Entries |
Document | Title | Date |
20080205667 | Room acoustic response modeling and equalization with linear predictive coding and parametric filters - A method for determining coefficients of a family of cascaded second order Infinite Impulse Response (IIR) parametric filters used for equalizing a room response. The method includes determining parameters of each IIR parametric filter from poles or roots of a reasonably high-order Linear Predictive Coding (LPC) model. The LPC model is able to accurately model the low-frequency room response modes providing better equalization of loudspeaker and room acoustics, particularly at the low frequencies. Advantages of the method include fast and efficient computation of the LPC model using a Levinson-Durbin recursion to solve the normal equations that arise from the least squares formulation. Due to possible band interactions between the cascaded IIR parametric filters, the method further includes optimizing the Q value of each filter to better equalize the room response. | 08-28-2008 |
20080212798 | System and Method for Intelligent Equalization - An equalizer/mixer receives an input signal from a musical source and equalizes the input signal based on the musical source using equalization parameters associated with the musical source. User-adjustable equalization controls may be applied where the equalization parameters defining the controls are associated with the musical source. | 09-04-2008 |
20080240467 | FREQUENCY-WARPED AUDIO EQUALIZER - In certain embodiments, an improved audio equalization filter can be generated by frequency warping one or more digital filters having a plurality of frequency bands. Frequency warping can include, for example, transforming at least some of the frequency bands of the one or more digital filters into lower frequency bands. As a result, in various implementations the audio equalization filter may be more accurate than certain currently-available IIR equalization filters. The audio equalization filter may also be more computing-resource efficient than certain currently-available FIR equalization filters. | 10-02-2008 |
20080260179 | Active loudspeaker - In an active loudspeaker, a bridge rectifier generates an unregulated DC voltage from the AC mains power supply voltage. A self-oscillating pulse modulator generates a pulse-modulated switch signal that is controlling a power switch stage which is switching the unregulated DC voltage. The power-switched signal is low-pass filtered and driving a high-impedance speaker element. The amplifier is compensating fluctuations and noise in the unregulated DC voltage by feeding a reference signal back to the pulse modulator which is derived by voltage-dividing the power-switched signal. The audio volume is regulated by adjusting the voltage divider ratio of the feedback signal. The loudspeaker provides galvanic voltage isolation of the speaker inputs through an audio input stage to comply with safety regulations. This invention eliminates the regulated switched-mode power supply in active loudspeaker applications and thereby reduces cost and weight. | 10-23-2008 |
20080292114 | Audio reproducing apparatus - An audio signal processing apparatus includes a harmonic overtone adder and an equalizer. The harmonic overtone adder includes a high-pass filter for extracting from an audio signal higher than a first predetermined frequency, a filter for extracting a frequency component lower than half a second predetermined frequency, an harmonic overtone generator for generating a frequency-doubled harmonic overtone component from an output from the filter, and a first combining unit for combining the frequency component output from the high-pass filter and the harmonic overtone component output from the harmonic overtone generator. The equalizer includes a level detector for detecting a level of an overtone component contained in an output from the first combining unit, a gain controller for controlling dynamically the level of the harmonic overtone component contained in the output from the first combining unit, and a second combining unit for combining the output from the first combining unit with the harmonic overtone component. | 11-27-2008 |
20090074206 | METHOD OF ENHANCING SOUND FOR HEARING IMPAIRED INDIVIDUALS - A portable assistive listening system for enhancing sound for hearing impaired individuals includes a fully functional hearing aid and a separate handheld digital signal processing (DSP) device. The focus of the present invention is directed to the handheld DSP device and a unique method of processing incoming audio signals. The DSP device includes a programmable digital signal processor, a UWB transceiver for communicating with the hearing aid and/or other wireless audio sources, an LCD display, and a user input device (keypad). The handheld device is user programmable to apply different sound enhancement algorithms for enhancing sound signals received from the hearing aid and/or other audio source. The handheld device is capable of receiving audio signals from multiple sources, and gives the user control over selection of incoming sound sources and selective enhancement of sound. Specifically, the invention focuses on a proprietary method of adjusting the master volume and equalization of each of the multiple incoming audio signals prior to mixing all of the enhanced signals for output. Applying a Master Volume and Equalization setting prior to mixing provides for a more evenly enhanced sound and better overall sound intelligibility, as well as reducing processing power drained from the system during use. | 03-19-2009 |
