EP0482699A2 - Methode und Einrichtung zur Kodierung und Dekodierung eines abgetasteten Analogsignals mit Wiederholungseigenschaften - Google Patents

Methode und Einrichtung zur Kodierung und Dekodierung eines abgetasteten Analogsignals mit Wiederholungseigenschaften Download PDF

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Publication number
EP0482699A2
EP0482699A2 EP91202675A EP91202675A EP0482699A2 EP 0482699 A2 EP0482699 A2 EP 0482699A2 EP 91202675 A EP91202675 A EP 91202675A EP 91202675 A EP91202675 A EP 91202675A EP 0482699 A2 EP0482699 A2 EP 0482699A2
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EP
European Patent Office
Prior art keywords
values
amplitudes
transmitted
samples
signal
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Application number
EP91202675A
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English (en)
French (fr)
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EP0482699B1 (de
EP0482699A3 (en
Inventor
John Gerard Beerends
Frank Muller
Robertus Lambertus Adrianus Van Ravesteijn
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Koninklijke PTT Nederland NV
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Koninklijke PTT Nederland NV
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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

Definitions

  • the invention relates to a method for coding a sampled analog signal having a repetitive nature, in which the sampled signal is split into consecutive segments each containing a predetermined number of samples; in which a short-term prediction analysis is performed on said segments and in which the coefficients determined in said analysis are transmitted and are also fed to a short-term prediction filter, in which a long-term prediction analysis is performed on the residual signal available at the output of said filter and the information determined in said analysis is also transmitted, and in which the information present in the residual signal is coded and transmitted.
  • the invention also relates to a method for decoding a signal coded in the manner described above, in which the long-term prediction analysis information received and the other information received from the residual signal are combined and the combined signal, together with the short-term prediction analysis coefficients received, is fed to an inverse short-term prediction filter at whose output a series of samples is delivered which forms a reconstruction of the sampled analog signal.
  • the invention also relates to a device for coding and decoding by the method described above.
  • LPC linear predictive coding
  • LTP long-term prediction
  • the object of the invention is to provide a method for very efficiently transmitting, i.e. with a small number of bits/sec, the information relevant to the human ear in the residual signal remaining after applying the STP principle without the quality, experienced by the listener, of the speech reconstructed by the decoder at the receiving side being impaired.
  • the method for coding according to the invention is characterised in that the residual signal is transformed to the frequency domain, in that the amplitudes of at least a number of the frequency components obtained in transforming to the frequency domain are combined in a manner such that the frequencies associated with the combined amplitudes are situated equidistantly on a linear Bark scale, and in that a signal is transmitted which is representative of said combined amplitudes.
  • the method for decoding according to the invention is characterised in that the original amplitudes in the frequency domain are reconstructed from the combined amplitude values received, in that the information transmitted as a result of the long-term prediction analysis is used to calculate the phase values associated with said amplitudes, and in that the calculated phase values, together with the associated amplitudes, are transformed to the time domain.
  • the residual signal is coded perceptively, which means that only that information is transmitted which is relevant for differences in the decoded received signal which can be detected by the human ear.
  • the present invention makes use of the insight known for some time that human hearing functions in fact as a chain consisting of a number of filters having adjacent frequency bands but having different bandwidths, the so-called critical bands or Barks, the bandwidth of such critical bands being much smaller for low frequencies than for high frequencies.
  • a frequency scale formed in accordance with this insight is referred to as a linear Bark scale.
  • Bark scale For a further explanation of the principle of the Bark scale, reference is made to B. Scharf and S. Buus, "Stimulus, Physiology, Thresholds" in L. Kaufman, K.R. Boff and J. P Thomas, editors, Handbook of Perception and Human Performance, chapter 14, pages 1-43, Wiley, New York, 1986.
  • An analog signal delivered by a microphone 1 is limited in bandwidth by a low-pass filter 2 and converted in an analog/digital converter 3 into a series of amplitude and time-discrete samples which are representative of the analog signal.
  • the output signal of the converter 3 is fed to the input of a short-term analysis unit 4 and to the input of a short-term prediction filter 5.
  • These two units cater for the abovementioned short-term prediction (STP) on segments of, for example, 160 samples and the analysis unit 4 provides an output signal in the form of short-term prediction filter coefficients which are quantised, coded and transmitted to the decoder unit shown in Figure 1 b.
  • STP short-term prediction
  • the structure and the function of the filter 5 and the unit 4 are well known to those skilled in the art in the field of speech coding and are of no further importance for the essence of the present invention, so that a further explanation can be omitted.
  • the STP-filtered signal is fed to a long-term prediction (LTP) analysis unit 6.
  • LTP long-term prediction
  • this analysis unit an LTP analysis is applied twice per segment of 160 samples in a manner such as that described, for example, in Dutch Patent Application 9001985.
  • a search is always made, in accordance with a particular search strategy, for a segment which is as similar as possible in a signal period preceding said segment having a particular duration and a signal is transmitted in coded form which is representative of the number of samples D situated between the starting instant of the segment found and the starting instant of the segment to be coded.
