WO2004040552A1 - Transcodeur et procede de conversion par codeur - Google Patents

Transcodeur et procede de conversion par codeur Download PDF

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Publication number
WO2004040552A1
WO2004040552A1 PCT/JP2003/012859 JP0312859W WO2004040552A1 WO 2004040552 A1 WO2004040552 A1 WO 2004040552A1 JP 0312859 W JP0312859 W JP 0312859W WO 2004040552 A1 WO2004040552 A1 WO 2004040552A1
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WO
WIPO (PCT)
Prior art keywords
signal
gain
unit
voiced
output
Prior art date
Application number
PCT/JP2003/012859
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English (en)
Japanese (ja)
Inventor
Kazunori Ozawa
Original Assignee
Nec Corporation
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 Nec Corporation filed Critical Nec Corporation
Priority to CA002504174A priority Critical patent/CA2504174A1/fr
Priority to DE60321712T priority patent/DE60321712D1/de
Priority to EP03751372A priority patent/EP1564723B1/fr
Priority to AU2003271119A priority patent/AU2003271119A1/en
Publication of WO2004040552A1 publication Critical patent/WO2004040552A1/fr
Priority to US11/118,346 priority patent/US7486719B2/en
Priority to HK05109774A priority patent/HK1077913A1/xx

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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
    • G10L19/16—Vocoder architecture
    • G10L19/173—Transcoding, i.e. converting between two coded representations avoiding cascaded coding-decoding
    • 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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/038—Speech enhancement, e.g. noise reduction or echo cancellation using band spreading techniques

