EP2589045B1 - Adaptive lineare prädiktive codierung/decodierung - Google Patents

Adaptive lineare prädiktive codierung/decodierung Download PDF

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EP2589045B1
EP2589045B1 EP11737984.2A EP11737984A EP2589045B1 EP 2589045 B1 EP2589045 B1 EP 2589045B1 EP 11737984 A EP11737984 A EP 11737984A EP 2589045 B1 EP2589045 B1 EP 2589045B1
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Prior art keywords
filter
current block
signal
backward
modified
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French (fr)
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EP2589045A1 (de
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Pierrick Philippe
David Virette
Claude Lamblin
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Orange SA
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Orange 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
    • G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • 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
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • 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
    • G10L19/18—Vocoders using multiple modes

Definitions

  • the object of the invention relates to the field of encoding / decoding audio data.
  • the invention may relate to the coding of sounds with alternating speech and music.
  • CELP Code-excited linear prediction
  • This number of coefficients P is chosen in order to correctly model the formational structure of the speech signal. Since the speech signal generally has four formants in the frequency band 0 to 4 kHz, ten filter coefficients correctly model this structure (two coefficients are necessary to model each formant).
  • an LPC order of 16 coefficients is typically employed.
  • the power of the residual signal r n can be small and its spectrum flat, by a judicious choice of the coefficients a i .
  • the residual signal is then easier to code than the signal s n itself. It can be easily modeled by a harmonic signal, highly periodic, as represented on the figure 2 , or X ( f ) is the spectrum of the original signal s (black line) and E ( f ) is the spectrum of the residual signal r (gray line).
  • the coefficients a i are typically calculated by a correlation measurement on the signal s n (and by applying a Levinson-Durbin type algorithm to invert the Wiener-Hopf equations).
  • the quality of the coding according to the AMR WB + is satisfactory for the audio signals consisting of speech mixtures with background noise or speech with musical background, so typically for the signals where the speech dominates in energy.
  • the envelope transmitted in LPC form is a relevant parameter since the signal consists mainly of speech which is described well thanks to a LPC envelope of a given order.
  • the envelope indeed describes the formants (related to the resonance frequencies of the vocal tract) as a function of the number of coefficients chosen.
  • the LPC envelope estimated and transmitted to the encoder is no longer sufficient.
  • the audio signal is then often too complex to be limited, for example, to five formants, and its evolution over time makes a fixed number of coefficients is not adapted.
  • the use of past information does not make it possible to anticipate the evolutions of the audio signal because using a backward predictor is relevant for a stationary signal but the spectrum at a given frame is precisely modeled and can be used to a next frame only if the statistical and especially spectral properties of the signal remain stable. Otherwise, the estimated LPC filter is irrelevant to the frame under consideration and the residual signal remains difficult to encode. The rear predictor loses all its interest.
  • a solution recommended in the known state of the art of the application for Patent No. FR 2 762 464 is therefore to use a switching between a prediction filter "before", calculated on the current frame, and a prediction filter, calculated on the previously received signal.
  • the encoder analyzes the signal and decides whether the signal is stationary or not. If the signal is stationary, the back filter is used. Otherwise, a forward filter with few coefficients is transmitted to the decoder.
  • Such an embodiment makes it possible to precisely control the quality of the signal residual to be encoded. It is implemented in the ITU-T G.729-E standard, in which a decision on stationarity of the signal results in a "back" estimated filter with 30 coefficients, or an estimated “forward” filter with 10 coefficients.
  • the present invention improves the situation.
  • the invention has many advantages: it makes it possible in particular not to switch abruptly from a rear filter to a front filter, but may for example offer the possibility of a transition by such a modified filter, in particular between the use of a rear filter and that of a front filter. It also avoids the passage through a filter before few coefficients to code a stationary signal with a complex envelope while it is only slightly disturbed by non-stationarity.
  • Another advantage is to enrich a rear filter by producing an optimum coding quality without necessarily transmitting a complete front filter, in particular with as many coefficients as for example a front filter.
  • Another advantage is to allow more choice to the encoder with different categories of filters: rear, front and modified.