  • the output signal of the STP filter unit 5 is referred to as the residual signal and, according to the invention, said residual signal is transmitted in coded form in a manner such that only the information which, seen perceptively, is relevant is transmitted.
  • the segments of 160 samples in said residual signal are divided into 8 subsegments of 30 samples in the circuit 7. This is done by first dividing the segment supplied into eight subsegments of 20 samples and then completing these at the leading edge with the ten last samples of the previous subsegment. This implies that the last ten samples of every segment have to be stored in order to also be able to complete the first subsegment of the subsequent segment.
  • Every subsegment of 30 samples is multiplied in a circuit 8 by a window function such as, for example, a cosine function.
  • the window function is so chosen that, for every sample in the overlapping parts of the subsegments, the sum of the squares of the two multiplication factors is unity. The reason that this has to be the case for the squares is that the multiplication by the window function takes place both in the coding unit and in the decoding unit shown in Figure 1 b.
  • a Discrete Fourier Transform (DFT) is performed on the windowed subsegment in a circuit 9, 16 different frequency components being obtained for every subsegment.
  • the amplitudes A of the components 1 to 13 inclusive are calculated in a circuit 10.
  • the components 0, 14 and 15 can be ignored because they are situated outside the frequency band of 300 - 3,400 Hz chosen for speech communication. If a greater or a smaller frequency band is relevant, the number of amplitude components taken into consideration can be adjusted accordingly.
  • four so-called Bark amplitude components are calculated in a circuit 11. These are amplitudes associated with frequencies which are situated equidistantly on a linear Bark scale.
  • the Bark amplitude components B 1 to B 4 inclusive can, for example, be calculated as follows from the DFT amplitudes A 1 to A13 3 inclusive:
  • a gain factor G is calculated as a scaling value in circuit 12 from the four Bark amplitude components in accordance with:
  • the application of the scaling value G has the advantage that the scaled amplitudes can be coded more efficiently.
  • the value of G is quantised in a circuit 13 and then transmitted to the decoding unit. If the scale factor G has been calculated, every Bark component is divided by the quantised gain factor G in a circuit 14. The result of this division is quantised in a circuit 15, coded and then also transmitted to the decoding unit.
  • circuits 12, 13 and 14 can be omitted and the four calculated values for the Bark amplitude components can be transmitted directly after quantisation in circuit 15.
  • the four scaled Bark amplitude components are multiplied in a multiplier 18 by the gain factor, G, decoded in a circuit 17, as a result of which the reconstructed Bark amplitude components to inclusive are obtained. This is of course not applicable if no scaling factor is used in the coding unit.
  • the amplitudes in the frequency domain to inclusive are calculated by means of the following formulae
  • the amplitudes and the phases are required.
  • the phases are determined in the following manner with the aid of the LTP information decoded in a circuit 23 and consisting of the sample spacing D.
  • the 120 most recent samples of the reconstructed STP residue such as are present at the output of the circuit 22 to be discussed in greater detail below are stored in each case.
  • the subsegment is determined which is situated at a spacing of D samples in the past with respect to the present subsegment and this subsegment is multiplied in a circuit 25 by the same window function as was used in the circuit 8 in the coder unit.
  • a DFT is then applied to said subsegment in a circuit 26, after which the phases of the 13 components considered can be calculated in a circuit 27. With the aid of the phases determined in this way and the amplitudes already calculated, an IDFT is performed in the circuit 20, the amplitudes of A o , A,4, and being set equal to zero.
  • the last ten samples in this subsegment are stored.
  • the first twenty samples form a portion of the reconstruction of a segment of the STP residue.
  • a completely reconstructed segment of the STP residue is obtained, and this is situated ten samples in the past with respect to the segment on which the STP analysis has been performed in the coding unit.
  • An inverse STP filtering is performed on this segment in a filter circuit 28 in a manner known per se with the aid of the STP coefficients received, the filter coefficients from the previous segment being used for the first ten samples.
  • the output signal of the filter 28 is converted in a digital/analog converter 29 into an analog signal which is fed via a low-pass filter 30 to a loudspeaker 31 which gives a high-fidelity reproduction of the speech signal supplied to the microphone 1, it having been possible to transmit said speech signal in coded form with a low number of bits due to the measures according to the invention.
  • a circuit 23' can be included between the circuits 23 and 24 to first subject the value of D received by the decoder additionally to a number of operations in order to obtain an optimum value of D for the reconstruction of the speech signal. These may be three consecutive operations.
  • the interpolation is carried out because the spacing D is determined in the coding unit twice per segment. Without interpolation, decoding of four consecutive subsegments would be carried out with the same value of D. If no fundamental regularity is present in the signal in the coding unit, a regularity would consequently wrongly be provided in the decoder during four subsegments. This problem is overcome by the interpolation.
  • the calculated spacing D corresponds as wall as possible with the actual repetition spacing present in the signal. If, however, said spacing D is less than 30, D is multiplied by an integer which is chosen in a manner such that the result is as a minimum equal to 30. This is necessary because all the samples of a subsegment at a spacing of less than 30 with respect to the present segment have not yet been reconstructed, so that they can therefore not be used to calculate the phases.