20090074207 | Mobile communication device capable of setting tone color and method of setting tone color - A mobile communication device and a method of setting tone color, which allow a user to set the tone color of received sound. Provided are a normal mode, which sets the equalizer using GCF standards stored in an internal memory or equalizer setting values selected by a provider, a country-specific mode, which uses country-specific setting, and a user mode, in which a user can set frequency-specific gains of the received sound, and one mode is selected from the provided mode, so that the tone color of the received sound can be adjusted according to the selection. Telephone speech quality can be optimized for user preference, network environments and language characteristics. | 03-19-2009 |
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 |
20090220108 | PROCESSING OF AN AUDIO SIGNAL FOR PRESENTATION IN A HIGH NOISE ENVIRONMENT - In accordance with the invention, audio signals are specially processed for sound presentation in a high noise environment. The electrical signal representative of the sound is first subjected to equalization to preferentially reduce the magnitude of bass signals while increasing the magnitude of treble signals. The equalized signal is then compressed, and the compressed signal is subjected to “mirror image” equalization which increases the magnitude of bass signals while reducing the magnitude of treble signals. The resulting signal fed to the speakers provides a sound presentation of compressed volume range and a bass-rich sound spectrum. It is particularly useful for providing quality sound presentation in a high noise environment. | 09-03-2009 |
20090232329 | EQUALIZATION METHOD USING EQUAL LOUDNESS CURVE, AND SOUND OUTPUT APPARATUS USING THE SAME - An equalization method using equal loudness curve, and a sound output apparatus using the same are disclosed. An audio signal is equalized using a standard loudness curve so as to increase a loudness of an audio signal. A sound quality can be significantly enhanced with the enhancement of a sound cleanness and quality when being adapted to a communication apparatus such as a telephone terminal, a transmitter or a repeater. When it is adapted to an audio apparatus such as a MP3, a common audio system, a car audio system, etc., it is possible to enhance a sound cleanness and quality, and a hearing ability. It can be used as a hearing aid. | 09-17-2009 |
20090290725 | AUTOMATIC EQUALIZER ADJUSTMENT SETTING FOR PLAYBACK OF MEDIA ASSETS - Systems and methods are disclosed in which correspondences with content or other media assets can be established such that a media player or portable media device can automatically modify or adjust an equalizer setting based on information associated with the content or other media assets. The media player may automatically adjust one or more equalizer settings based on genre, artist, album, or the like. In some embodiments, metadata associated with content or other media assets can be analyzed to determine normalized data thereby potentially grouping content into supersets. Based on the normalized data, the media player may automatically adjust equalizer settings for each superset or grouping of content. Correspondences with one or more accessories may be established such that the media player can automatically modify or adjust an equalizer setting based on the one or more accessories. | 11-26-2009 |
20090296959 | MISMATCHED SPEAKER SYSTEMS AND METHODS - According to various embodiments, an audio system having high quality sound and a high frequency response is provided. The audio system comprises one or more speakers with mismatched components. Although the combination of mismatched components may result in a diminished frequency response, digital signal processing may compensate for the physical deficiencies of a driver of the speaker. In some embodiments, an audio system comprises a speaker and a signal processor. The speaker may comprise one or more mismatched speaker components which are operably coupled to each other such that the components, together, have a low frequency response. The signal processor may operably couple to the one or more mismatched speaker components. The signal processor may be configured to process an input signal and to drive the speaker using the processed signal such that the speaker has a higher frequency response than the low frequency response. | 12-03-2009 |
