EP1564723A1 - Transcodeur et procede de conversion par codeur - Google Patents

Transcodeur et procede de conversion par codeur Download PDF

Info

Publication number
EP1564723A1
EP1564723A1 EP03751372A EP03751372A EP1564723A1 EP 1564723 A1 EP1564723 A1 EP 1564723A1 EP 03751372 A EP03751372 A EP 03751372A EP 03751372 A EP03751372 A EP 03751372A EP 1564723 A1 EP1564723 A1 EP 1564723A1
Authority
EP
European Patent Office
Prior art keywords
signal
output
unit
voiced
gain
Prior art date
Legal status (The legal status 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 status listed.)
Granted
Application number
EP03751372A
Other languages
German (de)
English (en)
Other versions
EP1564723A4 (fr
EP1564723B1 (fr
Inventor
Kazunori Nec Corporation Ozawa
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
NEC Corp
Original Assignee
NEC Corp
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 Corp filed Critical NEC Corp
Publication of EP1564723A1 publication Critical patent/EP1564723A1/fr
Publication of EP1564723A4 publication Critical patent/EP1564723A4/fr
Application granted granted Critical
Publication of EP1564723B1 publication Critical patent/EP1564723B1/fr
Anticipated expiration legal-status Critical
Expired - Fee Related legal-status Critical Current

