprenant :
    l'utilisation d'un filtre de synthèse et d'une excitation pour le filtre de synthèse, où l'excitation est générée par un gain et un vecteur de livre de codes fixe, ainsi qu'un gain de livre de codes adaptatif et un délai de tonie ;
    lorsque qu'une trame reçue est effacée, la fourniture de la dissimulation de la trame effacée en utilisant un filtre de synthèse de remplacement généré à partir du filtre de synthèse d'une trame antérieure et une excitation de remplacement générée à partir d'un gain de livre de codes fixe d'une trame antérieure avec un vecteur aléatoire ainsi qu'un gain de livre de codes adaptatif d'une trame antérieure et un délai de tonie ; et
    l'affaiblissement de ladite excitation de remplacement avant le filtrage de synthèse par multiplication avec un facteur de gain d'affaiblissement qui est ajusté en fonction du nombre de trames effacées successives, comme suit :
    a) lorsqu'un compteur d'affaiblissement est supérieur à zéro, ajuster le facteur de gain d'affaiblissement en le multipliant par un premier facteur qui est inférieur à 1,0, et
    b) lorsque ledit compteur d'affaiblissement n'est pas supérieur à zéro, ajuster le facteur de gain d'affaiblissement en le multipliant par un second facteur supérieur à 1,0, mais limiter le facteur de gain d'affaiblissement à 1,0 ;
    où ledit compteur d'affaiblissement est mis à jour à chaque trame :
    (i) en étant défini sur 4 lorsque le nombre de trames effacées consécutives est au moins 2,
    (ii) en étant réduit de 1 lorsque ledit compteur d'effacement est positif que le nombre de trames effacées consécutives est au maximum de 1 et que ledit gain de livre de codes adaptatif est inférieur à 1,0, ou
    (iii) n'est pas changé dans le cas contraire.
  2. Le procédé de la revendication 1, où :
    (a) ledit filtrage de synthèse comprend un coefficient de filtre de synthèse dérivé des coefficients de filtre desdits intervalles antérieurs dans le temps.
  3. Le procédé de la revendication 1, qui comprend en outre :
    (a) la création d'une excitation pour un intervalle effacé de parole numérique codée, par une somme d'une contribution de livre de codes adaptatif et une contribution de livre de codes fixe, où la contribution du livre de codes adaptatif est dérivée d'une excitation et d'une tonie ainsi que du premier gain des intervalles antérieurs dans le temps de ladite parole numérique codée avec ladite tonie à variation aléatoire, et ladite contribution de livre de codes fixe est dérivée d'un second gain desdits intervalles antérieurs dans les temps ; et
    (b) le filtrage de ladite excitation.
  4. Le procédé de la revendication 3, où :
    (a) ledit filtrage inclut un affaiblissement suivi d'une synthèse avec des coefficients de filtre de synthèse dérivés des coefficients de filtre de synthèse desdits intervalles antérieurs dans le temps.
  5. Le procédé de la revendication 4, qui comprend en outre :
    (a) la détermination des coefficients de filtre de synthèse pour ledit intervalle à partir des versions étendues de bande passante des coefficients de filtre de synthèse des intervalles antérieurs dans le temps desdites paroles numériques codées.
  6. Un décodeur de signaux codés, CELP le décodeur comprenant :
    un décodeur de livre de codes fixe ;
    un décodeur de livre de codes adaptatif;
    un générateur d'excitation couplé auxdits décodeurs ;
    un filtre de synthèse couplé audit générateur d'excitation, où, lorsqu'une trame reçue est effacée, ledit générateur d'excitation est utilisable pour générer une excitation de remplacement depuis les sorties dudit décodeur de livre de codes fixe et dudit décodeur de livre de codes adaptatif pour les trames antérieures, et ledit filtre de synthèse est utilisable pour générer des coefficients de filtre de remplacement à partir des coefficients de filtre pour les trames antérieures ; et
    un amplificateur de gain d'affaiblissement (gE) couplé entre une sortie dudit générateur d'excitation et une entrée dans ledit filtre de synthèse, où, lorsqu'une trame reçue est effacée, ledit amplificateur de gain d'affaiblissement est utilisable pour affaiblir ladite excitation de remplacement en l'amplifiant à l'aide d'un gain d'amplification conforme au procédé de la revendication 1.
  7. Le décodeur de la revendication 6, où :
    (a) ledit décodeur de livre de codes fixe et ledit décodeur de livre de codes adaptatif génèrent tous deux lesdites sorties de remplacement en répétant les sorties pour la trame antérieure.
  8. Un produit de programme informatique comprenant des éléments de programme informatique lisibles par un ordinateur ou une machine pour configurer un ordinateur de sorte qu'il implémente le procédé de l'une des revendications 1 à 5.
  9. Un produit de programme informatique comprenant des éléments de programme informatique lisibles par un ordinateur ou une machine qui peuvent être traduits pour configurer un ordinateur de sorte qu'il implémente le procédé de l'une des revendications 1 à 5.
  10. Un support porteur comportant un programme informatique conforme à la revendication 8 ou 9.
EP00310360A 1999-11-23 2000-11-22 Procédé de dissimulation de pertes de trames de parole Expired - Lifetime EP1103953B1 (fr)

