EP1125283B1 - Verfahren zur quantisierung der parameter eines sprachkodierers - Google Patents

Verfahren zur quantisierung der parameter eines sprachkodierers Download PDF

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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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pitch
frame
transmitted
values
filters
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EP1125283A1 (de
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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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  • Audiology, Speech & Language Pathology (AREA)
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Claims (12)

  1. Verfahren zur Kodierung und Dekodierung von Sprache für die Sprachübertragung unter Verwendung eines Vokoders mit sehr geringem Datendurchsatz, der einen Analyseteil (4, ..., 10) für die Kodierung und Übertragung der Parameter des Sprachsignals und einen Syntheseteil (11, ..., 16) für den Empfang und die Dekodierung der übertragenen Parameter sowie für die Rekonstruktion des Sprachsignals durch Verwendung von Synthesefiltern mit linearer Vorhersage enthält, wobei das Verfahren darin besteht, die Parameter zu analysieren, die den Pitch (8), die Übergangsfrequenz auf die stimmhaften Laute (9), die Energie (10) und die spektrale Hüllkurve des Sprachsignals (5) beschreiben, indem das Sprachsignal in aufeinanderfolgende Rahmen einer bestimmten Länge zerschnitten wird, dadurch gekennzeichnet, daß das Verfahren darin besteht, die Parameter in N aufeinanderfolgenden Rahmen zur Bildung eines Überrahmens zusammenzufassen (17), eine vektorielle Quantifizierung der Übergangsfrequenzen zu den stimmhaften Lauten während jedes Überrahmens durchzuführen (18) und ohne Verschlechterung nur die am häufigsten vorkommenden Konfigurationen zu übertragen, während die am wenigsten häufig vorkommenden Konfigurationen durch die hinsichtlich des absoluten Fehlers nächstliegende Konfiguration unter den häufiger vorkommenden ersetzt werden, den Pitch (19) zu kodieren, indem nur ein Pitchwert für jeden Überrahmen skalar quantifiziert wird, die Energie (20) zu kodieren, indem nur eine verringerte Anzahl von Werten durch Zusammenfassung dieser Werte in durch vektorielle Quantifizierung quantifizierte Unterpakete ausgewählt werden, wobei die nicht übertragenen Energiewerte im Syntheseteil durch Interpolation oder Extrapolation ausgehend von den übertragenen Werten wiedergewonnen werden, und durch vektorielle Quantifizierung (21) die Parameter der spektralen Hüllkurve für die Kodierung der Synthesefilter mit linearer Vorhersage zu kodieren, indem nur eine bestimmte Anzahl von Filtern ausgewählt wird, während die nicht übertragenen Parameter durch Interpolation oder Extrapolation ausgehend von den Parametern der übertragenen Filtern konstruiert werden.
  2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß der quantifizierte Pitchwert entweder der letzte Pitchwert der vollkommen stimmhaften stabilen Zonen oder ein durch die Übergangsfrequenz auf die stimmhaften Laute gewichteter Mittelwert in den Zonen ist, die nicht vollkommen stimmhaft sind.
  3. Verfahren nach Anspruch 2, dadurch gekennzeichnet, daß, wenn der Pitchwert der letzte eines Überrahmens ist, das Verfahren darin besteht, die anderen Werte durch Interpolation zu erzeugen.
  4. Verfahren nach Anspruch 3, dadurch gekennzeichnet, daß der im Syntheseteil verwendete Pitchwert derjenige des dekodierten Pitch ist, der durch einen Multiplikationskoeffizienten modifiziert wurde, um ein leichtes Tremolo in der rekonstruierten Sprache zu erzeugen.
  5. Verfahren nach einem beliebigen der Ansprüche 1 bis 4, dadurch gekennzeichnet, daß die Parameter über N=3 aufeinanderfolgende Rahmen zusammengefaßt sind.