20100046772 | SOUND LEVEL CONTROL - A system and method for controlling a sound level of a mobile audio device are disclosed herein. In accordance with at least some embodiments, a system includes a transducer, a phase estimator, and a sound level control. The transducer converts an electrical signal applied to the transducer into audible sound. The phase estimator estimates a phase difference between a voltage and a current of the electrical signal applied to the transducer. The sound level control controls the loudness of sound produced by the transducer based, at least in part on the estimated phase difference. | 02-25-2010 |
20100086149 | ACOUSTIC PROCESSING SYSTEM AND METHOD FOR ELECTRONIC APPARATUS AND MOBILE TELEPHONE TERMINAL - An acoustic processing system for an electronic device includes: an equalizer processor which boosts or cuts a digital signal which has been subjected to an acoustic process for each of frequency components; a low-pass filter which extracts a low-range signal from the signal from the equalizer processor and outputs the signal; a high-pass filter which extracts a high-range signal from the signal from the equalizer processor and outputs the signal; a high-frequency compressor which compresses the high-range signal; an adder which generates an addition of the low-range signal and the compressed high-range signal; and a speaker which outputs a sound in accordance with the addition of the signal. | 04-08-2010 |
20100104114 | TIMBRAL CORRECTION OF AUDIO REPRODUCTION SYSTEMS BASED ON MEASURED DECAY TIME OR REVERBERATION TIME - The invention relates to a method and system for use in directly adjusting the timbre of a reproduced audio signal in any closed or partially enclosed space according to the measured reverberation time or other function describing the decay of sound within the space. The measurement of the reverberation time and the correction of the timbre are performed by a system that can be incorporated within the installed audio reproduction system, although a separate measuring system could alternatively be used. The measurement of decay time or reverberation time for the space is by known methods. The invention centres around the calculation and application of a correction filter determined directly from the measured decay time or reverberation time for the space. | 04-29-2010 |
20100119082 | Pitch Detection Apparatus and Method - Band-pass filter suppresses frequency components of a sound signal that are lower than a low-side cutoff frequency and that are higher than a high-side cutoff frequency. Pitch detection section detects a pitch of the sound signal having been processed by the band-pass filter. Target setting section variably sets a low-side target value lower than the pitch detected by the pitch detection section and a high-side target value higher than the detected pitch. Filter control section causes the low-side cutoff frequency to approach the low-side target value over time and causes the high-side cutoff frequency to approach the high-side target value over time. In this way, a pass band of the band-pass filter can be smoothly variably controlled in accordance with pitch change of the sound signal that is an object of pitch detection. | 05-13-2010 |
20100128902 | COMBINATION EQUALIZER AND CALIBRATOR CIRCUIT ASSEMBLY FOR AUDIO SYSTEM - A combination equalizer and calibrator circuit assembly includes a calibrator formed of a frequency control circuit, a signal generator, an amplifier circuit connected with a microphone, a compare circuit and a display circuit and connected to the front end of an equalizer set. Calibration is made through the calibrator prior to audio system reproduction, wherein the frequency control circuit provides a reference frequency and different test frequencies for causing the signal generator to generate respective sound signals for output through a speaker; the microphone picks up these sound signals for comparison with respective reference values by the compare circuit; and the display circuit displays the respective comparison results. By means of adjustment through respective adjust circuits, signals of test frequencies are standardized so that the music played through the audio system is well calibrated without sound spectrum distortion due to space or audio system discrepancy. | 05-27-2010 |
20100158274 | Increased dynamic range microphone - Disclosed herein are apparatus, method, and computer program product whereby a device receives an acoustic signal, where the acoustic signal has a variable sound pressure level. A device outputs an electrical signal from an input audio transducer, where the input audio transducer is connected to a supply voltage. The device determines a distortion level of the electrical signal; and increases or decreases the supply voltage based on the distortion level. | 06-24-2010 |
