| Patent application number | Description | Published |
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| 20080285775 | SOUND TUNING METHOD - The invention relates to a method for automated tuning of a sound system, the sound system comprising delay lines, equalizing filters, and at least two loudspeakers, the method comprising the steps of reproducing a useful sound signal through the loudspeakers, measuring sound pressure values at least one location, providing a target transfer function for tuning the delay lines and the equalizing filters of the sound system, the target transfer function representing a desired transfer characteristics of the sound system, adjusting the delay of the delay lines, and adjusting amplitude responses of the equalizing filters such, that the actual transfer characteristics of the sound system approximates the target function. | 11-20-2008 |
| 20090034747 | AUDIO ENHANCEMENT SYSTEM AND METHOD - A system and method for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient sound in the listening environment, comprises producing an audio sound in the time domain from an electrical sound signal in the time domain. The electrical sound signal in the time domain is transformed into an electrical sound signal in the frequency domain and the electrical sound signal in the frequency domain is retransformed into an audio sound in the time domain. The total sound level in the environment is measured and a signal representative thereof is generated. The audio sound signal and the total sound signal are processed to extract a signal representing the ambient sound level within the environment, and equalization is performed in the frequency domain to adjust the output from the audio sound signal to compensate for the ambient noise level. | 02-05-2009 |
| 20090086990 | ACTIVE NOISE CONTROL USING BASS MANAGEMENT - An active noise cancellation system reduces, at a listening position, the power of a noise signal being radiated from a noise source to the listening position. The system includes an adaptive filter that receives a reference signal representing the noise signal, and provides a compensation signal. A bass management unit receives the compensation signal and applies a phase shift to the compensation signal to provide a phase shifted compensation signal. A first acoustic radiator receives the phase shifted compensation signal and radiates audio indicative thereof to the listening position. A second acoustic radiator receives the compensation signal and radiates audio indicative thereof to the listening position. The transfer function characteristics from the input of the bass management system to the listening position approximately matches a desired transfer function. | 04-02-2009 |
| 20090086995 | AUTOMATIC BASS MANAGEMENT - A method for an automatic equalization of sound pressure levels in at least one listening location, where the sound pressure is generated by a first and at least a second loudspeaker, comprising supplying an audio signal of a programmable frequency to each loudspeaker, where the audio signal supplied to the second loudspeaker is phase-shifted by a programmable phase shift relative to the audio signal supplied to the first loudspeaker, and where the phase shifts of the audio signals supplied to the other loudspeakers thereby are initially zero or constant; measuring the sound pressure level at each listening location for different phase shifts and for different frequencies; providing a cost function dependent on the sound pressure level; and searching a frequency dependent optimal phase shift that yields an extremum of the cost function, thus obtaining a phase function representing the optimal phase shift as a function of frequency. | 04-02-2009 |
| 20090214058 | MIXING SYSTEM - A system automatically mixes a first audio signal and a second audio signal. The system includes a correlator that determines whether the first signal and the second signal are correlated according to a predetermined correlation criterion. If the predetermined correlation criterion is fulfilled, the correlator determines whether the first and the second signal are delayed. A delay circuit compensates for the delay between the first signal and the second signal. A mixer mixes the first signal and the second signal that includes a compensation. | 08-27-2009 |
| 20090220098 | ADAPTIVE BASS MANAGEMENT - The invention relates to a method for adapting sound pressure levels in at least one listening location, the sound pressure being generated by a first and a second loudspeaker, each loudspeaker having a supply channel arranged upstream thereto, where at least the supply channel of the second loudspeaker modifies the phase of an audio signal transmitted therethrough according to a phase function. The method includes supplying an audio signal to the supply channels and thus generating an acoustic sound signal; measuring the acoustic sound signal at each listening location and providing corresponding electrical signals representing the measured acoustic sound signal; estimating updated transfer characteristics for each pair of loudspeaker and listening location; calculating an optimum offset phase function based on a mathematical model using the estimated transfer characteristics; updating the phase function by superposing the optimal offset phase function thereto. | 09-03-2009 |
