Apparatus and method for processing an audio signal using a harmonic post-filter.
Abstract
An apparatus is provided for processing an audio signal having associated thereto a pitch mismatch information and a gain information, comprising a domain converter (100) for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a subsequent harmonic filter (104) for filtering the second domain representation of the audio signal, wherein the subsequent filter is based on a transfer function comprising a numerator and a denominator, wherein the numerator comprises a Gain value indicated by the gain information,

Term
8.8 yearsleft in the term
Expires 24 July 2035.
- Priority
- Filed
- Granted
- Today
- Expires
9 claims: 4 independent, 5 dependent
- 1RS1V1NDICACIONES IMPÍ£í DE I.A FKíjEííEDaS mmABTAEAL 1.8 Vi MX/a/2017/001242 1. Un aparato para procesar una señal de audio que tiene asociada con el mismo una información de desfase de 5 tono y una información de ganancia, que comprende:un conversón de dominio (100) para convertir una primera representación de dominio de la señal de audio en una segunda representación de dominio de la señal de audio;y un filtro posterior de armónicos (104) para filtrar la 10 segunda representación de dominio de la señal de audio, en la que el filtro posterior está basado en una función de transferencia que comprende un numerador y un denominador, en la que el numerador comprende un valor de ganancia indicado por la información de ganancia, y en la que el denominador 15 comprende una parte entera de un desfase de tono indicado por la información de desfase de tono y un filtro de pulsaciones múltiples que depende de una parte fraccionaria del desfase de tono.
- 2El aparato de acuerdo con la reivindicación 1, en 20 el que la. función de transferencia, del filtro posterior comprende, en el numerador, un filtro FIR de pulsaciones múltiples adicional para una parte fraccionaría cero del desfa.se de tono. I ΜΡΙ ίΝόΤίΓυΤό MEBCANCl· .....óiliSiJ 1.8 «iiílilíí Vi DE I.A XWÍÍÜfiJj IDDLKTSiAL
- 3El aparato de acuerdo con la reivindicación 1 o 2, en el que el denominador comprende un producto entre el filtro de pulsaciones múltiples y el valor de ganancia.
- 4El aparato de una de las reivindicaciones 5 precedentes, en el que el numerador además comprende un producto de un primer valor escalar y un segundo valor escalar, en el que el denominador comprende el segundo valor escalar y no el primer valor escalar, en el que el primer y segundo valores escalares están predeterminados y tienen 10 valores mayores que 0 y menores que, y en el que el segundo valor escalar es menor que el primer valor escalar.
- 5El aparato de acuerdo con la reivindicación 4, que además comprende:un controlador de filtro (114) configurado para fijar el 15 segundo valor escalar dependiendo de una secuencia de bits, por medio de la que se opera el conversor de frecuenciatiempo (100), en el que el segundo valor escalar se fija a un primer valor, cuando la secuencia de bits tiene un primer valor, en el que el segundo valor escalar se fija a un 20 segundo valor, cuando la secuencia, de bits tiene un segundo valor, en el que el segundo valor de la secuencia de bits es menor que el primer valor de la secuencia de bits, y en el MX/a/2017/001242 que el segundo valor del segundo valor escalar es mayor que I MPI ÍNSÍTÍirTCLlUlKJhÍlA.Ní) ..... CE I.A PRijElíEQftíl mrncofibu 1.8 «iiílilíí Vi el primer valor del segundo valor escalar.
- 6El aparato de acuerdo con la reivindicación 4 o 5, en el que el primer valor escalar se fija entre 0,6 y 1,0 y en el que el segundo valor escalar se fija entre 0,1 y 0,5.
- 7El aparato de una de las reivindicaciones precedentes, en el que el filtro posterior tiene la función de transferencia H(z) en una representación de polo cero sobre la base de la siguiente ecuación:H(z} = 1- apgB(z,Qj 1- pgB(z,T fr )z~ T ™t en la que a es un primer valor escalar, en la que β es un segundo valor escalar, en la que B(z,0) es un filtro de pulsaciones múltiples para una parte fraccionaría cero desfase de tono, en la que B(z,T £r ) es un filtro de pulsaciones múltiples que depende de la parte fraccionaria del desfase de tono, en la que Ti„ t es la parte entera del desfase de tono, en la que T £r es la parte fraccionaria del desfase de tono, y en la que g es el valor de ganancia indicado por la información de ganancia z es una variable en un plano z.
- 8El aparato de una de las reivindicaciones MX/a/2017/001242 precedentes, en el que el filtro de pulsaciones múltiples es IMPIC^l ίΝϊΤΓΤυΓΟ MEXICANO lli·........................·β*|θ DE LA BEOE'EEDAD ΙβΒΪΒΙϊ ................ un filtro de respuesta de impulso finito (FIR, por sus siglas en inglés) y tiene por lo menos tres pulsaciones.
- 9El aparato de una de las reivindicaciones precedentes, 5 en el que el filtro de pulsaciones múltiples en el denominador comprende cuatro pulsaciones, en las que, para una parte fraccionaria cero, la primera pulsación se encuentra entre 0,0 y 0,1, la segunda pulsación se encuentra entre 0,2 y 0,3, la tercera pulsación se encuentra entre 0,5 10 y 0,6, y la cuarta pulsación se encuentra entre 0,2 y 0,3, en el que el filtro de pulsaciones múltiples comprende, para una primera parte fraccionaria, cuatro pulsaciones de filtro, en las que la primera pulsación se encuentra entre 0,0 y 0,1, la segunda pulsación se encuentra entre 0,3 y 0,4, 15 la tercera pulsación se encuentra entre 0,45 y 0,55, y la cuarta pulsación se encuentra entre 0,1 y 0,2, en el que el filtro de pulsaciones múltiples comprende, para una segunda parte fraccionaria, cuatro pulsaciones de filtro, en las que la primera pulsación se encuentra entre 20 0,0 y 0,1, la segunda pulsación se encuentra entre 0,35 y 0,45, la tercera pulsación se encuentra entre 0,35 y 0,45, y la cuarta pulsación se encuentra entre 0,0 y 0,1, MX/a/2017/001242 IMPIC^l ίΝϊΤΓΤίΓΤΟ MEOCANÍ) lli·........................·β|θ DELA FAOPLLDAD tliBlilLs IR.........f'Ul en el que el filtro de pulsaciones múltiples comprende, para una tercera parte fraccionaria, cuatro pulsaciones de filtro, en las que la primera pulsación se encuentra entre 0,1 y 0,2, la segunda pulsación se encuentra entre 0,45 y 5 0,55, la tercera pulsación se encuentra entre 0,3 y 0,4, y la cuarta pulsación se encuentra entre 0,0 y 0,1, en el que la tercera parte fraccionaria es mayor que la segunda parte fraccionaria, y en el que la segunda parte fraccionaria es mayor que la primera parte fraccionaria. 