20100166221 | Sound setting apparatus and sound setting method - A sound setting apparatus includes: a manipulating unit capable of making an input manipulation in upward, downward, left, right, and rotation directions; a band setting unit that sets each of individual frequency bands corresponding to low, midrange, and high frequencies in controlling an equalizer, in response to an input manipulation to the manipulating unit in the left and right directions; a level setting unit that sets an output level of each of the bands set by the band setting unit, in response to an input manipulation in the upward and downward directions; a center frequency setting unit that sets a center frequency of each of the bands set by the band setting unit, in response to an input manipulation in the rotation directions; and a display unit that collectively displays setting images to set the band, the output level, and the center frequency on a screen on the same layer. | 07-01-2010 |
20100166222 | SYSTEM AND METHOD FOR DIGITAL SIGNAL PROCESSING - The present invention provides for methods and systems for digitally processing an audio signal. In various embodiments, a method comprises receiving a profile comprising a plurality of filter equalizing coefficients, configuring a plurality of filters of a graphic equalizer with the plurality of filter equalizing coefficients from the profile, receiving a first signal for processing, adjusting the plurality of filters using a first gain, equalizing the first signal using the plurality of filters of the graphic equalizer, outputting the first signal, receiving a second signal for processing, adjusting the plurality of filters, previously configured with the filter equalizing coefficients from the profile, using a second gain, equalizing the second plurality of frequencies of the second signal with the plurality of filters of the graphic equalizer, and outputting the second signal. | 07-01-2010 |
20100202630 | METHOD AND SYSTEM FOR APPROXIMATING GRAPHIC EQUALIZERS USING DYNAMIC FILTER ORDER REDUCTION - Improved approaches to flexibly implementing graphic equalizers on media players are disclosed. These approaches provide dynamic order reduction of a multi-band graphic equalizer so that equalizer effects can be timely performed with only limited computational resources. In one embodiment, a media player receives a media item and associated equalizer settings for a multi-band graphic equalizer. The media player can then automatically (i.e., without user action) approximate the multi-band graphic equalizer with the equalizer settings for the media item using a fewer number of filters. Fewer filters means order reduction, and thus reduction in computational requirements. After the multi-band graphic equalizer is approximated, the media player can present the media item to its user in accordance with the reduced complexity, approximated equalizer. | 08-12-2010 |
20100215193 | Speaker Distortion Deduction System and Method - Many speakers, especially small speakers are susceptible to distortion if too much power is applied in certain vulnerable frequency bands. The distortion can be prevented by applying equalization to the audio signal driving the speaker. An equalizer can be configured to suppress the audio signal in the vulnerable frequency bands. The equalizer monitors the power in the vulnerable frequency bands and suppresses those vulnerable frequency bands only when they have sufficient power to distort. In this fashion, undesired audio effects due to the equalization can be avoided. | 08-26-2010 |
20100232624 | Method and System for Virtual Bass Enhancement - Techniques for enhancing bass effects in an audio signal are described. According to one embodiment, an audio input signal is filtered to produce a low frequency component thereof (a low frequency signal of the audio input signal). The low frequency signal expressed in time domain is transformed to a corresponding spectrum expression in frequency domain. A fundamental frequency signal of the low frequency signal in the frequency domain is determined to generate a plurality of harmonics that are then transformed back to the time domain. Both the audio input signal (delayed) and the harmonics are synthesized to produce an audio output signal whose bass is greatly enhanced. | 09-16-2010 |
20100266143 | FREQUENCY-WARPED AUDIO EQUALIZER - In certain embodiments, an improved audio equalization filter can be generated by frequency warping one or more digital filters having a plurality of frequency bands. Frequency warping can include, for example, transforming at least some of the frequency bands of the one or more digital filters into lower frequency bands. As a result, in various implementations the audio equalization filter may be more accurate than certain currently-available IIR equalization filters. The audio equalization filter may also be more computing-resource efficient than certain currently-available FIR equalization filters. | 10-21-2010 |