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  • Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Computational Linguistics (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Analogue/Digital Conversion (AREA)
  • Electrically Operated Instructional Devices (AREA)
EP91202675A 1990-10-23 1991-10-16 Methode und Einrichtung zur Kodierung und Dekodierung eines abgetasteten Analogsignals mit Wiederholungseigenschaften Expired - Lifetime EP0482699B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
NL9002308 1990-10-23
NL9002308A NL9002308A (nl) 1990-10-23 1990-10-23 Werkwijze voor het coderen en decoderen van een bemonsterd analoog signaal met een herhalend karakter en een inrichting voor het volgens deze werkwijze coderen en decoderen.

Publications (3)

Publication Number Publication Date
EP0482699A2 true EP0482699A2 (de) 1992-04-29
EP0482699A3 EP0482699A3 (en) 1992-08-19
EP0482699B1 EP0482699B1 (de) 1997-08-20

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EP91202675A Expired - Lifetime EP0482699B1 (de) 1990-10-23 1991-10-16 Methode und Einrichtung zur Kodierung und Dekodierung eines abgetasteten Analogsignals mit Wiederholungseigenschaften

Country Status (11)

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EP (1) EP0482699B1 (de)
JP (1) JP2958726B2 (de)
AT (1) ATE157188T1 (de)
CA (1) CA2053133C (de)
DE (1) DE69127339T2 (de)
DK (1) DK0482699T3 (de)
ES (1) ES2106051T3 (de)
FI (1) FI105623B (de)
NL (1) NL9002308A (de)
NO (1) NO305188B1 (de)
PT (1) PT99294A (de)

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5899966A (en) * 1995-10-26 1999-05-04 Sony Corporation Speech decoding method and apparatus to control the reproduction speed by changing the number of transform coefficients
EP1005020A2 (de) * 1998-11-27 2000-05-31 Matsushita Electronics Corporation Teilband Audiokodierer und diesen benutzendes drahtloses Mikrofon
EP1113432A2 (de) * 1999-12-24 2001-07-04 International Business Machines Corporation Verfahren und System zur Erkennung von identischen digitalen Daten

Families Citing this family (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH07261797A (ja) * 1994-03-18 1995-10-13 Mitsubishi Electric Corp 信号符号化装置及び信号復号化装置
FI116992B (fi) 1999-07-05 2006-04-28 Nokia Corp Menetelmät, järjestelmä ja laitteet audiosignaalin koodauksen ja siirron tehostamiseksi
CN114519996B (zh) * 2022-04-20 2022-07-08 北京远鉴信息技术有限公司 一种语音合成类型的确定方法、装置、设备以及存储介质