| 20100146026 | SUB-BAND SIGNAL PROCESSING - An apparatus for sub-band processing of an input signal includes an analysis filter bank, signal processors and a synthesis filter bank. The analysis filter bank includes first and second signal branches for decomposing the input signal into two sub-band signals. The first signal branch includes a decimation filter connected upstream of a down-sampling unit and a basis filter. The second branch includes an all-pass filter and a subtractor that is connected downstream of the all-pass filter and the basis filter in the first signal branch via an up-sampling unit and a subsequent interpolation filter. At least one of the decimation filter and the interpolation filter is an infinite impulse response filter, and the all-pass filter has a phase response that compensates for a phase response of at least one of the decimation filter and the interpolation filter. | 06-10-2010 |
| 20100189275 | PASSENGER COMPARTMENT COMMUNICATION SYSTEM - A communication system for a passenger compartment includes at least two microphone arrays arranged within first and second regions, respectively, in the passenger compartment, and at least two loudspeakers and a signal processor connected to the microphone arrays and to the loudspeaker. Each microphone array has at least two microphones and provides an audio signal. Each loudspeaker is located within a different one of the first and the second regions. The signal processor processes the audio signal from the microphone array within the first region and provides the processed audio signal to the loudspeaker located within the second region. | 07-29-2010 |
| 20100195844 | ADAPTIVE NOISE CONTROL SYSTEM - An active noise cancellation system includes an adaptive filter, a signal source, an acoustic actuator, a microphone, a secondary path and an estimation unit. The adaptive filter receives a reference signal representing noise, and provides a compensation signal in response to the received reference signal. The signal source provides a measurement signal. The acoustic actuator radiates the compensation signal and the measurement signal to the listening position. The microphone receives a first signal that is a superposition of the radiated compensation signal, the radiated measurement signal, and the noise signal at the listening position, and provides a microphone signal in response to the received first signal. The secondary path includes a secondary path system that represents a signal transmission path between an output of the adaptive filter and an output of the microphone. The estimation unit estimates a transfer characteristic of the secondary path system in response to the measurement signal and the microphone signal. | 08-05-2010 |
| 20100215185 | ACOUSTIC ECHO CANCELLATION - An input signal is supplied to a loudspeaker-room-microphone system having a transfer function and that provides an output signal. An adaptive filter unit models the transfer function of the loudspeaker-room-microphone system and provides an approximated output signal, where the output signal and the approximated output signal are subtracted from each other to provide an error signal. The modeling of the transfer function of the loudspeaker-room-microphone system in the adaptive filter comprises transforming the input signal and the error signal from the time domain into the spectral domain; delaying of the input signal in the frequency domain to generate multiple differently delayed input signals in the frequency domain; adaptive filtering of each one of the multiple differently delayed input signals in the frequency domain according to the error signal in the spectral domain; summing up of the filtered differently delayed input signals in the frequency domain to generate the approximated output signal in the frequency domain; and transforming the approximated output signal from the spectral domain into the time domain. | 08-26-2010 |
| 20100226501 | BACKGROUND NOISE ESTIMATION - In a system for estimating the power spectral density of acoustical background noise when the level of a smoothed power spectral density signal increases, an increment value is increased, starting from a minimum increment value, by a predetermined amount until a maximum increment value is reached if at the same time the value of the power spectral density currently determined in a new calculation cycle is larger than the estimate value of the power spectral density of the background noise determined in the previous calculation cycle. For cases in which the level of the smoothed power spectral density decreases, the amplitude of the decrement value is increased, starting from a minimum decrement value, by a predetermined amount until a maximum decrement value is reached if at the same time the value of the power spectral density currently determined in a new calculation cycle is smaller than the estimate value of the power spectral density of the background noise determined in the previous calculation cycle. | 09-09-2010 |