10 10. El aparato de una de las reivindicaciones precedentes, en el que el filtro posterior se encuentra configurado para tener una inclinación espectral negativa para compensar una pérdida de energía por medio del filtro posterior de 15 armónicos, o en el que el filtro posterior se encuentra configurado para suprimir una cantidad de energía entre armónicos en un cuadro, en el que la cantidad de energía suprimida es menor que el 20% de una energía total de la representación tiempo™ 20 dominio en el cuadro. 11, El aparato de una de las reivindicaciones MX/a/2017/001242 precedentes. IMPIC^l ¡ΝϊΤΓΤυΓΟΜίι::ΠΑΝΟ t·)........................·β|θ DE Li» ΓΚΟίΡΙΕΐΙΑΕ) llOalils I··............. índusi b¡aI: .....•wllr’*·»....... en el que el conversor de dominio es un conversor de frecuencia-tiempo, en el que el primer dominio es un dominio de frecuencia y el segundo dominio es un dominio de tiempo, o en el que el conversor de dominio es un convertidor de MX/a/2017/001242 5 tiempo residual LPC, en el que el primer dominio es un dominio residual LPC y el segundo dominio es un dominio de tiempo. 12. Un método para procesar una señal de audio que tiene asociada con el mismo una información de desfase de 10 tono y una información de ganancia, que comprende: la conversión (100) de una representación de frecuencia de la señal de audio en una representación tiempo-dominio de la señal de audio;y la filtración de la representación tiempo-dominio de la 15 señal de audio por medio de un filtro posterior de armónicos (104), en el que el filtro posterior está basado en una función de transferencia que comprende un numerador y un denominador, en la que el numerador comprende un valor de ganancia indicado por la información de ganancia, y en la que 20 el denominador comprende una parte entera de un desfase de tono indicado por la información de desfase de tono y un filtro de pulsaciones múltiples que depende de una parte fraccionaría del desfase de tono. IMPIC^l ¡ΝϊΤΓΤϋΤΟ MEXiCANt) lll·........................·β|θ DELA ΗΟΙΤΗΜΪ) «»· jl» 1 ·............. TNDlsnuÁL: .....ÑSÍIBrt(»....... 13, Un sistema para procesar una señal de audio que comprende un codificador para la codificación de una señal de audio y un decodíficador que comprende un procesador, el procesador comprende: 5 un conversor de dominio (100) para convertir una representación de frecuencia de la señal de audio en una representación tiempo-dominio· de la señal de audio;y un filtro posterior de armónicos (104) para filtrar la representación tiempo-dominio· de la señal de audio, 10 en el que el filtro posterior está basado en una función de transferencia que comprende un numerador y un denominador, en la que el numerador comprende un valor de ganancia indicado por una información de ganancia, y en el que el denominador comprende una parte entera de un desfase de tono 15 indicado por una información de desfase de tono y un filtro de pulsaciones múltiples que depende de una parte fraccionaria del desfase de tono, 14, El sistema de acuerdo· con la reivindicación 13, en el que el codificador comprende un calculador de desfase de 20 tono· (402, 404, 406) para el cálculo de una parte entera y una parte fraccionaria del desfase de tono y un calculador de ganancia (410, 412) para el cálculo del valor de ganancia, y un formador de señal codificada (414) para la generación de MX/a/2017/001242 IMPIC^l ¡ΝϊΤΓΓΙΤΟΜΙΕΪΕΓΑΓίΟ β·:........................·β*|θ DE LA ΓΪΕΙΕΕΕίίΑυ ΙβΗΪΒΕϊ ................. una señal codificada (102) que comprende la información de desfase de tono y la información de ganancia. 15. Un método para procesar una señal de audio que comprende un método de codificación de una señal de audio y 5 un método de decodificación que comprende: la conversión (100) de una representación de frecuencia de la señal de audio en una representación tiempo-dominio de la señal de audio;y la filtración de la representación tiempo-dominio de la 10 señal de audio por el uso de un filtro posterior de armónicos (104), en la que el filtro posterior está basado en una función de transferencia que comprende un numerador y un denominador, en la que el numerador comprende un valor de ganancia indicado por una información de ganancia, y en la 15 que el denominador comprende una parte entera de un desfase de tono indicado por una información de desfase de tono y un filtro de pulsaciones múltiples que depende de una parte fraccionaria del desfase de tono. 16. Un
Independent claims9
309 paragraphs in 12 sections, as filed
PATENT TITLE No. 360555
Owner (s): FRAUNHOFER-GESELLSCHAFT ZUR FORDERUNG DER ANGEWANDTEN
FORSCHUNG EV
Address: Hansastrasse, 27 c, 80686, Munich, GERMANY
Name: APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL BY THE USE OF A HARMONICS AFTER FILTER.
Classification: CIP: G10L19 / 26; G10L19 / 00: G10L19 / 025; G10L19 / 035; G10L21 / 02
CPC: G10L19 / 26; G10L19 / 025; G10L19 / 0212: G10L21 / 02
Inventors): EMMANUEL RAVELLI; CHRISTIAN HELMRICH; GORAN MARKOVIC; MATTHIAS
NEUSINGER; SASCHA DISCH; MANUEL JANDER: MARTIN DIETZ
REQUEST
Number: International Presentation Date:
MX / a / 2017/001242 July 24, 2015
PRIORITY
Country: Date: Number:
EP July 28, 2014 14178820.8
Validity: Twenty years Expiration Date: July 24, 2035 Issue Date: November 7, 2018
The reference patent is granted based on articles 1<sup>to</sup>. 2nd section V, 6th section III, and 59 of the Industrial Property Law.
In accordance with article 23 of the Industrial Property Law, this patent has a validity of twenty years, non-extendable, counted from the filing date of the international application and will be subject to the payment of the fee to keep the rights in force. .
Whoever signs this title does so based on the provisions of articles 6 ° Factions lli and 7 "bis 2 of the Industrial Property Law (Official Gazette of the Federation (DOF) 06/27/1991. Amended ei 02 / 08/1994. 10/25/1996, 12/26/1997. 05/17/1999. 01/26/2004. 16/00/2005, 01/25/2: 006, 05/06/2008, 06 / 01/2010, 06/18/2010, 06/18/2010. 01/27/20 * 2. 04/09/2012, 06/01/2016 and 03/13/2018); articules 1 °. 3rd section V subsection a), 4th and 12th sections I and III of the Regulations of the Mexican Institute of Industrial Property (DOF 12/14/1999, amended on 07/01/2002, 07/15/2004, 28 / 07/2004 and 07/09/2007 !; articles 1, 3, 4 ', 5 fraction V subsection a). 16 sections I and ttl and 30 of the Organic Statute of; Mexcano Institute of Industrial Property (DOF 12/27/1999. Amended on 10/10/2002, 07/29/2004, 08/04/2004 and 09/13/2007); 1 “, 3 and 5 subsection a) of the Agreement that delegates powers to the Directors. Deputy Generals, Coordinator. Divisional Directors, Heads of Regional Offices, Divisional Deputy Directors, Departmental Coordinators and other subordinates of the Mexican Institute of Industrial Property. (DOF, 12/15/1999, amended on 02/04/2000, 07/29/2004, 08/04/2004 and 09/13/2007).