Definitions

  • the present invention relates to a transcoder for mutually converting a code according to a first encoding method and a code according to a second encoding method, and when converting a first code into a second code, a frequency band of a signal.
  • Transcoder that expands and converts
  • a method is known in which the frequency band of an audio signal encoded and reproduced at a low bit rate is extended on the receiving side without transmitting auxiliary information for band extension from the transmitting side (for example, see Non-Patent Document 1). ).
  • Non-Patent Document 1 P. Jax, P. Vary, "Wideband extension of telephone speech using hidden markov model,” Pro IEEE Speech Coding Workshop, pp. 133-13, 2000
  • the receiving side searches for filter coefficients after band expansion using an HMM (Hidden Markov Model).
  • the band of the signal is extended when converting from the first code to the second code.
  • Non-Patent Document 1 The conventional method described in the above-mentioned document by P. Jax and P. Vary (Non-Patent Document 1) requires modeling of the spectral envelope of wideband speech and the filter coefficients by HMM, as follows. There were serious problems.
  • a main object of the present invention is to provide a transcoder for mutually converting between a code according to the first encoding method and a code according to the second encoding method, when converting from the first code to the second code. It is another object of the present invention to provide a transcoder and a code conversion method capable of performing band expansion conversion with good sound quality with a relatively small amount of operation when converting a signal after expanding the signal band.
  • a transcoder is a transcoder for mutually converting between a code according to a first encoding method and a code according to a second encoding method, and according to the first aspect.
  • a spectrum parameter calculation unit that receives a code coded by the first coding method and decodes the data by the first coding method to calculate a spectrum parameter representing the spectrum characteristics.
  • a noise generation unit that generates a noise signal; a coefficient calculation unit that calculates a fill coefficient after shifting the frequency of the spectrum parameter; and a gain unit that provides an appropriate gain to the output of the noise generation unit
  • a synthesis filter unit that passes an output of the gain unit through a synthesis filter configured using the coefficient to reproduce a band-extended signal; and converts a sampling frequency of the input signal
  • An adder for adding and outputting the output signal of the data section is configured to output a second code by encoding the output signals of the adders second encoding method.
  • the transcoder according to the present invention is a transcoder for mutually converting between a code according to a first coding scheme and a code according to a second coding scheme
  • a spectrum parameter calculation unit that receives a code coded by the coding scheme, decodes the code by the first coding scheme, and calculates a spectrum parameter representing a spectrum characteristic; and a pitch period from the input signal.
  • an adaptive codebook section for generating an adaptive codebook component based on the pitch period and a past sound source signal; a noise generating section for generating a noise signal; and shifting the frequency of the spectral parameter over and over.
  • a coefficient calculation unit for obtaining a filter coefficient in step (1), and an appropriate gain given to at least one of the output of the noise generation unit and the output of the adaptive codebook unit, followed by adding A gain section for outputting a No., synthetic fill constructed using the coefficients
  • a synthesis filter for inputting the sound source signal in the evening and reproducing the band extension signal
  • an adder for converting a sampling frequency of the reproduction signal and adding an output signal of the synthesis filter to output the converted signal
  • the output signal of the adder is encoded by a second encoding method to output a second code.
  • a transcoder is a transcoder for mutually converting between a code according to a first encoding scheme and a code according to a second encoding scheme
  • a spectrum parameter calculation unit that receives a code coded according to a coding scheme, decodes the code according to a first coding scheme, and calculates a spectrum parameter representing a spectrum characteristic, and calculates a pitch period from the input signal.
  • An adaptive codebook unit that calculates and generates an adaptive codebook component based on the pitch period and a past sound source signal; a noise generation unit that generates a noise signal; and a frequency shift of the spectral parameter.
  • a coefficient calculation unit for obtaining a filter coefficient; and an appropriate gain given to at least one of the output of the noise generation unit and the output of the adaptive codebook unit, followed by addition.
  • a gain section that outputs a signal, a pitch pre-filter is applied to the sound source signal using the pitch period, and the pitch pre-filter output signal is input to a synthesis filter configured using the coefficients to generate a band extension signal.
  • a synthesis filter section for reproducing, an adder for converting the sampling frequency of the reproduction signal, and adding and outputting an output signal of the synthesis filter section, and a second encoding method for outputting the output signal of the adder. And outputs the second code.
  • the output of the adaptive codebook section may be passed through a low-pass filter having a predetermined power-off frequency.
  • a post-filter is configured using a weighting coefficient obtained by weighting the coefficient, and a band extension signal is reproduced by passing an output signal of the synthesis filter unit through the BOST filter. Good.
  • a transcoding method of a transcoder for mutually converting a code according to a first encoding method and a code according to a second encoding method, Decoding the code coded according to the first decoding method according to the first decoding method and outputting a decoded signal;
  • a step of decoding a code coded by the first coding method by a first decoding method and outputting a decoded signal a step of decoding a code coded by the first coding method by a first decoding method and outputting a decoded signal
  • a code encoded by the first encoding method is decoded by a first decoding method, and a decoded signal is output.
  • the sound source signal from the gain section is subjected to a prefilling process using the pitch period in a pitch prefilling process, and
  • the method may further include a step of inputting an output signal from the pitch prefill.
  • the output signal of the synthesis filter unit is added to a post-filter configured using a weighting coefficient obtained by weighting the filter coefficient from the counting calculation unit. May be input.