  • Enrichment parameters include the coefficients of a modifying filter, and the modified filter is constructed by a combination of back filter and modifier filter.
  • This combination may be, in an exemplary embodiment described below, a convolution of the rear filter by the modifying filter. Alternatively, in another space, it may be a multiplication for example, or other.
  • Such an embodiment has the advantage of allowing a simplification of the calculation operations with a decoder receiving the aforementioned parameters.
  • This criterion can for example take into account a stationarity of the signal between the past block and the current block, for the choice of one of the filters from a rear filter, a front filter and a modified filter.
  • the predetermined criterion may comprise an estimate of a prediction gain based on a ratio between the power of the signal in the current block and the power of a residual signal after filtering this signal using each of the rear filters. , before and modified. Such an embodiment will be described in detail later, particularly with reference to Figures 4 and 5 .
  • the decoding method may then comprise a step in which, for the decoding of at least one given current block, the predicted filter thus modified is used instead.
  • this combination may consist of a multiplication or convolution (or other) of the rear filter by the modifying filter.
  • the decoder may also use a rear filter or a front filter, according to the information received from the encoder.
  • the back filter can be reconstructed based on previously decoded data. For example, it is possible to use the residual signal that the decoder has received from the encoder, for a past block, if the order of the rear filter to be reconstructed is higher than a previously constructed filter for this past block.
  • the "filter order" information may be transmitted directly from an encoder to the decoder, or may consist of implicit information.
  • the decoder can be programmed to compute a back filter at N1 coefficients if a modified filter is to be constructed and calculate a back filter at N2 coefficients for example if it is intended to use only a back filter for decoding.
  • the invention also proposes an encoding device according to claim 14, a decoding device according to claim 15 and a computer program according to claim 16.
  • the invention proposes a combination of rear filter and a modifying filter chosen to complement each other and to create a modified filter of better quality than the rear filter, since it is a version of the rear filter, enriched. by an update resulting from the characteristics taken from the current block.
  • the signal envelope is precisely described (for any type of signal), with an optimal transmission rate, whether in the form of a front filter, a rear filter or again a modified filter.
  • the transition between filter is smooth compared to the prior art and thus avoids the discontinuity effect described above with reference to the prior art.
  • the coding quality resulting from the use of the invention is then improved.
  • LPC Linear Predictive Coding
  • This technique can therefore be of the CELP type, for example according to the G.729, AMR or AMR-WB standards, or else a complementary coding transform can be used, for example in the sense of the G.718, G.729.1, AMR standards.
  • WB +, MPEG-D "Unified Speech and Audio Coding").
  • the number P denotes the number of non-zero coefficients. It is called "the order of the filter". Usually, a good number for a narrow-band speech signal (sampled at 8 kHz) is 10. This order can be increased, however, to better model the spectrum. signal and in particular to accentuate the precision of its envelope. It can also be increased if the signal sampling frequency is higher.
  • the coefficients a i of the LPC filter are commonly estimated by identifying the audio signal and its prediction in the least squares sense. We therefore seek the coefficients a i minimizing the quadratic error of the passed audio signal, through the filter A ( z ). It is therefore sought to minimize the power of the signal r n . This power is estimated over a certain period representing a number of samples N. The coefficients are therefore valid for this period of time.
  • This estimation of LPC filter coefficients is thus carried out by estimating the autocorrelation terms of the signal x n , and by solving the equations of Yule Walker or Wiener Hopf, typically by a fast algorithm of the Levinson Durbin type, as described by example in the reference: " Linear prediction a tutorial review ", John Makhoul, Proceedings of the IEEE, 63 (5): 561-580, April 1975 .
  • the estimation of the coefficients of the LPC filter can be performed on the current signal x n , on a frame representing a set of samples, or on a version of the signal x m (m ⁇ n) resulting from a previous local decoding ( complete or partial) of the signal in coded form.
  • Local decoding is obtained by decoding the encoded parameters at the encoder. This local decoding makes it possible to recover, at the level of the coder, the information that can be used by the decoder in the same way.
