WO2001024164A1 - Codeur vocal, decodeur vocal et procede de codage et de decodage de la parole - Google Patents

Codeur vocal, decodeur vocal et procede de codage et de decodage de la parole Download PDF

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Publication number
WO2001024164A1
WO2001024164A1 PCT/JP2000/006542 JP0006542W WO0124164A1 WO 2001024164 A1 WO2001024164 A1 WO 2001024164A1 JP 0006542 W JP0006542 W JP 0006542W WO 0124164 A1 WO0124164 A1 WO 0124164A1
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WO
WIPO (PCT)
Prior art keywords
quantization
spectrum envelope
speech
information
fundamental frequency
Prior art date
Application number
PCT/JP2000/006542
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English (en)
Japanese (ja)
Inventor
Tadashi Yonezaki
Original Assignee
Matsushita Electric Industrial Co., Ltd.
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Matsushita Electric Industrial Co., Ltd. filed Critical Matsushita Electric Industrial Co., Ltd.
Priority to EP00961220A priority Critical patent/EP1132891A1/fr
Priority to AU73212/00A priority patent/AU7321200A/en
Publication of WO2001024164A1 publication Critical patent/WO2001024164A1/fr

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Classifications

    • 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

Definitions

  • Speech coding apparatus speech decoding apparatus, and speech coding / decoding method
  • the present invention relates to a speech encoding device, a speech decoding device, and a speech coding / decoding method used for a communication device of a wireless communication system such as a mobile phone and a mobile phone.
  • FIG. 1 is a block diagram showing the configuration of a conventional speech encoding device and speech decoding device.
  • spectrum envelope analysis section 11 estimates spectral envelope information of an input speech signal.
  • the spectrum envelope quantization unit 12 quantizes the spectrum envelope information estimated by the spectrum envelope analysis unit 11.
  • the inverse filter 13 filters the inverse characteristic of the frequency characteristic of the spectrum envelope information quantized by the spectrum envelope quantization unit 12 with respect to the input audio signal, and removes the spectrum envelope component.
  • This signal is considered to mimic the sound source signal generated in the vocal cords during the vocalization process.
  • this signal is referred to as “sound source signal”.
  • Excitation codebook 14 stores signals having flat frequency characteristics.
  • Source coding section 15 searches source codebook 14 for a signal closest to the source signal, and outputs the code (hereinafter referred to as “source code”).
  • the multiplexing unit 16 receives the spectrum envelope output from the spectrum envelope quantization unit 12.
  • a code indicating the quantized value of the link information and the excitation code output from excitation encoding section 15 are multiplexed as a code sequence and transmitted to the communication channel.
  • demultiplexing section 21 separates the received code string into a code indicating a quantized value of spectrum envelope information and an excitation code.
  • excitation codebook 22 In excitation codebook 22, the same signals as in excitation codebook 14 are stored. Sound source selecting section 23 selects and extracts a signal corresponding to the received sound source code from sound source codebook 22.
  • the synthesis filter 24 filters the signal extracted by the sound source selection unit 23 so as to have the frequency characteristic of the received spectrum envelope information, and outputs a decoded voice.
  • the spectral envelope information having different dynamic ranges and quantization characteristics of the signal is separated from the source signal, and the quantizer corresponding to each characteristic is separated.
  • high-quality speech codec is realized.
  • An object of the present invention is to provide a speech coding apparatus, a speech decoding apparatus, and a speech codec method capable of implementing high-quality speech decoding even when transmitting information at a low bit rate. is there.
  • FIG. 1 is a block diagram illustrating a configuration of a conventional speech encoding device and a conventional speech decoding device.
  • FIG. 2 is a block diagram illustrating a configuration of the speech encoding device and the speech decoding device according to the first embodiment of the present invention.
  • FIG. 3 is a block diagram showing an internal configuration of a spectrum envelope quantization unit of the speech coding apparatus according to Embodiment 2 of the present invention.
  • FIG. 4 is a block diagram showing an internal configuration of a spectrum envelope quantization unit of the speech coding apparatus according to Embodiment 3 of the present invention.
  • FIG. 5 is a model diagram of a spectrum envelope curved surface according to Embodiment 3 of the present invention
  • FIG. 6 is an internal configuration of a spectrum envelope quantization unit of a speech encoding device according to Embodiment 4 of the present invention.
  • FIG. 7 is a block diagram showing an internal configuration of a spectrum envelope quantization unit of a speech coding apparatus according to Embodiment 5 of the present invention.
  • FIG. 8 is a block diagram showing an internal configuration of a model applicator of a speech coding apparatus according to Embodiment 6 of the present invention.
  • FIG. 9 is a block diagram illustrating an internal configuration of a parameter quantizer of a speech coding apparatus according to Embodiment 7 of the present invention.
  • FIG. 10 is a block diagram showing an internal configuration of a parameter quantizer in a speech coding apparatus according to Embodiment 8 of the present invention.
  • FIG. 11 is a block diagram showing an internal configuration of a parameter overnight quantizer of a speech coding apparatus according to Embodiment 9 of the present invention
