EP1619665A1 - Verfahren, Vorrichtung zur Sprachkodierung in einem mobilen Kommunikationsendgerät mittels PLP - Google Patents

Verfahren, Vorrichtung zur Sprachkodierung in einem mobilen Kommunikationsendgerät mittels PLP Download PDF

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Publication number
EP1619665A1
EP1619665A1 EP05015989A EP05015989A EP1619665A1 EP 1619665 A1 EP1619665 A1 EP 1619665A1 EP 05015989 A EP05015989 A EP 05015989A EP 05015989 A EP05015989 A EP 05015989A EP 1619665 A1 EP1619665 A1 EP 1619665A1
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EP
European Patent Office
Prior art keywords
signal
plp
coefficient
input signal
voiced
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EP05015989A
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English (en)
French (fr)
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EP1619665B1 (de
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Chan-Woo Kim
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LG Electronics Inc
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LG Electronics Inc
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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
    • G10L19/04—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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/93—Discriminating between voiced and unvoiced parts of speech signals

Definitions

  • the present invention relates to a coding of a mobile communications terminal, and particularly; to a voice coding apparatus and method using a Perceptual Linear Prediction (PLP).
  • PLP Perceptual Linear Prediction
  • mobile communications terminals have provided data communications using numbers, characters, symbols, and the like, and multimedia communications including various image signals as well as voice communications.
  • a plurality of terminal users receive radio channels allocated thereto from a system and transmit and receive required data using radio resources.
  • the radio channels have limited bandwidths in order for the plurality of users to use the radio channels at the same time, and accordingly a data bit rate of each user is deservedly limited.
  • a speech coding using a generic audio coding, a Pulse Code Modulation (PCM), and an Adaptive Delta Pulse Code Modulation(ADPCM) are effectively used at a high-bit rate over 16Kbps, and a Code Excited Linear Prediction (CELP) and other various variations are effectively used at a medium-bit rate at a range of 2.4Kbps to 16Kbps.
  • a coding method using LD-CELP, CS-ACELP, VSELP and MELP and a wideband speech coding can be used at the medium-bit rate.
  • LPC Linear Predictive Coding
  • RELP Residual Excited Linear Predictive
  • Cepstral vocoder have many advantages at a low-bit rate at a range of 75bps to 2.4Kbps.
  • Fig. 1 illustrates a structure of the related art LPC encoder.
  • the related art LPC encoder includes: a correlator 10 for calculating an autocorrelation value r x [n] of an input signal x[n]; an LP coefficient calculator 11for calculating an LP coefficient a L and a gain G by processing the autocorrelation value r x [n]; a V/UV determining unit 12 for determining whether the input signal x[n] is a voiced V signal or a unvoiced UV signal; a pitch calculator 13 for calculating a pitch P of the corresponding signal when the input signal x[n] is the voice V signal; a parameter coding unit 14 for outputting a bit stream by coding the LP coefficient an, the gain G and the pitch P received from the LP coefficient calculator 11 and the pitch calculator 13 according to a V/UV indication bit outputted from the V/UV determining unit 12.
  • the correlator 10 autocorrelates an input signal x[n].
  • the LP coefficient calculator 11 processes an autocorrelation value r x [n] calculated by the correlator 10 so as to calculate an LP coefficient a n and a gain G.
  • the V/UV determining unit 12 determines whether the input signal x[n] is a voiced V signal or a unvoiced UV signal to output a V/UV indication bit, and then outputs only the voiced V signal.
  • the pitch calculator 13 calculates a pitch P of the voiced V signal which is outputted from the V/UV determining unit 12.
  • the parameter coding unit 14 outputs a bit stream by coding (encoding by a low-bit rate) the LP coefficient a n , the gain G, and the pitch P received from the LP coefficient calculator 11 and the pitch calculator 13.
  • a controller processes the bit stream to thusly output it to a radio (wireless) unit (not shown).
  • the radio unit converts the signal outputted from the control unit into a radio (wireless) signal and transmits the converted radio signal.
  • a mobile communications terminal performs the LPC coding to transmit an audio signal by a low-bit rate.
  • a linear predication coefficient is generally used, which does not consider human auditory sensing features. Therefore, for the related art LPC coding operated using the low-bit rate, a compression efficiency is not very high (i.e., 1200Kbps to 2400Kbps) and good sound quality can not be obtained.
  • an object of the present invention is to provide a voice coding apparatus and method of a mobile communications terminal capable of improving compression efficiency and sound quality by performing an LPC coding using a PLP coefficient.
  • a Linear Predictive Coding (LPC) encoder of a mobile communications terminal comprising: a Perceptual Linear Prediction (PLP) coefficient calculator for calculating a PLP coefficient and a gain by processing an input signal; a V/UV determining unit for determining whether the input signal is a voiced signal or a unvoiced signal, and thusly outputting the determination signal and the voiced signal when the input signal is the voiced signal; a pitch calculator for calculating a pitch of the input signal outputted from the V/UV determining unit; and a parameter coding unit for performing a low-bit rate coding using the PLP coefficient, the gain, and the pitch on the basis of the determination signal.
  • PLP Perceptual Linear Prediction
  • a low-bit rate voice coding method of a mobile communications terminal comprising: calculating a Perceptual Linear Prediction (PLP) coefficient and a gain by processing an input signal; determining whether the input signal is a voiced signal and a unvoiced signal, and thereby outputting a determination bit value and the voiced signal when the input signal is determined as the voiced signal; calculating a pitch of the input signal outputted from a V/UV determining unit; and performing a low-bit rate coding using the PLP coefficient, the gain and the pitch on the basis of the determination bit value.
