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WO2007069369A1 - Dispositif de traitement de signal et méthode de traitement de signal - Google Patents

Dispositif de traitement de signal et méthode de traitement de signal Download PDF

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Publication number
WO2007069369A1
WO2007069369A1 PCT/JP2006/315932 JP2006315932W WO2007069369A1 WO 2007069369 A1 WO2007069369 A1 WO 2007069369A1 JP 2006315932 W JP2006315932 W JP 2006315932W WO 2007069369 A1 WO2007069369 A1 WO 2007069369A1
Authority
WO
WIPO (PCT)
Prior art keywords
signal
signal processing
bits
output
filter
Prior art date
Application number
PCT/JP2006/315932
Other languages
English (en)
Japanese (ja)
Inventor
Yasushi Sato
Original Assignee
Kyushu Institute Of Technology
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 Kyushu Institute Of Technology filed Critical Kyushu Institute Of Technology
Publication of WO2007069369A1 publication Critical patent/WO2007069369A1/fr

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Classifications

    • G—PHYSICS
    • G11—INFORMATION STORAGE
    • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
    • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
    • G11B20/10—Digital recording or reproducing
    • G11B20/10527—Audio or video recording; Data buffering arrangements
    • G—PHYSICS
    • G11—INFORMATION STORAGE
    • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
    • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
    • G11B20/10—Digital recording or reproducing
    • G11B20/10527—Audio or video recording; Data buffering arrangements
    • G11B2020/10537—Audio or video recording
    • G11B2020/10546—Audio or video recording specifically adapted for audio data

