EP2113913B1 - Verfahren und System zur Wiederherstellung von Niedrigfrequenzen in einem Audiosignal - Google Patents

Verfahren und System zur Wiederherstellung von Niedrigfrequenzen in einem Audiosignal Download PDF

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Publication number
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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signal
frequency
audio signal
low
compression
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French (fr)
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EP2113913A1 (de
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Julien De Muynke
Benoit Pochon
Guillaume Pinto
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Parrot SA
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Parrot SA
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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
    • 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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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Tone Control, Compression And Expansion, Limiting Amplitude (AREA)
  • Signal Processing Not Specific To The Method Of Recording And Reproducing (AREA)
  • Stereophonic System (AREA)

Claims (10)

  1. Verfahren zur Wiederherstellung von Niedrigfrequenzen eines Audiosignals am Ausgang einer Tonwiedergabevorrichtung (11, 12), die eine niedrige Grenzfrequenz (F0) aufweist, umfassend die folgenden Schritte:
    - Filtern des Audiosignals mit Hilfe eines Tiefpassfilters (101) mit einer Grenzfrequenz im Wesentlichen gleich der Grenzfrequenz (F0) der Tonwiedergabevorrichtung;
    - Bestimmung einer aus dem durch den Tiefpassfilter gefilterten Audiosignal wiederherzustellenden Grundfrequenz;
    - Erzeugung eines harmonischen Signals (Sharm), das der wiederherzustellenden Grundfrequenz zugeordnet ist;
    - Erfassung einer zeitlichen Hülle (env(t)) des durch den Tiefpassfilter gefilterten Audiosignals; und
    - Wiedereinleitung des harmonischen Signals in das Audiosignal durch Addition nach Multiplikation dieses harmonischen Signals (Sharm) mit der angepassten zeitlichen Hülle (envadapt(t)),
    gekennzeichnet durch einen Schritt der dynamischen Anpassung der zeitlichen Hülle (env(t)) in Abhängigkeit von dem betreffenden Frequenzband, die durch Kompression/Expansion (122a) der zeitlichen Hülle (env(t)) durchgeführt wird, mit einer Kontrolle des Schrittes der Kompression/Expansion durch Retroaktionsschleife (122b), die bedingend nach Vergleichen des Niveaus des komprimierten/expandierten Signals mit einer vorbestimmten Schwelle (S) erfolgt,
    und dass das harmonische Signal in Phase wieder in das Audiosignal eingeleitet wird.
  2. Verfahren nach Anspruch 1, bei dem die Kontrolle des Schrittes der Kompression/Expansion durch Retroaktionsschleife die dynamische Veränderung mindestens eines Parameters des Merkmals (D) der Kompression/Expansion in Abhängigkeit vom Niveau des komprimierten/expandierten Signals umfasst.
  3. Verfahren nach Anspruch 2, bei dem die dynamische Veränderung des Parameters eine auf iterative Weise oder in aufeinander folgenden Schritten durchgeführte Veränderung ist.
  4. Verfahren nach Anspruch 3, bei dem der Schritt der Veränderung des Parameters im Falle von hohen Niveaus des Niveaus des komprimierten/expandierten Signals, die höher als eine gegebene Schwelle sind, größer als der Schritt der Veränderung dieses selben Parameters im Falle von niedrigen Niveaus des komprimierten/expandierten Signals, die über einer gegebenen Schwelle liegen, ist.
  5. Verfahren nach Anspruch 2, bei dem der mindestens eine Parameter die Position des invariablen Punktes (I) des Merkmals der Kompression/Expansion ist.
  6. Verfahren nach Anspruch 5, bei dem das Merkmal der Kompression/Expansion ein lineares Merkmal (D) für Eingänge/Ausgänge, die im logarithmischen Maßstab ausgedrückt sind, ist.
  7. Verfahren nach Anspruch 6, bei dem das Gefälle (•) des Merkmals der Kompression/Expansion bei der Veränderung des Parameters konstant gehalten wird.
  8. Verfahren nach den Ansprüchen 5 und 6 in Kombination, bei dem die Veränderung der Position des invariablen Punktes (I) durch Veränderung der Ordinate am Ursprung (•) des linearen Merkmals erfolgt.
  9. Verfahren nach Anspruch 8, bei dem die Veränderung der Ordinate am Ursprung des linearen Merkmals eine durch Mindest- und Höchstwerte begrenzte Veränderung ist.
  10. Modul zur Wiederherstellung von Niedrigfrequenzen eines Audiosignals (Sin) am Ausgang einer Vorrichtung (11, 12) zur Tonwiedergabe, die eine Grenzfrequenz (F0) für die Niedrigfrequenzen aufweist, für den Einsatz des Verfahrens nach einem der vorhergehenden Ansprüche, wobei dieses Modul Folgendes umfasst:
    - einen Tiefpassfilter (101), der das Audiosignal (Sin) mit einer Grenzfrequenz im Wesentlichen gleich der Grenzfrequenz (F0) der Tonwiedergabevorrichtung (11, 12) filtern kann;
    - einen ersten Bearbeitungsabschnitt (110) für das durch den Tiefpassfilter gefilterte Audiosignal, der dazu bestimmt ist, ein harmonisches Signal (Sharm) zu erzeugen, das mindestens einer im Audiosignal wiederherzustellenden Grundfrequenz zugeordnet ist, wobei der erste Abschnitt (110) einen Block (112) umfasst, der die Grundfrequenz bestimmen kann;
    - einen zweiten Bearbeitungsabschnitt (120) für das durch den Tiefpassfilter gefilterte Audiosignal, umfassend einen Detektor (121) der zeitlichen Hülle des Signals; und
    - eine Schaltung, die das harmonische Signal durch Addition in das Audiosignal nach Multiplikation dieses harmonischen Signals (Sharm) mit der angepassten zeitlichen Hülle (envadapt(t)) in Phase wieder einleiten kann, wobei das Modul dadurch gekennzeichnet ist, dass
    - der zweite Abschnitt (120) ferner eine Schaltung (122) zur Anpassung der zeitlichen Hülle in Abhängigkeit von ihrem momentanen Niveau aufweist, umfassend eine Kompressions-/Expansionseinrichtung (122a) für die zeitliche Hülle und eine Kontrollschleife (122b) für die Kompressions-/Expansionseinrichtung (122a) durch Retroaktion in Abhängigkeit vom Niveau des komprimierten/expandierten Signals;
    - die Kontrolle durch die Retroaktionsschleife bedingend nach Vergleichen des Niveaus des komprimierten/expandierten Signals mit einer vorbestimmten Schwelle (S) erfolgt; und
    - das harmonische Signal in Phase wieder in das Audiosignal eingeleitet wird.
EP09290310A 2008-04-29 2009-04-29 Verfahren und System zur Wiederherstellung von Niedrigfrequenzen in einem Audiosignal Active EP2113913B1 (de)

