EP1103953B1 - Verschleierungsverfahren bei Verlust von Sprachrahmen - Google Patents

Verschleierungsverfahren bei Verlust von Sprachrahmen

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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
Authority
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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English (en)
French (fr)
Other versions
EP1103953A3 (de
EP1103953A2 (de
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 date
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Publication of EP1103953A3 publication Critical patent/EP1103953A3/de
Application granted granted Critical
Publication of EP1103953B1 publication Critical patent/EP1103953B1/de
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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
    • G10L19/005—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)
  • Laminated Bodies (AREA)
  • Materials For Medical Uses (AREA)
  • Catalysts (AREA)

Claims (10)

  1. Verfahren zum Decodieren digitaler Sprache, wobei das Verfahren umfasst:
    Verwenden eines Synthesefilters und einer Erregung für das Synthesefilter, wobei die Erregung aus einem Verstärkungsfaktor für festes Codebuch und einem Vektor sowie aus einem Verstärkungsfaktor für adaptives Codebuch und einer Tonhöhenverzögerung erzeugt wird;
    dann, wenn ein empfangener Rahmen gelöscht wird, Verbergen des gelöschten Rahmens unter Verwendung eines Ersatz-Synthesefilters, das aus einem Synthesefilter eines früheren Rahmens und aus einer Ersatzerregung erzeugt wird, die aus einem Verstärkungsfaktor für festes Codebuch eines früheren Rahmens zusammen mit einem Zufallsvektor und aus einem Verstärkungsfaktor für adaptives Codebuch eines früheren Rahmens und einer Tonhöhenverzögerung erzeugt wird; und
    Stummschalten der Ersatzerregung vor der Synthesefilterung durch Multiplikation eines Stummschaltungs-Verstärkungsfaktors, der entsprechend der Anzahl aufeinander folgender Rahmenlöschungen eingestellt wird, indem:
    (a) dann, wenn ein Stummschaltungszähler größer als null ist, der Stummschaltungs-Verstärkungsfaktor durch Multiplizieren mit einem ersten Faktor, der kleiner als 1,0 ist, eingestellt wird, und
    (b) dann, wenn der Stummschaltungszähler nicht größer als null ist, der Stummschaltungs-Verstärkungsfaktor durch Multiplizieren mit einem zweiten Faktor, der größer als 1,0 ist, eingestellt wird, wobei jedoch der eingestellte Stummschaltungs-Verstärkungsfaktor auf 1,0 begrenzt wird; wobei der Stummschaltungszähler bei jedem Rahmen aktualisiert wird, indem:
    (i) er gleich 4 gesetzt wird, wenn die Anzahl aufeinander folgender gelöschter Rahmen wenigstens 2 ist,
    (ii) er um 1 dekrementiert wird, wenn der Stummschaltungszähler positiv ist und die Anzahl aufeinander folgender gelöschter Rahmen höchstens 1 ist und der Verstärkungsfaktor für adaptives Codebuch kleiner als 1,0 ist, oder
    (iii) er andernfalls nicht geändert wird.
  2. Verfahren nach Anspruch 1, bei dem:
    (a) die Synthesefilterung Synthesefilter-Koeffizienten enthält, die aus Filterkoeffizienten von zeitlich vorhergehenden Intervallen abgeleitet sind.
  3. Verfahren nach Anspruch 1, das ferner umfasst:
    (a) Bilden einer Erregung für ein gelöschtes Intervall codierter digitaler Sprache durch eine Summe eines Beitrags eines adaptiven Codebuchs und eines Beitrags eines festen Codebuchs, wobei der Beitrag des adaptiven Codebuchs aus einer Erregung und einer Tonhöhe und einem ersten Verstärkungsfaktor von Intervallen, die der codierten digitalen Sprache bei der Tonhöhe mit zufälligem Jitter zeitlich vorhergehen, abgeleitet wird und der Beitrag des festen Codebuchs aus zweiten Verstärkungsfaktoren der zeitlich vorhergehenden Intervalle abgeleitet wird; und
