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WO1993004529A1 - Procede et dispositif de filtrage numerique - Google Patents

Procede et dispositif de filtrage numerique Download PDF

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Publication number
WO1993004529A1
WO1993004529A1 PCT/SE1992/000521 SE9200521W WO9304529A1 WO 1993004529 A1 WO1993004529 A1 WO 1993004529A1 SE 9200521 W SE9200521 W SE 9200521W WO 9304529 A1 WO9304529 A1 WO 9304529A1
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WO
WIPO (PCT)
Prior art keywords
filter
branches
signal
sampling frequency
sampling
Prior art date
Application number
PCT/SE1992/000521
Other languages
English (en)
Inventor
Jiri Klokocka
Original Assignee
Jiri Klokocka
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Jiri Klokocka filed Critical Jiri Klokocka
Publication of WO1993004529A1 publication Critical patent/WO1993004529A1/fr

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Classifications

    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H17/00—Networks using digital techniques
    • H03H17/02—Frequency selective networks
    • H03H17/06—Non-recursive filters
    • H03H17/0621—Non-recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing
    • H03H17/0635—Non-recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing characterized by the ratio between the input-sampling and output-delivery frequencies
    • H03H17/0685—Non-recursive filters with input-sampling frequency and output-delivery frequency which differ, e.g. extrapolation; Anti-aliasing characterized by the ratio between the input-sampling and output-delivery frequencies the ratio being rational
    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03G—CONTROL OF AMPLIFICATION
    • H03G5/00—Tone control or bandwidth control in amplifiers
    • H03G5/005—Tone control or bandwidth control in amplifiers of digital signals

