EP2656344A1 - Improved filtering in the transformed domain - Google Patents
Improved filtering in the transformed domainInfo
- Publication number
- EP2656344A1 EP2656344A1 EP11817369.9A EP11817369A EP2656344A1 EP 2656344 A1 EP2656344 A1 EP 2656344A1 EP 11817369 A EP11817369 A EP 11817369A EP 2656344 A1 EP2656344 A1 EP 2656344A1
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- equalization
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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
-
- 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
- G10L19/26—Pre-filtering or post-filtering
-
- 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
- G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
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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
- G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
- G10L19/0204—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 spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
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- 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
- G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
- G10L19/0212—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 spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
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- 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
- G10L19/02—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 spectral analysis, e.g. transform vocoders or subband vocoders
- G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
Definitions
- the present invention relates to the filtering of digital data, especially the filtering of digital audio data.
- rate-reduction audio coding requires post-processing filtering of audio decoding that can manifest itself in various forms:
- SBR processing modulates high frequencies in low frequency areas and adjusts signal energy to frequency. This adjustment makes it possible, after decoding, to obtain a signal similar to the original signal (signal before coding).
- the PS process recreates from a mono signal two composite signals that are tuned into frequency energy so that, again, the decoded signal resembles the original reference signal.
- Expanding MPS processing is based on the generation of N signals from M transmitted sound channels (with N> M).
- Critical sampling is an important property in rate reduction coding. Indeed, to maintain a good transmission efficiency, it should not transmit more processed samples than there were in the time domain. For this reason, in current flow rate encoders, only critical sampling transforms are employed. These are, for example, MDCT transforms (for "Modified Discrete Cosine Transform") which are typically real-factor transforms.
- the filtering then consists of a simple multiplication by transformed coefficient (equalization).
- a filter-adapted transformation for example a short-term Fourier transform or another complex-valued transformation, for example complex filters of the PQMF type for "Pseudo Quadrature Mirror Filters
- the filtering then consists of a simple multiplication by transformed coefficient (equalization).
- the present invention improves the situation. To this end, it proposes a method of processing a signal in the form of successive blocks of samples, comprising a filtering in a transformed subband domain.
- the method comprises: an equalization treatment applied to a current block in the transformed domain, and a filter adjustment processing applied in the transformed domain to at least one block adjacent to the current block.
- block is meant any succession of samples, such as a frame, or a sub-frame in certain types of signal formats.
- the invention provides improved filtering in the transformed domain.
- This approach is advantageously not very complex because the treatment remains in the field of the initial transform.
- One advantage provided is a limitation of audible folding components, while faithfully maintaining the filtering characteristic initially desired.
- the adjustment processing filtering is performed by a matrix applied to said at least one block adjacent to the current block, the matrix having upper and lower diagonals identical to the near sign.
- the method then comprises a preliminary step of optimizing parameters of the equalization and the adjustment of the filtering, by estimating a folding induced by the equalization.
- Folding is preferentially estimated in a domain obtained from an inverse transform of the domain of the subbands (for example in the time domain).
- the equalization and the filtering adjustment in the transformed domain comprise: an equalization process applied to a current block, a filter adjustment processing applied to at least one block preceding the current block in time, and a filter adjustment processing applied to at least one block following the current block in time.
- an equalization process applied to a current block a filter adjustment processing applied to at least one block preceding the current block in time
- a filter adjustment processing applied to at least one block following the current block in time it is proposed to rely both on the previous block, but also on the block immediately following the current block.
- equalization and filtering adjustment includes applying a matrix system comprising: a first matrix applied to a signal vector representing the current block, a second matrix applied to a signal vector representing the previous block, and a third matrix applied to a signal vector representing the next block.
- the third matrix is the transpose of the second matrix.
- the invention proposes in particular symmetrical structures (for example by filtering in a random-modulated cosine-modulated filter bank) that make it possible to obtain simple functions to be performed.
- the blocks are transformed in the subband domain by at least one modulated transform, for example of the MDCT type.
- the transform may be of the modulated transform type, with complex values (for example of the MCLT type, or else of the PQMF type).
- the equalization and the filtering adjustment comprise the application of a matrix system comprising at least:
- the first matrix (T 0 ) applied to the signal vector of the current block comprises, as the only non-zero elements, a succession of identical elements A, in the diagonal of the matrix, followed by an element AB for a given subband and d.