20110038490 | SYSTEM FOR INCREASING PERCEIVED LOUDNESS OF SPEAKERS - A system can be provided for increasing loudness of an audio signal to present a perceived loudness to a listener that is greater than a loudness provided natively by a loudspeaker. The system can include one or more of the following: a frequency suppressor, a loudness adjuster, an equalizer, and a distortion control module. The frequency suppressor can increase headroom in the audio signal by filtering out low and/or high frequencies. The loudness adjuster can calculate a loudness of the audio signal and apply a gain to the audio signal to increase the loudness. The equalizer can further increase headroom by attenuating portions of a passband of the loudspeaker's frequency response. The distortion control module can induce partial harmonic distortion in the audio signal to further increase loudness. | 02-17-2011 |
20110091051 | PORTABLE COMPUTER ELECTRICAL GROUNDING AND AUDIO SYSTEM ARCHITECTURES - A portable computing device having a substantially non-conducting outer housing and alternative electrical grounding and audio system architectures is disclosed. The device can be a laptop computer having a main logic board, a keyboard assembly, an audio source positioned below the keyboard assembly, and an equalizer electrically coupled to the audio source, with each of these components being electrically coupled to a universal grounding structure. The audio source emits sound waves that are propagated through the keyboard assembly and between gaps between keyboard keys and the outer housing. Settings for the equalizer can be selected to account for sound absorption and amplification characteristics of the sound waves along these sound transmission paths. The universal grounding structure includes a plurality of separate ground components that are electrically intercoupled, each being substantially smaller than the overall portable computing device, and also includes an electromagnetic interference shield around the main logic board. | 04-21-2011 |
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 |
20110268293 | System and Method for Processing Signals to Enhance Audibility in an MRI Environment - A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the MRI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor. | 11-03-2011 |
20110317853 | HEYSER SPIRAL LOW FREQUENCY CORRECTION OF FIR FILTERS - A method of operating a loudspeaker includes providing a digital audio signal and identifying a target transfer function to be applied to the signal. At least one coefficient of an FIR filter is generated. The generating includes performing Heyser spiral curve fitting, and fitting a three-dimensional curve based on a magnitude and phase of a target transfer function. The digital audio signal is filtered through the FIR filter. The filtered signal is inputted into the loudspeaker. | 12-29-2011 |
20120039485 | HIGH FIDELITY PHONOGRAPHIC PREAMPLIFIER FEATURING SIMULTANEOUS FLAT AND PLAYBACK COMPENSATION CURVE CORRECTION OUTPUTS - An audio playback system and method including an analog preamplifier configured to produce simultaneous “flat” and response-compensated outputs, to enable playback of a phonograph recording via an all-analog signal pathway, while optionally providing for the simultaneous digital recording and/or monitoring of the “flat,” uncompensated signal. | 02-16-2012 |
20120057724 | AUTOMATIC AUDIO SYSTEM EQUALIZING - An automated process for equalizing an audio system and an apparatus for implementing the process. An audio system includes a microphone unit, for receiving the sound waves radiated from a plurality of speakers, acoustic measuring circuitry, for calculating frequency response measurements; a memory, for storing characteristic data of the loudspeaker units and further for storing the frequency response measurements; and equalization calculation circuitry, for calculating an equalization pattern responsive to the digital data and responsive to the characteristic data of the plurality of loudspeaker units. Also described is an automated equalizing system including a acoustic measuring circuitry including a microphone for measuring frequency response at a plurality of locations; a memory, for storing the frequency responses at the plurality of locations; and equalization calculation circuitry, for calculating, from the frequency responses, an optimized equalization pattern. | 03-08-2012 |
20120063616 | DYNAMIC COMPENSATION OF AUDIO SIGNALS FOR IMPROVED PERCEIVED SPECTRAL IMBALANCES - An input audio signal is equalized to form an output audio signal on the basis of an intended listening sound pressure level, the output capabilities of a particular playback device, and unique hearing characteristics of a listener. An intended listening level is first determined based on the properties of the audio signal and a mastering sound level. The intended listening level is used to determine an optimal sound pressure level for the particular playback device based on its capabilities and any master volume gain. These two levels are used to determine how much louder to make individual frequencies based on data pertaining to human auditory perception, either standardized or directly measured. The audio is further compensated on the basis of hearing loss data, again either standardized or directly measured, after optionally extending the signal bandwidth. The final, compensated audio signal is sent to the playback device for playback. | 03-15-2012 |