Definitions

  • the present invention relates to a filtering procedure of a kind stated in the ingress to claim 1.
  • the invention relates even to a digital filter arrangement comprising a known time discrete filter of the transversal type, also named Finiry Impulse Response filter (FIR-fHter), of a kind as stated in the ingress to the claim 8.
  • FIR-fHter Finiry Impulse Response filter
  • US-A-3992582 reveals a filter device aiming at giving a "Concert Hall” character to a signal by delaying the signals lower frequencies longer.
  • That known device a number of parallel connected part-filters with a large number of delay stages and whose sampling frequencies are variable, is indicated.
  • That known filter device aims for giving the reverberation course a "softer tail", where that reverberation course is primarily determined by the choice of sampling frequencies.
  • a further connection is required (column 4, last paragraph) which gives a very limited control over the produced reverberation course.
  • the aim of the invention is to assign a process and a filter device that offers full freedom to create filter characteristics and that is possible to realize with today's computer technology at a cost that is low enough to make the filter arrangement acceptable to the industry.
  • a further purpose of the invention is to show a filter device that can be supplied with characteristics that have the same resonances as a model, such as another filter device, a particular violin sound board, a particular acoustic environment or as a preferred guitar amplifier's loudspeaker.
  • the aim includes assigning a filter device that can be given such characteristics by feeding a set of coefficients from a data file. That data file is stored in a registering medium that either contains artificially created characteristics or a true reflexion of characteristics from a real concert hall or similar.
  • the aim includes also that the invented procedure and the invented filter device will be able to very accurately give an input signal characteristics that conform with the set of coefficients contained in the data file. Instead of feeding the set of coefficients from a data file, the set of coefficients can be fed as output data from a program execution or another way.
  • nature sounds e.g. water splash
  • synthetic sounds e.g. noise generated by a synthesizer
  • mathemati ⁇ cally defined functions e.g. a sine sweep
  • hand-drawn functions e.g. with a mouse on a computer monitor
  • impulse response to virtual environment e.g. a CAD- designed concert hall
  • various combinations of both the artistically created and the real measured impulse response e.g. a sound collage
  • the sound passing through the inventive device inherits the properties of the sound that the data file's impulse response represents. That means for example a musical instrument with a dry and sterile sound can receive sound properties from another instrument or from a whole orchestra or from a fantasy sound. In practice, this can be most easily done using a very short piece of recording of a well sounding orchestra on a compact disc (CD), e.g. 1000 samples following one after another, as coefficients for the inventive device impulse response.
  • CD compact disc
  • inventive filter device is described here in the form of interconnected elements or structural blocks but the professional would realize that the inventive device can be realized in any form, e.g. as a super computer whose program would build the described structure, or as a special hardware that is optimized for this aim, or as a combination of those options described above, or as a general computer working in non-realtime.
  • sampling converters their function is to see to it that the time discrete digital signal is transmitted free of distortion. That means that the time discrete signal at the input of the transversal filter shall have a sampling frequency that the transversal filter is ready to receive. By the same token, the signal at the input of the summation circuit shall have a sampling frequency that the summation circuit can receive.
  • Sampling frequencies for the input signal and output signal of the part-filters can vary.
  • a DAT-player sends a digital audio-signal with a sampling frequen ⁇ cy of 48kHz to the inventive device and the processed sound is then sent with a new sampling frequency of 44.1kHz from the inventive device's output to another DAT- player.
  • This way the sound is processed and at the same time the sampling frequency is converted from the DAT-player's usual 48kHz to CD-player's standard 44.1kHz. Therefore it is advisable to arrange one or two sampling converters also in the upper filter branch.
  • the mixing device to mix the signals from the part-filters can be a pure adder or can have the form of a mixer with several outputs.
  • the branches responsible for the so called early reflections are mixed to one output and the so called cluster of the reverberation is mixed to another output to achieve a deeper spatial experience with the basis of one single impulse response.
  • Signals from the various branches are then weighted by individual factors to the various mixer out- puts.
  • the part-filters' output signals can run loudspeakers, whose sounds are summarised in the listening room.
  • the description of the invention refers to a mono-design.
  • a mono-design only one single impulse response is used.
  • the various branches give origin to various time sectors of the device's impulse response. These time sectors can overlap each other partly but not entirely.
  • a number of impulse responses is used. This can be achieved by connecting a number of devices of mono-design to an input signal, where outputs from those devices make up the various channels' output signals.