  • the performance of the LPC filter can then be evaluated by estimating the power of the residual signal (i.e. the signal strength resulting from the filtering of the original signal of the current frame by the LPC filter considered).
  • the ratio of the power of the original signal divided by the power of the residual signal gives a quantity called "prediction gain", often expressed in dB.
  • the LPC filters are estimated in forward mode, on the current frame, and in backward mode on the decoded previous frame. Their own prediction gain is then calculated.
  • the gain of the front LPC filter is always better than the gain of the rear LPC filter for a given order.
  • the rear LPC filter is not adapted to treat the current frame, but rather the previous frame.
  • it often happens (as in the case presented here as an example), in particular when the signal is stationary, that the gain of a rear LPC filter is greater than the prediction gain of a front LPC filter. lower order.
  • the prediction gain is greater in the backward mode with an order 24 than in the forward mode with an order of 10 or 16.
  • the filter f10 requires the transmission of its coefficients to the decoder, while the filter b24 is computable to the decoder without the need to transmit additional information.
  • the filter b24 has a much lower prediction gain than the prediction gain of the filter f24 (before filter of the same length).
  • This filter A hereinafter called “modified filter” is then used by the (possibly weighted) coder to calculate the residue.
  • An inverted version (1 / A (z) ) of this filter is used at the decoder to reshape the spectrum of the signal.
  • the modifying filter can be calculated conventionally by the Levinson Durbin algorithm acting on the signal derived from the filtering of the signal of the current frame by the determined back filter.
  • the modifying filter can be determined on the basis of an analysis of a residual signal obtained after filtering the current block by a back filter calculated for a past block.
  • the modifying filter (M) can be estimated by "deconvolution".
  • the modifying filter can be estimated, according to this first option, by deconvolving a front filter adapted for filtering the current block, by a back filter calculated for a past block.
  • the modifying filter can be estimated by a least squares Wiener identification method in which the autocorrelation terms of the filter are calculated.
  • rear r 0 , r 1 , r q-1
  • cross-correlation between the target front filter and the rear filter c 0 , c 1 ...
  • this second option can be implemented by least squares identification, by calculating terms of autocorrelation of coefficients of the back filter and intercorrelation between the modified filter and the back filter.
  • the second option can be executed in practice by a fast algorithm (of the type used for the identification of the LPC coefficients and based on the autocorrelation of the signal). Nevertheless, the first option of deconvolution can be as advantageous.
  • the filter M obtained by any of these techniques is then typically quantified in a form suitable for the transmission of the LPC filter coefficients (for example using a conversion of the LSF, LSP or ISF type (for "line spectral frequencies", or “peers”)). Once quantized, these coefficients are convoluted with the rear filter B to obtain a filter A (z) which can be reproduced identically to the decoder.
  • the filter A is of a higher order than the filter F (thus rendering its estimate expensive for the decoder since it involves the estimation of the filter B and the decoding of the filter M), the filter A is selected only if its gain of prediction is much higher than the filter F (a few dB).
  • An embodiment presented hereinafter therefore considers the calculation of a plurality of rear filters, before and modifying.
  • the number of front filters is not necessarily the same as the number of back filters.
  • a set of quantized modifying filters is calculated according to the method presented above. It is advisable to choose modifying filters with orders identical to the orders of the filters before F already calculated ( pf 0 , pf 1 , pf 2 , pf 3 ).
  • the convolution of the rear filters (B) and the modified filters (M) then gives a set of combined filters (A) whose performances are compared with those of the front filters (in particular those of the front filters having an order identical to the modified filter M).
  • a filter modifying (M 1,0 ) of order pf 0 is calculated to obtain a first filter A 0 .
  • a modifying filter (M 2,0 ) of order pf 0 is calculated to obtain a second filter A 1 .
  • a modifying filter (M 3.0 ) of order pf 0 is calculated to obtain a third filter A 2 .
  • a filter modifying (M 3.1 ) of order pf 1 is calculated to obtain a fourth filter A 3 .
  • the filters A 0 , A 1 and A 2 therefore have an identical transmission cost because they require the transport of pf 0 coefficients. This transmission cost can be considered identical to that of the filter F 0 .
