EP0475520A2 - Verfahren und Einrichtung zur Kodierung eines Analogsignals mit Wiederholeigenschaft - Google Patents

Verfahren und Einrichtung zur Kodierung eines Analogsignals mit Wiederholeigenschaft Download PDF

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Publication number
EP0475520A2
EP0475520A2 EP91202274A EP91202274A EP0475520A2 EP 0475520 A2 EP0475520 A2 EP 0475520A2 EP 91202274 A EP91202274 A EP 91202274A EP 91202274 A EP91202274 A EP 91202274A EP 0475520 A2 EP0475520 A2 EP 0475520A2
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EP
European Patent Office
Prior art keywords
segment
samples
coded
signal
partial
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EP91202274A
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English (en)
French (fr)
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EP0475520A3 (en
EP0475520B1 (de
Inventor
Adrianus Alphonsius Maria Van Der Krogt
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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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00Speech 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
    • G10L19/04Speech 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
    • G10L19/08Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/09Long term prediction, i.e. removing periodical redundancies, e.g. by using adaptive codebook or pitch predictor

Definitions

  • the invention relates to a method for coding a sampled analog signal having a repetitive nature, in which, for a signal segment to be coded consisting of a predetermined first number of samples, a search is always made in a preceding segment containing a predetermined second number of samples which is greater than the first number of samples for a signal segment which is as similar as possible by always comparing the signal segment to be coded, in steps of one sample interval, with a segment containing the first number of samples which forms part of the segment containing the second number of samples, and in which the difference signal is determined between the found, most similar segment and the segment to be coded as well as the difference between a reference time instant in the segment to be coded and a reference time instant in the found, most similar segment, expressed in the number of samples D between the two time instants.
  • LPC linear predictive coding
  • LTP long-term prediction
  • LTP for a signal segment to be coded, a search is made for a segment with the greatest possible similarity in a signal period, preceding the said segment, having a particular duration and a signal which is representative of the difference between the segment to be coded and the found segment, and also a signal which is representative of the time duration which has elapsed since the found segment, is transmitted in coded form, which can result in an appreciable reduction of the information to be transmitted.
  • HLTP high-resolution LTP
  • the object of the invention is to provide a method in which an improvement in the LTP principle is obtained in that the chance that the segment with the greatest similarity in a preceding period having a particular duration is found increases considerably without the number of operations needed in doing this increasing to the same extent as in the case of HLTP.
  • Figure 1a shows, in the time domain, an example of the sampled version of a signal having a strongly repetitive nature, such as a speech signal.
  • a search is always made in a preceding period of 15 ms for the segment of likewise 5 ms having the greatest similarity and that the sampling frequency is 8kHz.
  • segment A the segment to be coded in this connection
  • segment B the period of 15 ms segment B
  • the wanted segment with the greatest similarity segment C are shown in Figure 1a.
  • LTP the principle of LTP is that, before the transmission of segment A, no signals are transmitted which are directly related to the samples in segment A, but firstly signals which are related to the sample values which are produced if the difference signal between segment A and segment C is determined, and in the second place, signals which are related to the time difference between segment A and segment C, expressed, for example, in the number of samples D between the beginning of segment A and the beginning of segment C.
  • segment A can now be formed because segment C is in principle already known in the decoder, for example because the samples over the preceding 15 ms are always stored in a memory so that the samples from the segment can be read out of the memory with the received signal which is representative of the difference D in number of samples between the beginning of the segments A and C, after which the segment A can be formed with the signal, also received, which is representative of the difference between the sample values of segment A and segment C.
  • the complexity of the LTP principle can be defined as follows. In segment A, 40 samples are present,and in segment B 120 samples. The segment B therefore has to be investigated in 81 steps by "shifting" the entire segment A in steps equal to a sample interval over segment B and at every step determining the degree of agreement, expressed in a correlation value, by means of correlation techniques.
  • the segment B of 1440 samples now has to be searched in 961 steps by again shifting segment A oversegment B.
  • R(k) the above formula explained for the LTP principle can be used.
  • it is not necessary to calculate a correlation value also for the intermediate sample values, so that for every value of k (k 0 ... 960) 79 operations are necessary, just as in the case of LTP.
  • the spacing D between the beginning of segment A and the beginning of the segment C found can again be expressed in the number of samples between the two time instants (not more than 961) and can thus be reproduced in 10 bits.
  • the segment C with the greatest similarity is sought in the manner to be described below, which is less complex than the HLTP principle, the chance that the segment found is actually the segment with the greatest similarity being appreciably greater than in the case of the LTP principle.
  • the segment C having the greatest similarity to the segment A is first sought, according to the LTP principle explained above, in a segment B preceding a segment A to be coded.
  • This segment C is situated at a number D of samples from the segment A.
  • the found segment C ends up at a spacing equal to (D * Ob) from segment A.
  • the complexity of the method according to the invention with respect to that of LTP and HLTP can be calculated as follows: In the method according to the invention, if the same time duration of the segments A and B (5 and 15 ms, respectively) and the same sampling frequency (8 kHz) is assumed as in the example of the LTP principle described above, 6399 operations are necessary to find the segment C1.