  • the input signal is connected to two devices which deliver the signals for the left and right channel respectively.
  • a stereo reverberation is achieved by providing the two devices with impulse responses from the same hall that are taken between different points for each channel respectively.
  • branches are then mixed to right and left channel respectively.
  • some branches can have the same passage times since they can give origin to the same time sectors of the device's impulse response.
  • the signal can be delayed in one or more steps in one, a few or all part-filter branches.
  • the first taps in the transversal filters can be omitted. Through this technique it is possible to reduce the number of coefficients at the transversal filters and thus also reduce the cost of the device.
  • external connections can be arranged between points of free choice in the structure of the filter device.
  • a feedback can be introduced from the device's output to its input.
  • Another, more particular way is to introduce feedback in the delay at the part-filter branch that is responsible for the last part of the filter device's impulse response. Through repeating the last part of the impulse response with a decreasing volume, the device's impulse response is extended without the impulse response's beginning being affected by the feedback. When required, the signal in this filter branch may require to be delayed in several steps.
  • the sampling converters for the part-filter branch that is controlled by the highest sampling frequency can be moved outside the parallel connection of the part-filter branches, whereby lower requirements can be made for the other sampling converters.
  • Sampling frequencies belonging to a particular part-filter branch are obtained preferably through multiplication or division of the sampling frequency for the filter device's input signal, with a constant that preferably equals to a fractional number.
  • a part-filter can work with a sampling frequency that equals to V3 of the sampling frequency of the filter device's input signal.
  • sampling frequencies for all part filters but one can be lower than the sampling frequency of either the filter device's input signal or output signal.
  • the total number of utilized taps or coefficients for all part-filter branches' filtering of transversal filter type amounts to at least 5000.
  • the total number of coefficients is at least 15000 and most preferably in the magnitude of 50000.
  • the number of part-filters is at least 2, preferably 3 and most preferably at least 4
  • the set of coefficients can be obtained in the following way: A file with sampled impulse response is read while the first sample is used as the first coefficient to the first part-filter, the second sample is used as the second coefficient to the first part-filter, and so on until all coefficients in the first part-filter have been defined.
  • the following samples, read from the file, undergo a conversion of sampling fre ⁇ quency to second part-filter's sampling frequency after which the converted samples are assigned, in the same way as in the case of the first part-filter, to the second part-filter's coefficients.
  • Obtaining coefficients for the Ibird and following part-fil ⁇ ters is analogous to obtaining of coefficients for the second part-filter.
  • the procedure and the filter device according to the invention makes it possible to register the response characteristics of a real environment such as a specific concert hall with a very high accuracy, while the registering is in the form of a set of coefficients of the number mentioned above.
  • a set of coefficients enables the inventive filter device to give a signal a character that is typical for the registered environment/concert hall with a very high accuracy.
  • Another appUcation for the filtering procedure and the filter device according to the invention is the extinction of feedback in a sound system. It is a well-known problem that the sound from the loudspeaker is acoustically fed back into the microphone. When the amplification of the sound system exceeds a certain value, the sound system starts oscillating and one or several tones can be heard from the loudspeaker.
  • the inventive device can be adjusted to suppress the feedback sound from the loudspeaker and thus even permit a higher amplification to the sound system without a risk of feedback.
  • a procedure respectively a filter device according to the invention can for example be used for the following alternatives:
  • Figure 1 shows a filter device according to the invention.
  • Figure 2 shows an arrangement for copying an original filter with the filter device according to the invention.
  • Figure 3 shows the filter device according to the invention supplemented with a loudspeaker-microphone unit.
  • Figure 4 shows a device for eliminating acoustic re-coupling (feedback and ringings) with the filter device according to the invention.
  • Figure 5 shows filter device according to Figure 1 and a digital unit.
  • Figure 6 shows another realization of the filter device according to Figure 1.
  • Figure 7 shows a feedback in a part-filter branch included in the inventive filter device.
  • Figure 8 shows another variant of feedback in a part-filter branch included in the inventive filter device.
  • Figure 9 illustrates a summation device in the form of a mixer for the filter device's output signal.
  • Fig. 1 shows a filter device according to the invention, at the input I of which the signal can be regarded to be given a certain sampling frequency.
  • a number of parallel branches 11-13 Connected to the input is a number of parallel branches 11-13, that also connect to a mixing or summation device 2.