Definitions

  • the present invention relates to a signal processing device and a signal processing method suitable for use when, for example, expanding the number of bits of a quantized acoustic signal. Specifically, the number of bits can be expanded well while blocking the harmonic components due to the expanded bits.
  • CD compact disc
  • DZA Digital to Analog
  • the present invention has been made in view of such problems, and an object of the present invention is to reduce an information signal quantized with a small number of bits, such as recording on a compact disc.
  • Bi In addition to expanding the number of mobile stations, the deterioration of information such as the occurrence of noise during the expansion is prevented.
  • the present application provides a signal processing device for extending the number of bits of a quantized signal.
  • Filter means that expands the number of bits of the quantized original signal and blocks the harmonic component due to the extended bits, and the output of the filter means is a signal in which any lower bits of the extension are added to the original signal.
  • the filter unit includes a variable filter including a delay unit and a correction unit that controls a coefficient thereof, and the correction unit is determined by a tendency of the original signal.
  • the error ⁇ between the output of the low-pass filter whose cutoff frequency is controlled and the output of the variable filter is calculated according to the external signal generated, and the coefficient of the variable filter is controlled using a correction algorithm that minimizes the error ⁇ . It is characterized by that.
  • the moving correction average value of the peak is calculated with a time constant longer than the quantization sampling with respect to the output of the first arithmetic means, and the calculated movement is calculated.
  • the amplitude of the output of the first calculation means is controlled by the corrected average value, and the second calculation means adds to the output of the filter means.
  • the original signal is a signal obtained by quantizing an acoustic signal.
  • the original signal is a signal obtained by quantizing the acoustic signal, and the tendency of the original signal includes the distinction between the conversation of the acoustic signal and the music, and the song or music. It is characterized by being Jeyangle.
  • a signal processing method for extending the number of bits of a quantized signal comprising: The number of bits is expanded and the harmonic components due to the expanded bits are cut off. The signal with the harmonic component cut off is also subtracted by subtracting the signal power obtained by adding any lower bits of the extension to the original signal. The number of bits is expanded by adding the signal to the signal whose harmonic components are cut off. It is characterized by obtaining a quantized signal.
  • the number of bits of an information signal quantized with a small number of bits, such as recording on a compact disc is expanded, and noise is generated during the expansion. Therefore, it is possible to provide a signal processing apparatus and a signal processing method that can be applied as they are to recording on a conventional compact disc or the like.
  • FIG. 1 is a block diagram showing a configuration of an embodiment of an acoustic signal processing device to which a signal processing device and a signal processing method according to the present invention are applied.
  • FIG. 2 is a block diagram showing the configuration of the main part.
  • FIG. 3 is a diagram for explaining that.
  • FIG. 4 is a diagram for explaining that.
  • FIG. 5 is a diagram for explaining this.
  • FIG. 6 is a diagram for explaining that.
  • FIG. 7 is a diagram for explaining that.
  • FIG. 8 is a waveform diagram for explaining the effect.
  • FIG. 1 is a block diagram showing a configuration of an embodiment of an acoustic signal processing apparatus to which a signal processing apparatus and a signal processing method according to the present invention are applied. .
  • a 16-bit PCM signal is supplied to the input terminal 1.
  • a signal from the input terminal 1 is supplied to the variable filter 3 through a predetermined delay means 2.
  • a signal from the input terminal 1 is supplied to the computing means 5 through the low-pass filter (LPF) 4, and a difference from the output of the variable filter 3 is calculated and supplied to the coefficient correcting means 6. Then, the coefficient obtained by the coefficient correcting means 6 is supplied to the variable filter 3.
  • LPF low-pass filter
  • the signal from the input terminal 1 is supplied to the lower bit tracking circuit 8 through the predetermined delay means 7, and, for example, 8 bits of the value 0 are inserted into the lower order of the 16-bit input signal to obtain 24 bits. It is said that This 24-bit signal is supplied to the arithmetic means 9 and the output of the variable filter 3 is output. The difference from the force is calculated. Then, this difference value (24 bits) is added to the output of the variable filter 3 by the addition circuit 11 through the weighting circuit 10 and taken out to the output terminal 12.
  • the difference value from the calculation means 9 is supplied to the peak detection circuit 13, and the peak detected signal is supplied to the limiter circuit 15 through the low-pass filter (LPF) 14, and the signal corresponding to the lower 8 bits is deleted. Is done.
  • the weighting circuit 10 performs weighting based on the output of the limiter circuit 15. As a result, the number of bits of the information signal is expanded and correction processing for information deterioration due to the expansion is performed.
  • variable filter 3 is an FIR type digital filter as shown in FIG. 2, for example, and is supplied to a plurality of unit delay means Z-1 in which input signals are connected in cascade.
  • the outputs of each stage are added by the adding means (+) through the weighting circuit h.
  • This added value is taken out as an output value y (n), and a difference from the target value d (n) is calculated by the calculation means 5 and the weighting circuit h is calculated by the coefficient correction means 6 so that this difference value is minimized.
  • the coefficient of is obtained.
  • the output value y (n) is expressed as follows.
  • the coefficient correction means 6 determines the coefficient so that is minimized. [0021] Therefore, the condition that the mean square error e is minimum is
  • the low-pass filter (LPF) output is a signal obtained by deleting the bits below the original quantization with a limiter from the difference signal between the input signal and the output of the low-pass filter (LPF). Add to.
  • the waveform is smoothed as shown in (b) of FIG. 5 by the signal power low pass filter (LPF) quantized as shown in (a) of B of FIG.
  • the high-frequency component deleted by the low-pass filter (LPF) is added, so that this kind of sound can be prevented.
  • the low-pass filter (LPF) band has high sound quality of 24 bits, and high sound is added through as it is.
  • the quantization noise is high even in the case of force continuity signals that are extracted by an adaptive filter (LMS) focusing on the continuity signals.
  • LMS adaptive filter
  • the low-pass filter (LPF) is used to remove high-frequency quantization noise and correct the adaptive filter (LMS) coefficient.
  • the cutoff frequency of the low-pass filter (LPF) is generally about 1 OkHz when the sampling frequency force is 4.1 kHz, that is, half the frequency of Nyquist.
  • the cut-off frequency of the low-pass filter is the original acoustic signal conversation and music.
  • changes such as narrowing the passband can be made.
  • better sound quality improvement can be performed according to the genre of music.
  • the music information of a compact disc is 16 bits. Therefore, an adaptive filter is formed by the delay unit + variable filter unit, and the variable filter is controlled by the least square method so as to approach the signal source expanded to 24 bits by removing harmonics from the reference signal through LPF. As a result, for continuous (highly reproducible signals) due to the delay, the filtering effect is used to interpolate from 16 bits to 24 bits. However, since this sound is a reproducible signal, it becomes a muffled sound.
  • the lower 8 bits of the source signal delayed by the variable filter processing are added (the lower 8 bits are 0), and the difference from the adaptive filter is calculated.
  • This difference signal can be expanded to 24 bits by not adding it in the case of a signal with a magnitude less than the force quantization error, which is a high-frequency signal, and the conventional signal is added to the high-frequency information. Therefore, it can solve the problem.
  • the peak detection in FIG. 1 is a pop-up because a scene of sudden addition or supple force occurs if it is determined whether or not the force to add is based on the magnitude of the quantization error alone. There is a risk of noise. Therefore, the peak error is calculated as 1.0 at maximum, and the amount of error is multiplied by the amount of error, so the high frequency signal can be added smoothly, eliminating pop noise. be able to.
  • the sound source of a compact disk (CD) is 16 bits, which is insufficient in terms of high sound quality.
  • 24-bit DACs can be used at low cost, so there are many 24-bit sound sources. Therefore, the present invention has an effect of improving the sound quality by predicting and extending the lower 8 bits, and can easily cope with a conventional CD.
  • FIG. 8 the effect of the present invention is described with reference to waveforms.
  • a in Fig. 8 shows the waveform immediately after the start of calculation, and here the quantization noise is generated.
  • B in the figure shows the waveform after 100 ms from the start of the calculation, which shows that the quantization noise is reduced.
  • the number of bits of the information signal quantized with a small number of bits, for example, recording on a compact disc is expanded, and information such as the occurrence of noise is generated during the expansion.
  • a signal processing apparatus and a signal processing method that can be directly applied to recording on a conventional compact disc or the like.
  • the signal processing device expands the number of bits of the quantized signal, and extends the number of bits of the quantized original signal and uses the expanded bits.
  • Filter means for cutting off harmonic components first calculation means for subtracting the signal power obtained by adding an arbitrary lower bit of the extension to the original signal, and filtering the output of the first calculation means
  • the second calculation means for adding to the output of the means the number of bits can be expanded well.
  • the signal processing method extends the number of bits of the quantized signal, and extends the number of bits of the quantized original signal.
  • the harmonic component due to the bit is cut off, the signal with the harmonic component cut off is also subtracted from the original signal, and the signal power with any lower bits added to the extension is subtracted, and the subtracted signal is subtracted from the harmonic component.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Signal Processing (AREA)
  • Signal Processing For Digital Recording And Reproducing (AREA)

