EP2113913B1 - Method and system for reconstituting low frequencies in an audio signal - Google Patents
Method and system for reconstituting low frequencies in an audio signal Download PDFInfo
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- EP2113913B1 EP2113913B1 EP09290310A EP09290310A EP2113913B1 EP 2113913 B1 EP2113913 B1 EP 2113913B1 EP 09290310 A EP09290310 A EP 09290310A EP 09290310 A EP09290310 A EP 09290310A EP 2113913 B1 EP2113913 B1 EP 2113913B1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
- H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
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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
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L2021/02082—Noise filtering the noise being echo, reverberation of the speech
Definitions
- the invention relates to a method and system for reconstructing low frequencies of an audio signal that can be used at the output of a sound reproducing device having a cut-off frequency for low frequencies.
- the invention finds a particularly advantageous application in the field of electro-acoustic equipment, including stereo speakers for the reproduction of musical works or even the speakers of personal computers (PC) for reproducing the soundtrack of video files.
- PC personal computers
- any acoustic speaker has a cutoff frequency for low frequencies below which it is no longer able to radiate energy.
- This cutoff frequency is directly related to the dimensions of the speaker, and more precisely to the size of the membrane. The smaller the speaker, the higher the cutoff frequency in the spectrum.
- a small enclosure will impose a natural attenuation to the low frequency content of a piece of music, and this to the detriment of the listener who can not benefit from this information and will therefore feel an unpleasant effect related to the loss serious sounds.
- a first solution to this difficulty consists in applying a filter to amplify the low frequencies attenuated by the acoustic enclosure, by mechanically forcing the speaker diaphragm to radiate these low frequencies.
- this solution presents real risks for the integrity of the loudspeaker. Indeed, the excursion of the membrane, that is to say the amplitude of its displacement relative to its equilibrium position, would become too important, to the point of damaging or even breaking it.
- Another solution is based on a psychoacoustic property of the human ear that makes it possible to perceive low frequencies even if they are not actually transmitted by a device belonging to the sound reproduction system, a loudspeaker speaker. for example.
- the US 5,930,373 A1 discloses such a method of generating harmonics relating to the low frequencies of the audio signal by means of a modulation system.
- the reference signal is multiplied by itself to obtain a double frequency signal, then again multiplied by itself to obtain a triple frequency signal, and so on.
- This known system has the advantage of being fast, because without significant delay, and not requiring frequency information.
- it has the disadvantage of being non-linear. Indeed, if the original audio signal contains a sum of frequencies, will be generated not only the harmonics of each of these frequencies but also harmonics intermodulation terms that may severely degrade the audio performance of the system.
- the presence of the low-pass filter introduces a variable phase shift which negatively interferes with the signal obtained at the output because the harmonic signal will not be reinjected in phase in the original audio signal. This produces unequal harmonic levels depending on the frequencies, as they are potentially lower for frequencies that are not in phase with those of the original signal.
- the US 2003/223588 A1 proposes in this regard a bass enhancement device where the envelope of the synthetic signal is adjusted by a compression / expansion system in which the slope and an offset are adjustable. The slope and the offset are adjusted simultaneously so that the average energy of the envelope is compensated, this simultaneous control being adjustable by a potentiometer or other means of manual adjustment.
- This system has the disadvantage of not being suitable for all types of input signals, especially if the aim is to obtain the most natural rendering possible tones, not to produce acoustic effects by generating components frequencies not contained in the original signal, as in the case of the US 2003/223588 A1 which essentially seeks to artificially expand the stereo field, increase the "brilliance" of the sound or introduce a distortion reminding the particular sound of tube amplifiers.
- an object of the invention is to propose a method of reconstituting low frequencies of an audio signal at the output of a sound reproduction device which respects the temporal variations of the original signal so as to preserve the nuances, and which also takes into account variations in human auditory perception with frequency.
- the invention provides for a dynamic adaptation of said time envelope as a function of the frequency band considered.
- the dynamic adaptation of the temporal envelope according to the frequency band makes it possible, in particular, to take into account the variations of the human auditory perception with the frequency, and the detection of the temporal envelope and its taking into account by multiplication with the signal generated harmonic allows to modulate the signal synthesized according to the temporal variations of the envelope.
- the adaptation step of the temporal envelope is performed by compression / expansion of the temporal envelope.
- the invention proposes to dynamically automate the adjustment of the offset of the envelope by a feedback loop on the value of the envelope (advantageously with different time constants on ascent and descent).
- the offset will automatically adjust, based on the average energy of the input signal, to a value that maximizes this energy within a defined limit.
- control of the compression / expansion step is performed conditionally after comparing the level of the compressed / expanded signal with respect to a predetermined threshold.
- the invention also relates, according to claim 10, to a module for reconstituting low frequencies of an audio signal for implementing the aforementioned method.
- FIG. 1 On the figure 1 is shown an architecture of a system 10 for reconstituting low frequencies in an audio signal, a stereo signal for example, said low frequencies to be reconstructed at the output of a sound reproducing device constituted by two loudspeakers 11, 12 , associated with each stereo output signal L out and R out , said speakers having a cutoff frequency F 0 low frequency of 120 Hz for example.
- the system of reconstitution of the figure 1 comprises a reconstitution module 100, also referred to as a module for generating "virtual basses", operating according to the "pitch" rendering principle explained above which consists, in essence, in processing an input signal S in resulting from the average of the input stereo signals L in and R in order to generate an output harmonic signal S out associated with at least one fundamental frequency lower than the cutoff frequency F 0 that is to be reconstructed at the output of the loudspeakers 11, 12 by rendering effect of "pitch".
- the harmonic output signal S out thus generated is fed back into phase at the output of the virtual bass generation module 100 in the original stereo signals L in and R in to form the stereo output signals L out and R out .
- said output harmonic signal S out by summation of three sinusoidal components of frequencies respectively equal to the first three harmonics of the low frequency signal to be reconstructed, namely the fundamental frequency, or first harmonic, and the next two higher harmonics, that is to say the double and triple harmonics of the fundamental frequency.
- the fundamental frequency or first harmonic
- the next two higher harmonics that is to say the double and triple harmonics of the fundamental frequency.
- other choices are possible, such as, for example, the use of the first four harmonics, the essential in all cases being that the generated harmonic signal contains at least two consecutive harmonics in order to perceive their difference, which is equal to the pitch.
- the cut-off frequency F 0 is 120 Hz
- the range of low frequencies that can benefit from a "pitch" reconstitution extends between 60 and 120 Hz.
- the harmonics to be considered are those at 60, 120, 180 Hz.