Links

Images

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 that performs inter-conversion between a code encoded in accordance with a first encoding method and a code encoded in accordance with a second encoding method, and more particularly to a transcoder that extends the frequency band of a signal when a first code is converted to a second code.
  • Non-Patent Document 1 A method is known that is used by the receiving side to extend the frequency band of a speech signal, which is encoded and reproduced at a low-bit rate, without transmitting auxiliary information for band extension from the sending side (for example, Non-Patent Document 1).
  • Non-Patent Document 1 P.Jax, P.Vary, "Wideband extension of telephone speech using hidden markov model," Proc. IEEE Speech Coding Workshop, pp. 133-135, 2000
  • the receiving side uses an HMM (Hidden Markov Model) to search for filter coefficients after band extension.
  • HMM Hidden Markov Model
  • Non-Patent Document 1 described above Non-Patent Document 1 described above
  • P.Jax and P.Vary which requires the spectrum envelope of a wideband speech and the HMM-based modeling of filter coefficients, has the following problems.
  • the HMM model parameters must be determined offline from a large-volume speech database in advance, and this processing requires long computation time and high costs.
  • the receiving side where the band is extended in real time must perform HMM-model-based search processing that requires a large amount of computation.
  • a transcoder that performs inter-conversion between a code encoded in accordance with a first encoding method and a code encoded in accordance with a second encoding method.
  • the transcoder comprises a spectrum parameter calculating unit that receives a code encoded by the first encoding method, decodes the received code by the first encoding method, and calculates a spectrum parameter representing spectrum characteristics; a noise generating unit that generates a noise signal; a coefficient calculating unit that shifts a frequency of the spectrum parameter and calculates filter coefficients; a gain unit that applies an appropriate gain to the output of the noise generating unit; a synthesis filter unit that lets the output of the gain unit pass through a synthesis filter, configured by the coefficients, and reproduces a band extended signal; and an adder that converts the sampling frequency of the input signal, adds up the converted signal and the output signal of the synthesis filter unit, and outputs the resulting signal, and then encodes the output signal of the adder in accordance with the second encoding method to output a second code.
  • a transcoder that performs inter-conversion between a code encoded in accordance with a first encoding method and a code encoded in accordance with a second encoding method.
  • the transcoder comprises a spectrum parameter calculating unit that receives a code encoded by the first encoding method, decodes the received code by the first encoding method, and calculates a spectrum parameter representing spectrum characteristics; an adaptive codebook unit that calculates a pitch period from the input signal and generates an adaptive codebook component based on the pitch period and a past sound source signal; a noise generating unit that generates a noise signal; a coefficient calculating unit that shifts a frequency of the spectrum parameter and calculates filter coefficients; a gain unit that applies an appropriate gain to at least one of the output signal of the noise generating unit and the output of the adaptive codebook unit and adds up the signals to output a sound source signal; a synthesis filter unit that lets the sound source signal pass through a synthesis filter configured by the coefficients to reproduce a band extended signal; and an adder that converts the sampling frequency of the reproduced signal and adds up the converted signal and the output signal of the synthesis filter unit and outputs the resulting signal, and then encodes the output signal of
  • a transcoder that performs inter-conversion between a code encoded in accordance with a first encoding method and a code encoded in accordance with a second encoding method.
  • the transcoder comprises a spectrum parameter calculating unit that receives a code encoded by the first encoding method, decodes the received code by the first encoding method, and calculates a spectrum parameter representing spectrum characteristics; an adaptive codebook unit that calculates a pitch period from the input signal and generates an adaptive codebook component based on the pitch period and a past sound source signal; a noise generating unit that generates a noise signal; a coefficient calculating unit that shifts a frequency of the spectrum parameter and calculates filter coefficients; a gain unit that applies an appropriate gain to at least one of the output of the noise generating unit and the output of the adaptive codebook unit and adds up the signals to output a sound source signal; a synthesis filter unit that lets the sound source signal pass through a pitch pre-filter using the pitch period and that lets the output signal of the pitch pre-filter pass through a synthesis filter configured by the coefficients to reproduce a band extended signal; and an adder that converts the sampling frequency of the reproduced signal and adds up the converted signal and the output
  • the transcoder may further comprise a low-pass filter with a predetermined cutoff frequency through which the output of the adaptive codebook unit passes.
  • the transcoder may further comprise a post filter which is configured by weighting coefficients generated by giving weight to the coefficients and through which the output signal of the synthesis filter unit passes to reproduce the band extended signal.
  • a code conversion method for use by a transcoder that performs inter-conversion between a code encoded in accordance with a first encoding method and a code encoded in accordance with a second encoding method.