20120140953 | MULTI-AMPLIFIER AUDIO MODULE FOR A MULTICHANNEL SPEAKER, AND MULTI-AMPLIFIER AUDIO MODULE HOUSING ASSEMBLY FOR SAME - The present invention relates to a multi-amplifier audio module for a multichannel speaker. The multi-amplifier audio module comprises a plurality of amplifier units connected to a power source and connected to each other to amplify an input audio signal. A plurality of speakers for outputting the amplified audio signal, an equalizer for adjusting the amplified audio signal for each frequency band, and a volume controller for controlling the volume of the amplified audio signal, are connected to each of the amplifier units. | 06-07-2012 |
20120148070 | PARAMETRIC SIGNAL PROCESSING SYSTEMS AND METHODS - A signal processing system for generating a parametric signal comprises an audio compressor, operable to compress a dynamic range of an audio input signal, and an equalization network, operable to equalize the audio signal. A low pass filter is operable to remove high portions of the audio signal and a high pass filter is operable to remove low portions of the audio signal. An oscillator circuit is operable to generate a carrier signal, and a modulation circuit is operable to combine the audio signal with the carrier signal to produce at least one modulated carrier signal. | 06-14-2012 |
20120177224 | SIGNAL PROCESSOR AND METHOD FOR COMPENSATING LOUDSPEAKER AGING PHENOMENA - A signal processor including an equalizer responsive to an input signal and to a parameter signal, said equalizer configured to provide an output signal to an electro-acoustical transducer for compensating a frequency response of said electro-acoustical transducer; a transducer element for monitoring at least a physical parameter of said electro-acoustical transducer, said transducer element configured to provide a transducer signal, a processor block responsive to said transducer signal, configured to provide said parameter signal. | 07-12-2012 |
20120207328 | DYNAMIC BASS EQUALIZATION WITH MODIFIED SALLEN-KEY HIGH PASS FILTER - A dynamic bass equalization circuit has an amplitude dependent gain that is dependent upon the audio electrical signal amplitude and a dynamically adjusted frequency response that varies with the amplitude dependent gain. In one implementation, the dynamic bass equalization circuit includes a Sallen-Key high pass filter that includes an amplifier with a negative feedback path. The dynamically adjusted frequency response is provided by a parallel pair of reversed diodes connected in the negative feedback path. | 08-16-2012 |
20130016857 | LOW FREQUENCY EQUALIZATION FOR LOUDSPEAKER SYSTEM - A method of optimizing the low frequency audio response emanating from a pair of low frequency transducers housed within a cabinet. The low frequency transducers are electrically connected to a power amplifier and source of audio content. The resonant frequency (Fs) and amplitude (Q) are characterized as to the high-pass pole of the low frequency transducers as they are mounted within the cabinet. An equalizer is placed between the amplifier and source of audio content for canceling the complex pole of the low frequency transducers and for establishing a new complex pole at a cut off frequency below which the sound generated by the low frequency transducers will diminish. | 01-17-2013 |
20130101137 | Adaptive Sound Field Control - The present invention relates to a method for controlling one or more loudspeakers provided in an enclosure, such as a listening room or an automobile cabin, the method comprising the steps of: (i) providing said one or more loudspeakers ( | 04-25-2013 |
20130108078 | METHOD AND DEVICE OF CHANNEL EQUALIZATION AND BEAM CONTROLLING FOR A DIGITAL SPEAKER ARRAY SYSTEM | 05-02-2013 |
20130148823 | SYSTEM AND METHOD FOR DIGITAL SIGNAL PROCESSING - The present invention provides methods and systems for digitally processing audio signals in broadcasting and/or transmission applications. In particular, the present invention includes a pre-transmission processing module which is structured and configured to generate a partially processed signal. A transmitter is then structured to transmit or broadcast the partially processed signal to a receiver, where the signal is then fed to a post-transmission processing module. The post-transmission processing module is structured and configured to further processes the signal based upon, for example, the listening environment, profile(s), etc. and generate a final output signal. | 06-13-2013 |