  • PLP Perceptual Linear Prediction
  • the voiced signal is a speech signal.
  • the PLP coefficient has about a 7 th degree for a 8 kHz sampling rate.
  • the present invention provides a low-bit rate voice coding using a Perceptual Linear Prediction (PLP) capable of performing a coding of a degree (an order) lower than that of a Linear Predictive Coding (LPC) in order to perform a voice coding having high compressibility.
  • PLP Perceptual Linear Prediction
  • LPC Linear Predictive Coding
  • the LP is classically well-known, so that a detailed derived formula therefor will not be described.
  • the LP basically refers to obtaining a LP coefficient a k so that a Mean Squared Error (MSE), namely, a value of e[n] can be a minimum value according to Formula (1) as follows.
  • MSE Mean Squared Error
  • the obtained LP coefficient a k has about 8 th to 12 th degrees (orders) for a 8 kHz sampling rate. Therefore, the obtained LP coefficient a k is used for various coding methods (e.g., LPC, CELP, MELP, RELP, etc) using a Linear Prediction (LP), which is disclosed in more detail in Speech coding and synthesis, Amsterdam, the Netherlands: Elsevier, 1995.
  • LPC Linear Prediction
  • the PLP was introduced on a paper of Hermansky in 1990 for the first time.
  • the PLP uses human auditory sensing features similar to the existing Mel-Frequency Cepstral Coefficient (MFCC). Therefore, the present invention performs a low-bit rate voice coding using the PLP coefficient in stead of using the LP coefficient upon performing the LPC for a low-bit rate.
  • MFCC Mel-Frequency Cepstral Coefficient
  • the present invention obtains spectrum using the PLP coefficient.
  • the PLP coefficient reflects a human auditory effect. Accordingly, in aspect of the MSE, a greater error may occur in the spectrum using the PLP coefficient than using the LP. However, the spectrum using the PLP coefficient may have a less error when considering the auditory effect. Also, for coefficient transmissions, in case of LPC, for a typical 8kHz sampling rate, transmissions of about a 10 th degree (order) are used, but for PLP, transmissions of about a 7 th degree (order) are used, thus the bit rate can be lowered.
  • Fig. 2 illustrates a construction of an LPC encoder using the PLP coefficient according to the present invention.
  • an LPC encoder using the PLP coefficient is constructed as same as the related art LPC encoder shown in Fig. 1, except of which the correlator 10 is not included and a PLP coefficient calculator 20 replaces the LP coefficient calculator 11.
  • the PLP coefficient calculator 20 processes a speech signal S[n] to calculate a PLP coefficient a P and a gain G in which the auditory effect is considered.
  • the PLP coefficient calculator 20 receives the speech signal S[n], so as to calculate the PLP coefficient a P and the gain G by sequentially performing operations shown in Fig. 3.
  • the PLP coefficient calculator 20 performs a fast Fourier transform (FFT) of the input signal, namely, the speech signal S[n].
  • FFT fast Fourier transform
  • a critical-bank integration and resampling processing is performed for the Fourier-transformed speech signal to thusly remove noise components from the speech signal S[n] by a frequency unit.
  • the PLP coefficient calculator 20 performs equalizing and loudness processing of the Fourier-transformed speech signal into sound components having magnitudes appropriate for human auditory sensing, and then the speech signal is matched with an output power to allow listening by humans.
  • the PLP coefficient calculator 20 When the power matching is completed, the PLP coefficient calculator 20 performs an inverse discrete Fourier transform of the corresponding speech signal to thereafter obtain a set of Linear equations from the corresponding speech signal. Therefore, the PLP coefficient calculator 20 performs a Cepstral Recursion processing for the set of Linear equations, and thus outputs Cepstral Coefficients of a PLP model, namely, the PLP coefficients ap. In other words, the PLP coefficient calculator 20 outputs to the parameter coding unit 23 a low degree (order) of the PLP coefficients a P and a gain G reflecting the human auditory sensing features as parameter values.
  • the V/UV determining unit 21 outputs a V/UV Indication bit and transfers the speech signal S[n] to the pitch calculator 22.
  • the pitch calculator 22 calculates a pitch P of the speech signal S[n].
  • the parameter coding unit 23 outputs a bit stream by coding (encoding by a low-bit rate) the V/UV Indication bit value, the PLP coefficient a P , the gain G and the pitch P received from the PLP coefficient calculator 20 and the pitch calculator 22.
  • a degree of the transmitted PLP coefficient a P is about a 7 th degree for a 8 kHz sampling rate.
  • a controller processes the bit stream and then outputs the processed bit stream to a radio (wireless) unit (not shown).
  • the radio unit converts the signal outputted from the controller into a radio signal (wireless signal) and transmits it.
  • the LPC is performed by using the PLP coefficient, and thus a compressibility can be improved and voice-grade signal can be transmitted by a more efficient low-bit rate.
  • a higher compressibility can be realized and a quality of signal with high sound quality can be expected by using the PLP coefficient as a parameter rather than using the existing LP coefficient.
  • the voice coding apparatus and method according to the present invention can be used for coding and decoding voice using a low-bit rate, or be used for a device which takes up a small area and performs a voice synthesis using PLP parameters.
  • the voice coding apparatus and method according to the present invention can be used for a speech coding for an application as much as a voice itself is not very important but enough to hear. Also, an effective voice conversation can be performed on the Internet which stores data by a high compressibility or requires a low-bit rate in an embedded system with a limited memory.
EP05015989A 2004-07-23 2005-07-22 Verfahren, Vorrichtung zur Sprachkodierung in einem mobilen Kommunikationsendgerät mittels PLP Not-in-force EP1619665B1 (de)

