EP1125283B1 - Procede de quantification des parametres d'un codeur de parole - Google Patents

Procede de quantification des parametres d'un codeur de parole Download PDF

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Publication number
EP1125283B1
EP1125283B1 EP99946281A EP99946281A EP1125283B1 EP 1125283 B1 EP1125283 B1 EP 1125283B1 EP 99946281 A EP99946281 A EP 99946281A EP 99946281 A EP99946281 A EP 99946281A EP 1125283 B1 EP1125283 B1 EP 1125283B1
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Prior art keywords
pitch
frame
transmitted
values
filters
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EP99946281A
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German (de)
English (en)
French (fr)
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EP1125283A1 (fr
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Philippe Thomson-CSF Prop. Intel. GOURNAY
Frédéric Thomson-CSF Prop. Intel. CHARTIER
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Thales SA
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Thales SA
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    • 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
    • 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
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/087—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters using mixed excitation models, e.g. MELP, MBE, split band LPC or HVXC
    • 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
    • G10L2019/0001—Codebooks
    • 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
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/93—Discriminating between voiced and unvoiced parts of speech signals

Definitions

  • the present invention relates to a method for coding the word. It applies in particular to the production of very low vocoders throughput, of the order of 1200 bits per second and implemented for example in satellite communications, internet telephony, static answering machines, voice pagers etc ...
  • the objective of these vocoders is to allow the reconstruction of a signal which is as close as possible to the sense of perception by the ear of the original speech signal, using the highest bit rate weak possible.
  • vocoders use a model fully configured speech signal.
  • the parameters used concern voicing which describes the periodic nature of sounds voiced or randomness of unvoiced sounds, frequency fundamental of voiced sounds still known by the term Anglo-Saxon "PITCH", the time evolution of the energy as well as the envelope signal spectral to excite and configure the synthesis filters.
  • voicing describes the periodic nature of sounds voiced or randomness of unvoiced sounds, frequency fundamental of voiced sounds still known by the term Anglo-Saxon "PITCH”, the time evolution of the energy as well as the envelope signal spectral to excite and configure the synthesis filters.
  • PITCH Anglo-Saxon
  • the filtering is carried out by a filtering technique numerical linear prediction.
  • MELP The new federal standard at 2400 bps, published in the IEEE International Conference on Acoustics, Speech, and Signal Processing, Kunststoff, April 1997, pp. 1591 - 1594.
  • a first technique is that of the vocoder segmental, two variants of which are those described by MM. B. Mouy, P. de la Noue and G. Goudezeune already cited, and that described by M. Y. Shoham titled "Very low complexity interpolative speech coding at 1.2 to 2.4 K bps ", published in IEEE International Conference on Acoustics, Speech, and Signal Processing, Kunststoff, April 1997, pp 1599 - 1602.
  • a second technique is that implemented in phonetic vocoders, which combine recognition principles and of synthesis. Activity in this area is rather at the stage of basic research, the targeted flows are generally very less than 1200 bits / s (typically 50 to 200 bits / s) but the quality obtained is rather bad and there is often no recognition of the speaker.
  • a description of these types of vocoders can be found in the article by MM. J. Cernocky, G. Baudoin, G. Chollet, having for title: "Segmental vododer - Going beyond the phonetic approch” published in IEE International Conference on Acoustics, Speech, and Signal Processing, Seattle, May 12 - 15 1998, pp. 605 - 698.
  • the object of the invention is to overcome the drawbacks mentioned.
  • the subject of the invention is a method of coding and speech decoding for voice communications using a very low bit rate vocoder including an analysis part for coding and transmitting the speech signal parameters and part summary for the reception and decoding of the transmitted parameters and the reconstruction of the speech signal by using synthesis filters to linear prediction of the type consisting in analyzing the parameters, describing pitch, voicing transition frequency, energy, and envelope spectral of the speech signal, by cutting the speech signal into frames successive of determined length characterized in that it consists of group the parameters on N consecutive frames to form a super-frame, to perform a vector quantization of the frequencies of transition of voicing during each super-frame, not transmitting without degradation that the most frequent configurations and replacing the less frequent configurations with the closest configuration in terms of absolute error among the most frequent, to code the pitch by scaling only one value for each super-frame, to code the energy by not selecting than a reduced number of values by grouping these values under packets quantified by vector quantization, the energy values not transmitted being recovered in the synthesis part by interpolation or
  • the method according to the invention uses a vocoder of type known by the Anglo-Saxon abbreviation HSX of "Harmonic Stochastic Excitation ", as a basis for the realization of a vocoder of good quality at 1200 bits / s.
  • the method according to the invention relates to the encoding of parameters that allow best reproduction with minimum flow all the complexity of the speech signal.
  • an HSX vocoder is a linear prediction vocoder which uses in its synthesis part a simple mixed excitation model, in which a periodic pulse train excites the low frequencies and a noise level excites the high frequencies a synthetic LPC filter.
  • FIG. 1 describes the principle of generation of the mixed excitation which comprises two filtering channels. The first channel 1 1 is energized by a periodic pulse train performs low pass filtering and the second channel 1 2 energized by a stochastic noise signal performs high pass filtering. The cutoff or transition frequency fc of the filters of the two channels is the same and has a variable position over time. The filters of the two channels are complementary.
  • a summator 2 adds the signals supplied by the two channels.
  • a gain amplifier 3 g adjusts the gain of the first filtering channel so that the excitation signal obtained at the output of the summator 2 is flat spectrum.
  • FIG. 2 A functional diagram of the vocoder analysis part is shown in Figure 2.
  • the speech signal is first filtered by a high pass filter 4 and then segmented into 22.5 ms frames, comprising 180 samples sampled at the frequency 8 KHz.
