EP0341129B1 - Verfahren und Einrichtung zur Kodierung von Sprachsignalenergie in Vokodern mit sehr niedriger Bitrate - Google Patents
Verfahren und Einrichtung zur Kodierung von Sprachsignalenergie in Vokodern mit sehr niedriger Bitrate Download PDFInfo
- Publication number
- EP0341129B1 EP0341129B1 EP89401169A EP89401169A EP0341129B1 EP 0341129 B1 EP0341129 B1 EP 0341129B1 EP 89401169 A EP89401169 A EP 89401169A EP 89401169 A EP89401169 A EP 89401169A EP 0341129 B1 EP0341129 B1 EP 0341129B1
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- 238000000034 method Methods 0.000 title claims description 17
- 239000013598 vector Substances 0.000 claims description 88
- 230000015654 memory Effects 0.000 claims description 19
- 239000011159 matrix material Substances 0.000 claims description 8
- 230000008569 process Effects 0.000 claims description 5
- 238000004458 analytical method Methods 0.000 claims description 3
- 230000008859 change Effects 0.000 claims description 3
- 229940050561 matrix product Drugs 0.000 claims 1
- 238000005259 measurement Methods 0.000 claims 1
- 230000000875 corresponding effect Effects 0.000 description 6
- 230000005540 biological transmission Effects 0.000 description 2
- 230000002596 correlated effect Effects 0.000 description 2
- 238000013139 quantization Methods 0.000 description 2
- 101000822695 Clostridium perfringens (strain 13 / Type A) Small, acid-soluble spore protein C1 Proteins 0.000 description 1
- 101000655262 Clostridium perfringens (strain 13 / Type A) Small, acid-soluble spore protein C2 Proteins 0.000 description 1
- 101000655256 Paraclostridium bifermentans Small, acid-soluble spore protein alpha Proteins 0.000 description 1
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- 238000012552 review Methods 0.000 description 1
- 230000009466 transformation Effects 0.000 description 1
- 238000000844 transformation Methods 0.000 description 1
- 230000001755 vocal effect Effects 0.000 description 1
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Classifications
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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
- G10L19/0018—Speech coding using phonetic or linguistical decoding of the source; Reconstruction using text-to-speech synthesis
Definitions
- the present invention relates to a method and a device for coding the energy of the voice signal in vocoders at very low bit rates.
- the voice signal is divided into time slots or windows of fixed lengths of approximately 20 milliseconds in the transmission vocoders, and each signal window is analyzed to extract the parameters necessary for controlling the digital filters of the vocoders of reception. These parameters are made up of control coefficients for the reception filters, the effective mean value VME of the voice signal and an indication of the nature of the voice signal, whether or not it is heard.
- the method of coding the VME parameter consists in quantifying the VME parameter on 32 values (0 to 31) according to a scale logarithmic standardized by the NATO standard "Stanag 4198" relating to linear predictive coding of order 10 of which a description appears in the article of M TREMAIN having for title "The Government Standard Linear Predictive Coding Algorithm-LPC 10" and which is published in Speech Technology April 1982 pages 40-49.
- the quantized VME signal is then coded on 11 bits during three consecutive windows.
- the effective mean value of the middle window is coded on 5 bits and that of each of the extreme windows is coded by a coding method. of the middle window.
- a description of this coding process can be found in an article published by the authors Wong D, Juang BH Gray AH in the journal IEEE Transactions on ASSP vol. 30, 1982 pages 770-780 entitled "An 800 bits / s Vector Quantization LPC Vocoder".
- the 11-bit coding of the VME parameter limits the possibilities of reducing vocoder bit rate, in particular at low bit rates lower than 800 bits / second.
- the object of the invention is to overcome the aforementioned drawback.
- the invention also relates to a device for implementing the above method.