- the bandwidth of the system 100 is thus "virtually" extended downwards to a new cut-off frequency F ' 0 equal to 60 Hz, as shown in figure 2 .
- the range of frequencies in the range [F ' 0 , F 0 ] is called FFR ( Fundamental Frequency Range ).
- the reconstitution module 100 will now be described in detail with reference to the figure 3 .
- the module 100 comprises as input a first low-pass filter 101 whose cutoff frequency is substantially equal to the cut-off frequency F 0 .
- This filter 101 is intended to perform a first partition of the FFR within all the frequencies contained in the input signal S in , and to limit the distortion phenomenon by aliasing . Then, the signal S in thus filtered is subsampled by the block 102, in order to reduce the complexity of the filtering while maintaining a sufficient resolution for the future estimation of the fundamental frequencies to be reconstructed.
- the signal S in thus filtered low-pass and under-sampled is then processed in parallel in two branches 110, 120, of the module 100.
- the first branch 110 aims to generate a harmonic signal S harm resulting from the synthesis of three sinusoidal components of respective frequencies equal to a fundamental frequency contained in the FFR and its first two higher harmonics.
- the second branch 120 is intended to construct a time envelope env adapt (t) for modulating the harmonic signal S harm so that the output signal S out reproduces the temporal variations of the original signal.
- the first processing branch 110 comprises a second low-pass filter 111 designed to delimit the FFR again and to eliminate from the original signal the frequencies extending outside the FFR.
- This filter 111 advantageously incorporates an all-pass stage making it possible to linearize the phase of the signal, by neutralizing the variable phase shift effect introduced by the low-pass filtering.
- the phase effect introduced by this linearization is corrected by a delay ⁇ introduced ( figure 1 ) on the original signal L in or R in before it is combined with the output harmonic signal S out synthesized by the module 100 and reinjected in phase with the original signal to form the output signals L out and R out .
- the fundamental frequencies contained in the FFR and which one seeks to reconstitute by "pitch" effect, are determined by means of a block of zero crossings of the signal coming from the second low-pass filter 111. More precisely, block 112 determines the duration of the fundamental periods between two zero crossings and deduces the corresponding fundamental frequencies.
- sine table or wavetable, stored in memory, which gives the values of a sinusoidal period.
- the sampling step is chosen so as to be compatible with the computing power of the microprocessor of the system 10, it being understood that the method implemented by the invention is a real-time method and that consequently it must not introduce delay between the signals.
- the sine table can have 4096 points over an entire period.
- the sinusoidal components supplied by the generator 113 are then subjected to a weighting operation performed by a circuit 114 consisting in assigning to each component an experimentally determined patch matching coefficient, in order to give the output signal S out a stamp close to that of the original signal.
- the circuit 114 receives from the block 112 a frequency information and operates the weighting of the harmonics, which depends on the frequency instantaneous, from coefficient tables indexed by the frequency detected.
- the weighting applied to the 60 Hz, 120 Hz and 180 Hz sinusoids will be different from that applied to 100 Hz, 200 Hz and 300 Hz sinusoids.
- the weighted sinusoidal components are summed at the output of the weighting circuit 114 by an adder circuit 115 to form the synthesized harmonic signal S harm containing the first three harmonics of the fundamental frequency to be reconstituted.
- the second branch 120 of the process extracts the temporal envelope env (t) from the filtered low-pass and subsampled signal coming from the block 102, by means of a detector of envelope 121 shown in figure 4 which, for this purpose, conventionally performs a least squared RMS ( Root Mean Square ) calculation consisting in raising the signal squared by the block 121a, filtering it through a low-pass filter 121b, and then taking the square root by block 121c.
- RMS Root Mean Square
- the synthesized harmonic signal S harm does not have the same spectral composition as the original low frequency signal since it is composed not only of the fundamental frequency but also of the first two higher harmonics.
- the human ear does not perceive all the frequencies with the same intensity, and the temporal variations of two sound signals are not perceived in the same way if their spectral content is different.
- the variations of the envelope env (t) must be adapted according to the FFR.
- this adaptation is made on the second branch 120 of processing by a circuit 122 capable of performing a compression / expansion operation according to the input / output response curve given on the figure 6 .
- the envelope env (t) being previously calculated in decibels, the lowest levels of the envelope below a given threshold -N dB for example -27 dB in the example illustrated, are attenuated, while the higher levels , greater than -N dB, are further increased.
- This adaptation based on a perceptual scale, makes it possible to give the signal thus generated temporal variations which will be perceived as similar to the temporal variations of the original signal, thus making it possible to guarantee that the generated timbre will be faithful to the original timbre.
- the matching circuit 122 is controlled by a feedback loop 122b as follows.
- the compression / expansion process schematized at 122a, will be applied to the detected envelope determined by the envelope detector 121, then this expanded envelope will be used to modulate the sum of the synthesized harmonics (since the rate of expansion is the same for all harmonics).
- the rate of expansion corresponds to the slope of the line D shown figure 6 (as indicated above, after studying the curves of isophony one can consider that this slope will be constant).
- the ordinate at the origin of this line D will be designated ⁇ , and will be a function of the desired invariant point I, which in the illustrated example figure 6 is located at (-27 dB, - 27 dB).
- the invention proposes to use an envelope level matching system, based on a feedback loop.
- the principle of this loop is to compare at a threshold S the instantaneous level of the expanded envelope delivered at the output of the compression / expansion module 122a. If this level is below the threshold, the parameter ⁇ is increased by a fixed step for the adaptation of the next sample. Conversely, if the instantaneous level of the expanded envelope is greater than the threshold S, ⁇ is reduced by a fixed step.
- the step of increase or decrease will not be the same in one case and in the other. Indeed, if the instantaneous level of the expanded envelope suddenly becomes very large - in the case of a percussion for example - it is necessary that the decrease of ⁇ intervenes very quickly, to avoid reaching excessively high levels. On the other hand, if the instantaneous level is weak, it is possible to increase ⁇ more gradually, all the more since the nuances of the original piece must be respected: the natural attenuation of the bass notes must be respected because, if ⁇ increased as fast as it decreased, the notes would never stop.
- a flag variable takes the value 0 or 1 according to the result of the comparison between the instantaneous level of the expanded envelope and the threshold S
- the effective compression zone i.e., the area where the output signal is attenuated with respect to the input signal
- the effective expansion area i.e., the area where the signal output is amplified with respect to the input signal
- the feedback loop thus makes it possible to compress or expand the envelope according to its instantaneous level, in order to homogenize the level of the low components reinjected into the original signal whatever the musical genre of the piece in question (time constants of the enslavement being chosen sufficiently weak not to affect the natural decay of the notes).