| 20100239098 | BACKGROUND NOISE ESTIMATION - A system for estimating the background noise in a loudspeaker-room-microphone system is presented herein where the loudspeaker is supplied with a source signal and the microphone picks up the source signal distorted by the room and provides a distorted signal. The system comprises an adaptive filter receiving the source signal and the distorted signal, and providing an error signal, a post filter connected downstream of the adaptive filter and a smoothing filter arrangement connected downstream of the adaptive filter. The smoothing filter arrangement includes a spectral domain smoothing filter and that provides a spectral domain estimated-noise signal, and a time domain smoothing filter and that provides a time domain estimated-noise signal. A scaling factor calculation unit receives signals indicative of the spectral domain estimated noise signal and the time domain estimated noise signal provides a scaling factor to a scaling unit that applies the scaling factor to the spectral domain estimated-noise signal to provide an enhanced spectral domain estimated-noise signal. | 09-23-2010 |
| 20110103590 | AUDIO SYSTEM PHASE EQUALIZATION - A method is provided for optimizing acoustic localization at one or more listening positions in a listening environment such as, but not limited to, a vehicle passenger compartment. The method includes generating a sound field with a group of loudspeakers assigned to at least one of the listening positions, the group of loudspeakers including first and second loudspeakers, where each loudspeaker is connected to a respective audio channel; calculating filter coefficients for a phase equalization filter; configuring a phase response for the phase equalization filter such that binaural phase difference (Δφ | 05-05-2011 |
| 20110150241 | GROUP-DELAY BASED BASS MANAGEMENT - The listening room comprises at least one loudspeaker and at least one listening position. The method comprises providing for each loudspeaker, a group delay response to be equalized associated with one pre-defined position within the listening room; calculating filter coefficients for all-pass filter(s) each arranged upstream to one corresponding loudspeaker, the all-pass filter(s) having a transfer characteristic such that the corresponding group delay response(s) match(es) a predefined target group delay response. The filter coefficients have a group delay response being confined by a frequency dependent group delay constraint that defines a frequency dependent interval exponentially decaying with increasing frequency. | 06-23-2011 |
| 20110206214 | ACTIVE NOISE REDUCTION SYSTEM - A system for actively reducing noise at a listening point, includes an earphone housing, a transmitting transducer, a receiving transducer and a controller. The transmitting transducer converts a first electric signal into a first acoustic signal, and radiates the first acoustic signal along a first acoustic path having a first transfer characteristic and along a second acoustic path having a second transfer characteristic. The receiving transducer converts the first acoustic signal and ambient noise into a second electrical signal. The controller compensates for the ambient noise by providing a noise reducing electrical signal to the transmitting transducer. The noise reducing electrical signal is derived from a filtered electrical signal that is provided by filtering the second electrical signal with a third transfer characteristic. The second and the third transfer characteristics together model the first transfer characteristic. | 08-25-2011 |
| 20110216807 | SUB-BAND ADAPTIVE FIR-FILTERING - A method for designing a set of sub-band FIR filters, where each FIR filter has a number of filter coefficients and is connected to an adjustable delay line. The method includes dividing an input signal into a number of sub-band signals, where a spectrum of the input signal comprises spectra of the sub-band signals; providing a respective goal sub-band signal for each sub-band dependent on a goal signal; filtering and delaying each sub-band signal using a corresponding FIR filter and delay line to provide filtered signals; providing error signals for each sub-band dependent on the filtered signals and the corresponding goal signals; adapting the filter coefficients of each sub-band FIR filter such that the respective filtered signal approximately matches a corresponding goal sub-band signal; and changing a respective delay of the delay line for each sub-band to reduce or increase a first quality criterion. | 09-08-2011 |
| 20120154037 | AMPLIFIER CURRENT CONSUMPTION CONTROL - The audio amplifier includes a variable gain amplifier receiving the input audio signal and providing the output signal, whereby the output signal corresponds to the input signal amplified by a limiter gain. The audio amplifier further includes a limiter gain calculation unit, thus the input signal is amplified by the limiter gain. A control unit receives a signal representative of the input signal and is configured to estimate, based on a mathematical model, the input current or the total output current of the audio amplifier thus providing an estimated current signal corresponding to (and resulting from) the output signal, whereby the limiter gain calculation unit is configured to calculate, dependent on the estimation, the limiter gain such that the actual input current or the total output current of the audio amplifier does not exceed a threshold current value. | 06-21-2012 |