  • an output of a periodic signal generating unit that generates a periodic signal using a pitch period is supplied to a gain unit instead of an output signal from the adaptive codebook unit. You may.
  • the signal when a code according to the first encoding method is input, converted to a code according to the second encoding method, and output, the signal is expanded after expanding the band of the signal.
  • Performance A high-frequency signal is generated by processing with a small amount of computation, and a sampling signal of a narrow-band input signal is added to the converted signal to generate a band-extended signal (for example, a 7 kHz band).
  • an adaptive codebook signal is generated using a delay calculated from a narrowband input signal based on a past sound source signal in a high frequency part, and is multiplied by an appropriate gain to generate a noise signal.
  • a pitch prefilter is used for a sound source signal using a delay, or a post-filter is provided by weighting a coefficient from a coefficient calculation circuit to provide a post-filter.
  • FIG. 1 is a diagram showing a configuration of a first embodiment of the present invention.
  • FIG. 2 is a diagram showing a configuration of the second exemplary embodiment of the present invention.
  • FIG. 3 is a diagram showing a configuration of a third exemplary embodiment of the present invention.
  • FIG. 4 is a diagram showing a configuration of a fourth embodiment of the present invention.
  • FIG. 5 is a diagram showing a configuration of a fifth embodiment of the present invention.
  • FIG. 6 is a diagram showing a configuration of a modification of the second embodiment of the present invention.
  • the first code is coded for a narrow band input signal in the 4 kHz band, and the transcoder converts this to a signal in the 5 kHz band or 7 kHz band. It is assumed that the band is extended and then the second code is used to obtain the second code.
  • FIG. 1 is a block diagram showing a configuration of a first embodiment of a transcoder according to the present invention.
  • the transcoder includes a first decoding circuit 105 Addition of spectrum parameter calculation circuit 100, noise generation circuit 120, coefficient calculation circuit 130, synthesis filter circuit 170, sampling frequency conversion circuit 180 , A second encoding circuit 195, a voiced / unvoiced discriminating circuit 200, a gain adjusting circuit 310, and a gain circuit 140.
  • the first decoding circuit 105 receives a code according to the first encoding method, decodes the code according to the first decoding method, and outputs a decoded signal x (n).
  • the spectral parameter calculation circuit 100 divides the decoded signal x (n) into frames (for example, 10 ms), and calculates a spectrum parameter of a predetermined order P for each frame.
  • the spectral parameter is a parameter that represents the spectral outline of the audio signal for each frame, and a well-known LPC (linear predictive coding) analysis or the like can be used for this calculation.
  • LSP Line Spectrum Pairs
  • Non-Patent Document 2 Sugamura, Itakura "Speech Information Compression Using Line Spectrum Pair (LSP) Speech Analysis and Synthesis", IEICE Transactions, J64_A, pp. 599-606, 1981
  • the coefficient calculation circuit 130 receives the spectral parameter data output from the spectrum parameter calculation circuit 100 and converts it into a coefficient of a band-extended signal.
  • well-known methods such as a method of simply shifting the frequency of LSP to a higher frequency, a nonlinear conversion method, and a linear conversion method can be used.
  • the existing frequency band of the LSP is shifted to a higher frequency band, and then converted to a linear prediction coefficient of order P.
  • the noise generation unit 120 generates a noise signal with the average amplitude normalized to a predetermined level, a band-limited noise signal for a time length equal to the frame length, and outputs the noise signal to the gain circuit 140.
  • white noise is used as an example of the noise signal, but another noise signal may be used.
  • the voiced / unvoiced discrimination circuit 200 inputs the narrowband input signal x (n) and outputs the signal for each frame. Determine if the issue is voiced or unvoiced.
  • a voiced / unvoiced determination for example, for a narrowband input signal x (n), a normalized autocorrelation function D (T) up to a predetermined delay time m is calculated according to equation (1). The maximum value of (T) is obtained, and if the maximum value of D (T) is larger than a predetermined threshold value, it is determined that the voiced voice is used.
  • the voiced / unvoiced discrimination circuit 200 outputs the voiced / unvoiced discrimination information to the gain adjustment circuit 210.
  • N is the number of samples for calculating the normalized autocorrelation.
  • the gain adjustment circuit 310 inputs voiced / unvoiced discrimination information from the voiced / unvoiced discrimination circuit 200, adjusts the gain given to the noise signal according to voiced / unvoiced, and outputs the same to the gain circuit 140. .
  • the gain circuit 140 inputs the gain from the gain adjustment circuit 310, multiplies the output signal of the noise generation circuit 120 by the gain, and outputs the result to the synthesis filter circuit 170.
  • the synthesis filter circuit 170 inputs the output signal from the gain circuit 140, and further inputs a coefficient of a predetermined order from the coefficient calculation section 130 to configure a filter, thereby expanding the band. It outputs the high-frequency signal y (n) necessary for the conversion.
  • the sampling frequency conversion circuit 180 up-samples the narrowband input signal x (n) to a predetermined sampling frequency, and outputs an up-sampled signal s (n).
  • the adder 190 adds the output signal y (n) of the synthesis filter circuit 170 to the output signal s (n) of the sampling frequency conversion circuit 180, and finally a signal whose band is extended. Form z (n) and output.
  • the second encoding circuit 195 receives the output signal z (n) of the adder 190, performs encoding by a second encoding method, obtains a second code, and outputs the second code.
  • FIG. 2 is a block diagram showing a configuration of the second exemplary embodiment of the present invention.
  • the transcoder according to the second embodiment of the present invention includes a first decoding circuit 105, a spectrum parameter calculation circuit 100, an adaptive codebook circuit 110, and a noise generation circuit 1 2 0, coefficient calculation circuit 130, gain circuit 340, synthesis filter circuit 170, sampling frequency conversion circuit 180, adder 160, adder 190, A second encoding circuit 195, a voiced / unvoiced discrimination circuit 200, and a gain adjustment circuit 210.
  • the same elements as those in FIG. 1 are denoted by the same reference numerals.
  • differences from the first embodiment will be mainly described, and descriptions of the same elements as those in FIG. 1 will be omitted as appropriate.