- the second matrix ( ⁇ ) applied to the signal vector of the adjacent block comprises as the only non-zero elements at least two elements of identical absolute value and of opposite signs, arranged on the diagonal of the matrix, respectively for the given subband and for the subband that follows the given subband.
- the present invention makes it possible to implement structures for correcting low-pass, band-pass or other filters in the domain with real or complex values, by means of simple functions as described below.
- the filtering includes a cutoff component beyond a subband corresponding to said given subband.
- the second and third matrices comprise a number of non-zero elements which is a function of a chosen degree of optimization of the parameters of the filter adjustment, minimizing the estimated folding.
- the invention proposes efficient structures in calculation, with a limited number of coefficients to be added. Better yet, it is possible to choose the number of matrix coefficients to manage according to a desired complexity, or alternatively a compromise between complexity and limitation of folding.
- the first matrix is expressed in a form:
- the coefficient -ai of the diagonal being applied for the given subband the third matrix being transposed from the second matrix.
- the coefficients a 0 , "i, a 2 , ⁇ 3 ⁇ 4, a 4 and a 5 are positive real numbers, the real ai, at least, being nonzero.
- a 0 can be zero for a given matrix, which can be offset by another matrix combined with this given matrix.
- the correction matrix system within the meaning of the invention comprises at least: a corresponding linear combination of first matrices applied to the signal vector of the current block, a linear combination second matrices applied to the signal vector of the previous block, and - a linear combination of third respective transposed matrices of the second matrices, applied to the signal vector of the next block.
- the approach in the sense of the invention can be generalized to any filtering and equalization functions, by using filtering adjustment coefficients adapted from an analysis of the distortion to be corrected.
- the present invention is also directed to a computer program comprising instructions for implementing the above method when this program is executed by a processor.
- An exemplary flowchart of the general algorithm of such a program is described below with reference to FIG. 13.
- the present invention also provides a device for processing a signal in the form of successive blocks of samples, comprising filtering means in a transformed subband domain. These means furthermore apply: an equalization process to a current block in the transformed domain, and a filter adjustment processing, in the transformed domain, to at least one block adjacent to the current block.
- FIG. 1A schematically illustrates a first processing performing a filtering S (z), then carrying out a direct transformation followed by an inverse transformation
- FIG. 1B schematically illustrates a second processing proceeding to a direct transformation, followed by the desired subband processing S sb (z), and finally realizing the inverse transform, to distinguish, with FIG. 1A, two approaches to polyphase systems
- FIG. 2 schematically illustrates the multiplication of a scalar in each subband of the transformed domain to represent any filtering
- FIG. 3 illustrates the appearance of a linear filtering (filter low pass) applied in matrix form in the transformed domain
- FIG. 4 details the frequency profile of the filter of FIG.
- FIG. 5 represents the distortion (on the ordinate), decreased by optimization of the equalization parameter to 0 (on the abscissa), in an embodiment without filtering adjustment
- FIG. 6 represents the frequency characteristics of the filter resulting from the optimization of FIG. 7 shows the frequency characteristics of the filter resulting from the optimization of the equalization and the filtering adjustment
- FIG. 8 represents the reduction of distortion observed due to the (aliased) folding in FIG. a function of the number of coefficients involved in the equalization and the filtering adjustment (abscissa)
- FIG. 9 illustrates the filtering, equalization and filtering adjustment function, performed using a set of coefficients in the case of a bandpass filter
- FIG. 10 illustrates the case of a complex-valued modulated transform (of the MCLT type)
- FIG. 10 illustrates the case of a complex-valued modulated transform (of the MCLT type)
- FIG. 12 compares the reduction of distortion observed due to the folding (ordinate) as a function of the number of coefficients involved in the equalization and the adjustment of filtering (abscissa), for a transform with real values (of type MDCT, in solid line) and for a complex valued transform (of type MCLT, in dotted lines), for a band-pass filtering, figure 13 summarizes the steps a method in the sense of the invention, in an exemplary embodiment, and Figure 14 schematically illustrates a device for implementing the invention, as an exemplary embodiment.
- FIGS. 1A and 1B The two treatments are respectively illustrated in FIGS. 1A and 1B.
- the analysis filter bank (or the direct transform) is expressed by its polyphase matrix in the order M, E (z).
- the synthesis filter bank (or inverse transform) is expressed by its polyphase matrix in the order M, R ( z ).