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Application Number Priority Date Filing Date Title
KR1020040057739A KR100619893B1 (ko) 2004-07-23 2004-07-23 휴대단말기의 개선된 저전송률 선형예측코딩 장치 및 방법

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EP1619665A1 true EP1619665A1 (de) 2006-01-25
EP1619665B1 EP1619665B1 (de) 2010-09-08

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EP (1) EP1619665B1 (de)
JP (1) JP2006039559A (de)
KR (1) KR100619893B1 (de)
CN (1) CN1737904A (de)
AT (1) ATE480852T1 (de)
DE (1) DE602005023385D1 (de)

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KR101475724B1 (ko) * 2008-06-09 2014-12-30 삼성전자주식회사 오디오 신호 품질 향상 장치 및 방법
KR20110001130A (ko) * 2009-06-29 2011-01-06 삼성전자주식회사 가중 선형 예측 변환을 이용한 오디오 신호 부호화 및 복호화 장치 및 그 방법
WO2023112226A1 (ja) * 2021-12-15 2023-06-22 株式会社Peco 動物患者用遠隔診察システム

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP1199812A1 (de) * 2000-10-20 2002-04-24 Telefonaktiebolaget Lm Ericsson Kodierung der akustischen Signale mit Verbesserung der Wahrnehmung
US20040128130A1 (en) * 2000-10-02 2004-07-01 Kenneth Rose Perceptual harmonic cepstral coefficients as the front-end for speech recognition

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20040128130A1 (en) * 2000-10-02 2004-07-01 Kenneth Rose Perceptual harmonic cepstral coefficients as the front-end for speech recognition
EP1199812A1 (de) * 2000-10-20 2002-04-24 Telefonaktiebolaget Lm Ericsson Kodierung der akustischen Signale mit Verbesserung der Wahrnehmung

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
"SPEECH CODING AND SYNTHESIS", 1995, ELSEVIER, article "Linear Prediction (LP), which is disclosed in more detail"
GUNAWAN W ET AL: "PLP coefficients can be quantized at 400 bps", 2001 IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING. PROCEEDINGS. (ICASSP). SALT LAKE CITY, UT, MAY 7 - 11, 2001, IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING (ICASSP), NEW YORK, NY : IEEE, US, vol. VOL. 1 OF 6, 7 May 2001 (2001-05-07), pages 77 - 80, XP010803089, ISBN: 0-7803-7041-4 *

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JP2006039559A (ja) 2006-02-09
ATE480852T1 (de) 2010-09-15
KR20060008078A (ko) 2006-01-26
EP1619665B1 (de) 2010-09-08
DE602005023385D1 (de) 2010-10-21
KR100619893B1 (ko) 2006-09-19
CN1737904A (zh) 2006-02-22

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