  • the transmission cost of the filter A 3 is comparable to the transmission cost of the filter F 1 .
  • the type of front / rear / combined filter can change from one frame to the next, depending on the choice made at the encoder.
  • a change is useful only beyond a certain threshold (for example 1 dB).
  • the filter coefficients are assumed to be quantized in their ISF form. They are grouped together to be coded together.
  • a typical configuration used in the AMR-WB encoder (3GPP) is repeated in this embodiment. It is 46 bits for 16 LPC coefficients represented as ISF. For 10 coefficients, we will use 18 bits for example.
  • Reading the 2-bit index_pb flag is associated with a corresponding number of filter coefficients. For example, the following association can be provided: Index_pb pB 0 4 1 8 2 16 3 32
  • index_pf flag can be represented on a single bit: Index_pf pB 0 10 1 16
  • the coefficients f n are interpreted as the coefficients of the filter modifying the rear filter. Otherwise the coefficients f n are interpreted as front filter coefficients.
  • the syntax presented above can be arranged, or even simplified, if we reduce the number of combinations.
  • the index_pb field can be omitted if only one possible back filter order is considered.
  • the order of the rear filter can be implicitly set to 16.
  • a single length can be envisaged, for example 16.
  • the decoder On decoding, the decoder, on reading the information indicating the use of the rear filter and its order, calculates the rear filter of the order indicated on the decoded samples beforehand.
  • the decoder On receipt of the presence indication and the order of a filter, it decodes the transmitted ISF indices to convert the filter into LPC filter coefficients. Of course, here, if only the back filter is signaled (without ISF indexes), the decoder understands that the filter used is finally only the rear filter (B). If the two filters are transmitted (with the ISF indices), the decoder understands that the filter A used is the "modified" filter (obtained by the convolution of the front and back filters (B * M), the filter M being interpreted as the modifying filter).
  • the decoder understands that the filter used is the front filter alone.
  • the present invention provides an alternative to the coding of the LPC envelope, a critical element for the quality of coding, especially in audio coding.
  • an alternative mode of coding the LPC envelope does not cause any difficulty compared to current techniques: the encoder can always choose the standard mode LPC forward, as a fallback position.
  • the decoder is able to use rear filters, especially when the signal is stationary. Nevertheless, he is also able to take advantage of both approaches by combining them.
  • the performance of the LPC filter is further increased by increasing its accuracy to produce improved quality.
  • the encoder device determines a prediction gain Gp for a given bit rate d, by considering several types of filters before F, rear B and modified A and retains in step 12 the filter having for example the best prediction gain, at this given rate d.
  • the best candidate filter is a modified filter A (step 13)
  • the construction thereof implies a filter modifying Mj, the order j of this modifying filter being able to be chosen according to the order i of the rear filter Bi on the base of which is built the modified filter A.
  • the coefficients of the filter modifying Mj and the order i of the filter Bi can then be sent to a decoder device D2.
  • the present invention also relates to a computer program comprising instructions for the implementation of these steps, when this program is executed by a processor, for example such a coding device D1.
  • a processor for example such a coding device D1.
  • the organigram shown on the Figure 6B can illustrate the general algorithm of such a program.
  • this rear filter Bi is calculated from previously decoded data (for example from a previous frame T n -1 ) and using the filter order i.
  • the modifying filter Mj and the rear filter Bi calculated in this way are combined (for example by convolution) to obtain in step 23 the modified filter A used for decoding the signal by the decoder device D2 (step 24). , for a current frame to be delivered T n .
  • the present invention also relates to a computer program comprising instructions for the implementation of these steps, when this program is executed by a processor, for example such a decoding device D2.
  • a processor for example such a decoding device D2.
  • the organigram shown on the Figure 7B can illustrate the general algorithm of such a program.
  • the criterion of choosing a filter illustrated by the figure 5 may not be simply limited to the best prediction gain for a given rate.
  • another criterion that may be taken into consideration may be the complexity of calculations to the encoder or decoder. So, with reference again to the figure 5 , the modified filters A 0 and A 2 are the best candidates for the flow d 0 . It will then be chosen preferentially the filter A 0 , less complex than the filter A 2 , and nevertheless offering the same performance in terms of prediction gain.