- the method according to the invention is based on the observation that the energy contained in the speech signal varies very slowly over time, so that the energies E0, E1 and E2 of the samples quantified in each signal window can be considered as strongly correlated to each other. It is indeed possible to observe, by considering only a very large number of groups of two successive windows, and by referencing the corresponding energy vector of each group in a vector space orthonormal to two dimensions where the energies E1 and E2 of each window represent the projections of the energy vector E of each group in the representative base of this space, the origins of the energy vectors E of all the groups being confused with that of the two-dimensional vector space, that the ends of the energy vectors E are distribute, as shown in Figure 1, in a domain (D) of the plane formed by the two vectors E1 and E2, substantially symmetrical with respect to a bisector E'1 of the angle (E1, E2) formed by the two vectors, being very elongated in the direction of the bisector and on the other hand flattened in the direction
- the unit vector of the first axis of inertia has as components (3 -1 ⁇ 2 , 3 -1 ⁇ 2 , 3 -1 ⁇ 2 ), the unit vector of the second axis of inertia has components (-2 -1 ⁇ 2 , 0, 2 -1 ⁇ 2 ) and the unit vector of the third axis of inertia has components (-6 -1 ⁇ 2 , 2x6 -1 ⁇ 2 , -6 -1 ⁇ 2 ).
- the unit vector of the first axis of inertia has as components (3 -1 ⁇ 2 , 3 -1 ⁇ 2 , 3 -1 ⁇ 2 )
- the unit vector of the second axis of inertia has components (-2 -1 ⁇ 2 , 0, 2 -1 ⁇ 2 )
- the unit vector of the third axis of inertia has components (-6 -1 ⁇ 2 , 2x6 -1 ⁇ 2 , -6 -1 ⁇ 2 ).
- the matrix [E '] has for column vectors the components E'0, E'1 and E'2, the matrix [E] has for column vectors the components E0, E1 and E2, and t P denotes the matrix transposed from P.
- the preceding transformations make it possible, by limiting the values of E'0 between 0 and 54, to code it on only 4 bits according to a linear scale comprised between these two values and by truncating the values E'1 and E'2 between the values -16 and +16, these can be coded respectively on 3 bits and 2 bits according to also a linear scale also included between these 2 values.
- the result is then the obtaining of 3 coded values (E''0, E''1 and E''2) on a total of only 9 bits instead of 11 in the prior art which is sufficient to ensure good quality 800 bit / s transmissions.
- the operations carried out are the reverse operations of coding.
- the method determines in a first step the vector of component E'0, E'1 and E'2 expressed in the base of the unit vectors of the axes principal of inertia. Then according to a second step, it multiplies the matrix P by the vector of components E'0, E'1, E'2 to obtain a vector of components E0, E1 and E2. Finally, according to a third step, it applies to the components E0, E1 and E2 the decoding law of the linear predictive coding standard of order 10, to obtain the three effective values VME0, VME1 and VME2 of the three consecutive signal windows processed.
- FIGS. 4 and 5 A corresponding coding device is shown in FIGS. 4 and 5.
- the device for measuring the energy of the samples of the voice signal comprises an accumulator circuit 1, represented inside a closed line in dotted lines, this circuit being coupled to two registers 2 and 3 connected in series.
- the accumulator circuit 1 consists, in a known manner, of an accumulator register 4 and of an adder circuit 5.
- Each sample S i of the voice signal is applied to a first operand input of the adder circuit 5 and is added to the content of the accumulator register 4 which is applied to the second operand input of the adder circuit 5.
- the accumulation of the samples S i of a window thus takes place in the accumulator register 4 throughout the duration of the window.
- the coding device also comprises three processing channels 7, 8 and 9 shown in interior of closed dotted lines.
- Channel 7 comprises, an attenuator circuit 10 of attenuation ratio 3 -1 ⁇ 2 , a limiter stage 11 and an encoder 12. All of the elements 10, 11, 12 are coupled together, in this order, and in series at the output of the adder circuit 6.
- Channel 8 comprises an amplifier circuit 13 of gain 3, coupled to an attenuator circuit 15 of attenuation ratio 6 -1 ⁇ 2 through a circuit subtractor 14.