  • FIG. 12 is a block diagram showing the internal configuration of the spectrum envelope configuration unit of the speech decoding device according to Embodiment 10 of the present invention.
  • FIG. 2 is a block diagram showing a configuration of the speech encoding device and the speech decoding device according to Embodiment 1 of the present invention.
  • speech analysis section 101 extracts a fundamental frequency and short-time spectrum envelope information from an input speech signal.
  • the fundamental frequency quantization unit 102 quantizes the fundamental frequency extracted by the speech analysis unit 101.
  • the matrix generation unit 103 generates a spectrum envelope surface on a time-frequency plane by arranging the short-time spectrum envelope information extracted by the speech analysis unit 101 along the time axis.
  • the spectrum envelope quantization unit 104 quantizes the spectrum envelope curved surface generated by the matrix generation unit 103.
  • the spectral envelope information is quantized as a continuous function on the time-frequency plane.If only the extracted spectral envelope is quantized, the spectral envelope information is quantized depending on the sound source information. This is because information that is the gist of the present invention cannot be separated in the quantization processing.
  • the multiplexing unit 105 includes a code indicating the quantized value of the spectrum envelope curved surface output from the spectrum envelope quantization unit 104 and the basic value output from the fundamental frequency quantization unit 102. A code indicating a quantized value of the frequency is multiplexed and transmitted to the communication path.
  • demultiplexing section 201 converts the received code string into a code indicating the quantized value of the spectrum envelope information and a code indicating the quantized value of the fundamental frequency. To separate.
  • the spectrum envelope configuration unit 202 reconstructs a quantized spectrum envelope surface from the received spectrum envelope information.
  • the speech synthesis unit 203 synthesizes and outputs a decoded speech by cutting out the spectrum envelope curved surface reconstructed by the spectrum envelope construction unit 202 based on the fundamental frequency information.
  • a fundamental frequency and short-time spectrum envelope information are extracted from an input speech signal input by a speech analysis unit 101 of the speech coding apparatus 100.
  • the extracted fundamental frequency is quantized by the fundamental frequency quantization unit 102.
  • the extracted short-time spectrum envelope information is arranged along the time axis in a matrix generation unit 103, and a spectrum envelope curved surface on a time-frequency plane is generated.
  • the spectrum envelope curved surface is quantized by a spectrum envelope quantization unit 104.
  • the quantized fundamental frequency and spectrum envelope curved surface are multiplexed by the multiplexing unit 105 and transmitted to the communication path. Then, the fundamental frequency and the spectrum envelope surface are received by the demultiplexing unit 201 of the speech decoding apparatus 200, and are separated into a quantized value of the spectrum envelope information and a quantized value of the fundamental frequency. .
  • the quantized value of the spectrum envelope information is input to the spectrum envelope construction unit 202, and the spectrum envelope curved surface is reconstructed in the spectrum envelope construction unit 202. Then, the speech synthesis unit 203 uses the reconstructed spectrum envelope surface as a basis. By extracting based on this frequency information, the decoded speech is synthesized and output.
  • FIG. 3 is a block diagram showing an internal configuration of the spectrum envelope quantization unit of the speech encoding device according to Embodiment 2 of the present invention.
  • the configuration of the speech coding apparatus according to the present embodiment is the same as the configuration of the speech coding apparatus shown in FIG. 2 of Embodiment 1, and a description thereof will be omitted.
  • the two-dimensional orthogonal transformer 310 performs two-dimensional orthogonal transformation on the spectrum envelope curved surface in the time axis direction and the frequency axis direction.
  • the parameter overnight quantizer 302 quantizes the transform coefficients obtained by the two-dimensional orthogonal transform process in the two-dimensional orthogonal transformer 301.
  • the parameter overnight quantizer 302 quantizes only the coefficient information of the low frequency component.
  • FIG. 4 is a block diagram showing the internal configuration of the spectrum envelope quantization unit of the speech encoding device according to Embodiment 3 of the present invention.
  • the configuration of the speech coding apparatus according to the present embodiment is shown in FIG. 2 of Embodiment 1. Since the configuration is the same as the configuration of the speech encoding device shown, the description is omitted.
  • the model applicator 311 models a spectrum envelope curved surface and extracts model parameters.
  • This model is a model of a spectrum envelope surface in the time-frequency space.For example, as shown in Fig. 5, the application of the all-pole model to both sections on the time axis of the spectrum envelope surface and its application It can be modeled by interpolation.
  • the parameter overnight quantizer 302 quantizes the model parameter overnight extracted by the model applicator 311.
  • the quantization efficiency of the spectral envelope surface can be improved, and high quality speech decoding can be achieved even when transmitting information at a low bit rate. can do.
  • FIG. 6 is a block diagram showing the internal configuration of the spectrum envelope quantization unit of the speech encoding device according to Embodiment 4 of the present invention.
