EP1383109A1 - Verfahren und Vorrichtung für breitbandige Sprachkodierung - Google Patents

Verfahren und Vorrichtung für breitbandige Sprachkodierung Download PDF

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Publication number
EP1383109A1
EP1383109A1 EP02015918A EP02015918A EP1383109A1 EP 1383109 A1 EP1383109 A1 EP 1383109A1 EP 02015918 A EP02015918 A EP 02015918A EP 02015918 A EP02015918 A EP 02015918A EP 1383109 A1 EP1383109 A1 EP 1383109A1
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Prior art keywords
filter
term
word
excitation
short
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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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Priority to EP02015918A priority Critical patent/EP1383109A1/de
Priority to EP20030291747 priority patent/EP1383111A2/de
Priority to US10/622,021 priority patent/US7254534B2/en
Publication of EP1383109A1 publication Critical patent/EP1383109A1/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
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders

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 i.e. 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 voice 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 directory DLT, also called “adaptive long-term dictionary", an associated long-term gain Ga, a short-term excitation word C j , extracted from an algebraic coded directory DCT, also known as “fixed coded directory” or “short dictionary algebraic term ", and an associated short-term gain Gc).
  • a long-term digital excitation word V i extracted from an adaptive coded directory DLT, also called “adaptive long-term dictionary", an associated long-term gain Ga
  • a short-term excitation word C j extracted from an algebraic coded directory DCT, also known as “fixed coded directory” or “short dictionary algebraic term ", 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.
  • An object of the invention is to reduce the harmonic noise and the high frequency noise.
  • the invention also aims to suppress noise from "whistling" type tainting the voiced speech frames.
  • Another object of the invention is to independently control 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 and a associated long-term gain, as well as a word of short-term excitement extract from an algebraic coded repertoire and short-term gain associated, and we update the adaptive coded directory from the word excerpt long term excitement and short term excitement word extract.
  • the invention here uses a "total correction" filter which combines a harmonic noise correction filter and a high frequency correction.
  • the invention thus improves the quality during voiced speech frames. Furthermore, the complexity of the encoder is reduced by merging the filter of harmonic correction and the high frequency correction filter.
  • the invention differs in particular from a solution described in an article by Kroon and Atal, entitled “Strategies for Improving the Performance of CELP Coders at Low Bit Rates ”, Proc., IEEE, Int. Conf. Acoustics, Speech, and Signal Processing, ICASSP'88, New York, USA, 1988, Pages 151-154, which offers filtering of adaptive dictionary made at the output of this dictionary and not not at the entrance according to the invention.
  • the prefiltering of the adaptive dictionary according to the invention presents in relation to the post-filtering of the article of Kroon and Atal, the advantage that filtering is taken into account when error minimization performed to choose excitation adaptive to the next frame. This is not the case for the solution of Kroon and Atal, since the proposed filtering takes place on the excitation chosen next. Also, to take into account the filtering in the minimization of the error, it would then be necessary to increase considerably complexity and filter out any excitement to be tested.
  • the summed word with a finite impulse response digital filter at linear phase having an order at least equal to 10.
  • the sampling frequency is 16 kHz
  • the invention also provides a control type solution gain, but totally different from that described in particular in the articles of Taniguchi and others and of Shoham.
  • the extraction of the short term excitation word comprises digital linear prediction filtering and the method includes an update of the state of the linear prediction filter with the word short-term excitation filtered by a filter whose coefficients depend on the value of the long-term gain, so that weaken the contribution of short-term excitement when the gain long-term excitement is above a predetermined threshold, for example equal to 1.
  • the solution according to the invention consists here to weaken the contribution of short-term excitement if the gain of long-term excitement is important.
  • this is the contribution of undiminished short-term excitement which is stored in the adaptive dictionary for updating. So the reduction occurs only on exit. Preserving the magnitude of the short-term contribution to be stored is important, since the richness of the adaptive dictionary is thus preserved for the lowest frequencies.
  • the gain correction must also be applied when reconstructing the signal at the decoder.
  • This filter can be of order 0 or of higher or equal order to 1. In the latter case, the filter of order greater than or equal to 1 can be finite impulse response.
  • the first coefficient B0 of the filter is equal to 1 / (1 + ⁇ . min (Ga, 1))
  • the second coefficient B1 of the filter is equal to ⁇ .min (Ga, 1) / (1 + ⁇ .min (Ga, 1))
  • ⁇ is a real number with a lower absolute value at 1
  • Ga is the long-term gain
  • min (Ga, 1) designates the minimum value between Ga and 1.
  • 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 of 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 digital prediction filter linear