  • the method comprises a step of decoding a code in accordance with a first decoding method and outputting a decoded signal, the code encoded by the first encoding method; a step of calculating a spectrum parameter from the decoded signal and outputting the spectrum parameter, the spectrum parameter representing spectrum characteristics; a step of shifting a frequency of the spectrum parameter, calculating filter coefficients, and outputting the calculated filter coefficients; a step of applying a gain to an output signal from a noise generating unit; a step of letting the output signal, to which the gain was applied, pass through a synthesis filter to output a signal of a band required for band conversion, the synthesis filter configured by the filter coefficients; a step of adding up a signal, which is generated by converting the decoded signal using a predetermined sampling frequency, and the output signal of the synthesis filter; and a step of encoding the addition result in accordance with the second encoding method to produce and output a second code.
  • a code conversion method comprising:
  • a code conversion method comprising:
  • the method may further comprise a step of performing pre-filtering processing for the sound source signal from the gain unit using the pitch period in the pitch pre-filter and letting the output signal from the pitch pre-filter pass through the synthesis filter circuit.
  • the method may further comprise a step of letting the output signal of the synthesis filter unit pass through a post filter configured by weighted coefficients generated by applying weight to the filter coefficients from the coefficient calculating unit.
  • the output of the periodic signal generation unit that generates the periodic signal using the pitch period may be supplied to the gain unit instead of the output signal from the adaptive codebook unit.
  • the present invention extends the band of the signal before conversion, generates a high-frequency signal through relatively small calculation, and adds up the resulting signal and the narrowband input signal, whose sampling frequency is converted, to produce a band extended signal (for example, 7 kHz band).
  • the present invention also generates an adaptive codebook signal using a delay calculated from the narrowband input signal based on a past sound source signal in the high-frequency part, multiplies the signal by an appropriate gain, and adds up the signal and the noise signal to generate a good sound-quality, band-extended signal when periodicity is required for a high-frequency signal such as a vowel sound.
  • the present invention may comprise a pitch pre-filter for the sound source signal using a delay or a post filter configured by giving weight to the coefficients from the coefficient calculation circuit to generate a better sound-quality, band-extended signal.
  • a first code is generated by encoding a narrowband input signal, 4 kHz in band, and that a transcoder extends this signal into a 5 KHz or 7 KHz band signal and encodes the signal by a second encoding method to produce a second code.
  • FIG. 1 is a block diagram showing the configuration of a first embodiment of a transcoder according to the present invention.
  • the transcoder comprises a first decoding circuit 105, a spectrum parameter calculation circuit 100, a noise generation circuit 120, a coefficient calculation circuit 130, a synthesis filter circuit 170, a sampling frequency conversion circuit 180, an adder 190, a second encoding circuit 195, a voiced/unvoiced discriminating circuit 200, a gain adjustment circuit 310, and a gain circuit 140.
  • the first decoding circuit 105 receives a code encoded by the first encoding method, decodes the received code in accordance with the first decoding method, and outputs a decoded signal x(n).
  • the spectrum parameter calculation circuit 100 divides the decoded signal x(n) into frames (for example, 10ms) and calculates the spectrum parameters of a predetermined order P for each frame.
  • the spectrum parameters are parameters representing the spectrum outline of the speech signal of each frame, and the known LPC (Linear Predictive Coding) analysis is used for this calculation.
  • LSP Line Spectrum Pair
  • Non-Patent Document 2 Sugamura, Itakura "Speech Information Compression by Line Spectrum Pair (LSP) Speech Analysis and Synthesis Method", Journal of Institute of Electronics, Information and Communication Engineers, J64-A, pp. 599-606, 1981
  • the coefficient calculation circuit 130 receives the spectrum parameters output from the spectrum parameter calculation circuit 100 and converts the parameters to the coefficients of a signal whose band is extended.
  • any of known methods such as the method for simply shifting the frequency of the LSP to a higher frequency, the non-linear conversion method, and the linear conversion method can be used.
  • the frequency band of the LSP is shifted to a higher frequency band using all or a part of LSP parameters, and the parameters are converted to the linear predictive coefficients of the order P and are output to the synthesis filter circuit 170.
  • the noise generation circuit 120 generates a noise signal, whose average amplitude is normalized to a predetermined level and whose band is limited, for the length of time equal to the frame length and outputs the generated noise signal to the gain circuit 140.
  • a white noise is used as an example of the noise signal in this embodiment, other noise signals may also be used.
  • the voiced/unvoiced discriminating circuit 200 receives the narrowband input signal x(n) and determines whether the signal of each frame is voiced or unvoiced. To determine whether the signal is voiced or unvoiced, the normalized auto-correlation function D(T) of the narrowband input signal x(n) is calculated up to a predetermined delay time m using expression (1) to find the maximum value of D(T). If the maximum value of D(T) is larger than a predetermined threshold value, the signal is determined to be voiced; otherwise the signal is determined to be unvoiced.