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Application Number Priority Date Filing Date Title
FR0802388A FR2930672B1 (fr) 2008-04-29 2008-04-29 Procede et systeme de reconstitution de basses frequences dans un signal audio

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EP2113913A1 EP2113913A1 (de) 2009-11-04
EP2113913B1 true EP2113913B1 (de) 2011-08-24

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US (1) US8213636B2 (de)
EP (1) EP2113913B1 (de)
AT (1) ATE521963T1 (de)
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FR (1) FR2930672B1 (de)

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CN108365837A (zh) * 2018-02-05 2018-08-03 中国电子科技集团公司第二十四研究所 消除脉冲信号通过隔直电容后基线变化的处理电路及方法
EP3755006A1 (de) 2019-06-19 2020-12-23 Faurecia Clarion Electronics Europe Autonomes audiosystem für die kopfstütze eines sitzes, entsprechende kopfstütze und entsprechendes fahrzeug

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US9060223B2 (en) 2013-03-07 2015-06-16 Aphex, Llc Method and circuitry for processing audio signals
CN108365837A (zh) * 2018-02-05 2018-08-03 中国电子科技集团公司第二十四研究所 消除脉冲信号通过隔直电容后基线变化的处理电路及方法
CN108365837B (zh) * 2018-02-05 2021-11-09 中国电子科技集团公司第二十四研究所 消除脉冲信号通过隔直电容后基线变化的处理电路及方法
EP3755006A1 (de) 2019-06-19 2020-12-23 Faurecia Clarion Electronics Europe Autonomes audiosystem für die kopfstütze eines sitzes, entsprechende kopfstütze und entsprechendes fahrzeug
FR3097711A1 (fr) 2019-06-19 2020-12-25 Parrot Faurecia Automotive Sas Système audio autonome pour appui-tête de siège, appui-tête de siège et véhicule associés
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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FR2930672B1 (fr) 2011-06-24
FR2930672A1 (fr) 2009-10-30
US8213636B2 (en) 2012-07-03
ES2372259T3 (es) 2012-01-17
EP2113913A1 (de) 2009-11-04
ATE521963T1 (de) 2011-09-15
US20090323983A1 (en) 2009-12-31

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