- the subtractor circuit 14 includes a first operand input, marked "+” which is connected to the output of the amplifier circuit 13 and a second operand input marked "-" which is connected to the output of the adder circuit 6 .
- Channel 9 comprises an attenuator circuit 16 of attenuation ratio 2 -1 ⁇ 2 coupled to the output of an adder circuit 17.
- a switch circuit 18 applies one or the other of the signals obtained at the output of channels 8 and 9 to the input of an encoder 19 through a limiter stage 20.
- the reception decoder is shown in FIG. 6. It comprises a set of three reception channels 21, 22 and 23 shown inside closed dotted lines.
- the first channel 21 comprises, connected in series, an attenuation circuit 24 of attenuation ratio 3 -1 ⁇ 2 and two subtractor circuits 25 and 26.
- the second channel 22 comprises connected in series, an attenuation circuit 27 of attenuation ratio 2 -1 ⁇ 2 , an adder circuit 28 and a subtractor circuit 29;
- the third channel 23 comprises, connected in series with an attenuation circuit 30 of attenuation ratio 6 -1 ⁇ 2 , an amplifier 31 of gain 2 and an adder circuit 32.
- the subtractor circuit 25 is connected by a first operand input marked "+” to the output of the attenuator circuit 24 and by a second operand input marked “-” to the output of the attenuator circuit 27.
- the result of the subtraction carried out by the subtractor circuit 25 is applied to a first operand input marked "+” of the subtractor circuit 26.
- the second operand input marked "-" of the subtractor circuit 26 is connected to the output of the attenuator circuit 30.
- the output of the subtractor circuit 26 supplies the energy E0 of the first window of the voice signal.
- the adder circuit 28 has a first operand input connected to the output of the attenuator circuit 27 and a second operand input connected to the output of the attenuator circuit 24.
- the result obtained at the output of the adder circuit 28 is applied to a first operand input marked "+" of the subtractor circuit 29.
- the second operand input marked "-" of the subtractor circuit 29 is connected to the output of the attenuator circuit 30.
- the energy E2 of the signal is obtained at the output of the subtractor circuit 29.
- the adder circuit 32 is connected by a first operand input to the output of the amplifier 31 and by a second operand input to the output of the attenuator circuit 24
- the energy E1 of the signal is obtained at the output of the adder circuit 23.
- a second variant of implementation of the method according to the invention may consist in perform, as shown in FIG. 7, a vector coding of the vector (E'0, E'1, E'2), by searching for the closest vector of the vector (E'0, E'1, E'2) among 2 N vectors whose ends would coincide with the nodes of a bounded subset of a face-centered cubic network, so as to obtain an encoding on N bits.
- This coding mode is carried out by the circuits of FIG.
- the read-only memory 33 contains the three components of the 2 N estimated vectors (E0, E1 and E2) and these are addressed by the N-bit address counter 24. Each of the components read from the memory 33 is applied respectively to a first operand input of the subtractor circuits 35 to 37.
- the components E0, E1 and E2 of the energy of the voice signal from each of the three windows are applied respectively to the second operand inputs of the subtractor circuits 35 to 37.
- the results of the subtractions carried out by the subtractor circuits 35 to 37 are applied respectively to the input of the circuits squared from 38 to 40, and the results of the squared elevations are applied to the inputs of the summing circuit 41.
- the sums of the squares of the differences between, each component (E0, E1, E2) of a vector representing the energies of the voice signal in three consecutive windows and the components E0, E1, E2 of an estimated vector addressed by the address counter 34, are applied successively by the output of the summing circuit 41 to a first comparison input of a comparator circuit 42 to be compared with the content of register 43 which is applied to the second comparison input of comparator 42.