  • the voiced/unvoiced discriminating circuit 200 outputs the voiced/unvoiced discrimination information to the gain adjustment circuit 210.
  • N in expression (1) is the number of samples used for calculating the normalized auto-correlation.
  • the gain adjustment circuit 310 receives the voiced/unvoiced discrimination information from the voiced/unvoiced discriminating circuit 200 and, according to whether the signal is voiced or unvoiced, adjusts the gain to be given to the noise signal and outputs the adjusted gain to the gain circuit 140.
  • the gain circuit 140 receives the gain from the gain adjustment circuit 310, multiples the output signal from the noise generation circuit 120 by the gain, and outputs the result to the synthesis filter circuit 170.
  • the synthesis filter circuit 170 receives the output signal from the gain circuit 140, receives the coefficients of a predetermined number of orders from the coefficient calculation circuit 130 to configure a filter, and outputs a high frequency signal y(n) required for band extension.
  • 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 up the output signal y(n) from the synthesis filter circuit 170 and the output signal s(n) from the sampling frequency conversion circuit 180, and forms and outputs a signal z(n) whose band has been extended.
  • the second encoding circuit 195 receives the output signal z(n) from the adder 190, encodes the signal in accordance with the second encoding method, and produces and outputs the second code.
  • the first embodiment is as described above.
  • FIG. 2 is a block diagram showing the configuration of a second embodiment of the present invention.
  • a transcoder in the second embodiment of the present invention comprises a first decoding circuit 105, a spectrum parameter calculation circuit 100, an adaptive codebook circuit 110, a noise generation circuit 120, a coefficient calculation circuit 130, a gain circuit 340, a synthesis filter circuit 170, a sampling frequency conversion circuit 180, an adder 160, an adder 190, a second encoding circuit 195, a voiced/unvoiced discriminating circuit 200, and a gain adjustment circuit 210.
  • the same reference numeral is used to denote the same element in FIG. 1.
  • the second embodiment of the present invention is similar to the first embodiment except that the adaptive codebook circuit 110 and the adder 160 are added to the configuration in FIG. 1.
  • the voiced/unvoiced discriminating circuit 200 receives the narrowband input signal x(n) and determines whether the signal of each frame is voiced or unvoiced. To determine whether the signal is voiced or unvoiced, the normalized auto-correlation function D(T) of the narrowband input signal x(n) is calculated up to a predetermined delay time m using expression (1) described above to find the maximum value of D(T). If the maximum value of D(T) is larger than a predetermined threshold value, the signal is determined to be voiced; otherwise the signal is determined to be unvoiced. The determination result is output to the gain adjustment circuit 210.
  • the voiced/unvoiced discriminating circuit 200 supplies the value of T, which maximizes the normalized auto-correlation function D(T), to the adaptive codebook circuit 110 as the pitch period T.
  • the adaptive codebook circuit 110 receives the delay T of the adaptive codebook from the voiced/unvoiced discriminating circuit 200, generates an adaptive code vector p(n) according to expression (2) shown below based on the past sound source signal v(n), and outputs the generated vector.
  • p ( n ) v ( n - T )
  • the gain adjustment circuit 210 receives the voiced/unvoiced discrimination information from the voiced/unvoiced discriminating circuit 200, adjusts the gain of the adaptive codebook signal and the gain of the noise signal according to whether the signal is voiced or unvoiced, and supplies the adjusted gain to the gain circuit 340.
  • the gain circuit 340 receives the gain from the gain adjustment circuit 210, multiplies the output signal of at least one of the adaptive codebook circuit 110 and the noise generation circuit 120 by the gain, and outputs the result to the adder 160.
  • the adder 160 adds up two types of signal (two signals generated by multiplying the output signal of at least one of the adaptive codebook circuit 110 and the noise generation circuit 120 by the gain) output from the gain circuit 340 and outputs the result to the synthesis filter circuit 170 and the adaptive codebook circuit 110.
  • the synthesis filter circuit 170 receives the output signal from the adder 160, receives the coefficients (filter coefficients) of a predetermined number of orders from the coefficient calculation circuit 130 to configure a filter, and outputs a high frequency signal y(n) required for band extension.
  • the transcoder in the second embodiment of the present invention generates the adaptive codebook signal using the delay, calculated from the narrowband input signal, based on the past sound source signal of a high frequency part, multiplies the generated adaptive codebook signal by an appropriate gain, and adds up the resulting signal and the noise signal. Therefore, the transcoder can generate a good sound-quality band-extended signal required when periodicity is required for a high-frequency signal such as a vowel sound.
  • the second embodiment is as described above.
  • a periodic signal generation circuit 115 may be provided as shown in FIG. 6 instead of the adaptive codebook circuit 110 in FIG. 2.
  • the periodic signal generation circuit 115 receives a pitch period from the voiced/unvoiced discriminating circuit 200 and, using the pitch period, generates a periodic signal and outputs it to the gain circuit 340.