- M represents the number of transformed coefficients (i.e. the number of frequency coefficients obtained by the transform).
- the polyphase decomposition of the modulated transforms is expressed by:
- the polyphase components of the transforms are also written as follows, based on the impulse responses of the analysis filters h a n for the subband k and the coefficient n. In this example, however, with no loss of generality, we restrict our to a transform whose impulse responses have a length 2M, such as the MDCT.
- an is a prototype (or window) analysis filter containing 2M samples, some of which may be null (especially those with higher indices).
- the filter h Sj is here a prototype filter (called” synthesis window ”) containing 2M samples, some of which may be zero (in particular those of the weakest indices).
- the reconstruction is perfect insofar as the modulations and the analysis and synthesis filters ensure the following conditions:
- the MDCT transform is therefore perfect reconstruction (at the cost of a delay of one frame, that is to say M samples, in the case of a signal comprising a succession of frames of M samples each).
- S lin (z) is a filter matrix taking the following circulating form (guaranteeing that it corresponds to a linear filter that can be made in the form of a convolution):
- the coefficients of the linear filter and S aiias (z) is any filter matrix that represents components corresponding to folds due to inversions in the signal along the time and / or frequency axis and creating additional associated components.
- S lin (z) we preferentially use an estimation in the sense of least squares, minimizing the power of the term S alias (z). We therefore try to observe the principal contribution of linear filtering present in the matrix S sb (z).
- the terms of the matrix S lin (z) can be calculated by estimating the mean of the diagonal terms of the matrix S (z), as follows:
- the folding component which contains the non-linear component filters, is calculated by the difference of the two matrices:
- the power of this matrix is deduced by summing the square of the coefficients of this matrix, in order to estimate the amount of folding created.
- a linear part representing a linear filter (corresponding to a conventional filtering function in signal processing); this filtering has the effect of modifying the spectrum of the signal by attenuating or amplifying the signal in certain frequencies, and A nonlinear part which contains folding components, considered undesirable, and a measure of the power of these undesirable components.
- Exemplary embodiments of the invention are now described using such a measurement.
- it is proposed to study the transfer function obtained in the time domain after multiplication of the MDCT components.
- An example of multiplication is the application of a multiplication by a scalar T k of each component resulting from the transformation MDCT, as illustrated in FIG. 2. This multiplication processing of each component T k is called equalization.
- An example of a feasible function for filtering is the following: , which amounts to writing:
- the position of the coefficient 0 (line i, column i) corresponds to that of the last coefficient at "1" in the uncorrected filtering matrix of the conventional low-pass filter and that of the coefficient a 0 corresponds to the line i + 1, column i + 1.
- the evolution of the aliasing distortion (aliasing) is measured.
- the distortion is then lowered to -29.16 dB, which is equivalent to an improvement of 4.47 dB compared to the current situation (in the sense of the state of the art).
- the filter resulting from this modification also has characteristics close to the desired initial filter, as illustrated in FIG.
- a first matrix ⁇ is proposed to be applied to the previous frame (in the form of a signal vector) and, because of the symmetry of this type of transform, a second matrix ⁇ which is deduced from the first matrix ⁇ , to be applied to the next frame.
- the second matrix ⁇ corresponds in particular to the transpose of the first matrix ⁇ .
- the matrix ⁇ is therefore intended to reduce the level of folding introduced by the matrix T 0 which performs an equalization function.
- the filter matrix in this embodiment of the invention, is therefore perfectly described by the following diagonal elements:
- the resulting linear filter is always very close to the desired function, as shown in FIG. 7 illustrating a low-pass filter of very similar characteristics.
- the content of the diagonal of the matrix T 0 applied to the current frame finally sets the desired frequency mask.
- the representation can be "degraded" by increasing the number of zero coefficients, for example by imposing a value of 5 to zero.
- the optimal subband filter solution introduces an aliasing level of -42.90 dB (instead of -45.31 dB).
- FIG. 8 shows an embodiment in which the state of the art advocating the use of 16 "classical” coefficients, it is proposed to add here between 1 and 29 non-zero coefficients (with the choice of a set of six basic coefficients at 0 , "i,” 5) to obtain a reduction in spectral aliasing power ranging from 4.47 to 20.6 dB.
- the example of 29 non-zero coefficients added in the matrices ⁇ , T 0 and ⁇ corresponds to the addition of: - the coefficient a 0 in the matrix T 0 ,
- a low-pass filter can be transposed to any form of filter, for example a band-pass filter.