  • the configuration of the speech coding apparatus according to the present embodiment is the same as the configuration of the speech coding apparatus shown in FIG. 2 of Embodiment 1, and a description thereof will be omitted.
  • the time axis orthogonal transformer 3221 performs orthogonal transformation in the time axis direction on the spectrum envelope curved surface.
  • the model applicator 3 1 1 applies a model according to the order on the time axis to the orthogonally transformed time axis conversion coefficient, and extracts parameters.
  • the parameter overnight quantizer 3 02 quantizes the model parameters extracted by the model applicator 3 1 1.
  • FIG. 7 is a block diagram showing the internal configuration of the spectrum envelope quantization unit of the speech encoding device according to Embodiment 5 of the present invention.
  • the configuration of the speech coding apparatus according to the present embodiment is the same as the configuration of the speech coding apparatus shown in FIG. 2 of Embodiment 1, and a description thereof will be omitted.
  • the time-axis orthogonal transformer 331 1 performs orthogonal transformation on the spectrum envelope curved surface in the time-axis direction, and models the orthogonally transformed time-axis transformation coefficient. Classify into those that do and those that do not. As this classification method, for example, there is a method of applying the all-pole model because the coefficient of the 0th order on the time axis is a spectrum envelope obtained by averaging the spectrum envelope surface, and not applying the model to the other coefficients.
  • the model applicator 3 1 1 applies a model corresponding to the order on the time axis to a part of the orthogonally transformed time axis transform coefficients, and extracts parameters.
  • the frequency axis orthogonal transformer 333 performs orthogonal transformation in the frequency axis direction for the time axis conversion coefficient to which no model is applied.
  • the parameter quantizer 302 quantizes the model parameters extracted by the model applicator 311 and the transform coefficients output from the frequency-axis orthogonal transformer 332.
  • FIG. 8 is a block diagram showing an internal configuration of a model applicator of a speech encoding device according to Embodiment 6 of the present invention.
  • model applicator 3 11 is the one shown in any of the above-described third to fifth embodiments.
  • the model parameter overnight estimator 4 01 applies the model to the input signal and Extract the evening.
  • the input signal is modeled using an all-pole model in consideration of the speech generation process.
  • the model cannot represent the zero included in the signal, and the model causes analysis distortion.
  • model error estimator 402 estimates the analysis distortion generated when applying the model and outputs it to the parameter quantizer.
  • FIG. 9 is a block diagram showing an internal configuration of a parameter quantizer in a speech coding apparatus according to Embodiment 7 of the present invention.
  • parameter quantizer 302 according to the present embodiment is as described in any of Embodiments 2 to 5 above.
  • the weight calculator 501 determines the quantization sensitivity for each quantization target value using the fundamental frequency information.
  • the fundamental frequency information For example, an example of a method of determining the quantization sensitivity in the weight calculator 501 will be described.
  • the spectrum envelope surface is cut out according to the fundamental frequency and connected on the time axis to generate a decoded speech.
  • the harmonic amplitude value of the fundamental frequency is important information as compared with other spectral amplitude values. Therefore, a weight coefficient surface is generated by weighting the harmonic amplitude value at the extracted spectrum envelope position.
  • the quantization sensitivity is determined for each quantization target value by performing conversion using the same method as that for obtaining the quantization target value and calculating a weight coefficient in the quantization target parameter space.
  • the weight calculator 502 uses the spectral envelope information for each quantization target value. Determine the quantization sensitivity.
  • Determine the quantization sensitivity an example of a method of determining the quantization sensitivity in the weight calculator 502 will be described.
  • the quantization sensitivity is determined for each quantization target value by performing conversion using the same method as that used to obtain the quantization target value and calculating the weighting coefficient in the parameter space to be quantized. . Since the adaptation method of the quantizer in the weight calculator 502 is also required in the decoding process, it is necessary to use the spectrum envelope information quantized in the previous frame in order to synchronize with the decoding process. desirable.
  • the statistic accumulator 503 stores a statistic for each quantization target value obtained in advance.
  • the quantization generator 504 includes the quantization sensitivity for the quantization target value output from the weight calculator 501 and the weight calculator 502, and the statistics stored in the statistic storage 503. Design quantizer from quantity.
  • the quantization step width is determined based on the variance and the quantization sensitivity.
  • the quantization sensitivity is large. That is, the quantization step width is set to be smaller than the quantization target value that is easily affected by the quantization error.
  • the quantizer 505 quantizes the value to be quantized based on the design result of the quantization generator 504.
  • the quantization sensitivity for each quantization target value is determined using two pieces of information of the fundamental frequency information and the spectrum envelope information.
  • the quantization sensitivity may be determined using one of the two pieces of information to design a quantizer.