  • the configuration of this modification is similar to that of the second embodiment except the periodic signal generation circuit 115.
  • FIG. 3 is a block diagram showing the configuration of a third embodiment of the present invention.
  • a transcoder in the third embodiment of the present invention comprises a first decoding circuit 105, a spectrum parameter calculation circuit 100, 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, an adder 190, a second encoding circuit 195, a voiced/unvoiced discriminating circuit 200, a gain adjustment circuit 210, and a pitch pre-filter circuit 400.
  • the same reference numeral is used to denote the same or equivalent element in FIG. 1 and FIG. 2. The following mainly describes the difference from the second embodiment and omits the description of the same elements as those in FIG. 1 and FIG. 2.
  • the pitch pre-filter circuit 400 is provided.
  • the gain circuit 300 receives a gain from the gain adjustment circuit 210, multiplies the output signals from the adaptive codebook circuit 110 and the noise generation circuit 120 by the gain and adds up the resulting two types of signal, and outputs the addition result to the pitch pre-filter circuit 400.
  • the pitch pre-filter circuit 400 receives the delay T (pitch period) from the voiced/unvoiced discriminating circuit 200, performs pitch-filtering for the sound source signal v(n) from the gain circuit 300 according to expression (3) given below, and outputs the result to the synthesis filter circuit 170.
  • v' ( n ) v ( n ) + ⁇ p ( n - T )
  • the transcoder in this embodiment uses the pitch pre-filter circuit 400 for the sound source signal using the delay and therefore can produce a good sound-quality band-extended signal.
  • the third embodiment is as described above.
  • a periodic signal generation circuit may be used also in this embodiment instead of the adaptive codebook circuit 110.
  • the periodic signal generation circuit receives the signal from the voiced/unvoiced discriminating circuit 200, calculates the pitch period, generates a periodic signal based on the pitch period, and outputs the generated periodic signal to the gain circuit 300.
  • FIG. 4 is a block diagram showing the configuration of a fourth embodiment of the present invention.
  • a transcoder in the fourth embodiment of the present invention comprises a first decoding circuit 105, a spectrum parameter calculation circuit 100, 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, a sampling frequency conversion circuit 180, an adder 190, a second encoding circuit 195, a voiced/unvoiced discriminating circuit 200, a gain adjustment circuit 210, and a low-pass filter circuit 500.
  • the same reference numeral is used to denote the same or equivalent element in FIG. 2. The following mainly describes the difference from the second embodiment and omits the description of the same elements as those in FIG. 2.
  • the low-pass filter circuit 500 that receives the output of the adaptive codebook circuit 110 is provided.
  • the low-pass filter (LPF) circuit 500 allows the low-frequency signal of the output signal from the adaptive codebook circuit 110 to pass and outputs the result to the gain circuit 340.
  • p'(n) p(n)*h(n)
  • the cutoff frequency of the low-pass filter circuit 500 is predetermined, for example, to be 6 kHz.
  • h(n) indicates the impulse response of the low-pass filter and the symbol "*" indicates convolution operation, respectively.
  • the fourth embodiment of the present invention is as described above.
  • a periodic signal generation circuit may be used also in the fourth embodiment of the present invention instead of the adaptive codebook circuit 110.
  • the periodic signal generation circuit receives the signal from the voiced/unvoiced discriminating circuit 200, calculates the pitch period, generates a periodic signal based on the pitch period, and outputs the generated periodic signal to the gain circuit 340.
  • FIG. 5 is a block diagram showing the configuration of a fifth embodiment of the present invention.
  • a transcoder in the fifth embodiment of the present invention comprises a first decoding circuit 105, a spectrum parameter calculation circuit 100, 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, an adder 190, a second encoding circuit 195, a voiced/unvoiced discriminating circuit 200, a gain adjustment circuit 210, a pitch pre-filter 400, and a post filter 600.
  • the same reference numeral is used to denote the same or equivalent element in FIG. 3.
  • the following mainly describes the difference from the third embodiment and omits the description of the same elements as those in FIG. 3.
  • the configuration of this embodiment is similar to that of the third embodiment except that the post filter 600 is added.
  • the post filter 600 receives coefficients (filter coefficients) from the coefficient calculation circuit 130, gives weight to the coefficients, performs post filtering according to expression (5), and outputs the resulting output to the adder 190.
  • y'(n) y(n) - ⁇ a i ⁇ 1 i y(n-i) + ⁇ a i ⁇ 2 i y'(n-i)
  • This embodiment uses the post filter 600 to generate a good sound-quality band-extended signal.
  • the fifth embodiment is as described above.
  • a periodic signal generation circuit may be used also in the fifth embodiment of the present invention instead of the adaptive codebook circuit 110.
  • the periodic signal generation circuit receives the signal from the voiced/unvoiced discriminating circuit 200, calculates the pitch period, generates a periodic signal based on the pitch period, and outputs the generated periodic signal to the gain circuit 300.
  • a good sound-quality, band-extended signal is generated according to the present invention as described above when code encoded in a first encoding method is converted to code encoded in a second encoding method.
  • the present invention is, therefore, advantageously applicable to a code conversion device such as a transcoder.