This document is signed with an advanced electronic signature (FIEL), based on articles 7 BIS 2 of the Industrial Property Law; 3 of its Regulations, and 1 section III, 2 section V, 26 BIS and 26 TER of the Agreement by! that the Hneamierrtos are established for the use of the Electronic Payment and Services Portal (PASE) of the Mexican Institute of Industrial Property, in the procedures indicated,
DIVISIONAL PATENT DIRECTOR NAHANNY CANAL REYES
<img file="MX360555B_D0001.tif" />
[) Original string:
NAHANNY MARISOL CANAL REYES | 00001000000403252793 | Tax Administration Service | 1695 |! | MX / 2019/227 | MX / a / 2017/001242 | PCT patent title | 1223 | GAGV | Pág (s) | zLmTkOJkAtXVZt4e513nM7 Y + O + O513nM7 Y + O + O = M7
Digital stamp:
ZFORkX3Q¡RBEERYIFxk1Z5K6qnsvsyaj1GtQOC + QzwhVsy9bPpc8aDS9kiYy2QcSwOOecOCVuEIEjlvlSZH7UyíFGo qzN0s / OoMwqOpVV4rtdnRHVfnYyShqa £ + RcBHW6zTO / IQJ18ueEvXBGVy88Wh¡Y5vx97Os7OS12acmsMZd4ll2gaj
ZcU6hguRcHwfOP8 + uLO10YumQjBwuxZb! KuZN6ODCbtrr + INORRnORI8 / mj; bmQaxBIIWvSLRXzBdttXdljiKBzpcNx3 fOR2MuYbjiHNwAOoc + 2Wxlx7MHz = Y5VLXM7MT / Y5VLXM7MHz
Arenal No. SSO, Floor 1, Pueblo Santa María Pepepán. Xochimllco, Mexico City.
(55) 53340700 www.gob.mxOmpi
<img file="MX360555B_D0002.tif" />
MX / 2019/227 ta <1 g «iiílilíí
IMPI ^ d iNSTn'tnró MixjiCANct flj · ........ Β (· β * | (
OF IA PRijElíEDdíi ___
APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL BY USE
FROM A HARMONICS AFTER FILTER
MX / a / 2017/001242
Descriptive memory
The present invention is related to audio processing and, in particular, to audio processing by the use of a harmonic post filter.
Transform-based audio codecs generally introduce interharmonic noise when processing harmonic audio signals, particularly at low bit streams.
This effect is further worsened when the transform-based audio codee operates at low delay, due to the poorer frequency resolution and / or selectivity introduced by a shorter transform size and / or poorer window frequency response.
This interharmonic noise is generally perceived as a very annoying artifact, significantly reducing the performance of the transform-based audio codec when subjectively evaluated on highly tonal audio material.
There are several solutions to improve the subjective quality of audio elbows based on transformation into harmonic audio signals ta <| g «« ¿lili
IMPÍ £ í>
ίΝϊΤΓΤυΓΟ MEXICAN ffjjj ........ Βΐ · β * ||
OF THE riiDEOÍMD
All of them are based on prediction-based techniques, either in the transformation domain or in the time domain.
Examples of transform domain approaches are:
• [1] H. Fuchs, Improving MPEG Audio Coding by Backward
Adaptive Linear Stereo Prediction, 99<sup>na</sup> AES Convention,
New York 1995, Preliminary Edition 4086.
• [2] L. Yin, M. Suonio, M. Váánánen, A New Backward
Predictor for MPEG Audio Coding, 103<sup>ra</sup> AES Convention,
New York 1997, Preliminary Edition 4521 • [3] Juba Ojanperá, Mauri Váánánen, Lín Yin, Long Term
Predictor for Transform Domain Perceptual Audio Coding,
107 “AES Convention, New York 1999, Preliminary Edition
5036.
Examples of time domain approaches are:
• [4] Philip J. Wilson, Harprit Chhatwal, Adaptive transform coder having long term predictor, US Patent 5,012,517, April 30, 1991.
• [5] Jeongook Song, Chang-Heon Lee, Hyen-0 Oh, Hong-Goo
Kang, Harmonio Enhancement in Low Bitrate Audio Coding
MX / a / 2017/001242 üsing and Efficient Long-Term Predictor, EURASIP
IMPIC ^ l ¡ΝϊΤΓΤυΓΟΜΕΪίΠΑΝΟ ll · ........................ · β * | θ
DELA ΓΪΟΙΡΙϊΙίίΐυ tliBlilE: GO ..............
ENDlSTiUAIL: ............
Journal on Advances in Signal Processing 2010.
• [6] Juin-Hwey Chen, Pitch-based pre-filtering and postMX / a / 2017/001242 filtering for compression of audio signs, US Patent 8,738,385, May 27, 2014.
An object of the present invention is to provide a
<td>concept</td><td>improved</td><td>for him</td><td>prosecution</td><td>of</td><td>a sign of</td>
<td>Audio.</td><td></td><td></td><td></td><td></td><td></td>
<td>East</td><td>target</td><td>is achieved</td><td>through</td><td>a</td><td>apparatus for</td>
<td>process</td><td>a signal</td><td>audio</td><td>in accordance with</td><td>the</td><td>vindication</td>
i, a method for processing an audio signal according to claim 12, a system according to claim 13, a method for operating a system according to claim 17 or a computer program, according to claim 18.
The present invention is based on the finding that the subjective quality of an audio signal can be substantially improved by a harmonic afterfilter having a transfer function comprising a numerator and a denominator. The numerator of the transfer function comprises a gain value indicated by a transmitted gain information and the denominator comprises an integer part of a pitch offset indicated by a pitch offset information and a multi-beat filter that
IMPIC ^ l ίΝϊΤΓΤϋΤΟ ΜΕΧΙΓΑΝΟ β ·: ........................ · β | θ
DELA FEDEIEIME) ΙβΗΪΙΙ ».................
INDOSi B¡AI: ............
it depends on a fractional part of the pitch offset.
Therefore, it is possible to eliminate the interharmonic noise introduced by a typical domain change audio decoder as an artifact. This harmonic post filter is particularly useful in that it is based on the transmitted information, i.e. the pitch gain and pitch offset that are available anyway in a decoder, as this information is received from a corresponding encoder. via a decoder input signal. Furthermore, the post filtering is of specific precision due to the fact that not only is the entire part of the pitch offset represented, but also the fractional part of the pitch offset is represented. The fractional part of the pitch shift can in particular be fed into the post filter via a multi-beat filter which has filter coefficients that actually depend on the fractional part of the pitch shift. This filter can be implemented as a FIR filter or it can also be implemented as any other filter such as an IIR filter or a different filter implementation. Any domain change such as a time for frequency change or an LPC for time change or a time for LPC change or a frequency for change of
MX / a / 2017/001242
MPI
ΙΝΑΤΕΤΑΤΟ MEXICAN FROM HAWLEIÍAD ilNDUSTiBIAI:
Time can be advantageously improved by means of the postfilter concept of the invention. Preferably, however, the domain change is a frequency change for time domain.