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Claims (16)

  1. Verfahren zur Codierung eines digitalen Audiosignals, umfassend eine Aneinanderreihung von aufeinanderfolgenden Datenblöcken aus einem prädiktiven Filter,
    dadurch gekennzeichnet, dass es die Verwendung eines modifizierten prädiktiven Filters (A) für die Codierung mindestens eines laufenden Blocks umfasst, wobei der modifizierte Filter (A) konstruiert ist durch die Kombination:
    - eines hinteren Filters (B), der für einen vergangenen Block, der dem laufenden Block vorangeht, berechnet wurde, und
    - eines modifizierenden Filters (M) mit Koeffizienten, die in Abhängigkeit von dem Signal im laufenden Block bestimmt werden.
  2. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass es für die Codierung eines laufenden Blocks eine Auswahl umfasst, die auf mindestens einem vorbestimmten Kriterium eines prädiktiven Filters basiert, unter mindestens:
    - einem hinteren Filter (B), der für einen vergangenen Block, der dem laufenden Block vorangeht, berechnet wurde, und
    - einem vorderen Filter (F), der für den laufenden Block angepasst ist, und
    - einem modifizierten Filter (A), der auf Basis eines hinteren Filters und in Abhängigkeit von dem Signal im laufenden Block geschätzt wird.
  3. Verfahren nach Anspruch 2, dadurch gekennzeichnet, dass das Kriterium einen stationären Charakter des Signals zwischen dem vergangenen Block und dem laufenden Block für die Auswahl eines der Filter unter einem hinteren Filter, einem vorderen Filter und einem modifizierten Filter berücksichtigt.
  4. Verfahren nach Anspruch 3, dadurch gekennzeichnet, dass das vorbestimmte Kriterium eine Schätzung einer Prädiktionsverbesserung basierend auf einem Verhältnis zwischen der Leistung des Signals im laufenden Block und der Leistung eines Restsignals nach dem Filtern dieses Signals unter Verwendung jedes der hinteren, vorderen und modifizierten Filter umfasst.
  5. Verfahren nach einem der Ansprüche 3 und 4, dadurch gekennzeichnet, dass das Kriterium ferner eine Anzahl von Parametern berücksichtigt, die an einen Decoder für die Decodierung eines laufenden Blocks zu senden sind und umfassend mindestens Koeffizienten, die der auszuwählende Filter enthält.
  6. Verfahren nach Anspruch 5 in Kombination mit Anspruch 4, dadurch gekennzeichnet, dass das vorbestimmte Kriterium eine Optimumsuche umfasst zwischen:
    - der Prädiktionsverbesserung, die der Filter bietet, und
    - einer Menge, die für die Übertragung der Parameter angepasst ist.
  7. Verfahren nach einem der vorhergehenden Ansprüche, dadurch gekennzeichnet, dass es die folgenden Schritte umfasst:
    a) Bestimmung einer Vielzahl von vorderen Filtern (F0, F1, F2, F3) mit jeweils unterschiedlichen Befehlen,
    b) Bestimmung einer Vielzahl von hinteren Filtern (B0, B1, B2, B3) mit jeweils unterschiedlichen Befehlen,
    c) Berechnen einer Vielzahl von modifizierten Filtern (A0, A1, A2, A3) mit jeweils unterschiedlichen Befehlen, die jeweils auf Basis eines in Schritt b) bestimmten hinteren Filters und in Abhängigkeit von dem Signal im laufenden Block geschätzt werden,
    d) für eine selbe Anzahl von an einen Decoder zu sendenden Parametern, wobei diese Anzahl in Abhängigkeit von den Filterbefehlen bestimmt wird, Vergleichen der Leistungen mindestens zweier Filter unter den vorderen Filtern, hinteren Filtern und modifizierten Filtern, und
    e) für die Codierung eines laufenden Blocks, Auswahl eines prädiktiven Filters, der die besten Leistungen nach dem Vergleich des Schrittes d) für eine gegebene Anzahl von an einen Decoder zu sendenden Parametern aufweist.