  • the second embodiment of the present invention includes an adaptive code book circuit 110 and an adder 160 in addition to the configuration of FIG.
  • the voiced / unvoiced discriminating circuit 200 receives the narrow-band input signal x (n) and discriminates whether the signal for each frame is voiced or unvoiced.
  • a voiced / unvoiced determination for example, for a narrowband input signal x (n), a normalized autocorrelation function D (T) up to a predetermined delay time m is calculated according to the above equation (1).
  • the maximum value of D (T) is obtained, and if the maximum value of D (T) is larger than a predetermined threshold, it is determined as voiced, otherwise it is determined to be unvoiced, and the result of the determination is determined by the gain adjustment circuit 210.
  • the voiced / unvoiced discrimination circuit 200 supplies the value of T that maximizes the normalized self-correlation function D (T) to the adaptive codebook circuit 110 as the pitch period T in the voiced frame. .
  • the gain adjustment circuit 210 inputs voiced / unvoiced discrimination information from the voiced / unvoiced discrimination circuit 200, and adapts the adaptive code according to whether it is voiced or unvoiced. The gain of the book signal and the gain of the noise signal are adjusted and supplied to the gain circuit 340.
  • the gain circuit 340 inputs the gain from the gain adjustment circuit 210 and the adaptive code
  • the output signal of at least one of the book circuit 110 and the noise generation circuit 120 is multiplied by a gain and output to the adder 160.
  • the adder 160 outputs two types of signals output from the gain circuit 340 (two signals obtained by multiplying at least one output signal of the adaptive code book circuit 110 and the noise generation circuit 120 by a gain). And outputs the result to the synthesis filter circuit 170 and the adaptive codebook circuit 110.
  • the synthesis filter circuit 170 inputs the output signal of the adder 160 and further inputs a predetermined order coefficient (filter coefficient) from the coefficient calculator 130 to form a filter. Outputs the signal y (n) in the high frequency range necessary for band extension.
  • an adaptive codebook signal is generated using a delay calculated from a narrowband input signal based on a past sound source signal in a high frequency part, and is multiplied by an appropriate gain.
  • a band extension signal with good sound quality can be generated when periodicity is required for a high frequency signal such as a vowel.
  • a structure including a periodic signal generation circuit 115 instead of the adaptive code book circuit 110 of FIG. It is good also as a result.
  • the periodic signal generating circuit 115 receives the pitch period from the voiced / unvoiced discriminating circuit 200, generates a periodic signal using the pitch period, and outputs the periodic signal to the gain circuit 340.
  • the configuration other than the periodic signal generation circuit 115 is the same as that of the second embodiment.
  • FIG. 3 is a block diagram showing a configuration of the third exemplary embodiment of the present invention.
  • a transcoder according to a third embodiment of the present invention includes a first decoding circuit 105, a spectrum parameter calculation circuit 100, and an adaptive codebook circuit 110.
  • a noise generation circuit 120 , a coefficient calculation circuit 130, a gain circuit 300, a synthesis filter circuit 170, a sampling frequency conversion circuit 180, and an adder 190.
  • FIG. 3 the same or equivalent elements as those in FIGS. 1 and 2 are denoted by the same reference numerals. In the following, the differences from the second embodiment will be mainly described, and the same as FIGS. 1 and 2 will be described. Description of the elements is omitted.
  • a pitch prefill circuit 400 is provided.
  • the gain circuit 300 inputs the gain from the gain adjustment circuit 210, multiplies the output signal of the adaptive codebook circuit 110 and the noise generation circuit 120 by the gain, and adds and adds two types of signals. The result is output to the pitch pre-filter circuit 400.
  • the pitch pre-filter circuit 400 receives the delay T (pitch cycle) from the voiced / unvoiced discrimination circuit 200 and receives the sound source signal v (n) from the gain circuit 300 to obtain the following equation (3). And then output to the composite fill circuit 170 after performing the pitch prefilling.
  • a periodic signal generation circuit may be used instead of the adaptive codebook circuit 110.
  • the periodic signal generation circuit inputs a signal from the voiced / unvoiced discrimination circuit 200, calculates a pitch period, generates a periodic signal based on the pitch period, and outputs the signal to the gain circuit 300.
  • FIG. 4 is a block diagram showing a configuration of a fourth embodiment of the present invention.
  • a transcoder according to a fourth embodiment of the present invention includes a first decoding circuit 105, a spectral parameter overnight calculation circuit 100, and an adaptive codebook circuit 110.
  • a noise generation circuit 120 a coefficient calculation circuit 130, a gain circuit 340, an adder 160, a synthesis filter circuit 170, and a sampling frequency conversion circuit 180.
  • FIG. 4 the same or equivalent elements as those in FIG.
  • a low-pass filter circuit 500 having an output of the adaptive codebook circuit 110 as an input is provided.
  • the low-pass filter (LPF) circuit 500 passes the output signal of the adaptive code book circuit 110 to the gain circuit 340 after passing the low-pass signal according to equation (4) according to equation (4). I do.
  • the cut-off frequency of the low-pass filter circuit 500 is determined in advance, and may be, for example, 6 kHz.
  • h (n) indicates the impulse response of the low-pass filter
  • the symbol "" indicates the convolution operation.
  • a periodic signal generation circuit can be used instead of the adaptive codebook circuit 110.
  • a signal is input from the voiced / unvoiced discriminating circuit 200 to calculate a pitch period, a periodic signal based on the pitch period is generated, and output to the gain circuit 340.
  • FIG. 5 is a block diagram showing a configuration of a fifth exemplary embodiment of the present invention.
  • a transcoder includes a first decoding circuit 105, a spectral parameter overnight calculation circuit 100, and an adaptive codebook circuit 110.
  • a noise generation circuit 120 includes a coefficient calculation circuit 130, a gain circuit 300, a composite filter circuit 170, a sampling frequency conversion circuit 180, and an adder 190.
  • FIG. 5 the same or equivalent elements as those in FIG.
  • a post-fill 600 is further provided in addition to the configuration of the third embodiment.
  • a periodic signal generation circuit can be used instead of the adaptive codebook circuit 110, similarly to the modification of the second embodiment.
  • a signal is input from the voiced / unvoiced discriminating circuit 200 to calculate the pitch period, generate a periodic signal based on the pitch period, and output it to the gain circuit 300.