Landscapes

  • 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)
EP03751372A 2002-10-31 2003-10-08 Transcodeur et procede de conversion par codeur Expired - Fee Related EP1564723B1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
JP2002317204A JP4438280B2 (ja) 2002-10-31 2002-10-31 トランスコーダ及び符号変換方法
JP2002317204 2002-10-31
PCT/JP2003/012859 WO2004040552A1 (fr) 2002-10-31 2003-10-08 Transcodeur et procede de conversion par codeur

Publications (3)

Publication Number Publication Date
EP1564723A1 true EP1564723A1 (fr) 2005-08-17
EP1564723A4 EP1564723A4 (fr) 2005-12-21
EP1564723B1 EP1564723B1 (fr) 2008-06-18

Family

ID=32211714

Family Applications (1)

Application Number Title Priority Date Filing Date
EP03751372A Expired - Fee Related EP1564723B1 (fr) 2002-10-31 2003-10-08 Transcodeur et procede de conversion par codeur

Country Status (9)

Country Link
EP (1) EP1564723B1 (fr)
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)
WO (1) WO2004040552A1 (fr)

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2007064256A2 (fr) * 2005-11-30 2007-06-07 Telefonaktiebolaget Lm Ericsson (Publ) Conversion efficace d'un flux vocal
RU2454737C2 (ru) * 2008-02-19 2012-06-27 Сименс Энтерпрайз Коммьюникейшнз Гмбх Унд Ко.Кг Способ и средство для декодирования информации о фоновом шуме

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
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
CN107369454B (zh) 2014-03-21 2020-10-27 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

Family Cites Families (8)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP3297156B2 (ja) * 1993-08-17 2002-07-02 三菱電機株式会社 音声判別装置
JP3483958B2 (ja) * 1994-10-28 2004-01-06 三菱電機株式会社 広帯域音声復元装置及び広帯域音声復元方法及び音声伝送システム及び音声伝送方法
JP3189614B2 (ja) * 1995-03-13 2001-07-16 松下電器産業株式会社 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
JP4296622B2 (ja) * 1998-10-26 2009-07-15 ソニー株式会社 エコー消去装置及び方法、並びに音声再生装置
JP4135242B2 (ja) * 1998-12-18 2008-08-20 ソニー株式会社 受信装置及び方法、通信装置及び方法
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
US6260009B1 (en) * 1999-02-12 2001-07-10 Qualcomm Incorporated CELP-based to CELP-based vocoder packet translation

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
JAX P ET AL: "Wideband extension of telephone speech using a hidden Markov model" IEEE WORKSHOP ON SPEECH CODING. PROCEEDINGS. MEETING THE CHALLENGES OF THE NEW MILLENNIUM, 17 September 2000 (2000-09-17), pages 133-135, XP002185445 *
See also references of WO2004040552A1 *

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2007064256A2 (fr) * 2005-11-30 2007-06-07 Telefonaktiebolaget Lm Ericsson (Publ) Conversion efficace d'un flux vocal
WO2007064256A3 (fr) * 2005-11-30 2007-12-13 Ericsson Telefon Ab L M Conversion efficace d'un flux vocal
EP2276023A3 (fr) * 2005-11-30 2011-10-05 Telefonaktiebolaget LM Ericsson (publ) Conversion efficace d'un flux de parole
CN101322181B (zh) * 2005-11-30 2012-04-18 艾利森电话股份有限公司 有效的语音流转换方法及装置
US8543388B2 (en) 2005-11-30 2013-09-24 Telefonaktiebolaget Lm Ericsson (Publ) Efficient speech stream conversion
RU2454737C2 (ru) * 2008-02-19 2012-06-27 Сименс Энтерпрайз Коммьюникейшнз Гмбх Унд Ко.Кг Способ и средство для декодирования информации о фоновом шуме

Also Published As

Publication number Publication date
JP2004151424A (ja) 2004-05-27
KR100715014B1 (ko) 2007-05-09
AU2003271119A1 (en) 2004-05-25
DE60321712D1 (de) 2008-07-31
CN100498933C (zh) 2009-06-10
EP1564723A4 (fr) 2005-12-21
CN1708786A (zh) 2005-12-14
JP4438280B2 (ja) 2010-03-24
EP1564723B1 (fr) 2008-06-18
WO2004040552A1 (fr) 2004-05-13
HK1077913A1 (en) 2006-02-24
KR20050061579A (ko) 2005-06-22
CA2504174A1 (fr) 2004-05-13