20130163783 | SYSTEMS, METHODS, AND APPARATUS TO FILTER AUDIO - Systems, methods, and apparatus to filter audio are disclosed. An example device includes first and second audio speakers having first audio characteristics, a third audio speaker having second audio characteristics, wherein the third speaker is positioned between the first and second audio speakers, a first audio filter to process an audio input signal to have a first frequency response including a first cutoff frequency, the first audio filter to output a first audio output signal to the first audio speaker, and a second audio filter to process the audio input signal to have a second frequency response to compensate for interference between the first and second frequency responses caused by a position of the first audio speaker relative to the second audio speaker. | 06-27-2013 |
20130163784 | BASS ENHANCEMENT SYSTEM - A bass enhancement system can provide an enhanced bass effect for speakers, including relatively small speakers. The bass enhancement system can apply one or more bass enhancements to an input audio signal. For example, the bass enhancement system can exploit how the human ear processes overtones and harmonics of low-frequency sounds to create the perception that non-existent (or attenuated) low-frequency sounds are being emitted from a loudspeaker. The bass enhancement system can generate harmonics of at least some low-frequency fundamental frequencies in one embodiment. Playback of at least some harmonics of a low-frequency fundamental frequency can cause a listener to perceive the playback of the low-frequency fundamental frequency. Advantageously, in certain embodiments, the bass enhancement system can generate these harmonics without performing processing-intensive pitch-detection techniques or the like to identify the fundamental frequencies. | 06-27-2013 |
20130163785 | APPARATUS AND METHOD FOR GENERATING VIBRATION BASED ON SOUND CHARACTERISTICS - A method and apparatus for generating vibration based on sound characteristics in a mobile terminal are provided. The method includes converting audio data into an audio signal upon generation of a sound play request; determining whether to generate vibration based on a sound volume of the audio signal; setting an actuator to be driven for the audio signal among at least two actuators based on frequency distribution characteristics of the audio signal if it is determined upon determining to generate the vibration; and driving the actuator being set for the audio signal when outputting the audio signal. | 06-27-2013 |
20130208917 | Automatic Equalization Using Adaptive Frequency-Domain Filtering and Dynamic Fast Convolution - Frequency-domain techniques are used for adaptive equalization that is responsive to spectral magnitude characteristics but not sensitive to phase characteristics of system response. Signal correlation may be used to improve adaptation accuracy when significant levels of ambient sounds are present. A preferred filter implementation uses convolution-based block transforms and cross-fade windows. | 08-15-2013 |
20130230190 | ELECTRONIC DEVICE AND METHOD FOR OPTIMIZING MUSIC - In a method for optimizing music using an electronic device, music genres and optimization parameters of each of the music genres are preset. The method classifies songs stored in a storage device of the electronic device according to the music genres. When a song is playing using a music player of the electronic device, a music genre of the song is determined. The method further optimizes the song using an equalizer of the music player according to the optimization parameters corresponding to the determined music genre. | 09-05-2013 |
20130243221 | METHOD AND SYSTEM OF EQUALIZATION PRE-PREOCESSING FOR SOUND RECEIVNG SYSTEM - An exemplary embodiment illustrates an equalization pre-processing method, adapted for characterizing a second sound receptor unit in a sound receiving system based on knowing the internal structure parameters of a first sound receptor unit. The method includes: measuring a first sensitivity response of the first sound receptor unit and a second sensitivity response of the second sound receptor unit; equalizing the second sensitivity response according to the first sensitivity response and obtaining the differences in the sensitivity response; conducting simulation to obtain the third sensitivity response associated with the first sound receptor unit in the given sound receiving system; compensating the third sensitivity response to generate the fourth sensitivity response associated with the second sound receptor unit in the sound receiving system according to the differences in the sensitivity response; analyzing the fourth sensitivity response to characterize the second sound receptor unit in the given sound receiving system. | 09-19-2013 |