- the register 43 On each comparison the content of register 43 is updated by the result of the summation obtained at the output of the summing circuit 41 if this result is less than the content existing in the register 43. In this way after each progression of the address counter 34, the register 43 keeps in memory the sum of the squares obtained from the summing circuit 41 which is the smallest among all the sums already made since the beginning of the addressing of the estimated vectors in the memory 33. In parallel with each update of the content of the register 43 the content of the register 44 is replaced by the address of the corresponding vector which has been read in the memory 33. The number of the VME vector coded on N bits is thus obtained directly in the register 44.
- FIG. 8 A third variant of implementation of the method according to the invention is shown in FIG. 8. As this third variant follows from the embodiment of the second variant described above, the homologous elements of FIG. 7 are represented in FIG. 8 with the same references.
- This third variant differs from the previous one in that the memory space 33 is divided into three memory sub-spaces 33 a , 33 b and 33 c .
- the first N / 3 bits of the address counter 34 address the subspaces 33 a
- the following N / 3 bits address the second subspace
- the remaining N / 3 bits address the subspace 33 c .
- the groups of the cubic network By assigning the groups of the cubic network to memory space 33 a, the subgroups to memory space 33 b , 2 N / 3 estimated energy vectors can be coded in memory space 33 a and 2 N / 3 estimated energy vectors can be coded in memory space 33 b .
- the 2 N / 3 remaining vectors are coded in memory space 33 c .
- For N 9, 8 groups are thus obtained, each comprising 8 subgroups of 8 vectors each.
- the energy of the voice signal with component E de, E1 and E2 is measured by circuits 35 to 43 relative to the energy of the corresponding estimated vectors formed successively through a multiplexer 45 by memories 33 a , 33 b and 33 c .
- the group, the subgroup and then the vector of the subgroup which has the energy closest to the component vector E0, E1 and E2 are thus defined successively.
- the group numbers in the group and the vector within a sub-group are recorded respectively in the register 44 which has in FIG. 8 the form of a bank of registers composed of the registers 44 a , 44 b and 44 c .
- AND gates 48, 49 and 50 allow the transfer of the group, sub-group and vector addresses within a group, each time the result of the comparison carried out by the comparator 42 indicates that the sum formed by the summator 41 is less than the contents of register 43.
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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)
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Claims (8)
- Verfahren zur Kodierung der Energie von Sprachsignalen in Vocodern sehr geringer Geschwindigkeit, die die Sprachsignale in aufeinanderfolgenden Fenstern analysieren, in jedem Fenster die mittlere effektive Energie (VME) der Tastproben des Sprachsignals messen (1, 2, 3) und diese Energie VME im Rahmen einer gegebenen Anzahl m von Energiepegeln quantifizieren, dadurch gekennzeichnet, daß das Verfahren darin besteht, in einem n-dimensionalen Vektorraum, der als erste Basis die Einheitsvektoren (e₁ bis en) besitzt, einen resultierenden Energievektor zu konstruieren, dessen Basiskomponenten (e₁ bis en) die Energien VME sind, die in n aufeinanderfolgenden Sprachsignal-Analysefenstern gemessen werden, in diesem Vektorraum eine Basisänderung durchzuführen, die als erste Hauptachse eine entsprechend der Summe der Einheitsvektoren der ersten Basis orientierte Achse besitzt, den resultierenden Energievektor in die neu erhaltene Basis zu projizieren (6, 7, 8, 9) und die auf die Hauptachse der neuen Basis projizierte Komponente des resultierenden Vektors mit q Bits der Art zu kodieren, daß 2q = m gilt, sowie die Komponenten des auf die n-1 anderen Hauptachsen des in der neuen Basis definierten Vektorraums projizierten Energievektors mit einer verringerten Anzahl von Bits kleiner als q zu kodieren.