- T k 1 for M / 8 ⁇ k ⁇ M / 4
- T k 0 for M 14 ⁇ k ⁇ M
- the filtering function performed is illustrated in FIG. 9 and the level of aliasing distortion corresponds to -21.68 dB.
- An exemplary embodiment relates to transformations MCLT (for "Modulated Complex Lapped
- an MCLT transform consists of two components:
- the matrices Si contain the terms in sine:
- the transform MCLT is thus to carry out two direct transforms:
- the filter matrix S sb has the same number of coefficients as for the MDCT transform.
- the number of coefficients to be applied in total is double, as illustrated in FIG. 11 (for a low-pass filter) and the figure
- the shape of the resulting filter may be of the type:
- any kind of equalization template can be reproduced by combining basic weighted pass functions (and / or low-pass functions), weighted, summing them.
- these gains are reflected on the main diagonal of the matrix T 0 .
- These gains can be adjusted to limit the discontinuities from one band to another (typically the values of the coefficients a 0 ).
- the respective gains are weighted by folding reduction templates defining a modification of the matrices ⁇ and ⁇ intended to weight the preceding and following frames in the transformed domain.
- the filtering is then applied.
- a spectrum is obtained in the MDCT transformed domain, and an inverse MDCT transformation is applied to obtain a time signal having the desired filtering characteristics.
- This approach extends to a complex valued transform (for example of the MCLT type) and more generally to any modulated transform, complex value or not.
- FIG. 13 summarizes the main steps of an exemplary embodiment of a method within the meaning of the invention.
- a first step 10 for a given filtering, for example for a bandpass (a low-pass or a high-pass being considered as special cases of bandpass), all the sets of coefficients are determined:
- This step 10 can be performed beforehand, "offline".
- a set of coefficients ⁇ 0 , ai, a h is chosen with i less than or equal to n, such that the index i complies with a compromise of complexity / quality of the filtering, set for example according to the computing capacity of a terminal or other, or setting a quality level depending on the sound coding quality.
- a coded sound signal is inevitably distorted: it is therefore unnecessary to reduce the folding to significant values below the noise level generated by the coding.
- the value of the index i can be determined at this step 11, as a function of the above-mentioned conditions, and in the following step 12, for example, from a memory, the set of coefficients corresponding to the value chosen for the index i.
- the elementary filterings (F 1 , F 2 ,..., F k ) constituting this given filtering, for example a low-pass F 1 (step 13), a pass, are determined.
- F 2 (thus forming a pass-band, by subtraction, with the elementary filtering F 1 ), a complex-party filtering F k (step 20), or others.
- the global filtering F may result from a linear combination of elementary filtering:
- the present invention also provides a computer program comprising instructions for implementing the method in the sense of the invention when the program is executed by a processor. It will be understood that FIG. 13 may correspond to a flowchart of the general algorithm of such a program.
- the present invention also provides a device for implementing the method and then comprising filtering means in a transformed subband domain. In particular, these means apply:
- a filter adjustment processing in the transformed domain, to at least one block adjacent to the current block.
- FIG. 14 shows an exemplary embodiment of such a device, with, in the example shown:
- a second optional input to receive conditions for determining the index i mentioned above, so as to choose a set of coefficients adapted to a compromise of complexity / quality of the filtering
- signal processing means such as, for example, a processor PROC and a working memory MEM,
- embodiments have been presented at three consecutive spectra resulting from the processing of successive frames by the matrices Ti, T 0 , ⁇ . Nevertheless, the number of frames to be processed may be greater if it is desired to produce longer finite impulse response filters.
- the equalization filtering adjustment processing can be performed. to apply to a number of frames before the current frame, different from the number of frames after the current frame. For example, it is possible to process only one frame adjacent to the current frame (anterior or posterior). In this case, an asymmetric linear filter is obtained.
- Processing matrices in particular matrices ⁇ and ⁇
- MDCT transforms in particular with regard to the position of non-null matrix elements
- these matrix forms are susceptible to variations for other types of transforms.
- the matrix ⁇ may not be in the form of the transpose of the matrix ⁇ for a type of transform different from the MDCT transform with Malvar filters as realized here.