Similar Documents

Publication Publication Date Title
JP2940005B2 (ja) 音声符号化装置
JP4270866B2 (ja) 非音声のスピーチの高性能の低ビット速度コード化方法および装置
JP4302978B2 (ja) 音声コーデックにおける擬似高帯域信号の推定システム
JP2005528647A (ja) 合成発話の周波数選択的ピッチ強調方法およびデバイス
JP3180762B2 (ja) 音声符号化装置及び音声復号化装置
JP4040126B2 (ja) 音声復号化方法および装置
US7486719B2 (en) Transcoder and code conversion method
TW463143B (en) Low-bit rate speech encoding method
JP3582589B2 (ja) 音声符号化装置及び音声復号化装置
JP3266178B2 (ja) 音声符号化装置
EP1619666B1 (fr) Decodeur vocal, programme et procede de decodage vocal, support d'enregistrement
EP1564723B1 (fr) Transcodeur et procede de conversion par codeur
JP4433668B2 (ja) 帯域拡張装置及び方法
JP3481027B2 (ja) 音声符号化装置
JP3319396B2 (ja) 音声符号化装置ならびに音声符号化復号化装置
KR0155798B1 (ko) 음성신호 부호화 및 복호화 방법
JP4287840B2 (ja) 符号化装置
JP3153075B2 (ja) 音声符号化装置
JP3598111B2 (ja) 広帯域音声復元装置
JP3063087B2 (ja) 音声符号化復号化装置及び音声符号化装置ならびに音声復号化装置
JP2853170B2 (ja) 音声符号化復号化方式
JP3560964B2 (ja) 広帯域音声復元装置及び広帯域音声復元方法及び音声伝送システム及び音声伝送方法
JP3598112B2 (ja) 広帯域音声復元方法及び広帯域音声復元装置
JP2004046238A (ja) 広帯域音声復元装置及び広帯域音声復元方法
JP2004341551A (ja) 広帯域音声復元方法及び広帯域音声復元装置

Legal Events

Date Code Title Description
PUAI Public reference made under article 153(3) epc to a published international application that has entered the european phase

Free format text: ORIGINAL CODE: 0009012

17P Request for examination filed

Effective date: 20050513

AK Designated contracting states

Kind code of ref document: A1

Designated state(s): AT BE BG CH CY CZ DE DK EE ES FI FR GB GR HU IE IT LI LU MC NL PT RO SE SI SK TR

AX Request for extension of the european patent

Extension state: AL LT LV MK

A4 Supplementary search report drawn up and despatched

Effective date: 20051107

RIC1 Information provided on ipc code assigned before grant

Ipc: 7G 10L 19/00 A

Ipc: 7G 10L 21/02 B

DAX Request for extension of the european patent (deleted)
RBV Designated contracting states (corrected)

Designated state(s): DE FR GB

REG Reference to a national code

Ref country code: HK

Ref legal event code: DE

Ref document number: 1077913

Country of ref document: HK

GRAP Despatch of communication of intention to grant a patent

Free format text: ORIGINAL CODE: EPIDOSNIGR1

GRAS Grant fee paid

Free format text: ORIGINAL CODE: EPIDOSNIGR3

GRAA (expected) grant

Free format text: ORIGINAL CODE: 0009210

AK Designated contracting states

Kind code of ref document: B1

Designated state(s): DE FR GB

REG Reference to a national code

Ref country code: GB

Ref legal event code: FG4D

REF Corresponds to:

Ref document number: 60321712

Country of ref document: DE

Date of ref document: 20080731

Kind code of ref document: P

REG Reference to a national code

Ref country code: HK

Ref legal event code: GR

Ref document number: 1077913

Country of ref document: HK

PLBE No opposition filed within time limit

Free format text: ORIGINAL CODE: 0009261

STAA Information on the status of an ep patent application or granted ep patent

Free format text: STATUS: NO OPPOSITION FILED WITHIN TIME LIMIT

26N No opposition filed

Effective date: 20090319

REG Reference to a national code

Ref country code: FR

Ref legal event code: PLFP

Year of fee payment: 14

REG Reference to a national code

Ref country code: FR

Ref legal event code: PLFP

Year of fee payment: 15

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: DE

Payment date: 20171004

Year of fee payment: 15

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: GB

Payment date: 20171004

Year of fee payment: 15

REG Reference to a national code

Ref country code: FR

Ref legal event code: PLFP

Year of fee payment: 16

PGFP Annual fee paid to national office [announced via postgrant information from national office to epo]

Ref country code: FR

Payment date: 20180913

Year of fee payment: 16

REG Reference to a national code

Ref country code: DE

Ref legal event code: R119

Ref document number: 60321712

Country of ref document: DE

GBPC Gb: european patent ceased through non-payment of renewal fee

Effective date: 20181008

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: DE

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20190501

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: GB

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20181008

PG25 Lapsed in a contracting state [announced via postgrant information from national office to epo]

Ref country code: FR

Free format text: LAPSE BECAUSE OF NON-PAYMENT OF DUE FEES

Effective date: 20191031