- This makes it possible to generate harmonic signals of relatively constant amplitude regardless of the original signal.
- a low-frequency low-frequency sound signal at low frequencies will still be significantly enhanced by the system, while a sound signal with a high-energy bass line will be boosted to a limited level, to maintain a natural look. .
- This method of adapting the envelope, combining a compression / expansion module with a feedback control loop, makes it possible to generate a signal that will be perceived as similar to the original signal if it was produced by one more acoustic speaker. large dimensions.
- the harmonic signal S harm synthesized in the first branch 110 is modulated by the adapted envelope env adapt (t) from the second branch 120, by multiplication effected at means of the circuit 103, then the signal is oversampled by a factor 10 by the block 105 to return to the initial sampling frequency. It may be advantageous to introduce at this stage a low-pass filter in the oversampling process, since this filter is linear phase, it does not introduce phase distortion that would defeat the objective sought signal reinjection signal synthesized in phase in the original signal.
- a limiter is used at the output of the reconstitution system 10, so that the signal sent back to the loudspeakers 11, 12 remains content on a 16-bit dynamic.
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Abstract
Description
L'invention concerne un procédé et un système de reconstitution de basses fréquences d'un signal audio, utilisable en sortie d'un dispositif de reproduction du son présentant une fréquence de coupure pour les basses fréquences.The invention relates to a method and system for reconstructing low frequencies of an audio signal that can be used at the output of a sound reproducing device having a cut-off frequency for low frequencies.
L'invention trouve une application particulièrement avantageuse dans le domaine des équipements électro-acoustiques, notamment les enceintes stéréo pour la reproduction d'oeuvres musicales ou encore les enceintes d'ordinateurs personnels (PC) pour la reproduction de la bande son de fichiers vidéo.The invention finds a particularly advantageous application in the field of electro-acoustic equipment, including stereo speakers for the reproduction of musical works or even the speakers of personal computers (PC) for reproducing the soundtrack of video files.
On sait que toute enceinte acoustique possède une fréquence de coupure pour les basses fréquences en dessous de laquelle elle n'est plus capable de rayonner de l'énergie. Cette fréquence de coupure est directement liée aux dimensions du haut-parleur, et plus précisément à la taille de la membrane. Plus le haut-parleur est petit et plus la fréquence de coupure est élevée dans le spectre. Ainsi, une enceinte de petites dimensions imposera une atténuation naturelle au contenu basse fréquence d'un morceau de musique, et ceci au détriment de l'auditeur qui ne pourra pas bénéficier de cette information et ressentira de ce fait un effet désagréable lié à la perte des sonorités graves.It is known that any acoustic speaker has a cutoff frequency for low frequencies below which it is no longer able to radiate energy. This cutoff frequency is directly related to the dimensions of the speaker, and more precisely to the size of the membrane. The smaller the speaker, the higher the cutoff frequency in the spectrum. Thus, a small enclosure will impose a natural attenuation to the low frequency content of a piece of music, and this to the detriment of the listener who can not benefit from this information and will therefore feel an unpleasant effect related to the loss serious sounds.
Une première solution à cette difficulté consiste à appliquer un filtre pour amplifier les basses fréquences atténuées par l'enceinte acoustique, en forçant mécaniquement la membrane du haut-parleur à rayonner ces fréquences graves. Cependant, cette solution présente de réels risques pour l'intégrité du haut-parleur. En effet, l'excursion de la membrane, c'est-à-dire l'amplitude de son déplacement par rapport à sa position d'équilibre, deviendrait trop importante, jusqu'à l'endommager, voire la rompre. Une autre solution repose sur une propriété psycho-acoustique de l'oreille humaine qui permet de percevoir des fréquences basses même si elles ne sont pas effectivement transmises par un dispositif appartenant à la chaîne de reproduction du son, un haut-parleur d'enceinte acoustique par exemple. Cet effet de perception de tonalité ("pitch") résiduel, généralement connu sous le terme anglo-saxon de Missing-Fundamental Effect, résulte du fait que la perception du "pitch" d'un signal sonore n'est pas seulement liée à la présence de la fréquence fondamentale dans le signal mais également à celle d'harmoniques supérieures de cette fréquence. En d'autres termes, si la fréquence fondamentale, de 100 Hz par exemple, est éliminée d'un signal tout en conservant ses harmoniques supérieures, à 200, 300, 400 Hz, ... le "pitch" perçu sera le même, car dans ce cas c'est en fait l'écart fréquentiel, ici de 100 Hz, entre les harmoniques supérieures qui fixe le "pitch" et procure à l'auditeur l'impression d'entendre un signal de "pitch" 100 Hz. Bien entendu, cette troncature du signal, ainsi privé de sa fréquence fondamentale, se traduit par un timbre différent, ce dernier étant déterminé notamment par les amplitudes relatives de l'ensemble des harmoniques.A first solution to this difficulty consists in applying a filter to amplify the low frequencies attenuated by the acoustic enclosure, by mechanically forcing the speaker diaphragm to radiate these low frequencies. However, this solution presents real risks for the integrity of the loudspeaker. Indeed, the excursion of the membrane, that is to say the amplitude of its displacement relative to its equilibrium position, would become too important, to the point of damaging or even breaking it. Another solution is based on a psychoacoustic property of the human ear that makes it possible to perceive low frequencies even if they are not actually transmitted by a device belonging to the sound reproduction system, a loudspeaker speaker. for example. This residual pitch perception effect, generally known by the Anglo-Saxon term Missing-Fundamental Effect, results from the fact that the perception of the "pitch" of a sound signal is not only related to the presence of the fundamental frequency in the signal but also to that of higher harmonics of this frequency. In other words, if the fundamental frequency, of 100 Hz for example, is eliminated from a signal while preserving its higher harmonics, at 200, 300, 400 Hz, ... the perceived "pitch" will be the same, because in this case it is in fact the frequency difference, here of 100 Hz, between the higher harmonics which fixes the "pitch" and gives the listener the impression of hearing a "pitch"
Il est donc possible de remédier à l'atténuation, totale ou partielle, des fréquences fondamentales des signaux audio inférieures à la fréquence de coupure, en générant en temps réel un signal harmonique synthétisé à partir d'harmoniques associées à chacune des fréquences fondamentales atténuées, et en réinjectant ce signal harmonique dans le signal audio original. Un comprend en effet que, même si la fréquence fondamentale d'un son est atténuée ou complètement absorbée, les harmoniques supérieures, situées au-dessus de la fréquence de coupure du dispositif de reproduction du son, pourront être transmises et reconstituer le "pitch" du son par l'effet de tonalité résiduelle expliqué plus haut.It is therefore possible to remedy the attenuation, in whole or in part, of the fundamental frequencies of the audio signals below the cutoff frequency, by generating in real time a harmonic signal synthesized from harmonics associated with each of the attenuated fundamental frequencies, and re-injecting this harmonic signal into the original audio signal. One understands that, even if the fundamental frequency of a sound is attenuated or completely absorbed, the higher harmonics, located above the cutoff frequency of the sound reproduction device, can be transmitted and reconstruct the "pitch" sound by the residual tone effect explained above.