  • Each branch 11-13 is shown to include a transversal part-filter 31-33 and a sampling converter 21-23 and 41-43 respectively is connected in each branch before and after part-filter 31-33 respectively.
  • the filter and its converters in each branch 11-13 are controlled by a sampling frequency converter 3 which receives the sampling frequency at the input and converts it to other frequencies for branch 11-13 respectively.
  • Converter 3 is preferably arranged to create sampling frequencies for each branch respectively that equals to a fractional number multi ⁇ plied by the sampling frequency of the input signal.
  • transversal filters 31-33 can be realized as a number of cascade- coupled transversal filters.
  • Branches 11-13 connect to summation device 2 which is shown to have an output U.
  • the summation device 2 can consist of a mixer, as Figure 9 schematically illustrates, where mixer 2' has several outputs Ul, U2.
  • mixer 2' has several outputs Ul, U2.
  • branches responsible for the so called early reflections can be mixed to an output Ul and the so called cluster of the reverberation is mixed to another output U2 to achieve a deeper spatial experience.
  • Signals from the various branches 11-13 are weighted with individual factors or numbers to the various outputs.
  • sampling converters in branch 11 can be omitted if transversal filter 31 is controlled by the sampling frequency at the input of the device and summation device 2 receives the same frequency.
  • the embodiment according to Figure 1 makes it possible to use other sampling frequen- cies for the input signal and output signal of the part-filters.
  • the entire impulse response from device 1 in Figure 1 is divided into a number of parts.
  • the first part of the impulse response is initiated in the first part-filter 31, the second part is initiated in the second part-filter 32, etc.
  • delay units are not compulsory, since the device can work correctly without them.
  • the purpose of delays is merely to reduce the number of taps at the filters in the lower branches of the device and thus reduce the cost of the device.
  • the delays can be placed at any point in respective branch. The delays can even become a part of the transversal filter or the sampling converters, for example by not utilizing a number of initial taps of the part-filters.
  • the number of taps (sample values of the impulse response) for achieving a rever ⁇ beration effect should be chosen as follows:
  • filter device 1 as per Figure 1 is shown connected to a registering device 20.
  • An original filter 21 that is to be copied has its input connected to an output U2 of registering device 20 and its output connected to an input 12 of registering device 20.
  • Registering device 20 emits a measuring signal, preferably a short pulse, giving a response at the output of filter 21 (at 12). If filter 21 is excited with an impulse emitted from registering device 20, then filter 21 leaves a so called impulse response at its output. In the most simple case the curve shape of the impulse response is converted to digital sample values that are used as the impulse response values of the inventive device. Sample no. 1 will then become impulse response value no. 1, sample no.2 will be value no.2 etc.
  • the registered sound can be processed in some way before it is used as the impulse response of the inventive device, such as being filtered, weighted with a window, time expanded or distorted. If filter 21 is excited with a measuring signal that is not equal to an impulse and if it is desirable that the inventive device emits a true image of the original filter 21, the response from filter 21 is first transformed to an impulse response before its sample values are used as the impulse response values of the inventive device.
  • filter 21 response can be done with sampling frequencies fl, £, 1/2 of the various part-filter branches in order to be immediately used as the impulse response of the various part-filters.
  • filter 21 response is resampled using for example a computer program for adjusting the response to the various filters' sampling frequencies.
  • filter device 1 as per Figure 1 is shown connected to a registering device 20. Furthermore, a unit with loudspeaker 30 and microphone 31 is added, with the loudspeaker input connected possibly via an power amplifier to output U2 of the registering device and microphone 31 output connected to input 12 of the registering device.
  • This device can be used if one, for example, desires to capture the acoustic pattern of a concert hall.
  • a short impulse from the registering device is emitted via loudspeaker 30 and is affected by the acoustics in the concert hall.
  • the pulse response is captured by microphone 31, then processed in the way described earlier and results in a number of digital sample value sets representing a number of part-impulse respon ⁇ ses belonging to each particular part-filter branch in filter device 1.
  • the first part-filter branch responding to the beginning of the impulse response, works with the highest sampling frequency and the following part-filter branches work with gradually lower sampling frequencies, thus fl >f2... >fn.
  • a sound effect may be desired that is soft in the beginning and the end but markedly hard in between.
  • the sampling frequency should be chosen according to pattern f 1 ⁇ £2 ⁇ f3 > f4 > f5 > f6.
  • Figure 4 shows a method for how the device according to the invention can be used for suppressing acoustic feedback in a sound system.
  • filter device 1 as per Figure 1 is used to simulate the feedback signal that is transmitted from loudspeaker 40 to microphone 41 and then subtract that artificial feedback signal (which is found at output Ul) from the real feedback signal coming from microphone 41 which is mixed with the actual sound to be amplified, for example speech or music.