Applications Claiming Priority (2)

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US16719799P 1999-11-23 1999-11-23
US167197P 1999-11-23

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EP1103953A2 EP1103953A2 (fr) 2001-05-30
EP1103953A3 EP1103953A3 (fr) 2002-09-11
EP1103953B1 true EP1103953B1 (fr) 2006-08-16

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Cited By (1)

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Publication number Priority date Publication date Assignee Title
CZ304196B6 (cs) * 2000-11-30 2013-12-27 Panasonic Corporation Vektorové kvantizační zařízení LPC parametru, kodér řeči a přijímací zařízení signálu řeči

Families Citing this family (10)

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Publication number Priority date Publication date Assignee Title
US6826527B1 (en) * 1999-11-23 2004-11-30 Texas Instruments Incorporated Concealment of frame erasures and method
US7835916B2 (en) 2003-12-19 2010-11-16 Telefonaktiebolaget Lm Ericsson (Publ) Channel signal concealment in multi-channel audio systems
SE527866C2 (sv) * 2003-12-19 2006-06-27 Ericsson Telefon Ab L M Kanalsignalmaskering i multikanalsaudiosystem
CN101138174B (zh) 2005-03-14 2013-04-24 松下电器产业株式会社 可扩展解码装置和可扩展解码方法
WO2010000303A1 (fr) * 2008-06-30 2010-01-07 Nokia Corporation Décodeur de parole avec dissimulation d'erreur
JP5111430B2 (ja) * 2009-04-24 2013-01-09 パナソニック株式会社 音声符号化装置、音声復号化装置、及びこれらの方法
EP2770504B1 (fr) * 2011-10-20 2016-11-30 LG Electronics Inc. Procédé de gestion d'un tampon de gigue, et tampon de gigue l'utilisant
JP5328883B2 (ja) * 2011-12-02 2013-10-30 パナソニック株式会社 Ｃｅｌｐ型音声復号化装置およびｃｅｌｐ型音声復号化方法
JP6516099B2 (ja) * 2015-08-05 2019-05-22 パナソニックＩｐマネジメント株式会社 音声信号復号装置および音声信号復号方法
CN110474815B (zh) * 2019-09-23 2021-08-13 北京达佳互联信息技术有限公司 带宽预测方法、装置、电子设备及存储介质

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CA2142391C (fr) * 1994-03-14 2001-05-29 Juin-Hwey Chen Reduction de la complexite des calculs durant l'effacement des trames ou les pertes de paquets
JP3155952B2 (ja) * 1995-05-22 2001-04-16 株式会社エヌ・ティ・ティ・ドコモ 音声復号装置

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CZ304196B6 (cs) * 2000-11-30 2013-12-27 Panasonic Corporation Vektorové kvantizační zařízení LPC parametru, kodér řeči a přijímací zařízení signálu řeči

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DE60030069D1 (de) 2006-09-28
ATE336780T1 (de) 2006-09-15
DE60030069T2 (de) 2007-02-22
EP1103953A2 (fr) 2001-05-30
JP2001154699A (ja) 2001-06-08
EP1103953A3 (fr) 2002-09-11

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