  8. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass der modifizierende Filter (M) durch Dekonvolution eines vorderen Filters (F), der zum Filtern des laufenden Blocks angepasst ist, durch den hinteren Filter, der für einen vergangenen Block berechnet wurde, geschätzt wird.
  9. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass der modifizierende Filter (M) auf Basis einer Analyse eines Restsignals, das nach dem Filtern des laufenden Blocks durch den hinteren Filter, der für einen vergangenen Block berechnet wurde, geschätzt wird.
  10. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass der modifizierende Filter (M) durch Identifikation im Sinne der kleinsten Quadrate geschätzt wird, wobei die Termini der Autokorrelation von Koeffizienten des hinteren Filters (B) und der Interkorrelation zwischen dem modifizierten Filter (A) und dem hinteren Filter (B) berechnet werden.
  11. Verfahren nach Anspruch 1, dadurch gekennzeichnet, dass es ferner eine Kommunikation zu einem Decoder von Informationen folgenden Typs umfasst:
    - Auswahl eines vorderen Filters für einen laufenden Block mit einer Übertragung von Parametern, die Koeffizienten des vorderen Filters darstellen,
    - oder Auswahl eines hinteren Filters oder eines modifizierten Filters für einen laufenden Block mit, im Falle einer Auswahl eines modifizierten Filters, einer Übertragung von Parametern, die Koeffizienten des modifizierenden Filters (M) darstellen.
  12. Verfahren zum Decodieren eines digitalen Audiosignals, umfassend eine Aneinanderreihung von aufeinanderfolgenden Datenblöcken, wobei das Verfahren einen prädiktiven Filter für die Decodierung eines laufenden Blocks verwendet, dadurch gekennzeichnet, dass es umfasst:
    - einen Empfang von Informationen, umfassend Filterkoeffizienten, die Anreicherungsparameter eines hinteren Filters darstellen, für die Berechnung eines modifizierten prädiktiven Filters (A), basierend auf der Kombination:
    * eines hinteren Filters (B), der für einen vergangenen Block, der dem laufenden Block vorangeht, berechnet wurde, und
    * eines modifizierenden Filters (M) mit Koeffizienten, die aus den empfangenen Informationen stammen,
    - und eine Verwendung des so modifizierten prädiktiven Filters (A) für die Decodierung mindestens eines laufenden Blocks.
  13. Verfahren nach Anspruch 12, dadurch gekennzeichnet, dass es für die Bestimmung des hinteren Filters (B) die folgenden Schritte umfasst:
    - Bestimmung eines Befehls des hinteren Filters (B) in Abhängigkeit von den empfangenen Informationen, und
    - Berechnen des hinteren Filters auf Basis von vorher decodierten Daten und unter Verwendung des Filterbefehls.
  14. Vorrichtung zur Codierung eines Signals für den Einsatz des Verfahrens nach einem der Ansprüche 1 bis 11, dadurch gekennzeichnet, dass sie mindestens umfasst:
    - Mittel, die zur Berechnung eines modifizierten prädiktiven Filters auf Basis eines hinteren Filters und eines modifizierenden Filters in Kombination mit dem hinteren Filter geeignet sind, wobei der modifizierende Filter Koeffizienten hat, die in Abhängigkeit von dem Signal im laufenden Block bestimmt werden, und
    - Mittel zum Codieren mindestens eines laufenden Blocks unter Verwendung des modifizierten prädiktiven Filters.
  15. Vorrichtung zur Decodierung eines Signals für den Einsatz des Verfahrens nach einem der Ansprüche 12 bis 13, dadurch gekennzeichnet, dass sie mindestens umfasst:
    - Mittel für den Empfang von Informationen, umfassend Filterkoeffizienten, die Anreicherungsparameter eines hinteren Filters darstellen, für die Berechnung eines modifizierten prädiktiven Filters (A),
    - Mittel, die für die Berechnung des modifizierten Filters angepasst sind, basierend auf:
    * einem hinteren Filter (B), der für einen vergangenen Block, der dem laufenden Block vorangeht, berechnet wurde, und
    * einem modifizierenden Filter (M) in Kombination mit dem hinteren Filter, mit Koeffizienten, die von den empfangenen Informationen stammen,
    - und Mittel zum Decodieren mindestens eines laufenden Blocks unter Verwendung des modifizierten prädiktiven Filters.