Ce procédé permettant d'étendre virtuellement vers le bas du spectre la bande passante d'un système électro-acoustique est désigné sous le terme de "génération de basses virtuelles".This method of virtually extending the bandwidth of an electro-acoustic system to the bottom of the spectrum is referred to as "virtual bass generation".
Dans ce contexte, le
On connaît également d'après le
Une autre technique encore est décrite dans le
Un autre problème tient au fait que le signal synthétisé présente des variations temporelles qui ne suivent pas fidèlement celles du signal original, ce qui a pour effet d'en altérer les nuances.Another problem is that the synthesized signal has time variations that do not faithfully follow those of the original signal, which has the effect of altering the nuances.
Le
Ce système présente l'inconvénient de ne pas être adapté à tous les types de signaux d'entrée, notamment si le but recherché est d'obtenir un rendu le plus naturel possible des tonalités, et non de produire des effets acoustiques en générant des composantes fréquentielles non contenues dans le signal original, comme dans le cas du
En effet, si l'on appliquait les enseignements de ce document à la reconstitution du "pitch" du son par l'effet de tonalité résiduelle expliqué plus haut, une ligne de basse de niveau modéré serait amplifiée d'une même valeur qu'une ligne de basse très forte, et l'effet serait perçu négativement par l'utilisateur.Indeed, if one applied the lessons of this document to the reconstitution of the "pitch" of the sound by the effect of residual tonality explained above, a line of bass of moderate level would be amplified of the same value as bass line very strong, and the effect would be perceived negatively by the user.
Un autre problème, commun à toutes les techniques décrites dans les documents présentés ci-dessus, tient au fait que ces techniques ne tiennent pas compte des variations de la perception auditive humaine avec la fréquence (effet dit de la "perception du loudness"). En effet, selon le niveau sonore et selon la fréquence, une même variation d'un signal acoustique ne produira pas la même variation d'intensité perçue. Par exemple, pour passer une variation d'intensité perçue de 40 à 50 phones, il faut augmenter le signal acoustique de presque 10 dB à 100 Hz, alors qu'il ne faut que 5 ou 6 dB supplémentaires à 50 Hz.Another problem, common to all the techniques described in the documents presented above, is that these techniques do not take into account the variations of the human auditory perception with the frequency (effect of the "perception of the loudness ") . Indeed, according to the sound level and according to the frequency, the same variation of an acoustic signal will not produce the same variation of perceived intensity. For example, to change the perceived intensity from 40 to 50 phones, the acoustic signal needs to be increased by almost 10 dB at 100 Hz, while it takes only 5 or 6 dB more at 50 Hz.
Aussi, un but de l'invention est de proposer un procédé de reconstitution de basses fréquences d'un signal audio en sortie d'un dispositif de reproduction du son qui respecte les variations temporelles du signal original de façon à en préserver les nuances, et qui tienne également compte des variations de la perception auditive humaine avec la fréquence.Also, an object of the invention is to propose a method of reconstituting low frequencies of an audio signal at the output of a sound reproduction device which respects the temporal variations of the original signal so as to preserve the nuances, and which also takes into account variations in human auditory perception with frequency.
Le procédé de l'invention est du type divulgué par le
- filtrage du signal audio au moyen d'un filtre passe-bas de fréquence de coupure sensiblement égale à ladite fréquence de coupure du dispositif de reproduction du son ;
- détermination d'une fréquence fondamentale à reconstituer à partir du signal audio filtré passe-bas ;
- génération d'un signal harmonique associé à ladite fréquence fondamentale à reconstituer,
- détection d'une enveloppe temporelle du signal audio filtré passe-bas ; et
- réinjection en phase dudit signal harmonique dans ledit signal audio par addition après multiplication de ce signal harmonique avec l'enve-loppe temporelle adaptée.
- filtering the audio signal by means of a low-pass filter of cutoff frequency substantially equal to said cutoff frequency of the sound reproduction device;
- determining a fundamental frequency to be reconstructed from the filtered low pass audio signal;
- generating a harmonic signal associated with said fundamental frequency to be reconstructed,
- detecting a temporal envelope of the filtered low-pass audio signal; and
- phase reinjection of said harmonic signal in said audio signal by addition after multiplication of this harmonic signal with the adapted temporal envelope.
L'invention prévoit d'opérer une adaptation dynamique de ladite enveloppe temporelle en fonction de la bande de fréquences considérée.The invention provides for a dynamic adaptation of said time envelope as a function of the frequency band considered.
L'adaptation dynamique de l'enveloppe temporelle en fonction de la bande de fréquences permet notamment de tenir compte des variations de la perception auditive humaine avec la fréquence, et la détection de l'enveloppe temporelle et sa prise en compte par multiplication avec le signal harmonique généré permet de moduler le signal synthétisé selon les variations temporelles de l'enveloppe.The dynamic adaptation of the temporal envelope according to the frequency band makes it possible, in particular, to take into account the variations of the human auditory perception with the frequency, and the detection of the temporal envelope and its taking into account by multiplication with the signal generated harmonic allows to modulate the signal synthesized according to the temporal variations of the envelope.
Selon l'invention, l'étape d'adaptation de l'enveloppe temporelle est réalisée par compression/expansion de l'enveloppe temporelle.According to the invention, the adaptation step of the temporal envelope is performed by compression / expansion of the temporal envelope.
Il a été en particulier constaté qu'il était préférable d'amplifier le gain de l'enveloppe lorsque la ligne de basse est faible ou modérée, afin que l'effet proposé soit toujours perçu positivement par l'utilisateur.In particular, it has been found that it is preferable to amplify the gain of the envelope when the bass line is weak or moderate, so that the proposed effect is always perceived positively by the user.