Thus, embodiments of the present invention reduce interarnion noise introduced by a transform audio code based on a long-term predictor working in the time domain. Contrary to [04] a
[6], where both a pre-filter before transform encoding and a post-filter after transform decoding are used, the present invention preferably applies a post-filter only.
In addition, it has been observed that the prefilter used in
[04] a. [6 ·] has the tendency to introduce instabilities in the input signal given to the transform encoder.
These instabilities are due to changes in gain and / or pitch shift from frame to frame. The transform encoder has difficulties in encoding such jitters, particularly at low bit streams, and one will sometimes introduce even more noise into the decoded signal compared to a situation without any
-j— -η -j— j— i pj -j— 1 -j ^ "Preferably, the present invention does not employ any
<img file="MX360555B_D0003.tif" />
MX / a / 2017/001242 ¡NJTrTirrO MEXICANO β ·: ........................ · βϊ | θ
DELA ΓΗΙΙίΡΙΕΙΜΕ) .......................
IN DES I ¡UAL ..... ΝΙβί '' '^ .......
pre-filter at all and therefore completely avoids the problems involved with a pre-filter.
Furthermore, the present invention is based on a post filter that is applied on the decoded signal after transform encoding. This post filter is based on a long-term prediction filter that represents the integer part and the fractional part of the pitch offset that reduces the inteharmonic noise introduced by the transform audio codec.
MX / a / 2017/001242
<td> 10</td><td>For</td><td>a</td><td>better</td><td>robustness, the</td><td>pitch offset and</td><td>the</td>
<td></td><td colspan="3">tone gain of</td><td>the parameters</td><td>after filter</td><td>I know</td>
<td></td><td>they estimate in</td><td>the</td><td>side</td><td>encoder</td><td>and are transmitted in</td><td>the</td>
<td></td><td>sequences</td><td>of</td><td>bits.</td><td>However, in</td><td colspan="2">other implementations,</td>
<td></td><td>the lag</td><td>of</td><td>tone j</td><td><sup>r</sup> the gain of</td><td colspan="2">tone can also be</td>
<td> 15</td><td>estimate in</td><td>the</td><td>side</td><td colspan="2">decoder based on</td><td>the</td>
<td></td><td colspan="2">audio signal</td><td colspan="2">decoded obtained</td><td>by a decoder</td><td>of</td>
<td></td><td>audio that</td><td colspan="2">understands</td><td>a conversion of</td><td colspan="2">• frequency-time for</td>
<td></td><td>turn into</td><td>a</td><td colspan="3">signal frequency representation</td><td>of</td>
<td></td><td colspan="2">audio in one</td><td colspan="2">representation time-</td><td>signal domain</td><td>of</td>
<td> 20</td><td>Audio.</td><td></td><td></td><td></td><td></td><td></td>
In a preferred embodiment, the numerator further comprises a multi-beat filter for a fractional zero part of the pitch offset in order to
IMPIC ^ l ¡ΝϊΤΓΤΙΓΤΟ MÍXiCANl) t ·) ........................ · β * | θ
DELA ΓίΟίΡΙΕΙΜίυ ΙβΒΪΒΕ: jl * ......... 1 <Λ
ENDOSTIUAIL: ..... Αϊβί '' '^ .......
compensate for a spectral tilt introduced by the multi-beat filter in the denominator, which depends on the fractional part of the pitch offset.
Preferably, the afterfilter is configured to suppress an amount of energy between harmonics in a frame, where the amount of energy suppressed is less than 20%: of a total energy of the time-domain representation in the frame .
In a further embodiment, the denominator comprises a product between the multitasking filter and the gain value.
In a further embodiment, the filter numerator further comprises a product of a first scalar value and a second scalar value, wherein the denominator comprises only the second scalar value rather than the first scalar value. These scalar values are set to default values and have values greater than 0 and less than
1; and, additionally, the second scalar value is less than the first scalar value. Therefore, it is possible to set in a very efficient way the energy removal characteristics that are typically unwanted and to further set the power of the filter, that is, how strong the filter is.
MX / a / 2017/001242 filter attenuates interharmonic artifacts in a signal from
IMPIC ^ l ¡ΝϊΤΓΤϋΤΟ MEXiCANt) t ·) ........................ i¡Sl & | »
OF THE FEDFEEEMD ΙβΒΪΒΕι jl * ..............
output of the transform domain decoder.
The apparatus further comprises, in a preferred embodiment, a filter controller for setting at least the second scalar value depending on a sequence of bits so that a higher value is set for a smaller sequence of bits and vice versa.
In addition, the filter controller is configured to select, depending on the fractional part of the pitch offset, the corresponding multi-beat filter in a signal-dependent manner in order to fix the signal adaptively to the harmonic afterfilter. , that is, dependent on the value of the fractional part actually provided on the pitch offset.
The preferred embodiments of the present invention are discussed below in the context of the accompanying figures, in which:
Fig, 1 illustrates an embodiment of an inventive apparatus for processing an audio signal;
Fig. 2 illustrates a preferred implementation of the harmonic afterfilter represented as z-domain transfer functions;
MX / a / 2017/001242
IMPÍ £ í>
iNjTrTuroMíi¡c * No flj · ........ ιΐ · β * | (
Fig. 3
Fig. 4
Fig. 5
Fig. 6
Fig. 7a
Fig. 7b
Fig. 7c «· <l8
DELA FBI'ItíiÁU IMbIBL. Jl · .............
illustrates a further preferred embodiment for the harmonic afterfilter represented by a transfer function in the z domain;
illustrates a preferred implementation of an encoder for generating an encoded signal to be decoded by means of a transform domain audio decoder illustrated in Fig. 1;
illustrates a preferred implementation of the multitasking filter as a FIR filter controlled by a filter controller;
illustrates a cooperation between the filter controller and a memory having previously stored beat weights depending on the fractional part;
illustrates a frequency response of a filter · having a value of zero;
illustrates a frequency response of a preferred harmonic backfilter having a value of 1;
illustrates a frequency response of a preferred harmonic afterfilter having a value of 0.8;
MX / a / 2017/001242
Fig. 8a illustrates a later harmonic embodiment that
MX / a / 2017/001242 ta <| 8 «« lili <sup>:</sup><n filter
IMPÍ £ í>
¡NSTrTuromexicani) flj · ........ ιΐ · β * | (
OF IA riiOE'lEÍMD ÚNDUSTBiAIL:
of a preferred has a β value equal to
0.4; Y
Fig. 8b illustrates a frequency response of a harmonic afterfilter having a β value of 0.2.