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FR1061187A FR2969804A1 (en) | 2010-12-23 | 2010-12-23 | IMPROVED FILTERING IN THE TRANSFORMED DOMAIN. |
PCT/FR2011/053024 WO2012085410A1 (en) | 2010-12-23 | 2011-12-16 | Improved filtering in the transformed domain |
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Publication number | Priority date | Publication date | Assignee | Title |
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EP2665208A1 (en) * | 2012-05-14 | 2013-11-20 | Thomson Licensing | Method and apparatus for compressing and decompressing a Higher Order Ambisonics signal representation |
CN104471641B (en) | 2012-07-19 | 2017-09-12 | 杜比国际公司 | Method and apparatus for improving the presentation to multi-channel audio signal |
CN103414678B (en) * | 2013-08-02 | 2016-08-03 | 浙江大学 | The transform domain equalization methods of dual-selection channel based on Vector OFDM |
CN105556600B (en) * | 2013-08-23 | 2019-11-26 | 弗劳恩霍夫应用研究促进协会 | The device and method of audio signal is handled for aliasing error signal |
EP3267646B1 (en) * | 2016-07-06 | 2021-06-02 | Nxp B.V. | Iq mismatch correction module |
CN112243125B (en) * | 2020-10-20 | 2022-07-12 | 浙江大华技术股份有限公司 | Video encoding method, electronic device, and storage device |
Family Cites Families (12)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
SE0101175D0 (en) * | 2001-04-02 | 2001-04-02 | Coding Technologies Sweden Ab | Aliasing reduction using complex-exponential-modulated filter banks |
KR20030070177A (en) * | 2002-02-21 | 2003-08-29 | 엘지전자 주식회사 | Method of noise filtering of source digital data |
AU2003260958A1 (en) * | 2002-09-19 | 2004-04-08 | Matsushita Electric Industrial Co., Ltd. | Audio decoding apparatus and method |
US7720644B2 (en) * | 2005-09-16 | 2010-05-18 | Agilent Technologies, Inc. | Method and apparatus of spectral estimation |
FR2899423A1 (en) * | 2006-03-28 | 2007-10-05 | France Telecom | Three-dimensional audio scene binauralization/transauralization method for e.g. audio headset, involves filtering sub band signal by applying gain and delay on signal to generate equalized and delayed component from each of encoded channels |
DE102006047197B3 (en) * | 2006-07-31 | 2008-01-31 | Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. | Device for processing realistic sub-band signal of multiple realistic sub-band signals, has weigher for weighing sub-band signal with weighing factor that is specified for sub-band signal around subband-signal to hold weight |
USRE50009E1 (en) * | 2006-10-25 | 2024-06-11 | Fraunhofer-Gesellschaft Zur Foerderung Der Angewandten Forschung E.V. | Apparatus and method for generating audio subband values and apparatus and method for generating time-domain audio samples |
EP2077550B8 (en) * | 2008-01-04 | 2012-03-14 | Dolby International AB | Audio encoder and decoder |
EP2099027A1 (en) * | 2008-03-05 | 2009-09-09 | Deutsche Thomson OHG | Method and apparatus for transforming between different filter bank domains |
TWI559679B (en) * | 2009-02-18 | 2016-11-21 | 杜比國際公司 | Low delay modulated filter bank and method for the design of the low delay modulated filter bank |
US9124233B2 (en) * | 2010-08-25 | 2015-09-01 | Vixs Systems, Inc | Audio equalizer and methods for use therewith |
WO2012050705A1 (en) * | 2010-10-14 | 2012-04-19 | Dolby Laboratories Licensing Corporation | Automatic equalization using adaptive frequency-domain filtering and dynamic fast convolution |
-
2010
- 2010-12-23 FR FR1061187A patent/FR2969804A1/en not_active Withdrawn
-
2011
- 2011-12-16 CN CN201180067872.4A patent/CN103384901B/en active Active
- 2011-12-16 EP EP11817369.9A patent/EP2656344B1/en active Active
- 2011-12-16 US US13/995,718 patent/US9847085B2/en active Active
- 2011-12-16 WO PCT/FR2011/053024 patent/WO2012085410A1/en active Application Filing
Also Published As
Publication number | Publication date |
---|---|
EP2656344B1 (en) | 2016-08-17 |
CN103384901A (en) | 2013-11-06 |
CN103384901B (en) | 2016-08-10 |
FR2969804A1 (en) | 2012-06-29 |
US20130282387A1 (en) | 2013-10-24 |
US9847085B2 (en) | 2017-12-19 |
WO2012085410A1 (en) | 2012-06-28 |
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