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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)
  • Quality & Reliability (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

L'invention concerne un transcodeur d'interconversion comprenant: un circuit de calcul de paramètre spectral (100) conçu pour calculer un paramètre spectral pour un signal x(n) obtenu par décodage d'un premier code; un circuit de calcul de coefficient (130) conçu pour recevoir un paramètre spectral et pour le convertir en un coefficient du signal soumis à l'extension de bande; un circuit générateur de bruit (120) conçu pour produire un signal de bruit dont la bande est limitée; un circuit de gain (140) conçu pour multiplier le signal de sortie du circuit de génération de bruit par un gain; un circuit à filtre de combinaison (170) conçu pour recevoir le signal de sortie provenant du circuit générateur de bruit (120) et le coefficient provenant du circuit de calcul de coefficient (130), et pour produire un signal bande haute y (n) pour une extension de bande; un circuit de conversion de fréquence d'échantillonnage (180) conçu pour produire un signal s(n) qui est le signal x(n) suréchantillonné à une fréquence d'échantillonnage prédéterminée; un additionneur (190) pour additionner le signal bande haute y (n) et le signal s (n) afin de former un signal à extension de bande z (n); et un second circuit de codage (195) conçu pour coder le signal à extension de bande z (n) d'après le second procédé de codage, et pour produire le résultat.
PCT/JP2003/012859 2002-10-31 2003-10-08 Transcodeur et procede de conversion par codeur WO2004040552A1 (fr)