FIG. 1 illustrates an apparatus for processing an audio signal having pitch offset information and gain information associated therewith. This gain information can be transmitted to a decoder 100 through a decoder input.
102 receiving an encoded signal or, alternatively, this information can be calculated in the decoder itself, when this information is not available.
However, for more robust operation, it is preferred to calculate the pitch offset information and the pitch gain information on the encoder side.
Decoder 100 comprises eg, a frequency-time converter for converting a time-frequency representation of the audio signal into a time-domain representation of the audio signal. Therefore, the decoder is not a pure time-domain speech codec, but comprises a pure transform domain decoder or a mixed transform domain decoder or ¡ΝϊΤΓΤυΓΟ MEXICAN β ·: ........ ................ · β * | θ
DELA HfflFEEÍiAE) iBilt. ...............
EN EilJS I ¡UAL ..... Ν! Βί · '' ^ .......
any other encoder that operates in a domain other than a time domain. Furthermore, it is preferred that the second domain is the time domain.
The apparatus further comprises a harmonic post filter 104 to filter the time-domain representation of the audio signal, and this harmonic post filter is based on a transfer function comprising a numerator and a denominator. In particular, the numerator comprises a gain value indicated by the gain information and the denominator comprises an integer part of a pitch offset indicated by the pitch offset information and, importantly, further comprises a multi-beat filter that it depends on a fractional part of the pitch offset.
A preferred implementation of this harmonic post-filter with a transfer function H (z) is illustrated in Fig. 2. This filter receives the output signal from decoder 106 and subjects this decoded output signal to a post-filtering operation to get a post filtered output signal 108. This post-filtered output signal can be extracted as the processed signal or it can also be processed by any method to eliminate any discontinuity introduced by means of the
MX / a / 2017/001242
IMPIC ^ l
ΙΝϊΤΓΤϋΤΟ MfciiCANt) t ·) ........................ · β | θ
DELA ΓΕΕΙΕΙΕΙΜίυ ΙβΒΪΒΧι jl * .............
iNDlJSI ¡UAIL: ..... Ν · β8 · '' ^ .......
post-filtering operation, which of course is signal dependent, that is, it can vary from frame to frame. This de-discontinuity operation can be any of the well-known de-discontinuity operations such as crossfade, which means that a previous frame is fused and at the same time a new frame gradually appears and preferably the characteristic fading is so that the fading factors ^ add to one for the entire
MX / a / 2017/001242
<td>the cross fade operation.</td><td>Nevertheless,</td><td>I know</td>
<td>can apply in the same way another</td><td>elimination</td><td>of</td>
<td>discontinuity such as a leak</td><td>low pass</td><td>or</td>
<td>LPC filtration.</td><td></td><td></td>
<td>The apparatus for processing a signal</td><td>illustrated audio</td><td>in</td>
the Fig. 1 further comprises a multi-push filter information storage 112 and a filter controller 114. In particular, the filter controller 114 receives the lateral information 116 from the decoder 100, and this lateral information may, for example, be the information of the pitch gain g and the pitch offset information, that is, the information in the integer part T<sub>int</sub> pitch shift and fractional part T<sub>fr</sub> pitch offset. This information is useful for setting the after filter
IMPIC ^ l ¡ΝϊΤΓΤίΓΤΟ MEXTCANt) ll · ........................ · β * | θ
OF THE BDEEEDad ΙβΒΪΒΧι jl * ..............
iNDUSI ¡(¡AI: ..... «βϊί · '' ^ .......
harmonics from frame to frame and, additionally, to select an information from the multipath filter B (z, T<sub>fr</sub>). Furthermore, additional information such as the bit sequence applied by the decoder or the sample rate underlying the decoded signal can also be used by the filter control 114 in order to in particular set the scalar values α, β for a certain encoder and / or decoder configuration with respect to bit sequence and sample rate.
Fig. 2 illustrates a pole / zero representation of a filter transfer function H (z) in the z domain in accordance with what is known in the art. Naturally, there are numerous other harmonic afterfilter representations, which are all filter representations, that can be converted to the z-domain pole / zero representation type. Therefore, the present invention is applicable for each filter, which can be described in any way by means of such a transfer function as illustrated in the specification.
Fig. 3 illustrates a preferred embodiment of the harmonic afterfilter again described as a transfer function in pole / zero notation in the z domain.
The filter can be described as follows:
MX / a / 2017/001242
IMPIC ^ l ¡ΝϊΤΓΤυΓΟΜίΐίΠΑΝΟ lli · ........................ · β | θ
OF THE FEDPÍEEMD ΙβΒΪΒΕι IR ..............
ENDlISTiUAI: ............
Hf) = <sup>1_</sup><sup>W</sup> 1 - β<sub>3</sub>Π (ζ, Τ ^ Ζ-Τΐηε with g the decoded gain, Tixt Y <sup>T</sup>fr the integer and fractional part of the decoded pitch lag, a and β two 5 scalars weighing the gain, and B (z, Ty<sub>r</sub>) a low-pass FIR filter whose coefficients depend on the fractional part of the decoded pitch shift.
It should be noted that B (z, 0) in the numerator of H (z) is used to compensate for the inclination introduced by B (z, T<sub>/ r</sub>).
β is used to control the power of the afterfilter. A β equal to 1 produces full effects, suppressing the maximum amount of energy possible between harmonics. A β equal to 0 disables the post filter. In general, a fairly low value is used so as not to suppress too much energy between harmonics. The value can also depend on the sequence of bits with a higher value to a lower sequence of bits, eg. 0.4 to a lower bit stream and 0.2 to a higher bit stream.
a is used to add a slight slope to the frequency response of H (z), to compensate for the slight loss in energy in the low frequencies. The
MX / a / 2017/001242 ίΝϊΤΓΤϋΤΟ MEiiCANt) ........................ · β | θ
DE LA FEDREEMD ΙβΒΪΒΕι jl * .............
INDIJSIBIAIL: ..... ΥϊΙ | 1 * “ί <<< .......
value of s is usually chosen close to 1, eg. 0.8.
An example of β (ζ, Γ<sub>Λ</sub>) in Fig. 6. The order and coefficients of β (ζ, Τ ^ can also depend on the bit sequence and the output sample rate. A different frequency response can be designed and tuned for each combination of the bit stream and the output sample rate.
In particular, even values for a between 0.6 and less than 1.0 have been found to be useful and that, additionally, values for β between 0.1 and 0.5 have proven useful as well.
Also, the multi-click filter can have a variable number of clicks. It has been found that for certain implementations, four keystrokes are sufficient, where one keystroke is z<sup>+1</sup>. However, smaller filters with only two clicks or even larger filters with more than four clicks are useful for certain implementations.