  • the sampling frequency is increased by a factor of 12, for example by always placing 11 samplevalues equal to 0 between two consecutive known samples and by only calculating the actual sample value for samples at predetermined positions, for example by means of interpolating 7 already known samples. These predetermined positions are the positions of the samples which are situated at a spacing Dd from the original samples in the segment A.
  • the sampling frequency of the segment A is also increased and, just as in the case of HLTP, this is done by always placing 11 samples having a value equal to 0 between two known samples. Segment A therefore consists of 480 samples, of which a maximum of 40 are not equal to 0.
  • this latter segment is in fact the wanted segment because the fundamental regularity P which is present in the signal and which, for example in the case of speech, is determined by the fundamental frequency of the vocal cords, is determined by the spacing D1 between segment A and segment C1 and not by the spacing D2 between segment A and segment C2.
  • This phenomenon may be due, for example, to the presence of noise.
  • the segment Cd which is situated nearest segment A that is to say the segment having the smallest value for D
  • the segment C1 is chosen. The method described above for seeking the most suitable segment C, taking account of the regularity P in the signal, is shown in a flow chart in Figure 2.
  • the invention therefore also relates to a method for coding a sampled analog signal having a repetitive nature, in which, for a signal segment to be coded consisting of a predetermined first number of samples, a search is always made in a preceding segment containing a predetermined second number of samples which is greater than the first number of samples for a signal segment which agrees as well as possible by always comparing the signal segment to be coded, in steps of one sample interval, with a segment containing the first number of samples which forms part of the segment containing the second number of samples, and in which the difference signal is determined between the found, most similar segment and the segment to be coded as well as the difference between a reference time instant in the segment to be coded and a reference time instant in the found, most similar segment, expressed in the number of samples D between the two time instants, characterised in that of the partial segments compared with the segment to be coded that segment is chosen as the partial segment with the greatest agreement which has a correlation value R with the samples of the segment to be coded for which it is
  • FIG. 3a shows a block diagram of a coding/decoding system for carrying out the method according to the invention in the case of a speech signal comprising a coding unit 10 and a decoding unit 30.
  • An analog signal delivered by a microphone 11 is limited in bandwidth by a low pass filter 12 and converted in an analog/digital convertor into a series of sampled values which are representative of the analog signal.
  • the output signal of the convertor 13 is fed to the inputs of a short-term prediction filter 14 and of a short-term analysis unit 15.
  • These two units provide the above-mentioned short-term prediction and the analysis unit 15 provides an output signal in the form of short-term prediction filter coefficients, which output signal is transmitted to the decoder 30.
  • the structure and the operation of the filter 14 and the unit 15 are well known to those skilled in the field of speech coding and are not of further importance for the essence of the present invention, so that a further explanation can be omitted.
  • Said unit 17 is shown in greater detail in Figure 3b and comprises a unit 18 for splitting off the segment A, possibly the output signal of unit 16 can also be used for this purpose, and also a unit 19 for splitting off the segment B.
  • the output signals of the units 18 and 19 are fed to a circuit 20 in which the correlation value R c1 is calculated for the segment C1 in the manner outlined above and also the value of D is determined.
  • the calculated value of D is transmitted to the decoder 30 and is also fed to a unit 21 which is designed to calculate the different values of Dd on the basis of the pre-chosen values for d and Ob.
  • the value of Dd and the segment B are fed to a unit 22 for the purpose of calculating the segments Cd.
  • the calculated Cd's are fed to a circuit 23 which calculates with the aid of formula (1) the correlation values R cd for the different segments Cd on the basis of the segment A also fed to it.
  • the correlation values R c1 and R cd are compared with one another (see also Figure 2), and d optimum is determined in the manner described above and transmitted to the decoder.
  • the optimum segment Cd determined in the unit 25 is subtracted sample by sample from corresponding samples of the segment A in a subtraction unit 26 and the resultant difference signal is quantified in a manner known per se in a unit 27 and coded in a unit 28 in order to be transmitted to the decoding unit 30.
  • the difference signal received is decoded in a decoder 31 while the segment Cd opt is reconstructed in a unit 32 from the received values of D and d opt and from the previously received and reconstructed signal segment B.
  • the decoded difference signal and the segment Cd opt are added sample by sample in order thus to reconstruct the segment A.
  • the reconstructed segment Aand the received short-term prediction filter coefficients are fed to an inverse short-term prediction filter which reconstructs the transmitted signal samples as well as possible in a manner known per se.
  • the output signal of the filter 34 is converted in a digital/analog convertor into an analog signal which is fed to a loudspeaker 37 via a pass filter 36.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Signal Processing (AREA)
  • Computational Linguistics (AREA)
  • Acoustics & Sound (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)
  • Investigating Or Analysing Biological Materials (AREA)
  • Investigating Or Analysing Materials By The Use Of Chemical Reactions (AREA)
  • Nitrogen Condensed Heterocyclic Rings (AREA)
EP91202274A 1990-09-10 1991-09-06 Verfahren und Einrichtung zur Kodierung eines Analogsignals mit Wiederholeigenschaft Expired - Lifetime EP0475520B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
NL9001985 1990-09-10
NL9001985A NL9001985A (nl) 1990-09-10 1990-09-10 Werkwijze voor het coderen van een analoog signaal met een herhalend karakter en een inrichting voor het volgens deze werkwijze coderen.