  • the subtraction is done by means of inverter 45 and balance control 42. All that remains after this subtraction is just the speech or music.
  • Registering device 20 sends a measuring signal via switch 46 in position 1 to loud ⁇ speaker 40.
  • the impulse response for the whole acoustic system (loudspeaker- room-microphone) is fed from microphone 41 to input 12 of the registering device, after which the curve shape of the impulse response is converted, in the manner described earher, to the digital sample values for the impulse response which is used in filter device 1 where the characteristics of the acoustic system are reconstructed.
  • acoustic sound from both the sound source 47 and from loudspeaker 40 is converted to an electric signal in the micro ⁇ phone 41.
  • filter device 1 creates a feedback signal that is, unlike the signal from the microphone, free from the direct signal of the sound source and that is inverted in inverter 45 and then used in the balancing circuit to extinct the acoustic feedback of the sound system.
  • Inverter 45 can be omitted if the sign of the impulse response of filter device 1 is reversed.
  • a so called effect processor such as a chorus machine or an echo unit, can be connected to the patch point 43 if required.
  • Filter device 1 and registering device 20 can use common parts, such as the same A/D and D/A converters.
  • outputs Ul and U2 can physically be one single output.
  • the device as per Figure 5 with a filter device 1 as per Figure 1, has a digital unit 50, for example a mass storage, a computer, an interface or an A/D converter arranged to feed out data in the form of sample values for part-impulse responses in each particular part-filter branch of filter device 1 to this device when required. Conse ⁇ quently, the mentioned sample values are intended for successive control of the functions of filter device 1 as the outer signal is fed into the filter device via input terminal II.
  • a digital unit 50 for example a mass storage, a computer, an interface or an A/D converter arranged to feed out data in the form of sample values for part-impulse responses in each particular part-filter branch of filter device 1 to this device when required. Conse ⁇ quently, the mentioned sample values are intended for successive control of the functions of filter device 1 as the outer signal is fed into the filter device via input terminal II.
  • the filter device in some context it is desirable to alter the impulse response of the filter device as time goes on for the reason of a smooth changeover from one impulse response to another or to achieve a chorus effect, pitch shift, some sort of modulation or another time variant filter effect. That is achieved by digital unit 50 transferring to filter device 1 new values for the momentary pattern of the impulse response at any point of time.
  • the device as per Figure 6 shows another design'of the device as per Figure 1. It is commonly known that a transversal filter can be arranged in such a way that it can leave beside the filtered (convoluted) signal itself also a purely delayed signal. This delayed signal is used in the uppermost branch of transversal filter 61 as the input signal to the second branch.
  • the need for a separate delay unit (shown dotted in Figure 1) is elmiinated.
  • the delayed signal from the second branch transversal filter 62 is used as the input signal to the third branch.
  • the delay in the third branch is thus distributed over several units, namely transversal filters 61 and 62.
  • the summation of the output signals of the part-filter branches can be distributed, too. In Figure 6 this is done in two summation devices 64 and 65.
  • sampling frequency of the time discrete signal can be also done in several steps.
  • the third part-filter branch in Figure 6 uses sampling converters 21, 22 and 23 to achieve a signal with a suitable sampling frequency at the input to filter 63.
  • a particular option for feedback in the inventive device is illustrated, which is a feedback in the delay at the branch that is responsible for the last part of the impulse response of the device.
  • This feedback can be shaped in several various ways and when required, the signal in this branch may need to be delayed in several steps.
  • Figure 7 shows the lowermost branch 13 (between sampling converters 23 and 43) which contains an adder 71 before transversal part-filter 63.
  • a feedback 70 takes the signal from the delay output of transversal part-filter 63 and carries it to summation device 71 via a feedback unit 72 which, in the simplest case, can be an attenuator with gaina, 0 ⁇ a ⁇ l.
  • FIG 8 shows another realization of the above mentioned feedback, through the introduction of an IIR (Infinity Impulse Response) filter 84 which is between transversal filter 33 and sampling converter 43 in the branch that is responsible for the last part of the impulse response.
  • the UR filter 84 can in fact be inserted at any point in that branch.
  • the UR filter consists of an adder 81 and a feedback branch 80 containing an attenuator 82 and a delay 83.
  • Figure 9 shows mixing or summation device 2' from Figure 1, extended with a number of outputs, two in this case.
  • the signal from the various part-filter branches is weighted with attenuators al-a3 and bl-b3 respectively and then added in adders ⁇ to output signals Ul and U2 respectively.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Computer Hardware Design (AREA)
  • Mathematical Physics (AREA)
  • Reverberation, Karaoke And Other Acoustics (AREA)
  • Auxiliary Devices For Music (AREA)