  • FIG. 10 is a block diagram showing the internal configuration of the parameter quantizer in the speech coding apparatus according to Embodiment 8 of the present invention.
  • parameter quantizer 302 according to the present embodiment is as described in any of Embodiments 2 to 5 above.
  • the error scale determiner 511 adaptively determines the quantization error scale on the spectrum envelope using the fundamental frequency information.
  • the error scale determiner 512 adaptively determines a quantization error scale on the spectrum envelope using the spectrum envelope information.
  • the error scale synthesizer 5 13 synthesizes the error scales obtained by the error scale determiner 5 11 and the error scale determiner 5 12 into one error scale.
  • the codebook 5 14 stores quantized values.
  • the spectrum envelope constructor 515 converts the quantized values stored in the codebook 514 into a spectrum envelope curved surface.
  • the spectrum envelope constructor 516 converts the quantization target value into a spectrum envelope curved surface.
  • the error calculator 517 is based on the error scale output from the error scale synthesizer 513, and the spectral envelope surface and the spectral envelope formed by the spectral envelope composer 515 are used. Calculate the error with respect to the spectrum envelope curved surface composed by the constructors 516.
  • the code selector 518 selects a code corresponding to the quantized value with the smallest error from the codebook 514 and outputs the selected code.
  • the error of the spectrum envelope curved surface is calculated on the time-frequency plane using the error scale at the time of quantization adapted to the fundamental frequency and the spectrum envelope information. Objective quantization distortion and auditory distortion can be reduced.
  • both the fundamental frequency and the spectrum envelope information are used.
  • the quantization error scale on the spectrum envelope is determined by using the above method, the quantization error scale may be determined for any one of them, and the error may be calculated.
  • FIG. 11 is a block diagram showing an internal configuration of a parameter quantizer in the speech coding apparatus according to Embodiment 9 of the present invention.
  • parameter quantizer 302 according to the present embodiment is as described in any of Embodiments 2 to 5 above.
  • the error function determiner 521 adaptively determines a quantization error weighting function on the spectrum envelope using the fundamental frequency information.
  • the error function determiner 522 adaptively determines a quantization error weight function on the spectrum envelope using the spectrum envelope information.
  • the error function synthesizer 5 2 3 synthesizes the quantization error weight function obtained by the error function determiner 5 2 1 and the error function determiner 5 2 2 into one error function.
  • the error function converter 524 defines an error measure over a quantization parameter for converting the quantization error weight function output from the error function synthesizer 523.
  • the codebook 525 stores quantized values.
  • the error calculator 526 calculates an error between the quantization target value and the quantized value stored in the codebook 525 based on the error measure output from the error function converter 524.
  • the code selector 527 selects a code corresponding to the quantized value that minimizes the error from the codebook 525 and outputs the selected code.
  • the objective of the synthesized speech signal can be reduced with a small amount of processing. Quantization distortion and audible distortion can be reduced.
  • the quantization error weight function on the spectrum envelope is determined for both the fundamental frequency and the spectrum envelope information, but the quantization error weight function is determined for one of them, and the error May be calculated. (Embodiment 10)
  • FIG. 12 is a block diagram showing the internal configuration of the spectrum envelope configuration unit of the speech decoding device according to Embodiment 10 of the present invention.
  • the configuration of the speech decoding apparatus according to the present embodiment is the same as the configuration of the speech decoding apparatus shown in FIG.
  • an envelope surface is generated on the decoding side by complementing a parameter that has not been received by using a parameter value that has been statistically obtained in advance.
  • the parameter storing unit 601 stores the parameter value statistically obtained in advance corresponding to each parameter not to be quantized. I have.
  • the spectrum envelope generator 602 generates a spectrum envelope curved surface based on the input spectrum envelope information.
  • the speech coding apparatus As described above, according to the speech coding apparatus, the speech decoding apparatus, and the speech coding / decoding method of the present invention, the spectrum envelope information and the sound source information are completely separated from each other, and the quantum of the spectrum envelope information is obtained. It is possible to realize speech codec processing that is not affected by quantization accuracy, and to realize highly efficient speech codec processing through a high-efficiency quantization method of spectrum envelope information that is effective in an analysis-synthesis model. Therefore, even when information is transmitted at a low bit rate, high-quality speech decoding can be realized.
  • the present specification is based on Japanese Patent Application No. 11-27551 19 filed on Sep. 28, 1999. This content is included here.
  • INDUSTRIAL APPLICABILITY The present invention is suitable for use in a communication terminal apparatus, which is a base station apparatus of a wireless communication system that performs wireless communication of voice data.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)