    (b) Filtern der Erregung.
  4. Verfahren nach Anspruch 3, bei dem:
    (a) das Filtern ein Stummschalten, gefolgt von einer Synthese mit Synthesefilter-Koeffizienten, die aus Synthesefilter-Koeffizienten zeitlich vorhergehender Intervalle abgeleitet werden, umfasst.
  5. Verfahren nach Anspruch 4, das ferner umfasst:
    (a) Bestimmen von Synthesefilter-Koeffizienten für das Intervall aus bandbreitenerweiterten Versionen von Synthesefilter-Koeffizienten von zeitlich vorhergehenden Intervallen der codierten digitalen Sprache.
  6. Decodierer für CELP-codierte Signale, wobei der Decodierer umfasst:
    einen Decodierer für festes Codebuch;
    einen Decodierer für adaptives Codebuch;
    einen Erregungsgenerator, der mit den Decodierern gekoppelt ist;
    ein Synthesefilter, das mit dem Erregungsgenerator gekoppelt ist, wobei der Erregungsgenerator dann, wenn ein empfangener Rahmen gelöscht wird, so betreibbar ist, dass er eine Ersatzerregung aus Ausgaben des Decodierers für festes Codebuch und des Decodierers für adaptives Codebuch für frühere Rahmen erzeugt, und das Synthesefilter so betreibbar ist, dass es Ersatz-Filterkoeffizienten aus Filterkoeffizienten früherer Rahmen erzeugt; und
    einen Stummschaltungsverstärkungsfaktor-Verstärker (gE), der zwischen einen Ausgang des Erregungsgenerators und einen Eingang des Synthesefilters geschaltet ist, wobei der Stummschaltungsverstärkungsfaktor-Verstärker dann, wenn ein empfangener Rahmen gelöscht wird, so betreibbar ist, dass er die Ersatzerregung durch Verstärken mit einem Verstärkungsfaktor entsprechend dem Verfahren nach Anspruch 1 stummschaltet.
  7. Decodierer nach Anspruch 6, bei dem:
    (a) der Decodierer für festes Codebuch und der Decodierer für adaptives Codebuch jeweils Ersatzausgänge erzeugen, indem sie die Ausgabe für den früheren Rahmen wiederholen.
  8. Computerprogrammprodukt, das computer- oder maschinenlesbare Computerprogrammelemente enthält, um einen Computer so zu konfigurieren, dass er das Verfahren nach einem der Ansprüche 1 bis 5 ausführt.
  9. Computerprogrammprodukt, das computer- oder maschinenlesbare Computerprogrammelemente enthält, die so übersetzbar sind, dass ein Computer so konfiguriert wird, dass das Verfahren nach einem der Ansprüche 1 bis 5 implementiert wird.
  10. Trägermedium, das ein Computerprogramm nach Anspruch 8 oder 9 trägt.
EP00310360A 1999-11-23 2000-11-22 Verschleierungsverfahren bei Verlust von Sprachrahmen Expired - Lifetime EP1103953B1 (de)

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US167197P 1999-11-23

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EP1103953A2 EP1103953A2 (de) 2001-05-30
EP1103953A3 EP1103953A3 (de) 2002-09-11
EP1103953B1 true EP1103953B1 (de) 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

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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
JP4846712B2 (ja) * 2005-03-14 2011-12-28 パナソニック株式会社 スケーラブル復号化装置およびスケーラブル復号化方法
WO2010000303A1 (en) * 2008-06-30 2010-01-07 Nokia Corporation Speech decoder with error concealment
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
CN103988255B (zh) * 2011-10-20 2016-03-02 Lg电子株式会社 管理抖动缓存器的方法和使用其的抖动缓存器
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 (en) * 1994-03-14 2001-05-29 Juin-Hwey Chen Computational complexity reduction during frame erasure or packet loss
US6085158A (en) * 1995-05-22 2000-07-04 Ntt Mobile Communications Network Inc. Updating internal states of a speech decoder after errors have occurred

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

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