Fig. 6 illustrates a preferred implementation of filters B (z) for different fractional values of pitch offset and, in particular, for pitch offset resolution of d. For this implementation, they are illustrated
MX / a / 2017/001242
I ΜΡΙ
MKOCANCb INSTRUCTION .....
<1.8 «i! Ilill 'Oí
CE IA XwííÜfiSj iNDUSTíUAL four different filter descriptions for the multi-pulse filter in the denominator of the harmonic afterfilter transfer function. However, it has been found that the filter coefficients do not necessarily have to indicate exactly the values illustrated in Fig. 6, but certain variations of +/- 0.05 may be useful in other implementations as well.
In particular, as illustrated in Fig. 1, the pulse weights illustrated in Fig. 6 are stored within memory 112 for multi-pulse filter information. Filter controller 114 receives fractional part T<sub>fc</sub> from line 116 of Fig. 1 and, in response to this value, is directed to memory 112 in order to retrieve, through a retrieval line
200 the specific filter information for the specific fractional part of the pitch offset. This information is then forwarded through an output line 202 to the harmonic afterfilter 104 so that the harmonic afterfilter is set correctly. A certain implementation of the multi-pulse FIR filter is illustrated in Fig. 5. The weight indication wi aw<sub>4</sub> corresponds to the notation in Fig. 6 and the filter driver 114 applies, in response to the fractional part
MX / a / 2017/001242
IMPIC ^ l ¡ΝϊΤΓΤϋΤΟ MESiCANt) ll · ........................ · β | θ
OF THE FEDFEEEMD ΙβΒΪΒΕι jl * .............
ENDUSTiUAI: ..... Μβϊί · '' ^ .......
current pitch offset the corresponding weights for a certain audio frame. The other portions such as delay portions 501, 502, 503 and combiner 505 can be implemented as illustrated. In this context, it is emphasized that the delay value 501 is, in the z notation a negative delay value, since an FIR filter representation has been found to have a negative delay value in addition to a positive delay value such as 503 and 504 is particularly useful.
A preferred encoder implementation having certain functional blocks and operating without any pre-filter is illustrated below in Fig. 4. The filter portion illustrated in Fig, 4 comprises a pitch estimator
402, a pitch refiner 404, a fractional part estimator 406, a transient detector 408, a gain estimator 410, and a gain quantizer 412.
The information provided by the gain quantizer
412, fractional part estimator 406, pitch refiner 4 04, and the decision bit generated by transient detector 408 are input to a coded signal former 414. The encoded signal former provides a coded signal 102, which then is input to decoder 100 illustrated in Fig. 1. The encoded signal
MX / a / 2017/001242
IMPIC ^ l ¡ΝϊΤΓΤϋΤΟ MEXiCANt) t ·) ........................ · β | θ
DELA FKDP3EEMD t | lR · !!!! » ...............
102 it will comprise additional signal information not illustrated in Fig. 4.
Subsequently, the functionality of the tone estimator 402 is described,
A pitch shift (whole part + fractional part) is estimated per frame (frame size eg 2Oras).
This is done in 3 steps to reduce complexity and improve estimation precision.
A pitch analysis algorithm is used that produces a smooth pitch evolution contour (eg open loop pitch analysis described in ITU-T Rec. G.718, section
6.6). This analysis is generally carried out on a subframe basis (subframe size eg 10ms), and produces an estimate of pitch lag per subframe. It should be noted that these pitch offset estimates have no fractional parts and are generally estimated on a subsampled signal (sample rate eg 6400Hz). The signal used can be any audio signal, eg.
an LPC weighted audio signal as described in ITU-T Rec. G.718, section 6.5.
The tone refiner operates as follows:
The final integer part of the pitch offset is estimated on an audio signal x [n] running on the
MX / a / 2017/001242
IMPIC ^ l ίΝϊΤΓΤυΓΟ MEXICAN t ·) ........................
DELA iBDETEDAD aBiBvISSLss ........... Β'βΐ ilNDUSTiBIAI: ..... ÑEillB · ^ .......
core encoder sample rate, which is generally higher than the sample rate of the subsampled signal used in a. (eg 12.8kHz, 16kHz, 32 kHz ...). The signal x [n] can be any audio signal through e j. an LPC weighted audio signal.
The integer part of the pitch offset is therefore the offset d<sub>m</sub> which maximizes the autocorrelation function
N
C (d) = x [n] x [n - d] n = o θ with d around a pitch shift T estimated in step 1.a.
T-ff<sub>1</sub><d <T + d<sub>2</sub>
The fractional part estimator 406 operates as follows:
The fractional part is found by interpolating the autocorrelation function C (d) computed in step 2, b, and selecting the fractional pitch lag that maximizes the interpolated autocorrelation function. Interpolation can be carried out by using a low-pass FIR filter as described in eg. Rec. ITU-T G.718, section 6.6.7.
The transient detector 408 illustrated in FIG. 4 is
MX / a / 2017/001242
IMPIC ^ l ¡ΝϊΤΓΤϋΤΟ MEXiCANt) t ·) ........................ · β * | θ
DELA ΗΟΙΪΗΜΪ) tliBlilT. GO ......... f'Ol
TNDÜSTUAT ............
It is configured to generate a decision bit.
If the input audio signal does not contain any harmonic content, then no parameter is encoded in the bit stream. Only 1 bit is sent so that the decoder knows if it has to decode the postfilter parameters or not. The decision is made on the basis of various parameters:
MX / a / 2017/001242
to. Normalized correlation on estimated integer pitch lag in step lb
Ση = ο4η] χ [η-4η] corr. norm. = λ / Σ = 0 * Μ * Μ λ / Σ £ = οx [n - d<sub>m</sub>] x [n - d<sub>m</sub>]
The normalized correlation is 1 if the input signal is perfectly predictable by the entire pitch offset, and if it is not predictable at all. A high value (close to 1) will then indicate a harmonic signal. For a more robust decision, the normalized correlation of the past frame can also be used in the decision, e.g .:
2Q If (normal current (current) * normal current (previous))> 0.25, then the current frame contains some harmonic content (bit = l) ¡ΝϊΤΓΤυΓΟ MEXICAN β · ......... ................ · β * | θ
DELA riIDiPIEÍMD 1 «··! .. ...............
IN TWO ΤΒΙΛΙ: .....<sup>?</sup>W | Írt (».......
b. Characteristics computed by a transient detector (eg Temporal Fullness Measurement, Maximum Energy Change), to avoid post-filter activation on a signal containing a transient, eg.
If (Flatnesstemp> 3.5 or Max Energy change> 3.5) then set bit = 0 and do not send any parameter
In addition, the gain estimator 410 calculates a gain to be input to the gain quantizer 412.
The gain is generally estimated on the input audio signal at the center sample rate encoder, but can also be any audio signal such as the LPC weighted audio signal. This signal is denoted y [n] and can be the same or different from x [n].