  • Two analyzes by linear prediction are performed in 5 on each of the frames.
  • the semi-whitened signal obtained is filtered into four sub-bands.
  • a robust pitch 8 tracker uses the first sub-band.
  • the transition frequency f c between the low frequency band of voiced sounds and the high frequency band of unvoiced sounds is determined by the voicing rate measured at 9 in the four sub-bands.
  • the energy is measured and coded in step 10 in a pitch-synchronous manner, 4 times per frame.
  • the pitch tracker and the analyzer voicing 9 can be greatly improved when their decision is delayed by one frame, the resulting parameters, filter coefficients synthesis, pitch, voicing, transition frequency and energy are encoded with a delay frame.
  • the excitation signal of the synthesis filter is formed by the already shown in Figure 1 by the sum of a signal harmonic and of a random signal whose spectral envelopes are complementary.
  • the harmonic component is obtained by passing a pulse train at the pitch period in a precalculated bandpass filter 11.
  • the random component is obtained from a generator 12 combining an inverse Fourier transform and an overlap temporal.
  • the synthesis LPC filter 14 is interpolated 4 times per frame.
  • the perceptual filter 15 coupled to filter outlet 14 makes it possible to obtain a better reproduction of the nasal characteristics of the speech signal original.
  • the automatic gain control device allows ensure that the pitch-synchronous energy of the output signal is equal to the one that was transmitted.
  • Step 17 groups together the vocoder frames by N frames to form a super weft.
  • N a value of N equal to 3 can be chosen because it achieves a good compromise between the possible reduction of the flow binary and the delay introduced by the quantification process.
  • it is compatible with interlacing and coding techniques corrector of current errors.
  • the voicing transition frequency is coded in step 18 by vector quantization using only four values of frequency, 0.750.2000 and 3625 HZ for example. Under these conditions 6 bits at the rate of 2 bits per frame are sufficient to code each of the frequencies and exactly transmit the voicing configuration of three frames of a super frame.
  • 6 bits at the rate of 2 bits per frame are sufficient to code each of the frequencies and exactly transmit the voicing configuration of three frames of a super frame.
  • voicing patterns are very rare may consider that they are not necessarily characteristic of the evolution of the normal speech signal, because they do not seem to participate intelligibility, nor the quality of the restored speech. This is the case with example when a frame is completely voiced from 0 Hz up to 3,625 Hz and that it is between two completely non- voiced.
  • the table in Figure 5 shows a distribution of voicing configuration on three successive frames, calculated on a database of 123,158 speech frames.
  • the 32 least frequent configurations account for only 4% of all the frames, partially or totally voiced.
  • Degradation obtained by replacing each of these configurations with the closest, in terms of absolute error, of the 32 most represented configurations is imperceptible. This shows that it is possible to save a bit by vectorially quantizing the voicing transition frequency on a great frame.
  • a vector quantification of the configurations of voicing is shown in the table referenced 22 in Figure 6.
  • the table 22 is organized so that the mean square error produced by an error on an addressing bit is minimal.
  • the value of the pitch decoded for the three frames of the current superframe is equal to the weighted average value quantified.
  • the advantage of carrying out a scalar quantification of values of pitch is that it limits the problem of propagation of errors on the train binary.
  • the coding schemes 2 and 3 are sufficiently close each other to be insensitive to bad decoding of the voicing frequency.
  • the encoding of the energy is carried out in step 20. It takes place from the as shown in the table referenced 23 in Figure 7 using a vector quantization method of the type described in RM Gray's article, titled “Vector Quantization", published in IEEE ASP Magazine, vol. 1, pp 4-29, April 1984. Twelve values of energy numbered from 0 to 11 are calculated for each superframe by the analysis part and only six energy values among the twelve are transmitted. This leads to construct two vectors of three values by the analysis part. Each vector is quantized on six bits. Two bits are used to transmit the selection scheme number used. then decoding in the synthesis part, the energy values which have not quantified are recovered by interpolation.
  • the bits giving the diagram number transmitted are not considered sensitive, since an error on their value only slightly alters the time evolution of the energy value.
  • the vector quantization table of energies is organized so that the mean square error produced by an error on an address bit is minimal.
  • the coding of the coefficients modeling the envelope of the signal speech takes place by vector quantization in step 21.
  • This coding allows to determine the coefficients of the digital filters used in the part synthesis.
  • Six LPC filters with 10 coefficients numbered from 0 to 5 are calculated at each superframe by the analysis part and only 3 filters among the 6 are transmitted.
  • the six vectors are transformed into six vectors of 10 pairs of LSF spectral lines following for example the process described in the article by M F. ITAKURA, entitled "Line Spectrum Representation of Linear Predictive Coefficients "and published in the Journal Acoustics Sociaty America, vol.57, P.S35, 1975. Pairs of lines spectral are encoded by a technique similar to that used work for the coding of energy.
  • the process is to select three LPC filters, and to quantify each of the vectors on 18 bits in using for example a loop predictive vector quantizer open, with a prediction coefficient equal to 0.6, of type SPLIT -VQ relating to two sub-packets of 5 consecutive LSFs to which it is allocated to each 9 bits. Two bits are used to transmit the number of the selection scheme used.
  • a filter LPC is not quantified, its value is estimated from that of the filters LPC quantified by linear interpolation for example, or by extrapolation by duplicating for example the previous LPC filter.
  • a vector quantization process by packets can be constituted as described in the article by MM K.K. PALIWAL, BS. ATAL, having for title "Efficient Vector Quantization of LPC Parameters at 24 bits / frame "and published in IEEE transaction on Speech and Audio Processing, Vol. 1, January 1993.
  • Bit allocation for the transmission of LSF parameters, of energy, pitch and voicing that results from the method of coding implemented by the invention is represented in the table of Figure 9 in the context of a 1200 bit / s vocoder in which the parameters are coded every 67.5 ms; 81 bits being available at each super frame to encode the signal parameters. These 81 bits break down into 54 LSF bits, 2 bits for decimating the diagram of LSF, 2 times 6 bits for energy, 6 bits for pitch and 5 bits for voicing.