Abstract

Procédé servant à filtrer un signal d'entrée représentant un son, de façon à apporter une modification souhaitée à la nature du son et comprenant l'introduction du signal d'entrée dans plusieurs branches de filtres parallèles, la convolution du signal dans les branches et le mélange des signaux de sortie des branches. Pendant le procédé, on utilise des signaux numériques temporels discrets. Les convolutions sont exécutées avec différentes fréquences d'échantillonnage des signaux se trouvant dans les différentes branches, les prises utilisées pour la convolution sont maintenues à une distance correspondante à une période d'échantillonnage les unes par rapport aux autres, et les convolutions dans toutes les branches sont exécutées avec 5000 prises au minimum. Un dispositif de filtrage servant à la mise en application du procédé comprend plusieurs filtres partiels de type transversal branchés en parallèle, ainsi qu'un dispositif mélangeur servant à mélanger les signaux de sortie des filtres partiels. Les prises utilisées des filtres partiels sont placées à une distance correspondant à une période d'échantillonnage les unes par rapport aux autres. Les filtres partiels dans leur totalité possèdent au moins 5000 prises et sont conçus pour fonctionner avec une fréquence d'échantillonnage différente.
PCT/SE1992/000521 1991-08-12 1992-07-17 Procede et dispositif de filtrage numerique WO1993004529A1 (fr)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
SE9102333-3 1991-08-12
SE9102333A SE9102333D0 (sv) 1991-08-12 1991-08-12 Foerfarande och anordning foer digital filtrering

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WO1993004529A1 true WO1993004529A1 (fr) 1993-03-04

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Cited By (15)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2001090927A1 (fr) * 2000-05-19 2001-11-29 Philipson Lars H G Procede et dispositif pour processus de convolution
WO2016198481A3 (fr) * 2015-06-09 2017-01-12 Cirrus Logic International Semiconductor Limited Filtre à réponse impulsionnelle finie hybride
US9666176B2 (en) 2013-09-13 2017-05-30 Cirrus Logic, Inc. Systems and methods for adaptive noise cancellation by adaptively shaping internal white noise to train a secondary path
US9704472B2 (en) 2013-12-10 2017-07-11 Cirrus Logic, Inc. Systems and methods for sharing secondary path information between audio channels in an adaptive noise cancellation system
US9711130B2 (en) 2011-06-03 2017-07-18 Cirrus Logic, Inc. Adaptive noise canceling architecture for a personal audio device
US9721556B2 (en) 2012-05-10 2017-08-01 Cirrus Logic, Inc. Downlink tone detection and adaptation of a secondary path response model in an adaptive noise canceling system
US9773490B2 (en) 2012-05-10 2017-09-26 Cirrus Logic, Inc. Source audio acoustic leakage detection and management in an adaptive noise canceling system
US9773493B1 (en) 2012-09-14 2017-09-26 Cirrus Logic, Inc. Power management of adaptive noise cancellation (ANC) in a personal audio device
US9807503B1 (en) 2014-09-03 2017-10-31 Cirrus Logic, Inc. Systems and methods for use of adaptive secondary path estimate to control equalization in an audio device
US9824677B2 (en) 2011-06-03 2017-11-21 Cirrus Logic, Inc. Bandlimiting anti-noise in personal audio devices having adaptive noise cancellation (ANC)
US9955250B2 (en) 2013-03-14 2018-04-24 Cirrus Logic, Inc. Low-latency multi-driver adaptive noise canceling (ANC) system for a personal audio device
US10013966B2 (en) 2016-03-15 2018-07-03 Cirrus Logic, Inc. Systems and methods for adaptive active noise cancellation for multiple-driver personal audio device
US10026388B2 (en) 2015-08-20 2018-07-17 Cirrus Logic, Inc. Feedback adaptive noise cancellation (ANC) controller and method having a feedback response partially provided by a fixed-response filter
US10219071B2 (en) 2013-12-10 2019-02-26 Cirrus Logic, Inc. Systems and methods for bandlimiting anti-noise in personal audio devices having adaptive noise cancellation
US10468048B2 (en) 2011-06-03 2019-11-05 Cirrus Logic, Inc. Mic covering detection in personal audio devices