The prediction yp [n] of y [n] is danced first by filtering y [n] with the following filter
P (z) = B (z, 7><sub>r</sub>) z-<sup>T</sup>"With T<sub>int</sub> the integer part of the pitch offset (estimated in lb) and B (z, 7}<sub>r</sub>) a low-pass FIR filter whose coefficients depend on the fractional part of the pitch shift Tj<sub>r</sub> (estimated at 1.c.).
An example of B (z) when the offset resolution of
MX / a / 2017/001242 tone is M:
OR
T<sub>fr</sub> = - B (z) = O.OOOOz<sup>2</sup> + 0.2325z<sup>_1</sup> + 0.5349z ° + 0.2325Z<sup>1</sup> l> r = ξ B (z) = 0 »0152z<sup>2</sup> + 0.3400Z-<sup>1</sup> + 0.5094z ° + 0.13532<sup>1 </sup>2
T<sub>fr</sub> = - B (z) = 0.0609z<sup>-2</sup> + 0.4391z<sup>_1</sup> + 0.4391z ° + 0.0609Z<sup>1 </sup>' 4
Tf<sub>r</sub> = - B (z) = 0, Í353z<sup>-2</sup> + 0.5094z<sup>-1</sup> + 0.3400z ° + 0.01522<sup>1</sup>
The gain g is then computed as follows:
MX / a / 2017/001242 <sub>=</sub> Zw = oyWypW <sup>9</sup> and is limited between 0 and 1.
Finally, the gain is quantified eg. over 2 bits, by using eg. uniform quantification.
If the gain is quantized to 0, then no parameter is encoded in the sequence of bits, only the only one<sup>:</sup> decision bit (bit — O).
As noted above, the post filter is applied on the output audio signal after the transform decoder. It processes the signal on a frame-by-frame basis, with the same frame size as used on the encoder side such as 20ms. Of
0 According to what is illustrated, it is based on a long-term prediction filter H (z) whose parameters are determined from the estimated parameters on the side of the
IMPIC ^ l iNjTrTuroMiEi¡c * No t ·) ........................ isSEÍjoi
DELA ΓΪΕΙΕΕΕίίΑυ ElHlii!.: ........... 1 <Λ encoder and are decoded from the bit stream. This information comprises the decision bit, the pitch offset, and the gain. If the decision bit is 0, then the pitch offset and gain are not decoded and assumed to be 0 are not written to the bit stream at all.
According to what has been discussed, if the filter parameters are different from one frame to the next frame, a discontinuity can be introduced in the boundary between the two frames. To avoid discontinuity, a discontinuity eliminator such as a cross-fader or any other implementation for that purpose is applied.
Furthermore, different ways of attaching the harmonic afterfilter are illustrated in Fig. 7a to 8b. The graphs illustrate the frequency domain transfer function. The horizontal axis is related to the normalized frequency 1 and the vertical axis is the magnitude of the filter response in dB. It is emphasized that in all the illustrations except Fig. 7b, the filter introduces an amplification for low frequencies, that is, a certain positive dB magnitude value.
In particular, Fig, 7a illustrates a transfer function, which implements the filter in Fig. 3, with the certain values of the parameters as indicated
MX / a / 2017/001242
ΜΡΙ! NjTrTuroMfciic * No DE LEE ΗΕΟΕ'ΕΕΒΜ) previously. Furthermore, the value a, that is, the first scalar value is set to 0. Fig. 7b illustrates a similar situation, but now with a value equal to 1. The other parameters are identical to Fig. 7a.
Fig. 7c illustrates a further implementation where ex equals 0.8 which has a slight tilt and a boost of the lower frequencies. Again, Fig.
has the same other parameters as indicated<sup>:</sup> in Fig. 7a * It becomes clear<sup>!</sup> that ex equal to 1 eliminates skew and all harmonic frequencies have a gain of 1. The drawback of this configuration is a loss of energy in the frequencies between the harmonics.
Therefore, a value of a equal to 0.8 is preferred as in Fig * 7o. This value adds a slight tilt: compared to the situation of a equals 1 in Fig. 7b.
In order to compensate for the loss of energies in the frequencies between the harmonics, this slight inclination is preferably used »
<img file="MX360555B_D0004.tif" />
MX / a / 2017/001242
<td>Besides, the</td><td>Fig,</td><td>8a and</td><td>8b illustrate</td><td>configurations</td><td>of</td>
<td>20 filters for a</td><td>value</td><td>from to</td><td>equal to 0.8 and</td><td colspan="2">different values</td>
<td>β, that is, a</td><td>value</td><td>β of (</td><td>3,4 in Fig.</td><td>8a and a β value</td><td>of</td>
0.2 in Fig. 8b. It becomes clear that a β value of 0.4 has a stronger post-filtration effect compared to
I ΜΡΙ ίΝΊΤίΓυϊΓό MIÜOCANCb .....
<1.8 Vi
CE LA fmtufiSj INDUSTRIAL with a β value of 0.2 and therefore a β value of 0.4 is used in lower bit sequences in order to eliminate the interharmonic noise introduced by such low bit sequences.
On the other hand, β equal to. 0.2 has a less strong effect in suppressing energy between harmonics and therefore this β value is preferred for higher bit sequences due to the fact that in such higher bit sequences, there is not as much interharmonic noise.
Although some aspects have been described in the context of an apparatus, it is clear that these aspects also represent a description of the corresponding method, in which a block or device corresponds to a method step or a characteristic of a method step. Similarly, aspects described in the context of a method step also represent a description of a corresponding block or element or characteristic of a corresponding apparatus. Some or all of the steps in the method can be executed by means of (or by the use of) a hardware apparatus, such as a microprocessor, a programmable computer, or an electronic circuit. In some embodiments, some one or more of the most important steps
MX / a / 2017/001242 of the method can be executed by such apparatus.
IMPIC ^ l ¡ΝϊΤΓΤϋΤΟ MEOCANÍ), 11 · ........................ · β * | θ
DELA HiOE'EEDM) aBiBRISÍSIss .................
iNDlJSTBEAL ..... 'ÁSIIIggf' i ·· .......
The inventive transmitted or encoded signal may be stored on a digital storage medium or it may be transmitted on a transmission medium such as a wireless transmission medium or a wired transmission medium such as the Internet.
Depending on certain implementation requirements, embodiments of the invention can be implemented in hardware or software. The application can be carried out by the use of a digital storage medium, for example a floppy disk, a DVD, a Blu-Ray, a CD, a ROM, a PROM, and
EPROM, an EEPROM or a FLASH memory, having electronically readable control signals stored therein, which cooperate (or are capable of cooperating) with a computer system programmed in such a way that the respective method is carried out. Therefore, the digital storage medium can be computer readable.
Some embodiments according to the invention comprise a data carrier having electronically readable control signals, which are capable of cooperating with a programmable computer system, such that one of the methods described herein is carried out.