Priority Applications (6)

Application Number Priority Date Filing Date Title
CA002504174A CA2504174A1 (fr) 2002-10-31 2003-10-08 Transcodeur et procede de conversion par codeur
DE60321712T DE60321712D1 (de) 2002-10-31 2003-10-08 Transkoder und kodierkonvertierungsverfahren
EP03751372A EP1564723B1 (fr) 2002-10-31 2003-10-08 Transcodeur et procede de conversion par codeur
AU2003271119A AU2003271119A1 (en) 2002-10-31 2003-10-08 Transcoder and coder conversion method
US11/118,346 US7486719B2 (en) 2002-10-31 2005-05-02 Transcoder and code conversion method
HK05109774A HK1077913A1 (en) 2002-10-31 2005-11-03 Transcoder and coder conversion method

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
JP2002-317204 2002-10-31
JP2002317204A JP4438280B2 (ja) 2002-10-31 2002-10-31 トランスコーダ及び符号変換方法

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US11/118,346 Continuation US7486719B2 (en) 2002-10-31 2005-05-02 Transcoder and code conversion method

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JP (1) JP4438280B2 (fr)
KR (1) KR100715014B1 (fr)
CN (1) CN100498933C (fr)
AU (1) AU2003271119A1 (fr)
CA (1) CA2504174A1 (fr)
DE (1) DE60321712D1 (fr)
HK (1) HK1077913A1 (fr)
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WO2007064256A2 (fr) * 2005-11-30 2007-06-07 Telefonaktiebolaget Lm Ericsson (Publ) Conversion efficace d'un flux vocal
DE102008009720A1 (de) * 2008-02-19 2009-08-20 Siemens Enterprise Communications Gmbh & Co. Kg Verfahren und Mittel zur Dekodierung von Hintergrundrauschinformationen
CN104751849B (zh) 2013-12-31 2017-04-19 华为技术有限公司 语音频码流的解码方法及装置
CN107369453B (zh) 2014-03-21 2021-04-20 华为技术有限公司 语音频码流的解码方法及装置
CN105869653B (zh) * 2016-05-31 2019-07-12 华为技术有限公司 话音信号处理方法和相关装置和系统

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JP2000200099A (ja) * 1998-10-26 2000-07-18 Sony Corp エコ―消去装置及び方法、並びに音声再生装置
JP2000206995A (ja) * 1999-01-11 2000-07-28 Sony Corp 受信装置及び方法、通信装置及び方法

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JPH0756598A (ja) * 1993-08-17 1995-03-03 Mitsubishi Electric Corp 有声音・無声音判別装置
JPH08123495A (ja) * 1994-10-28 1996-05-17 Mitsubishi Electric Corp 広帯域音声復元装置
JPH08248997A (ja) * 1995-03-13 1996-09-27 Matsushita Electric Ind Co Ltd 音声帯域拡大装置
JPH0918347A (ja) * 1995-06-28 1997-01-17 Oki Electric Ind Co Ltd 音声符号化方式変換装置
JP2000200099A (ja) * 1998-10-26 2000-07-18 Sony Corp エコ―消去装置及び方法、並びに音声再生装置
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DE60321712D1 (de) 2008-07-31
CN100498933C (zh) 2009-06-10
JP4438280B2 (ja) 2010-03-24
AU2003271119A1 (en) 2004-05-25
EP1564723B1 (fr) 2008-06-18
EP1564723A4 (fr) 2005-12-21
EP1564723A1 (fr) 2005-08-17
JP2004151424A (ja) 2004-05-27
KR20050061579A (ko) 2005-06-22
CA2504174A1 (fr) 2004-05-13
KR100715014B1 (ko) 2007-05-09
CN1708786A (zh) 2005-12-14
HK1077913A1 (en) 2006-02-24

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