  6. Verfahren nach Anspruch 5, dadurch gekennzeichnet, daß es vier Übergangsfrequenzen zur Stimmhaftigkeit gibt, die vektoriell mithilfe einer Quantifizierungstabelle (22) mit 32 Frequenzkonfigurationen in Dreiergruppen kodiert werden.
  7. Verfahren nach einem beliebigen der Ansprüche 5 und 6, dadurch gekennzeichnet, daß es darin besteht, die Energie viermal je Rahmen zu messen, wobei nur sechs der zwölf Werte eines Überrahmens in Form von zwei Vektoren von drei Werten übertragen werden (23).
  8. Verfahren nach Anspruch 7, dadurch gekennzeichnet, daß es darin besteht, die Energie gemäß vier Schemata zu kodieren, (23), die je zwei Vektoren zusammenfassen, wobei ein erstes Schema definiert wird, wenn die zwölf Energievektoren im Überrahmen stabil sind, während die übrigen Schemata für jeden der Rahmen definiert werden, und das Schema zu übertragen, das den quadratischen Gesamtfehler minimiert.
  9. Verfahren nach Anspruch 8, dadurch gekennzeichnet, daß
    im ersten Schema nur die Energiewerte mit den Nummern 1, 3 und 5 des ersten Vektors und die mit den Nummern 7, 9 und 11 des zweiten Vektors übertragen werden,
    im zweiten Schema nur die Energiewerte mit den Nummern 0, 1 und 2 das ersten Vektors und die mit den Nummern 3, 7 und 11 des zweiten Vektors übertragen werden,
    im dritten Schema nur die Energiewerte mit den Nummern 1, 4 und 5 des ersten Vektors und die mit den Nummern 6, 7 und 11 des zweiten Vektors übertragen werden,
    und im vierten Schema nur die Energiewerte mit den Nummern 2, 5 und 8 des ersten Vektors und die mit den Nummern 9, 10 und 11 des zweiten Vektors übertragen werden.
  10. Verfahren nach einem beliebigen der Ansprüche 1 bis 9, dadurch gekennzeichnet, daß es darin besteht, die Kodierparameter für die Filter mit linearer Vorhersage gemäß vier Schemata so auszuwählen, daß entweder die Zonen, für die die spektrale Hüllkurve stabil ist, oder die Zonen, für die die spektrale Hüllkurve rasch im Verlauf der Rahmen 1, 2 oder 3 eines Überrahmens verliert, möglichst gut kodiert werden.
  11. Verfahren nach Anspruch 10, dadurch gekennzeichnet, daß es darin besteht, im Syntheseteil sechs Filter mit linearer Vorhersage und 10 Koeffizienten mit den Nummern 0 bis 5 zu verwenden (24) und
    in einem ersten Schema nur die Koeffizienten der Filter 1, 3 und 5 zu übertragen, wenn die spektrale Hüllkurve stabil ist,
    in einem zweiten Schema entsprechend dem ersten Rahmen nur die Koeffizienten der Filter 0, 1 und 4 zu übertragen,
    in einem dritten Schema entsprechend dem zweiten Rahmen nur die Koeffizienten der Filter 2, 3 und 5 zu übertragen,
    in einem vierten Schema entsprechend dem dritten Rahmen nur die Koeffizienten der Filter 1, 4 und 5 zu übertragen,
       wobei das tatsächlich übertragene Schema dasjenige ist, das den quadratischen Gesamtfehler minimiert, und wobei die nicht übertragenen Koeffizienten der Filter im Syntheseteil durch Interpolation oder Extrapolation berechnet werden.
  12. Verfahren nach einem beliebigen der Ansprüche 1 bis 11, dadurch gekennzeichnet, daß die Koeffizienten LSF der Synthesefilter über 54 kodiert sind, denen noch zwei Bits für die Übertragung der Dezimierungsschemata hinzugefügt werden, daß die Energie mit zwei mal sechs Bits kodiert wird, denen noch zwei Bits für die Übertragung der Dezimierungsschemata hinzugefügt werden, daß der Pitch über sechs Bits kodiert wird und die Übergangs frequenz auf die stimmhaften Laute über fünf Bits kodiert wird, was insgesamt 81 Bits für die Überrahmen von 67,5 ms ergibt.
EP99946281A 1998-10-06 1999-10-01 Verfahren zur quantisierung der parameter eines sprachkodierers Expired - Lifetime EP1125283B1 (de)

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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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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 (de) 2001-08-22
TW463143B (en) 2001-11-11
JP2002527778A (ja) 2002-08-27
US6687667B1 (en) 2004-02-03

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