Abstract

L’objet est d’augmenter le nombre de bits de signaux d’information tout en évitant une dégradation du signal à cette occasion. Pour cela, un signal d’une borne d’entrée (1) est fourni à un filtre variable (3) par un moyen de retard prédéterminé (2) puis à un moyen opérationnel (5) par un filtre passe bas arbitraire (4) et la différence par rapport à la sortie du filtre variable (3) est calculée et fournie à un moyen de correction de coefficient (6). Le coefficient déterminé par le moyen de correction de coefficient (6) est fourni au filtre variable (3). En outre, le signal de la borne d’entrée (1) est fourni par un moyen de retard prédéterminé (7) et un circuit d’addition des bits de poids le plus faible (8) à un moyen opérationnel (9) et sa différence avec la sortie du filtre variable (3) est calculée. Cette différence est ajoutée par un circuit de pondération (10) à la sortie d’un circuit d’addition (11) et extraite par une borne de sortie (12). En outre, la différence du moyen opérationnel (9) est fournie à un circuit limiteur (15) par un circuit de détection de crête (13) et un filtre passe bas (14) et le signal correspondant à la valeur la moins significative est supprimé, si bien que la pondération est réalisée au niveau du circuit de pondération (10).
PCT/JP2006/315932 2005-12-14 2006-08-11 Dispositif de traitement de signal et méthode de traitement de signal WO2007069369A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
JP2005360789A JP5103606B2 (ja) 2005-12-14 2005-12-14 信号処理装置
JP2005-360789 2005-12-14

Publications (1)

Publication Number Publication Date
WO2007069369A1 true WO2007069369A1 (fr) 2007-06-21

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WO (1) WO2007069369A1 (fr)

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2009104710A1 (fr) * 2008-02-21 2009-08-27 株式会社ケンウッド Appareil, programme et procédé de conversion des données
JP2010114553A (ja) * 2008-11-05 2010-05-20 Mitsubishi Electric Corp 音声信号処理装置及び方法
EP2244261A3 (fr) * 2009-04-13 2011-12-28 Panasonic Corporation Extension de la profondeur de bits de données audionumériques

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH03267808A (ja) * 1990-03-16 1991-11-28 Toshiba Corp ディジタル信号処理回路
JPH05304474A (ja) * 1991-05-18 1993-11-16 Nippon Columbia Co Ltd ディジタルアナログ変換装置
JPH08223036A (ja) * 1995-02-16 1996-08-30 Hitachi Ltd 周波数シンセサイザ
JPH0964750A (ja) * 1995-08-18 1997-03-07 Victor Co Of Japan Ltd 音響信号処理装置

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH03267808A (ja) * 1990-03-16 1991-11-28 Toshiba Corp ディジタル信号処理回路
JPH05304474A (ja) * 1991-05-18 1993-11-16 Nippon Columbia Co Ltd ディジタルアナログ変換装置
JPH08223036A (ja) * 1995-02-16 1996-08-30 Hitachi Ltd 周波数シンセサイザ
JPH0964750A (ja) * 1995-08-18 1997-03-07 Victor Co Of Japan Ltd 音響信号処理装置

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2009104710A1 (fr) * 2008-02-21 2009-08-27 株式会社ケンウッド Appareil, programme et procédé de conversion des données
JP2009198834A (ja) * 2008-02-21 2009-09-03 Kenwood Corp データ変換装置、プログラム、及び方法
US8368569B2 (en) 2008-02-21 2013-02-05 Kabushiki Kaisha Kenwood Data converting device, program and method
JP2010114553A (ja) * 2008-11-05 2010-05-20 Mitsubishi Electric Corp 音声信号処理装置及び方法
EP2244261A3 (fr) * 2009-04-13 2011-12-28 Panasonic Corporation Extension de la profondeur de bits de données audionumériques
US8443017B2 (en) 2009-04-13 2013-05-14 Panasonic Corporation Digital data processor

Also Published As

Publication number Publication date
JP5103606B2 (ja) 2012-12-19
JP2007166315A (ja) 2007-06-28

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