| 20120177221 | AUDIO ENHANCEMENT SYSTEM - A system for enhancing the sound signal produced by an audio system in a listening environment by compensating for ambient noise in the listening environment is provided. The system receives an electrical sound signal and generates a sound output therefrom. A total sound signal is sensed representative of the total sound level in the environment, where the total sound level includes both the sound output from the audio system and the ambient noise within the environment. The system extracts an ambient noise signal representative of the ambient noise in the environment from the total sound signal in response to the total sound signal and to a reference signal derived from the electrical sound signal. The system extracts the ambient noise signal using an adaptive filter with an adaptive step size. The system generates a control signal in response to the ambient noise signal and adjusts the sound output of the audio system to compensate for the ambient noise level in response to the control signal. The system calculates a step size for controlling the adaptive step size of the adaptive filter. | 07-12-2012 |
| 20120183150 | SOUND TUNING METHOD - The invention relates to a method for automated tuning of a sound system, the sound system comprising delay lines, equalizing filters, and at least two loudspeakers, the method comprising the steps of reproducing a useful sound signal through the loudspeakers, measuring sound pressure values at least one location, providing a target transfer function for tuning the delay lines and the equalizing filters of the sound system, the target transfer function representing a desired transfer characteristics of the sound system, adjusting the delay of the delay lines, and adjusting amplitude responses of the equalizing filters such, that the actual transfer characteristics of the sound system approximates the target function. | 07-19-2012 |
| 20120308029 | ADAPTIVE FILTERING SYSTEM - An audio system with at least one audio channel may include a digital audio processor in which at least one digital filter is implemented for each channel. The digital filter of each channel may include: an analysis filter bank configured to receive a broad-band input audio signal and divide the input audio signal into a plurality of sub-bands, a sub-band filter for each sub-band. and a synthesis filter bank configured to receive the filtered sub-band signals and combine them for providing a broad-band output audio signal. A delay is associated with each sub-band signal, the delay of one of the sub-band signals being applied to the broad-band input audio signal upstream of the analysis filter bank and the residual delays being applied to the remaining sub-band signals downstream of the analysis filter bank. | 12-06-2012 |
| 20120308036 | SPEED DEPENDENT EQUALIZING CONTROL SYSTEM - A speed dependent equalizing control system for automated design of gain and equalization filter parameters can be used for volume and velocity dependent equalization of audio signals reproduced in a vehicle. The system is configured to develop volume-dependent power spectral density estimations based on a test signal received at a number of different volume levels, and develop non-acoustical parameter-dependent power spectral density estimations based on received noise received at a number of different non-acoustical measurement values representing different states of the vehicle. In one example, the non-acoustical measurement values are different velocities, or speeds, of the vehicle. The system may generate filter parameters of a parameterized equalization filter based on a target equalization curve developed by summation of the volume-dependent power spectral density estimates and the non-acoustical measurement-dependent power spectral density estimates. | 12-06-2012 |
| 20130028435 | NOISE REDUCING SOUND-REPRODUCTION - An active noise reduction system includes an earphone with a cup-like housing, and a transmitting transducer, which converts electrical signals into acoustical signals and is arranged at an aperture of the housing. A receiving transducer converts acoustical signals into electrical signals, and is arranged proximate the transmitting transducer. A duct includes an end acoustically coupled to the receiving transducer, another end located proximate the transmitting transducer. An acoustical path extends from the transmitting transducer to a listener's ear, and has a first transfer characteristic. Another acoustical path extends from the transmitting transducer through the duct to the receiving transducer, and has a second transfer characteristic. A control unit generates a noise reducing electrical signal that is supplied to the transmitting transducer. This signal is derived from the receiving-transducer signal and filtered with a third transfer characteristic. The second and third transfer characteristics together model the first transfer characteristic. | 01-31-2013 |