Ainsi, contrairement au procédé de compression/expansion proposé par le
Toujours selon l'invention, le contrôle de l'étape de compression/expansion est opéré conditionnellement après comparaison du niveau du signal comprimé/expansé par rapport à un seuil prédéterminé.Still according to the invention, the control of the compression / expansion step is performed conditionally after comparing the level of the compressed / expanded signal with respect to a predetermined threshold.
Selon diverses caractéristiques subsidiaires avantageuses :
- ce contrôle comprend la modification dynamique d'au moins un paramètre de la caractéristique de compression/expansion en fonction du niveau du signal comprimé/expansé ;
- cette modification dynamique est opérée de manière itérative, par pas successifs, le pas de modification dudit paramètre en cas de niveaux forts, supérieurs à un seuil donné, du niveau du signal comprimé/expansé étant supérieur au pas de modification de ce même paramètre en cas de niveaux faibles, supérieurs à un seuil donné, du signal comprimé/expansé ;
- le paramètre en question est la position du point invariant de la caractéristique de compression/expansion ;
- la caractéristique de compression/expansion est une caractéristique linéaire, pour des entrées/sorties exprimées en échelle logarithmique ;
- la pente de la caractéristique de compression/expansion est maintenue constante lors de la modification du paramètre ;
- la modification de la position du point invariant de la caractéristique de compression/expansion est opérée par modification de l'ordonnée à l'origine de ladite caractéristique linéaire, cette modification étant de préférence limitée par des valeurs minimale et maximale.
- this control comprises dynamically modifying at least one parameter of the compression / expansion characteristic as a function of the level of the compressed / expanded signal;
- this dynamic modification is performed iteratively, in successive steps, the step of modifying said parameter in case of strong levels, greater than a given threshold, the level of the compressed / expanded signal being greater than the step of modifying this parameter in case low levels, above a given threshold, of the compressed / expanded signal;
- the parameter in question is the position of the invariant point of the compression / expansion characteristic;
- the compression / expansion characteristic is a linear characteristic, for inputs / outputs expressed in logarithmic scale;
- the slope of the compression / expansion characteristic is kept constant when the parameter is changed;
- the modification of the position of the invariant point of the compression / expansion characteristic is effected by modifying the ordinate at the origin of said linear characteristic, this modification being preferably limited by minimum and maximum values.
L'invention concerne également, selon la revendication 10, un module de reconstitution de basses fréquences d'un signal audio pour la mise en oeuvre du procédé précité.The invention also relates, according to
On va maintenant décrire un exemple de mise en oeuvre du dispositif de l'invention, en référence aux dessins annexés où les mêmes références numériques désignent d'une figure à l'autre des éléments identiques ou fonctionnellement semblables.
- La
figure 1 est un schéma de l'architecture générale d'un système de reconstitution de basses fréquences conforme à l'invention. - La
figure 2 représente l'extension de bande-passante réalisée par le système de lafigure 1 . - La
figure 3 est un schéma détaillé du module de reconstitution de basses fréquences du système de lafigure 1 . - La
figure 4 est un bloc-diagramme du détecteur d'enveloppe temporelle du module de lafigure 3 . - La
figure 5 est un schéma du compresseur/expanseur du circuit d'adaptation d'enveloppe du module de lafigure 3 . - La
figure 6 est un diagramme de réponse du compresseur/expanseur de - la
figure 5 . - La
figure 7 illustre la manière dont évolue l'ordonnée à l'origine β du compresseur/expanseur de lafigure 5 , de façon différenciée dans le sens de l'augmentation et de la diminution, et avec application de seuils minimum et maximum. - Les
figures sont des diagrammes de réponse du compresseur/expanseur de la8a et 8bfigure 5 , respectivement dans une configuration de gain minimal et de gain maximal, montrant la manière dont la caractéristique est modifiée en fonction du niveau du gain appliqué par le compresseur/expanseur.
- The
figure 1 is a diagram of the general architecture of a low frequency reconstruction system according to the invention. - The
figure 2 represents the bandwidth extension achieved by the system of thefigure 1 . - The
figure 3 is a detailed diagram of the low frequency reconstitution module of the system of thefigure 1 . - The
figure 4 is a block diagram of the time envelope detector of the module of thefigure 3 . - The
figure 5 is a diagram of the compressor / expander of the envelope adaptation circuit of the module of thefigure 3 . - The
figure 6 is a response chart of the compressor / expander of - the
figure 5 . - The
figure 7 illustrates the way in which the ordinate β of the compressor / expander of thefigure 5 , in a differentiated way in the sense of increase and decrease, and with application of minimum and maximum thresholds. - The
Figures 8a and 8b are compressor / expander response diagrams from thefigure 5 in a configuration of minimum gain and maximum gain, respectively, showing how the characteristic is changed according to the gain level applied by the compressor / expander.
La description qui va suivre en regard des dessins annexés, donnée à titre d'exemple non limitatif, fera bien comprendre en quoi consiste l'invention et comment elle peut être réalisée.The following description with reference to the accompanying drawings, given by way of non-limiting example, will make it clear what the invention is and how it can be achieved.
Sur la
Le système de reconstitution de la
Dans la suite de cette description, on choisira de générer ledit signal harmonique de sortie Sout par sommation de trois composantes sinusoïdales de fréquences respectivement égales aux trois premières harmoniques du signal basse fréquence à reconstituer, à savoir la fréquence fondamentale, ou première harmonique, et les deux harmoniques supérieures suivantes, c'est-à-dire les harmoniques double et triple de la fréquence fondamentale. Bien entendu, d'autres choix sont possibles comme, par exemple, l'utilisation des quatre premières harmoniques, l'essentiel dans tous les cas étant que le signal harmonique généré contienne au moins deux harmoniques consécutives de manière à percevoir leur écart, lequel est égal au "pitch".In the remainder of this description, it will be chosen to generate said output harmonic signal S out by summation of three sinusoidal components of frequencies respectively equal to the first three harmonics of the low frequency signal to be reconstructed, namely the fundamental frequency, or first harmonic, and the next two higher harmonics, that is to say the double and triple harmonics of the fundamental frequency. Of course, other choices are possible, such as, for example, the use of the first four harmonics, the essential in all cases being that the generated harmonic signal contains at least two consecutive harmonics in order to perceive their difference, which is equal to the pitch.