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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)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
  • Executing Machine-Instructions (AREA)
  • Machine Translation (AREA)
  • Devices For Executing Special Programs (AREA)
EP99946281A 1998-10-06 1999-10-01 Procede de quantification des parametres d'un codeur de parole Expired - Lifetime EP1125283B1 (fr)

Applications Claiming Priority (3)

Application Number Priority Date Filing Date Title
FR9812500 1998-10-06
FR9812500A FR2784218B1 (fr) 1998-10-06 1998-10-06 Procede de codage de la parole a bas debit
PCT/FR1999/002348 WO2000021077A1 (fr) 1998-10-06 1999-10-01 Procede de quantification des parametres d'un codeur de parole

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US (1) US6687667B1 (xx)
EP (1) EP1125283B1 (xx)
JP (1) JP4558205B2 (xx)
KR (1) KR20010075491A (xx)
AT (1) ATE222016T1 (xx)
AU (1) AU768744B2 (xx)
CA (1) CA2345373A1 (xx)
DE (1) DE69902480T2 (xx)
FR (1) FR2784218B1 (xx)
IL (1) IL141911A0 (xx)
MX (1) MXPA01003150A (xx)
TW (1) TW463143B (xx)
WO (1) WO2000021077A1 (xx)

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FR2815457B1 (fr) * 2000-10-18 2003-02-14 Thomson Csf Procede de codage de la prosodie pour un codeur de parole a tres bas debit
KR100355033B1 (ko) * 2000-12-30 2002-10-19 주식회사 실트로닉 테크놀로지 선형예측 분석을 이용한 워터마크 삽입/추출 장치 및 그방법
CA2388439A1 (en) * 2002-05-31 2003-11-30 Voiceage Corporation A method and device for efficient frame erasure concealment in linear predictive based speech codecs
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CN101009096B (zh) * 2006-12-15 2011-01-26 清华大学 子带清浊音模糊判决的方法
EP2122610B1 (en) * 2007-01-31 2018-12-26 Telecom Italia S.p.A. Customizable method and system for emotional recognition
KR101317269B1 (ko) 2007-06-07 2013-10-14 삼성전자주식회사 정현파 오디오 코딩 방법 및 장치, 그리고 정현파 오디오디코딩 방법 및 장치
ES2650492T3 (es) * 2008-07-10 2018-01-18 Voiceage Corporation Dispositivo y método de cuantificación de filtro LPC de múltiples referencias
US9947340B2 (en) * 2008-12-10 2018-04-17 Skype Regeneration of wideband speech
GB2466201B (en) * 2008-12-10 2012-07-11 Skype Ltd Regeneration of wideband speech
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AU768744B2 (en) 2004-01-08
DE69902480D1 (de) 2002-09-12
CA2345373A1 (fr) 2000-04-13
MXPA01003150A (es) 2002-07-02
JP4558205B2 (ja) 2010-10-06
ATE222016T1 (de) 2002-08-15
AU5870299A (en) 2000-04-26
FR2784218A1 (fr) 2000-04-07
KR20010075491A (ko) 2001-08-09
IL141911A0 (en) 2002-03-10
WO2000021077A1 (fr) 2000-04-13
DE69902480T2 (de) 2003-05-22
FR2784218B1 (fr) 2000-12-08
EP1125283A1 (fr) 2001-08-22
TW463143B (en) 2001-11-11
JP2002527778A (ja) 2002-08-27
US6687667B1 (en) 2004-02-03

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