- Verfahren nach Anspruch 1, dadurch gekennzeichnet, daß der Vektorraum ein dreidimensionaler Raum ist und daß die Einheitsvektoren entlang der Hauptachsen der neuen Basis als Komponente in der Basis, die durch die die in drei aufeinanderfolgenden Fenstern gemessenen Energien bildenden Vektoren des dreidimensionalen Raums folgende Werte besitzen: (3-½, 3-½, 3-½) für die erste Hauptachse, (-2-½, 0, 2-½) für die zweite Hauptachse und (-6½, 2x6½, -6½) für die dritte Hauptachse.
- Verfahren nach Anspruch 2, dadurch gekennzeichnet, daß die Kodierung des resultierenden Energievektors, der auf die erste, zweite und dritte Hauptachse projiziert ist, die Länge von 4, 3 bzw. 2 Bits besitzt.
- Vorrichtung zur Durchführung des Verfahrens nach einem beliebigen der Ansprüche 1 bis 3 mit Mitteln zur Analyse der Sprachsignale in aufeinanderfolgenden Fenstern, Mitteln (1, 2, 3) zur Messung der mittleren effektiven Energie (VME) der Tastproben des Sprachsignals in jedem der Fenster und zur Quantifizierung der Energie im Rahmen einer gegebenen Anzahl m von Pegeln, dadurch gekennzeichnet, daß einerseits Mittel zur Matrixberechnung (6, 7, 8, 9), um auf eine Vektorbasis, deren erste Hauptachse eine entsprechend der Summe der Einheitsvektoren (e₁ bis en) einer ersten Basis ausgerichtete Achse besitzt, einen in der Basis (e₁ bis en) konstruierten Energievektor zu projizieren, dessen Komponenten in dieser Basis die in n aufeinanderfolgenden Meßfenstern des Sprachsignals gemessenen Energiewerte VME sind, und andererseits Kodiermittel (12, 19) vorgesehen sind, um mit q Bits die Komponente des auf die Hauptachse der neuen Basis projizierten Energievektors mit q Bits, derart, daß 2q = m gilt, sowie mit einer verringerten Anzahl von Bits kleiner als q die Komponenten des auf die n-1 anderen Hauptachsen des in der neuen Basis definierten Vektorraums projizierten Energievektors zu kodieren.
- Vorrichtung nach Anspruch 4, dadurch gekennzeichnet, daß für n = 3 die Mittel zur Matrixberechnung (6, 7, 8, 9; 33...44) das folgende Matrixprodukt bilden:
- Vorrichtung nach Anspruch 4, dadurch gekennzeichnet, daß sie Mittel (35, ...44) aufweist, um in einer Gruppe von 2N Vektoren, die in einem Speicher (33) vorher registriert wurden und deren Enden mit den Knoten einer begrenzten Untereinheit eines kubischen Netzes mit zentrierter Seite koinzidieren, den Vektor zu suchen, dessen Komponenten in der neuen Basis die den Komponenten E₀, E₁ und E₂ der in drei aufeinanderfolgenden Fenstern des Sprachsignals gemessenen Energien am nächsten kommen, so daß sich der Kode des resultierenden Vektors mit N Bits ergibt.
- Vorrichtung nach Anspruch 6, dadurch gekennzeichnet, daß der Speicher (33) in N/3 Gruppen mit N/3 Untergruppen von je N/3 Vektoren strukturiert ist.
- Vorrichtung nach einem beliebigen der Ansprüche 6 und 7, dadurch gekennzeichnet, daß sie aufweist- einen Adressenzähler, um die vorher im Speicher (33) registrierten Vektoren zu adressieren,- Subtrahierkreise (35, 36, 37), um die von N Fenstern gelieferten N effektiven Mittelwerte des Sprachsignals mit den N entsprechenden Komponenten der aus dem Speicher (33) ausgelesenen Vektoren zu vergleichen,- und einen Entscheidungskreis (42, 43, 44), um den aus dem Speicher (33) ausgelesenen Vektor zu erfassen, der die den N gemessenen effektiven Mittelwerten des Sprachsignals nächstliegenden gelesenen Komponenten besitzt.