En conséquence, dans le cas envisagé ici, si la fréquence de coupure F0 est de 120 Hz, la plage de basses fréquences pouvant bénéficier d'une reconstitution par effet de "pitch" s'étend entre 60 et 120 Hz. Pour une fréquence fondamentale à reconstituer de 60 Hz, les harmoniques à considérer sont celles à 60, 120, 180 Hz. La bande passante du système 100 est donc "virtuellement" étendue vers le bas jusqu'à une nouvelle fréquence de coupure F'0 égale à 60 Hz, comme le montre la
Le module de reconstitution 100 va maintenant être décrit en détail en référence à la
Le module 100 comprend en entrée un premier filtre passe-bas 101 dont la fréquence de coupure est sensiblement égale à la fréquence de coupure F0. Ce filtre 101 est destiné à effectuer une première partition du FFR au sein de toutes les fréquences contenues dans le signal d'entrée Sin, et à limiter le phénomène de distorsion par repliement (aliasing). Puis, le signal Sin ainsi filtré est sous-échantillonné par 10 par le bloc 102, afin de réduire la complexité du filtrage tout en conservant une résolution suffisante pour l'estimation à venir des fréquences fondamentales à reconstituer.The
Le signal Sin ainsi filtré passe-bas et sous-échantillonné est ensuite traité parallèlement dans deux branches 110, 120, du module 100.The signal S in thus filtered low-pass and under-sampled is then processed in parallel in two
La première branche 110 a pour but de générer un signal harmonique Sharm résultant de la synthèse de trois composantes sinusoïdales de fréquences respectives égales à une fréquence fondamentale contenue dans le FFR et ses deux premières harmoniques supérieures.The
La deuxième branche 120 a pour but de construire une enveloppe temporelle envadapt(t) destinée à moduler le signal harmonique Sharm de façon à ce que le signal de sortie Sout reproduise les variations temporelles du signal original. Le signal de sortie Sout résulte donc, en particulier, de la multiplication par le circuit multiplicateur 103 du signal harmonique Sharm par l'enveloppe envadapt(t) :
Comme le montre la
Ce filtre 111 incorpore avantageusement un étage passe-tout permettant de linéariser la phase du signal, en neutralisant l'effet de déphasage variable introduit par le filtrage passe-bas. L'effet de phase introduit par cette linéarisation est corrigé par un retard τ introduit (
Les fréquences fondamentales, contenues dans le FFR et que l'on cherche à reconstituer par effet de "pitch", sont déterminées au moyen d'un bloc 112 de passages par zéro du signal issu du deuxième filtre passe-bas 111. Plus précisément, le bloc 112 détermine la durée des périodes fondamentales entre deux passages par zéro et en déduit les fréquences fondamentales correspondantes.The fundamental frequencies, contained in the FFR and which one seeks to reconstitute by "pitch" effect, are determined by means of a block of zero crossings of the signal coming from the second low-
Pour chaque fréquence fondamentale déterminée par le bloc 112, un générateur 113 d'harmoniques fournit ensuite trois composantes sinusoïdales à la fréquence fondamentale elle-même (n=1), ainsi qu'aux deux harmoniques supérieures (n=2, n=3). Ces trois composantes sinusoïdales sont construites à partir d'une même table, dite "table de sinus" ou encore wavetable, stockée en mémoire, qui donne les valeurs d'une période de sinusoïde. Pour plus de détail sur cette technique, on pourra se référer à l'article
En pratique, le générateur 113 construit, à partir de la période fondamentale, les composantes sinusoïdales d'échantillon en échantillon en progressant selon un pas régulier dans la table. En fonction de la période détectée, le générateur 113 calcule un certain pas pour construire la composante à la fréquence fondamentale (n=1) et, partant du premier échantillon, il incrémente l'indice de ce pas afin de déterminer l'échantillon suivant. Le pas d'échantillonnage est choisi de manière à être compatible avec la puissance de calcul du microprocesseur du système 10, étant entendu que le procédé mis en oeuvre par l'invention est un procédé en temps réel et qu'en conséquence il ne doit pas introduire de retard entre les signaux. A titre d'exemple, la table de sinus peut comporter 4096 points sur une période entière.In practice, the generator 113 builds, from the fundamental period, the sinusoidal components from sample to sample while progressing in a regular step in the table. Based on the detected period, the generator 113 calculates a certain step to build the fundamental frequency component (n = 1) and, starting from the first sample, it increments the index of this step in order to determine the next sample. The sampling step is chosen so as to be compatible with the computing power of the microprocessor of the
Les deux harmoniques supérieures (n=2, n=3) sont générées de la même façon en prenant pour pas respectif le double et le triple du pas correspondant à la fréquence fondamentale.The two higher harmonics (n = 2, n = 3) are generated in the same way, taking for each step the double and triple of the pitch corresponding to the fundamental frequency.
On peut voir sur la
Les composantes sinusoïdales pondérées sont sommées en sortie du circuit de pondération 114 par un circuit additionneur 115 pour former le signal harmonique synthétisé Sharm contenant les trois premières harmoniques de la fréquence fondamentale à reconstituer considérée.The weighted sinusoidal components are summed at the output of the
Parallèlement à la génération des harmoniques dans la première branche 110, la seconde branche 120 du traitement extrait l'enveloppe temporelle env(t) du signal filtré passe-bas et sous-échantillonné issu du bloc 102, au moyen d'un détecteur d'enveloppe 121 représenté à la
Par ailleurs, il faut remarquer que le signal harmonique synthétisé Sharm n'a pas la même composition spectrale que le signal de basse fréquence original puisqu'il se compose non seulement de la fréquence fondamentale mais aussi des deux premières harmoniques supérieures. Or, l'oreille humaine ne perçoit pas toutes les fréquences avec la même intensité, et les variations temporelles de deux signaux sonores ne sont pas perçues de la même façon si leur contenu spectral est différent. Afin de tenir compte de cette contrainte, les variations de l'enveloppe env(t) doivent être adaptées en fonction du FFR.Moreover, it should be noted that the synthesized harmonic signal S harm does not have the same spectral composition as the original low frequency signal since it is composed not only of the fundamental frequency but also of the first two higher harmonics. However, the human ear does not perceive all the frequencies with the same intensity, and the temporal variations of two sound signals are not perceived in the same way if their spectral content is different. To take account of this constraint, the variations of the envelope env (t) must be adapted according to the FFR.
Conformément à la
Pour simplifier la réalisation du circuit, et sans que cela ait d'incidence notable sur les résultats obtenus, on peut faire, dans la gamme de fréquences analysées (typiquement 40-120 Hz) la double approximation suivante :
- le taux d'expansion, c'est-à-dire le facteur par lequel il faut multiplier une variation x donnée sur le signal original, exprimée en décibels, pour obtenir la même variation d'intensité perçue sur le signal harmonique, exprimée en phones, est constant pour une harmonique considérée ; et
- le taux d'expansion ne dépend pas non plus de l'ordre de l'harmonique considérée (alors que, théoriquement, il augmenterait avec l'ordre de l'harmonique).