Publications (3)

Publication Number Publication Date
EP0475520A2 true EP0475520A2 (de) 1992-03-18
EP0475520A3 EP0475520A3 (en) 1992-09-30
EP0475520B1 EP0475520B1 (de) 1996-04-24

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EP91202274A Expired - Lifetime EP0475520B1 (de) 1990-09-10 1991-09-06 Verfahren und Einrichtung zur Kodierung eines Analogsignals mit Wiederholeigenschaft

Country Status (11)

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EP (1) EP0475520B1 (de)
JP (1) JP2640595B2 (de)
AT (1) ATE137354T1 (de)
CA (1) CA2050979C (de)
DE (1) DE69119005T2 (de)
DK (1) DK0475520T3 (de)
ES (1) ES2089113T3 (de)
FI (1) FI105624B (de)
NL (1) NL9001985A (de)
NO (1) NO302549B1 (de)
PT (1) PT98900B (de)

Cited By (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2001003122A1 (en) * 1999-07-05 2001-01-11 Nokia Corporation Method for improving the coding efficiency of an audio signal
US7388201B2 (en) 2005-05-13 2008-06-17 National University Of Singapore Radiation detector having coated nanostructure and method
CN117176177B (zh) * 2023-11-03 2024-02-06 金乡县林业保护和发展服务中心(金乡县湿地保护中心、金乡县野生动植物保护中心、金乡县国有白洼林场) 一种用于林业信息的数据共享方法及系统

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB2061071A (en) * 1979-09-28 1981-05-07 Hitachi Ltd Speech analyzer
EP0280827A1 (de) * 1987-03-05 1988-09-07 International Business Machines Corporation Verfahren zur Grundfrequenzbestimmung und Sprachkodierer unter Verwendung dieses Verfahrens

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB2061071A (en) * 1979-09-28 1981-05-07 Hitachi Ltd Speech analyzer
EP0280827A1 (de) * 1987-03-05 1988-09-07 International Business Machines Corporation Verfahren zur Grundfrequenzbestimmung und Sprachkodierer unter Verwendung dieses Verfahrens

Non-Patent Citations (2)

* Cited by examiner, † Cited by third party
Title
ICASSP'86 (IEEE-IECEJ-ASJ INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Tokyo, 7th - 11th April 1986), vol. 3, pages 1717-1720, IEEE, New York, US; T. MIYAMOTO et al.: "Single DSP 8kbps speech codec" *
ICASSP'90 (1990 INTERNATIONAL CONFERENCE ON ACOUSTICS, SPEECH, AND SIGNAL PROCESSING, Albuquerque, New Mexico, 3rd - 6th April 1990), vol. 2, pages 661-664, IEEE, New York, US; P. KROON et al.: "Pitch predictors with high temporal resolution" *

Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2001003122A1 (en) * 1999-07-05 2001-01-11 Nokia Corporation Method for improving the coding efficiency of an audio signal
US7289951B1 (en) 1999-07-05 2007-10-30 Nokia Corporation Method for improving the coding efficiency of an audio signal
US7457743B2 (en) 1999-07-05 2008-11-25 Nokia Corporation Method for improving the coding efficiency of an audio signal
US7388201B2 (en) 2005-05-13 2008-06-17 National University Of Singapore Radiation detector having coated nanostructure and method
CN117176177B (zh) * 2023-11-03 2024-02-06 金乡县林业保护和发展服务中心(金乡县湿地保护中心、金乡县野生动植物保护中心、金乡县国有白洼林场) 一种用于林业信息的数据共享方法及系统

Also Published As

Publication number Publication date
EP0475520A3 (en) 1992-09-30
FI914261A0 (fi) 1991-09-10
CA2050979A1 (en) 1992-03-11
JP2640595B2 (ja) 1997-08-13
NO913448D0 (no) 1991-09-03
ATE137354T1 (de) 1996-05-15
FI105624B (fi) 2000-09-15
EP0475520B1 (de) 1996-04-24
NO302549B1 (no) 1998-03-16
PT98900B (pt) 1999-02-26
PT98900A (pt) 1994-01-31
JPH05206955A (ja) 1993-08-13
FI914261A (fi) 1992-03-11
NO913448L (no) 1992-03-11
CA2050979C (en) 1996-05-21
DE69119005T2 (de) 1996-11-28
ES2089113T3 (es) 1996-10-01
NL9001985A (nl) 1992-04-01
DE69119005D1 (de) 1996-05-30
DK0475520T3 (da) 1996-09-02

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