  • the device comprises second updating means capable of updating the state of the linear prediction filter with short term excitation word filtered by a filter whose coefficient (s) depend on the value long-term gain, so as to weaken the contribution of short-term excitement when gaining long-term excitement is above a predetermined threshold.
  • 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 cascaded to a second perceptual weighting filter comprising a second formantic weighting filter, and by the fact that the denominator of the transfer function of the first filter of formantic weighting is equal to the numerator of the second filter formantic weighting.
  • 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 encoding device, or encoder, CD differs from that of the prior art as illustrated in FIG. 1 by the fact that the MAJ means of DLT adaptive long-term dictionary update feature a total correction filter FLCT connected between the output of a SM summer and DLT dictionary entry.
  • the two inputs of the summator SM respectively receive the product of the extracted word of long-term excitation v; by the associated long-term gain Ga, and the product of the extracted short-term excitation word c j by the associated gain Gc.
  • This FLCT total correction filter is a low pass filter generally having a cutoff frequency greater than quarter of the sampling frequency and less than half of it.
  • This filter is in the example described a digital filter with linear phase finite impulse response with order at less than 10.
  • a cutoff frequency of preferably will be used of the order of 6 kHz and a filter of order 20, which achieves a good trade-off between memory complexity and signal quality reconstituted vocal.
  • Harmonic noise is introduced by the contribution of long-term excitement and by repeating samples for values of the fundamental period (pitch) less than the length of a speech frame, here of 5 ms. This noise is also present for values of the fundamental period greater than the size of a frame. It is also linked to adaptive gain, extracts a single times per speech frame.
  • the total correction filter according to the invention therefore achieves the double harmonic correction and high correction function frequency. This allows an improvement in quality during voiced speech frames.
  • this filter i.e. at the input of the adaptive dictionary, allows filtering to be taken into account minimization of the error made to choose the excitation adaptive of the following speech frame.
  • the coder CD further comprises second updating means MAJ2 able to update the state of the linear prediction filter FP and the state of the filter perceptual weighting FPP with the short-term excitation word C j filtered by a filter which is represented here schematically by a gain Gc '.
  • This filter can be of order 0 and its gain Gc 'is less than the gain Gc.
  • this filter can be of finite impulse response and of order greater than or equal to 1, with for example a filter of finite impulse response of order 1.
  • the coefficients of this first order filter depend on the value long-term gain Ga, so as to weaken the contribution of short-term excitement when gaining long-term excitement Ga is greater than a predetermined threshold, for example equal to 1.
  • the transfer function of this filter is equal to B0 + B1 z -1 .
  • the first coefficient of the filter B0 can be determined by the formula (I) below. 1 / (1 + 0.98 min (Ga, 1)) while the second coefficient of filter B1 can be determined by formula (II) below. 0.98 min (Ga, 1) / (1 + 0.98 min (Ga, 1))
  • the attenuation occurs only on the signal preserving the magnitude of the short contribution term to store keeps the richness of the dictionary adaptive for the lowest frequencies.
  • the variant embodiment illustrated in FIG. 3 allows, in addition to the benefits of the total correction filter, to eliminate hissing noise on speech frames voiced.
  • 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.
  • the same FPP perceptual weighting filter is used for short-term research and for long-term research.
  • the transfer function W (z) of this FPP filter is given by the formula (III) below.
  • 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. 4 Such an embodiment is illustrated in FIG. 4, in which, compared to Figure 3, the unique FPP filter was replaced by a first formantic weighting filter FPP1 for long-term research, cascaded with a second filter of FPP2 formant weighting for short-term research.
  • 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 (IV) 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 the formula (V) 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 the formula (VI) 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 (VII) below. 1 AT ( z / ⁇ 22 )
  • FIG. 5 Such an embodiment is illustrated in FIG. 5, where one see that the use of the two form filters is taken into account combination with the use of the total correction filter.
  • 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. 6, 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)
EP02015918A 2002-07-17 2002-07-17 Verfahren und Vorrichtung für breitbandige Sprachkodierung Withdrawn EP1383109A1 (de)

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EP02015918A EP1383109A1 (de) 2002-07-17 2002-07-17 Verfahren und Vorrichtung für breitbandige Sprachkodierung
EP20030291747 EP1383111A2 (de) 2002-07-17 2003-07-15 Verfahren und Vorrichtung zur Sprachkodierung mit erweiterter Bandbreite
US10/622,021 US7254534B2 (en) 2002-07-17 2003-07-17 Method and device for encoding wideband speech

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US9947340B2 (en) * 2008-12-10 2018-04-17 Skype Regeneration of wideband speech
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
KR101920297B1 (ko) 2014-04-25 2018-11-20 가부시키가이샤 엔.티.티.도코모 선형 예측 계수 변환 장치 및 선형 예측 계수 변환 방법
CN107452391B (zh) 2014-04-29 2020-08-25 华为技术有限公司 音频编码方法及相关装置
US9959364B2 (en) * 2014-05-22 2018-05-01 Oath Inc. Content recommendations
CN106502799A (zh) * 2016-12-30 2017-03-15 南京大学 一种基于长短时记忆网络的主机负载预测方法

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