  16. Informatikprogramm, umfassend Instruktionen für den Einsatz des Codierungsverfahrens nach einem der Ansprüche 1 bis 11 und/oder für den Einsatz des Decodierungsverfahrens nach einem der Ansprüche 12 bis 13, wenn dieses Programm von einem Prozessor ausgeführt wird.
EP11737984.2A 2010-06-29 2011-06-17 Adaptive lineare prädiktive codierung/decodierung Active EP2589045B1 (de)

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FR1055206A FR2961937A1 (fr) 2010-06-29 2010-06-29 Codage/decodage predictif lineaire adaptatif
PCT/FR2011/051393 WO2012001260A1 (fr) 2010-06-29 2011-06-17 Codage/decodage predictif lineaire adaptatif

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US9583115B2 (en) * 2014-06-26 2017-02-28 Qualcomm Incorporated Temporal gain adjustment based on high-band signal characteristic
JP6517924B2 (ja) * 2015-04-13 2019-05-22 日本電信電話株式会社 線形予測符号化装置、方法、プログラム及び記録媒体

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US4328585A (en) * 1980-04-02 1982-05-04 Signatron, Inc. Fast adapting fading channel equalizer
US5533052A (en) * 1993-10-15 1996-07-02 Comsat Corporation Adaptive predictive coding with transform domain quantization based on block size adaptation, backward adaptive power gain control, split bit-allocation and zero input response compensation
JP3064947B2 (ja) * 1997-03-26 2000-07-12 日本電気株式会社 音声・楽音符号化及び復号化装置
FR2762464B1 (fr) * 1997-04-16 1999-06-25 France Telecom Procede et dispositif de codage d'un signal audiofrequence par analyse lpc "avant" et "arriere"
EP0932141B1 (de) * 1998-01-22 2005-08-24 Deutsche Telekom AG Verfahren zur signalgesteuerten Schaltung zwischen verschiedenen Audiokodierungssystemen
US6449590B1 (en) * 1998-08-24 2002-09-10 Conexant Systems, Inc. Speech encoder using warping in long term preprocessing
US7072832B1 (en) * 1998-08-24 2006-07-04 Mindspeed Technologies, Inc. System for speech encoding having an adaptive encoding arrangement
US6456964B2 (en) * 1998-12-21 2002-09-24 Qualcomm, Incorporated Encoding of periodic speech using prototype waveforms
US7302387B2 (en) * 2002-06-04 2007-11-27 Texas Instruments Incorporated Modification of fixed codebook search in G.729 Annex E audio coding
DE102004025471A1 (de) * 2004-05-21 2005-12-15 Micronas Gmbh Verfahren bzw. adaptives Filter zum Verarbeiten einer Folge aus Eingabe-Daten eines Funksystems
KR101393298B1 (ko) * 2006-07-08 2014-05-12 삼성전자주식회사 적응적 부호화/복호화 방법 및 장치
EP1883067A1 (de) * 2006-07-24 2008-01-30 Deutsche Thomson-Brandt Gmbh Verfahren und Vorrichtung für verlustfreie Kodierung eines Quellsignals, unter Benutzung eines verlustbehafteten kodierten Datenstroms und eines verlustfreien Erweiterungsdatenstroms.
JP4299323B2 (ja) * 2006-08-04 2009-07-22 株式会社日立国際電気 通信システム
EP2054878B1 (de) * 2006-08-15 2012-03-28 Broadcom Corporation Beschränkte und kontrollierte entschlüsselung nach paketverlust

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FR2961937A1 (fr) 2011-12-30
EP2589045A1 (de) 2013-05-08
WO2012001260A1 (fr) 2012-01-05
US9620139B2 (en) 2017-04-11
US20130103408A1 (en) 2013-04-25

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