| 20130028436 | NOISE REDUCING SOUND REPRODUCTION SYSTEM - A noise reducing sound reproduction system and method may be operable with an input signal supplied to a loudspeaker by which it is acoustically radiated. The signal radiated by the loudspeaker may be received by a microphone that is acoustically coupled to the loudspeaker via a secondary path and that provides a microphone output signal. From the microphone output signal a useful-signal can be subtracted to generate a filter input signal. The filter input signal is filtered in an active noise reduction filter to generate an error signal, and the useful-signal is subtracted from the error signal to generate the loudspeaker input signal. In addition, the useful-signal is filtered by one or more spectrum shaping filters prior to subtraction from the microphone output signal or the loudspeaker input signal or both. | 01-31-2013 |
| 20130101129 | ACTIVE NOISE REDUCTION - A noise reducing sound reproduction system comprises a loudspeaker that is connected to a loudspeaker input path and that radiates noise reducing sound. A microphone is connected to a microphone output path and picks up the noise or a residual thereof An active noise reduction filter is connected between the microphone output path and the loudspeaker input path, and the active noise reduction filter comprises at least one shelving filter. | 04-25-2013 |
| 20130195287 | DIGITAL EQUALIZING FILTERS WITH FIXED PHASE RESPONSE - An equalization filter structure for filtering an audio signal within an audio system is disclosed. The equalization filter comprises a first and a second shelving filter each having a fixed first and a fixed second phase response, each of which is determined by a respective cut-off frequency and Q factor which represent the transfer characteristic of the corresponding shelving filter. The first and the second shelving filters are coupled in series and each shelving filter comprises at least one fourth order low-pass filter having a cut-off frequency, a Q factor and a first broadband gain and further at least one fourth order high-pass filter having a second broadband gain and the same cut-off frequency and the same Q factor as the low-pass filter. The fourth order low-pass filter and the fourth order high-pass filter are connected in parallel, such that both filters receive the same input signal and the corresponding filtered signals are summed to form a respective shelving filter output signal. Each fourth order low-pass and high-pass filter is composed of a cascade of two second order low-pass or high-pass filters, respectively, and each second order filter has the same cut-off frequency and Q factor as the corresponding shelving filter. | 08-01-2013 |
| 20130216049 | LOUDSPEAKER OVERLOAD PROTECTION - A loudspeaker overload protection circuit and method receives at a compressor a signal representing the estimated loudspeaker power consumption; receives at the compressor a signal representing the nominal power of the loudspeaker; receives at the compressor an input audio signal from the signal source and supplying with the compressor an output audio signal to the loudspeaker; estimates from the output audio signal, (a) signal(s) that represent(s) the voltage and/or current supplied to the loudspeaker and a parameter that represents the ohmic resistance of the loudspeaker the power consumed by the loudspeaker; supplies a signal representing the estimated loudspeaker power consumption to the compressor; and attenuates the input audio signal when the signal representing the estimated loudspeaker power consumption exceeds the signal representing the nominal power of the loudspeaker. | 08-22-2013 |
| 20130308785 | ACTIVE NOISE REDUCTION - A noise reducing comprises a first microphone that picks up noise signal at a first location and that is electrically coupled to a first microphone output path; a loudspeaker that is electrically coupled to a loudspeaker input path and that radiates noise reducing sound at a second location; a second microphone that picks up residual noise from the noise and the noise reducing sound at a third location and that is electrically coupled to a second microphone output path; a first active noise reducing filter that is connected between the first microphone output path and the loudspeaker input path; and a second active noise reducing filter that is connected between the second microphone output path and the loudspeaker input path; in which the first active noise reduction filter is a shelving or equalization filter or comprises at least one shelving or equalization filter or both. | 11-21-2013 |
| 20140016792 | ENGINE SOUND SYNTHESIS SYSTEM - An engine sound synthesis system is operable to analyze sound. Operation of the system may include providing an input sound signal to be analysed and determining a fundamental frequency of the input signal from the input signal or from at least one guide signal. Furthermore, the frequencies of higher harmonics of the fundamental frequency are determined, thus determining harmonic model parameters. A harmonic signal based on the harmonic model parameters is synthesized and a residual signal is estimated by subtracting the harmonic signal from the input signal. Residual model parameters are estimated based on the residual signal. Furthermore, a corresponding method for synthesizing a sound signal is described. | 01-16-2014 |