- the rate of expansion, that is to say the factor by which it is necessary to multiply a given variation x on the original signal, expressed in decibels, to obtain the same variation of intensity perceived on the harmonic signal, expressed in phones is constant for a considered harmonic; and
- the rate of expansion does not depend either on the order of the harmonic considered (whereas, theoretically, it would increase with the order of the harmonic).
On choisira pour la valeur du taux d'expansion une moyenne des taux d'expansion pour toutes les fréquences, amplitudes et ordres d'harmonique considérés.One will choose for the value of the rate of expansion an average of the rates of expansion for all the frequencies, amplitudes and orders of harmonic considered.
Le processus de compression/expansion, schématisé en 122a, sera appliqué sur l'enveloppe détectée déterminée par le détecteur d'enveloppe 121, puis cette enveloppe expansée sera utilisée pour moduler la somme des harmoniques synthétisées (puisque le taux d'expansion est le même pour toutes les harmoniques).The compression / expansion process, schematized at 122a, will be applied to the detected envelope determined by the
Le taux d'expansion, désigné par la suite α, correspond à la pente de la droite D représentée
Si l'on souhaite que le système amplifie dans tous les cas le niveau sonore perçu des sons graves (c'est-à-dire même quand le niveau de l'enveloppe temporelle est inférieur à--N dB (-27 dB dans l'exemple illustré), et puisque α est fixé, il convient d'augmenter β d'une certaine valeur pour que la caractéristique D de compression/expansion passe au-dessus de la droite y = x de pente unité pour ce niveau faible de l'enveloppe. Inversement, dans le cas d'un niveau de basses important sur le signal original, il faut veiller à ne pas trop amplifier l'enveloppe.If it is desired that the system amplify in all cases the perceived loudness of the bass sounds (that is, even when the level of the time envelope is less than - -N dB (-27 dB in the illustrated example), and since α is fixed, it is necessary to increase β by a certain value so that the characteristic D of compression / expansion passes above the line y = x of unit slope for this low level of l Conversely, in the case of a significant bass level on the original signal, care should be taken not to over-amplify the envelope.
Pour obtenir ce résultat, l'invention propose d'utiliser un système d'adaptation du niveau de l'enveloppe, basé sur une boucle de rétroaction.To achieve this result, the invention proposes to use an envelope level matching system, based on a feedback loop.
Le principe de cette boucle, illustré
Le pas d'augmentation ou de diminution ne sera pas le même dans un cas et dans l'autre. En effet, si le niveau instantané de l'enveloppe expansée devient brusquement très grand - dans le cas d'une percussion par exemple -, il faut que la diminution de β intervienne très vite, pour éviter d'atteindre des niveaux excessivement importants. En revanche, si le niveau instantané est faible, il est possible d'augmenter β plus progressivement, d'autant plus qu'il convient de respecter les nuances du morceau original : l'atténuation naturelle des notes de basses doit être respectée car, si β augmentait aussi vite qu'il diminuait, les notes ne s'arrêteraient jamais.The step of increase or decrease will not be the same in one case and in the other. Indeed, if the instantaneous level of the expanded envelope suddenly becomes very large - in the case of a percussion for example - it is necessary that the decrease of β intervenes very quickly, to avoid reaching excessively high levels. On the other hand, if the instantaneous level is weak, it is possible to increase β more gradually, all the more since the nuances of the original piece must be respected: the natural attenuation of the bass notes must be respected because, if β increased as fast as it decreased, the notes would never stop.
La
Le principe d'incrémentation/décrémentation de β est le suivant : une variable flag prend la valeur 0 ou 1 en fonction du résultat de la comparaison entre le niveau instantané de l'enveloppe expansée et le seuil S, et le pas d'adaptation de β est calculé selon la formule :
x0 étant choisi en fonction du rapport souhaité entre l'augmentation et le pas de diminution de β, et coeff étant choisi en fonction de la vitesse d'adaptation souhaitée (si coeff est petit, β évoluera doucement alors qu'il évoluera rapidement avec coeff grand).The principle of incrementation / decrementation of β is as follows: a flag variable takes the
x 0 being chosen according to the desired ratio between the increase and the decrease step of β, and coeff being chosen as a function of the desired adaptation speed (if coeff is small, β will evolve slowly while it will evolve rapidly with coeff great).
Les variations de β vont se traduire par un déplacement du point invariant I de la caractéristique D de compression/expansion.The variations of β will result in a displacement of the invariant point I of the compression / expansion characteristic D.
Les
La zone de compression effective (c'est-à-dire la zone où le signal de sortie est atténué par rapport au signal d'entrée) et la zone d'expansion effective (c'est-à-dire la zone où le signal de sortie est amplifié par rapport au signal d'entrée) sont séparées par le point invariant I, les secteurs compris entre la caractéristique D et la droite de pente unité y = x définissant les régions de compression (en deçà du point I) et d'expansion (au-delà du point I).The effective compression zone (i.e., the area where the output signal is attenuated with respect to the input signal) and the effective expansion area (i.e., the area where the signal output is amplified with respect to the input signal) are separated by the invariant point I, the sectors between the characteristic D and the unit slope line y = x defining the compression regions (below point I) and expansion (beyond point I).
La boucle de rétroaction permet ainsi de compresser ou d'expanser l'enveloppe en fonction de son niveau instantané, afin d'homogénéiser le niveau des composantes basses réinjectées dans le signal original quel que soit le genre musical du morceau considéré (les constantes de temps de l'asservissement étant choisies suffisamment faibles pour ne pas affecter la décroissance naturelle des notes). Cela permet de générer des signaux harmoniques d'amplitude relativement constante quel que soit le signal original. Ainsi, un signal sonore basse fréquence de faible dynamique dans les basses fréquences sera quand même sensiblement renforcé par le système, tandis qu'un signal sonore avec une ligne de basse de forte énergie sera renforcé à un niveau limité, afin de conserver un rendu naturel.The feedback loop thus makes it possible to compress or expand the envelope according to its instantaneous level, in order to homogenize the level of the low components reinjected into the original signal whatever the musical genre of the piece in question (time constants of the enslavement being chosen sufficiently weak not to affect the natural decay of the notes). This makes it possible to generate harmonic signals of relatively constant amplitude regardless of the original signal. Thus, a low-frequency low-frequency sound signal at low frequencies will still be significantly enhanced by the system, while a sound signal with a high-energy bass line will be boosted to a limited level, to maintain a natural look. .