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB2060321A (en) * 1979-10-01 1981-04-29 Hitachi Ltd Speech synthesizer
US4742550A (en) * 1984-09-17 1988-05-03 Motorola, Inc. 4800 BPS interoperable relp system

Family Cites Families (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2892462B2 (ja) * 1990-08-27 1999-05-17 沖電気工業株式会社 コード励振線形予測符号化器

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB2060321A (en) * 1979-10-01 1981-04-29 Hitachi Ltd Speech synthesizer
US4742550A (en) * 1984-09-17 1988-05-03 Motorola, Inc. 4800 BPS interoperable relp system

Non-Patent Citations (5)

* Cited by examiner, † Cited by third party
Title
ADVANCED IN SPEECH CODING (IEEE WORKSHOP ON SPEECH CODING FOR TELECOMMUNICATIONS, Vancouver, CA, 5th - 8th September 1989), pages 81-86, Kluwer Academic Publishers, Dordrecht, NL; H. SUDA et al.: "An error protected transform coder for cellular mobile radio" *
ICASSP'85 (IEEE INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, Tampa, Florida, 26th - 29th March 1985), vol. 2, pages 509-513, IEEE, New York, US; H. HERMANSKY et al.: "Perceptually based linear predictive analysis of speech" *
ICASSP'86 (IEEE-IECEJ-ASJ INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, Tokyo, 7th - 11th April 1986), vol. 4, pages 3075-3078, IEEE, New York, US; B. MAZOR et al.: "Adaptive subbands excited transform (ASET) coding" *
ICASSP'88 (1988 INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH AND SIGNAL PROCESSING, New York, 11th - 14th April 1988), vol. 1, pages 263-266, IEEE, New York, US; B. FETTE et al.: "Experiments with a high quality, low complexity 4800 bps residual excited LPC (RELP) vocoder" *
ICC'87 (IEEE INTERNATIONAL CONFERENCE ON COMMUNICATIONS, SEATTLE, 7th - 10th June 1987), vol. 1, pages 418-424, IEEE, New York, US; Y. YATSUZUKA et al.: "Hardware implementation of 9.6/16 Kbit/s APC-MLQ speech codec and its application for mobile satellite communications" *

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5899966A (en) * 1995-10-26 1999-05-04 Sony Corporation Speech decoding method and apparatus to control the reproduction speed by changing the number of transform coefficients
EP1005020A2 (de) * 1998-11-27 2000-05-31 Matsushita Electronics Corporation Teilband Audiokodierer und diesen benutzendes drahtloses Mikrofon
EP1005020A3 (de) * 1998-11-27 2002-12-11 Matsushita Electric Industrial Co., Ltd. Teilband Audiokodierer und diesen benutzendes drahtloses Mikrofon
EP1113432A2 (de) * 1999-12-24 2001-07-04 International Business Machines Corporation Verfahren und System zur Erkennung von identischen digitalen Daten
EP1113432A3 (de) * 1999-12-24 2006-08-30 International Business Machines Corporation Verfahren und System zur Erkennung von identischen digitalen Daten

Also Published As

Publication number Publication date
DE69127339T2 (de) 1998-01-29
FI914993A0 (fi) 1991-10-23
JP2958726B2 (ja) 1999-10-06
CA2053133A1 (en) 1992-04-24
EP0482699B1 (de) 1997-08-20
NO305188B1 (no) 1999-04-12
PT99294A (pt) 1994-01-31
EP0482699A3 (en) 1992-08-19
NO914105L (no) 1992-04-24
JPH05268098A (ja) 1993-10-15
NL9002308A (nl) 1992-05-18
DK0482699T3 (da) 1998-03-30
ATE157188T1 (de) 1997-09-15
CA2053133C (en) 1996-05-21
NO914105D0 (no) 1991-10-18
FI914993A (fi) 1992-04-24
ES2106051T3 (es) 1997-11-01
FI105623B (fi) 2000-09-15
DE69127339D1 (de) 1997-09-25

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