| 20140037108 | AUTOMATIC LOUDNESS CONTROL - An improved automatic loudness control system and method comprise controlling gain/attenuation applied to an input audio signal and providing an output audio signal that is the amplified/attenuated input audio signal; evaluating an actual loudness of the input audio signal from the input audio signal and a desired loudness of the input audio signal from a volume control input; and evaluating the gain/attenuation applied to the input audio signal from the actual loudness and the desired loudness of the input audio signal. | 02-06-2014 |
| 20140177867 | SOUND CAPTURE SYSTEM - A sound capture system is disclosed that includes an open-sphere microphone array where at least four omnidirectional microphones providing at least four output signals are disposed around a point of symmetry and an evaluation circuit that is connected to the at least four microphones disposed around the point of symmetry and that is configured to superimpose the output signal of each of the at least four microphones disposed around the point of symmetry with the output signal of one of the other microphones to form at least four differential microphone constellations providing at least four output signals, each differential microphone constellation having an axis along which it exhibits maximum sensitivity. | 06-26-2014 |
| 20140348329 | SOUND SYSTEM FOR ESTABLISHING A SOUND ZONE - A system and method for acoustically reproducing k electrical audio signals (where k=2, 3, 4, . . . ) and establishing k sound zones are provided, in each of which one of k reception sound signals occurs that is an individual pattern of the reproduced and transmitted k electrical audio signals, comprising processing the k electrical audio signals to provide k processed electrical audio signals and converting the k processed electrical audio signals into corresponding k acoustic audio signals with k loudspeakers that are arranged at positions separate from each other and within or adjacent to the k sound zones. Each of the k acoustic audio signals is transferred according to a transfer matrix from each of the k loudspeakers to each of the k sound zones, where they contribute to the corresponding reception sound signals. Processing of the k electrical audio signals comprises inverse filtering according to three filter matrices, one of which is an i×i filter matrix, one is a j×j filter matrix and one is a k×k filter matrix, in which i, j| 11-27-2014 | |
| 20140348353 | SOUND SYSTEM FOR ESTABLISHING A SOUND ZONE - A system and method for acoustically reproducing at least two electrical audio signals and establishing at least two sound zones that are represented by individual patterns of reception sound signals includes processing the at least two electrical audio signals to provide processed electrical audio signals; converting the processed electrical audio signals into corresponding acoustic audio signals with at least two loudspeakers that are arranged at positions separate from each other; transferring each of the acoustic audio signals according to a transfer matrix from each of the loudspeakers to each of the sound zones where they contribute to the reception sound signals; and processing of the at least two electrical audio signals comprises inverse filtering according to a filter matrix. Inverse filtering is configured to compensate for the room transfer matrix so that each one of the reception sound signals corresponds to one of the electrical audio signals. | 11-27-2014 |
| 20140348354 | GENERATION OF INDIVIDUAL SOUND ZONES WITHIN A LISTENING ROOM - A sound reproduction system that is capable of providing at least two separate sound zones in one coherent listening room. In each sound zone, resulting acoustic signals substantially corresponds to a respective audio source signal associated with the same sound zone, and the contribution of audio source signals associated with a different sound zone to the resulting sound signal is minimized. | 11-27-2014 |
| 20140362998 | DIRECTIONAL CODING CONVERSION - A directional coding conversion method and system includes receiving input audio signals that comprise directional audio coded signals into which directional audio information is encoded according to a first loudspeaker setup and extracting the directional audio coded signals from the received input audio signals. The method and system further includes decoding, according to the first loudspeaker setup, the extracted directional audio coded signals to provide at least one absolute audio signal and corresponding absolute directional information and processing the at least one absolute audio signal and the absolute directional information to provide first output audio signals coded according to a second loudspeaker setup. | 12-11-2014 |
| 20140363027 | AUDIO SIGNAL MIXING - A system and a method for mixing at least two audio signals are provided that comprise transferring the audio signals with respective transfer functions, the audio signals each having an amplitude and a phase; adding the audio signals to provide an output signal representative of the mixed audio signals, the output signal having an amplitude and a phase; controlling at least one of the transfer functions of the signal lines so that the phase of the output signal is adapted to the phase of the audio signal with a higher signal strength than the other audio signal(s), the signal strengths corresponding to the amplitudes of the audio signals. | 12-11-2014 |