Cette méthode d'adaptation de l'enveloppe, combinant un module de compression/expansion avec une boucle de contrôle par rétroaction, permet de générer un signal qui sera perçu comme semblable au signal original si celui-ci était produit par une enceinte acoustique de plus grandes dimensions.This method of adapting the envelope, combining a compression / expansion module with a feedback control loop, makes it possible to generate a signal that will be perceived as similar to the original signal if it was produced by one more acoustic speaker. large dimensions.
Si l'on revient à la
Comme la réinjection du signal de sortie Sout filtré passe-haut et sur-échantillonné présente des risques de dépassement de la dynamique, on utilise un limiteur en sortie du système 10 de reconstitution, pour que le signal renvoyé aux haut-parleurs 11, 12 reste contenu sur une dynamique de 16 bits.Since the reinjection of the output signal S out filtered high-pass and oversampled presents risks of exceeding the dynamic range, a limiter is used at the output of the
Claims (10)
- Method for reconstituting low frequencies of an audio signal output from a sound reproduction device (11, 12) having a low cut-off frequency (F0), comprising steps for:- filtering the audio signal by means of a low-pass filter (101) with a cut-off frequency substantially equal to said cut-off frequency (F0) of the sound reproduction device;- determining a fundamental frequency to be reconstituted from the low-pass filtered audio signal;- generating a harmonic signal (Sharm) associated with said fundamental frequency to be reconstituted;- detecting a time envelope (env(t)) of the low-pass filtered audio signal; and- reinjecting said harmonic signal into said audio signal by addition after multiplication of this harmonic signal (Sharm) with the adapted time envelope (envadapt(t)),characterized by a step for dynamic adaptation of said time envelope (env(t)) according to the frequency band concerned, performed by compression/expansion (122a) of the time envelope (env(t)) with a control by feedback loop (122b) of said compression/expansion step applied conditionally after comparison of the level of the compressed/expanded signal with a predetermined threshold (S), and in that the harmonic signal is reinjected in phase into said audio signal.
- Method according to Claim 1, in which said control by feedback loop of the compression/expansion step comprises the dynamic modification of at least one parameter of the compression/expansion characteristic (D) according to the level of the compressed/expanded signal.
- Method according to Claim 2, in which said dynamic modification of said parameter is a modification applied iteratively, by successive steps.
- Method according to Claim 3, in which the modification step of said parameter in case of high levels, above a given threshold, of the level of the compressed/expanded signal is greater than the modification step of this same parameter in case of low levels, above a given threshold, of the compressed/expanded signal.
- Method according to Claim 2, in which said at least one parameter is the position of the invariant point (I) of the compression/expansion characteristic.
- Method according to Claim 5, in which said compression/expansion characteristic is a linear characteristic (D), for inputs/outputs expressed in logarithmic scale.
- Method according to Claim 6, in which the slope (•) of said compression/expansion characteristic is kept constant during the modification of said parameter.
- Method according to Claims 5 and 6 taken in combination, in which the modification of the position of said invariant point (I) is done by modifying the ordinate at the origin (•) of said linear characteristic.
- Method according to Claim 8, in which said modification of the ordinate at the origin of the linear characteristic is a modification limited by minimum and maximum values.
- Module for reconstituting low frequencies of an audio signal (Sin), at the output of a sound reproduction device (11, 12) having a cut-off frequency (F0) for said low frequencies, for implementing the method according to one of the preceding claims, this module comprising:- a low-pass filter (101) suitable for filtering said audio signal (Sin) with a cut-off frequency substantially equal to the cut-off frequency (F0) of said sound reproduction device (11, 12);- a first branch (110) for processing the low-pass filtered audio signal intended to generate a harmonic signal (Sharm) associated with at least one fundamental frequency to be reconstituted in the audio signal, said first branch (110) comprising a block (112) suitable for determining said fundamental frequency;- a second branch (120) for processing the low-pass filtered audio signal comprising a detector (121) for detecting the time envelope of said signal; and- a circuit suitable for reinjecting in phase said harmonic signal into said audio signal by addition after multiplication of this harmonic signal (Sharm) with the adapted time envelope (envadapt(t)),module being characterized in that:- the second branch (120) also comprises a circuit (122) for adapting said time envelope according to its instantaneous level, comprising a compressor/expander (122a) for the time envelope and a control loop (122b) for said compressor/expander (122a) based on feedback according to the level of the compressed/expanded signal;- the control by the feedback loop is applied conditionally after comparison of the level of the compressed/expanded signal with a predetermined threshold (S); and- the harmonic signal is reinjected in phase into said audio signal.
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2008
- 2008-04-29 FR FR0802388A patent/FR2930672B1/en not_active Expired - Fee Related
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2009
- 2009-04-29 AT AT09290310T patent/ATE521963T1/en not_active IP Right Cessation
- 2009-04-29 ES ES09290310T patent/ES2372259T3/en active Active
- 2009-04-29 US US12/432,250 patent/US8213636B2/en active Active
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US9060223B2 (en) | 2013-03-07 | 2015-06-16 | Aphex, Llc | Method and circuitry for processing audio signals |
CN108365837A (en) * | 2018-02-05 | 2018-08-03 | 中国电子科技集团公司第二十四研究所 | Eliminate processing circuit and method of the pulse signal by baseline variation after capacitance |
CN108365837B (en) * | 2018-02-05 | 2021-11-09 | 中国电子科技集团公司第二十四研究所 | Processing circuit and method for eliminating baseline change of pulse signal after pulse signal passes through blocking capacitor |
EP3755006A1 (en) | 2019-06-19 | 2020-12-23 | Faurecia Clarion Electronics Europe | Standalone audio system for seat headrest, seat headrest and associated vehicle |
FR3097711A1 (en) | 2019-06-19 | 2020-12-25 | Parrot Faurecia Automotive Sas | Autonomous audio system for seat headrest, seat headrest and associated vehicle |
US11259120B2 (en) | 2019-06-19 | 2022-02-22 | Faurecia Clarion Electronics Europe | Autonomous audio system for seat headrest, seat headrest and associated vehicle |
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ATE521963T1 (en) | 2011-09-15 |
ES2372259T3 (en) | 2012-01-17 |
FR2930672B1 (en) | 2011-06-24 |
US8213636B2 (en) | 2012-07-03 |
FR2930672A1 (en) | 2009-10-30 |
EP2113913A1 (en) | 2009-11-04 |
US20090323983A1 (en) | 2009-12-31 |
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