Abstract

Un analyseur vocal (101) intégré dans un codeur vocal (100) extrait d'un signal vocal d'entrée la fréquence fondamentale et les informations d'enveloppe spectrale. Un quantificateur (102) de fréquence fondamentale quantifie la fréquence fondamentale. Un générateur matriciel (103) dérive une enveloppe spectrale des informations d'enveloppe spectrale et un quantificateur (104) d'enveloppe spectrale quantifie l'enveloppe spectrale. Un multiplexeur (105) multiplexe l'enveloppe spectrale quantifiée et la fréquence fondamentale quantifiée pour la transmission. Dans un décodeur vocal (200), un dispositif (202) de restauration de l'enveloppe spectrale rétablit l'enveloppe spectrale quantifiée à partir des informations d'enveloppe spectrale et un synthétiseur vocal (203) extrait l'enveloppe spectrale sur la base des informations de fréquence fondamentale pour synthétiser la voix décodée. On peut ainsi effectuer un décodage de la parole de haute qualité pour une transmission à faible débit binaire.
PCT/JP2000/006542 1999-09-28 2000-09-25 Codeur vocal, decodeur vocal et procede de codage et de decodage de la parole WO2001024164A1 (fr)

Priority Applications (2)

Application Number Priority Date Filing Date Title
EP00961220A EP1132891A1 (fr) 1999-09-28 2000-09-25 Codeur vocal, decodeur vocal et procede de codage et de decodage de la parole
AU73212/00A AU7321200A (en) 1999-09-28 2000-09-25 Voice encoder, voice decoder, and voice encoding and decoding method

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
JP11/275119 1999-09-28
JP27511999A JP3360046B2 (ja) 1999-09-28 1999-09-28 音声符号化装置、音声復号化装置及び音声符復号化方法

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Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US10004506B2 (en) 2011-05-27 2018-06-26 Ethicon Llc Surgical system

Families Citing this family (2)

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Publication number Priority date Publication date Assignee Title
KR100738109B1 (ko) * 2006-04-03 2007-07-12 삼성전자주식회사 입력 신호의 양자화 및 역양자화 방법과 장치, 입력신호의부호화 및 복호화 방법과 장치
JP5799824B2 (ja) 2012-01-18 2015-10-28 富士通株式会社 オーディオ符号化装置、オーディオ符号化方法及びオーディオ符号化用コンピュータプログラム

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JPH01227200A (ja) * 1988-03-07 1989-09-11 Fujitsu Ltd ホルマント抽出装置

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JPH01227200A (ja) * 1988-03-07 1989-09-11 Fujitsu Ltd ホルマント抽出装置

Non-Patent Citations (1)

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Title
HIDEKI KAWAHARA: "A high quality speech analysis, modification and synthesis method STRAIGHT: Insights on auditory scene analysis", ACOUSTICAL SOCIETY OF JAPAN, vol. 9, 17 September 1997 (1997-09-17), pages 189 - 192, XP002933376 *

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US10004506B2 (en) 2011-05-27 2018-06-26 Ethicon Llc Surgical system

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AU7321200A (en) 2001-04-30
EP1132891A1 (fr) 2001-09-12
JP3360046B2 (ja) 2002-12-24
JP2001100798A (ja) 2001-04-13

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