Generally, the realizations of the present
MX / a / 2017/001242
2?
IMPIC ^ l ¡ΝϊΤΓΤϋΤΟ MIEOCANÍ) ll · ........................ · β * | θ
DELA PttCíiiFAECDUD ElOliils ...............
IN [(t, KTB¡AL ............
invention can be implemented as a computer program product with a program code, the program code is operative to carry out one of the methods when the computer program product is run on a computer. The program code can be stored, for example, on a machine-readable carrier.
Other embodiments comprise the computer program to carry out one of the methods described herein, stored on a machine-readable carrier.
In other words, an embodiment of the inventive method is therefore a computer program that has program code to carry out one of the methods described herein, when the computer program is run on a computer.
A further embodiment of the inventive method is therefore a data carrier (or a non-transitory storage medium such as a digital storage medium, or a computer-readable medium) comprising, recorded thereon, the computer program to carry out one of the methods described herein. The data carrier, digital storage medium, or recorded medium are typically tangible and / or non-transitory.
MX / a / 2017/001242 ¡ΝϊΤΓΤίΠΓΟ MfciiCANt) ll · ........................ · β * | θ
DELA ΓΪΕΙΡΙίίΜΐ) 1 «··!« I ·· .............
A further embodiment of the method of the invention is therefore a stream of data or a sequence of signals representing the computer program to carry out one of the methods described herein. The data stream or signal sequence may, for example, be configured to be transferred over a data communication connection, for example, over the internet.
A further embodiment comprises a processing means, eg, a computer or a programmable logic device, configured for, or adapted to, carry out one of the methods described herein.
A further embodiment comprises a computer that has the computer program installed therein to carry out one of the methods described herein.
A further embodiment according to the invention comprises an apparatus or a system configured to transfer (eg, in electronic or optical form) a computer program to carry out one of the methods described herein to a receiver. The receiver can, for example, be a computer, a mobile device, a memory device, or the like. The apparatus or system may, for example, comprise a file server for
MX / a / 2017/001242
IMPIC ^ l ¡NJTrTiLLOHfciECAEit) β ·: ........................ sigCto
DELA HOPUEDAD tliBlilL: Ι »..............
ΜΕΧΑΪΪΕΛΙ: ..... 'Wli'T, ».......
transfer the software to the recipient.
In some embodiments, a programmable logic device (eg, a field programmable gate array) can be used to perform some or all of the functionality of the methods described herein. In some embodiments, a field-programmable gate array may cooperate with a microprocessor in order to carry out one of the methods described herein. In general, the methods are preferably carried out by means of any hardware apparatus.
The above-described embodiments are merely illustrative for the principles of the present invention. It is understood that modifications and variations to the arrangements and details described herein will be apparent to others of skill in the art.
It is the intent, therefore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of description and explanation of the embodiments herein.
MX / a / 2017/001242 memory.
Contents12
12 sheets
Sheet 1 Sheet 2 Sheet 3 Sheet 4 Sheet 5 Sheet 6 Sheet 7 Sheet 8 Sheet 9 Sheet 10 Sheet 11 Sheet 12
44 members in 18 offices
Priority claims9
| Document | Office | Kind | Date |
|---|---|---|---|
| 14178820 | European Patent Office (EPO) | A | |
| 14178820 | European Patent Office (EPO) | A | |
| 141788208 | European Patent Office (EPO) | – | |
| 2015066998 | European Patent Office (EPO) | W | |
| 2015066998 | European Patent Office (EPO) | W | |
| 141788208 | – | – | – |
| EP20140178820 | – | – | – |
| PCTEP2015066998 | – | – | – |
| WO2015EP66998 | – | – | – |
Members44
| Document | Office | Kind | |
|---|---|---|---|
| EP2980799A1 | European Patent Office (EPO) | A1 | |
| CA2955255A1 | Canada | A1 | |
| WO2016016121A1 | World Intellectual Property Organization (WIPO) | A1 | |
| TW201618086A | Taiwan Province of China | A | |
| AR101340A1 | Argentina | A1 | |
| SG11201700696UA | Singapore | A | |
| AU2015295603A1 | Australia | A1 | |
| KR20170035987A | Republic of Korea | A | |
| CN106663444A | China | A | |
| US2017140769A1 | United States of America | A1 | |
| EP3175454A1 | European Patent Office (EPO) | A1 | |
| TWI590238B | Taiwan Province of China | B | |
| MX2017001242A | Mexico | A | |
| JP2017522604A | Japan | A | |
| BR112017001631A2 | Brazil | A2 | |
| AU2015295603B2 | Australia | B2 | |
| EP3175454B1 | European Patent Office (EPO) | B1 | |
| ES2676584T3 | Spain | T3 | |
| RU2665259C1 | Russian Federation | C1 | |
| PT3175454T | Portugal | T | |
| MX360555BThis record | Mexico | B | |
| PL3175454T3 | Poland | T3 | |
| KR101959211B1 | Republic of Korea | B1 | |
| US10242688B2 | United States of America | B2 | |
| CA2955255C | Canada | C | |
| US2019198034A1 | United States of America | A1 | |
| JP6546264B2 | Japan | B2 | |
| JP2019194716A | Japan | A | |
| MY179023A | Malaysia | A | |
| CN106663444B | China | B | |
| CN112420061A | China | A | |
| JP2021064009A | Japan | A | |
| US2021125624A1 | United States of America | A1 | |
| JP6877488B2 | Japan | B2 | |
| US11037580B2 | United States of America | B2 | |
| JP2023072014A | Japan | A | |
| US11694704B2 | United States of America | B2 | |
| JP7340553B2 | Japan | B2 | |
| US2023282223A1 | United States of America | A1 | |
| CN112420061B | China | B | |
| US12190897B2 | United States of America | B2 | |
| US2025118316A1 | United States of America | A1 | |
| JP7733688B2 | Japan | B2 | |
| JP2025131828A | Japan | A |
1 legal event, as the office reported them to INPADOC
Events
| Event | Code | |
|---|---|---|
| Grant or registrationFG | FG |
Numbers
- Publication
- 360555
- Publication, DOCDB
- 360555
- Publication, EPODOC
- MX360555
- Application
- 2017001242
- Application, DOCDB
- 2017001242
- Application, EPODOC
- MX20170001242
Titles2
- Spanish
- APARATO Y METODO PARA PROCESAR UNA SENAL DE AUDIO POR EL USO DE UN FILTRO POSTERIOR DE ARMONICOS
- English
- APPARATUS AND METHOD FOR PROCESSING AN AUDIO SIGNAL FOR THE USE OF A SUBSEQUENT FILTER OF HARMONICS
Classification
- CPC, 5
- G10L19/26
- G10L19/025
- G10L21/02
- G01L21/02
- G10L19/0212
- IPC, 5
- G10L19 26
- G10L19 00
- G10L19 025
- G10L19 035
- G10L21 02