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
FR8806002A FR2631146B1 (fr) | 1988-05-04 | 1988-05-04 | Procede et dispositif de codage de l'energie du signal vocal dans des vocodeurs a tres faibles debits |
FR8806002 | 1988-05-04 |
Publications (2)
Publication Number | Publication Date |
---|---|
EP0341129A1 EP0341129A1 (de) | 1989-11-08 |
EP0341129B1 true EP0341129B1 (de) | 1993-06-23 |
Family
ID=9365992
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
EP89401169A Expired - Lifetime EP0341129B1 (de) | 1988-05-04 | 1989-04-25 | Verfahren und Einrichtung zur Kodierung von Sprachsignalenergie in Vokodern mit sehr niedriger Bitrate |
Country Status (7)
Country | Link |
---|---|
US (1) | US5016278A (de) |
EP (1) | EP0341129B1 (de) |
JP (1) | JPH01319100A (de) |
CA (1) | CA1312380C (de) |
DE (1) | DE68907267T2 (de) |
ES (1) | ES2041425T3 (de) |
FR (1) | FR2631146B1 (de) |
Families Citing this family (8)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
US5255339A (en) * | 1991-07-19 | 1993-10-19 | Motorola, Inc. | Low bit rate vocoder means and method |
US5473731A (en) * | 1993-07-20 | 1995-12-05 | Intel Corporation | Lattice based dynamic programming classification system |
FR2738383B1 (fr) * | 1995-09-05 | 1997-10-03 | Thomson Csf | Procede de quantification vectorielle de vocodeurs bas debit |
FR2778041A1 (fr) * | 1998-04-24 | 1999-10-29 | Thomson Csf | Procede de neutrodynage du tube d'un emetteur |
US6192283B1 (en) | 1998-07-31 | 2001-02-20 | Siemens Energy & Automation, Inc. | Method and apparatus for adaptive control of a system or device |
FR2788390B1 (fr) | 1999-01-12 | 2003-05-30 | Thomson Csf | Emetteur de radiodiffusion en ondes courtes a haut rendement optimise pour les emissions de type numerique |
FR2790343B1 (fr) | 1999-02-26 | 2001-06-01 | Thomson Csf | Systeme pour l'estimation du gain complexe d'un canal de transmission |
FR2799592B1 (fr) | 1999-10-12 | 2003-09-26 | Thomson Csf | Procede de construction et de codage simple et systematique de codes ldpc |
Family Cites Families (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
DE2608244C2 (de) * | 1976-02-28 | 1981-09-24 | Licentia Patent-Verwaltungs-Gmbh, 6000 Frankfurt | Verfahren zur Analyse und Synthese des Differenzsignals bei Prädiktionsvocodern |
-
1988
- 1988-05-04 FR FR8806002A patent/FR2631146B1/fr not_active Expired - Lifetime
-
1989
- 1989-04-25 DE DE89401169T patent/DE68907267T2/de not_active Expired - Fee Related
- 1989-04-25 EP EP89401169A patent/EP0341129B1/de not_active Expired - Lifetime
- 1989-04-25 ES ES198989401169T patent/ES2041425T3/es not_active Expired - Lifetime
- 1989-05-01 US US07/345,231 patent/US5016278A/en not_active Expired - Fee Related
- 1989-05-03 CA CA000598628A patent/CA1312380C/fr not_active Expired - Fee Related
- 1989-05-06 JP JP1114012A patent/JPH01319100A/ja active Pending
Also Published As
Publication number | Publication date |
---|---|
EP0341129A1 (de) | 1989-11-08 |
DE68907267D1 (de) | 1993-07-29 |
JPH01319100A (ja) | 1989-12-25 |
FR2631146B1 (fr) | 1991-05-10 |
CA1312380C (fr) | 1993-01-05 |
DE68907267T2 (de) | 1993-09-30 |
FR2631146A1 (fr) | 1989-11-10 |
US5016278A (en) | 1991-05-14 |
ES2041425T3 (es) | 1993-11-16 |
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