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Publication number Priority date Publication date Assignee Title
GB2129638A (en) * 1982-09-20 1984-05-16 Nec Corp Segmented transversal filter
US4888808A (en) * 1987-03-23 1989-12-19 Matsushita Electric Industrial Co., Ltd. Digital equalizer apparatus enabling separate phase and amplitude characteristic modification
US5025472A (en) * 1987-05-27 1991-06-18 Yamaha Corporation Reverberation imparting device

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB2129638A (en) * 1982-09-20 1984-05-16 Nec Corp Segmented transversal filter
US4888808A (en) * 1987-03-23 1989-12-19 Matsushita Electric Industrial Co., Ltd. Digital equalizer apparatus enabling separate phase and amplitude characteristic modification
US5025472A (en) * 1987-05-27 1991-06-18 Yamaha Corporation Reverberation imparting device

Cited By (16)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2001090927A1 (fr) * 2000-05-19 2001-11-29 Philipson Lars H G Procede et dispositif pour processus de convolution
US10468048B2 (en) 2011-06-03 2019-11-05 Cirrus Logic, Inc. Mic covering detection in personal audio devices
US9711130B2 (en) 2011-06-03 2017-07-18 Cirrus Logic, Inc. Adaptive noise canceling architecture for a personal audio device
US9824677B2 (en) 2011-06-03 2017-11-21 Cirrus Logic, Inc. Bandlimiting anti-noise in personal audio devices having adaptive noise cancellation (ANC)
US10249284B2 (en) 2011-06-03 2019-04-02 Cirrus Logic, Inc. Bandlimiting anti-noise in personal audio devices having adaptive noise cancellation (ANC)
US9721556B2 (en) 2012-05-10 2017-08-01 Cirrus Logic, Inc. Downlink tone detection and adaptation of a secondary path response model in an adaptive noise canceling system
US9773490B2 (en) 2012-05-10 2017-09-26 Cirrus Logic, Inc. Source audio acoustic leakage detection and management in an adaptive noise canceling system
US9773493B1 (en) 2012-09-14 2017-09-26 Cirrus Logic, Inc. Power management of adaptive noise cancellation (ANC) in a personal audio device
US9955250B2 (en) 2013-03-14 2018-04-24 Cirrus Logic, Inc. Low-latency multi-driver adaptive noise canceling (ANC) system for a personal audio device
US9666176B2 (en) 2013-09-13 2017-05-30 Cirrus Logic, Inc. Systems and methods for adaptive noise cancellation by adaptively shaping internal white noise to train a secondary path
US10219071B2 (en) 2013-12-10 2019-02-26 Cirrus Logic, Inc. Systems and methods for bandlimiting anti-noise in personal audio devices having adaptive noise cancellation
US9704472B2 (en) 2013-12-10 2017-07-11 Cirrus Logic, Inc. Systems and methods for sharing secondary path information between audio channels in an adaptive noise cancellation system
US9807503B1 (en) 2014-09-03 2017-10-31 Cirrus Logic, Inc. Systems and methods for use of adaptive secondary path estimate to control equalization in an audio device
WO2016198481A3 (fr) * 2015-06-09 2017-01-12 Cirrus Logic International Semiconductor Limited Filtre à réponse impulsionnelle finie hybride
US10026388B2 (en) 2015-08-20 2018-07-17 Cirrus Logic, Inc. Feedback adaptive noise cancellation (ANC) controller and method having a feedback response partially provided by a fixed-response filter
US10013966B2 (en) 2016-03-15 2018-07-03 Cirrus Logic, Inc. Systems and methods for adaptive active noise cancellation for multiple-driver personal audio device

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