20130272541 | MOBILE COMMUNICATION DEVICE CAPABLE OF SETTING TONE COLOR AND METHOD OF SETTING TONE COLOR - A mobile communication device and a method of setting tone color, which allow a user to set the tone color of received sound. Provided are a normal mode, which sets the equalizer using GCF standards stored in an internal memory or equalizer setting values selected by a provider, a country-specific mode, which uses country-specific setting, and a user mode, in which a user can set frequency-specific gains of the received sound, and one mode is selected from the provided mode, so that the tone color of the received sound can be adjusted according to the selection. Telephone speech quality can be optimized for user preference, network environments and language characteristics. | 10-17-2013 |
20130287227 | MUSICAL DYNAMICS ALTERATION OF SOUNDS - An improved method and arrangement for altering musical dynamics of a sound S included in a sound signal is disclosed. The altering of the musical dynamics is performed by filtering and amplification of the sound signal. The filtering is performed by the use of a parametric equalizer, the parametric equalizer having a first gain G | 10-31-2013 |
20130336502 | AUDIO REPRODUCTION METHOD AND APPARATUS WITH AUTO VOLUME CONTROL FUNCTION - An audio reproduction method and apparatus, in which an audio volume is automatically controlled based on audio energy and human auditory characteristics. The audio reproduction method includes splitting a reproduction audio signal into audio signals corresponding to a plurality of frequency bands, extracting audio energy for each of the frequency bands, and comparing the audio energy for each of the frequency bands with a predetermined threshold and controlling the volume of the audio signal corresponding to each of the frequency bands. | 12-19-2013 |
20130343573 | SYSTEM FOR INCREASING PERCEIVED LOUDNESS OF SPEAKERS - A system can be provided for increasing loudness of an audio signal to present a perceived loudness to a listener that is greater than a loudness provided natively by a loudspeaker. The system can include one or more of the following: a frequency suppressor, a loudness adjuster, an equalizer, and a distortion control module. The frequency suppressor can increase headroom in the audio signal by filtering out low and/or high frequencies. The loudness adjuster can calculate a loudness of the audio signal and apply a gain to the audio signal to increase the loudness. The equalizer can further increase headroom by attenuating portions of a passband of the loudspeaker's frequency response. The distortion control module can induce partial harmonic distortion in the audio signal to further increase loudness. | 12-26-2013 |
20140003625 | System and Method for Device Playback Calibration | 01-02-2014 |
20140003626 | AUTOMATIC AUDIO EQUALIZATION USING HANDHELD MODE DETECTION | 01-02-2014 |
20140003627 | Operator Adjustable Full-bandwidth Audio Spectral Shifting Control with a Simple Listener Interface | 01-02-2014 |
20140044283 | Amplification System, Carrier Tracking Systems and Related Methods for use in Parametric Sound Systems - A signal processing system for generating an ultrasonic signal from an audio signal comprises a compressor, operable to compress the audio signal, and an equalization circuit, operable to equalize the audio signal. A modulation circuit is operable to combine the audio signal with a carrier signal to produce at least one modulated carrier signal. A voltage detection and control circuit is operable to: detect a voltage of the modulated carrier signal; compare the detected voltage of the modulated carrier signal to a desired voltage; and cause the voltage of the modulated carrier signal to be adjusted if the compared voltage differs by a predetermined amount from the desired voltage. | 02-13-2014 |
20140093097 | Amplifying loudspeaker apparatus - An amplifying loudspeaker apparatus for an existing host device comprises a sensor coil for receiving an audio signal in electromagnetic form from an internal loudspeaker of the host device; a pre-amplifier and equalizer module connected to the sensor coil for amplifying and equalizing the audio signal to a suitable level and frequency response; and an audio power amplifier and a loudspeaker to reproduce the sound for the audio signal. | 04-03-2014 |
20140119570 | Equalization of Speaker Arrays - Methods and apparatus are described by which equalization and/or bass management of speakers in a sound reproduction system may be accomplished. | 05-01-2014 |
20140153744 | Audio Precompensation Controller Design Using a Variable Set of Support Loudspeakers - Disclosed is a method and a system to determine an audio precompensation controller for an associated sound generating system including a total of N≧2 loudspeakers, each having a loudspeaker input. The audio precompensation controller has a number L≧1 inputs for L input signals and N outputs for N controller output signals, one to each loudspeaker. For each one of at least a subset of the N loudspeaker inputs, an impulse response is estimated at each measurement position. For each one of the L input signal(s), a selected one of the N loudspeakers is specified as a primary loudspeaker and a selected subset S including at least one of the N loudspeakers as support loudspeaker(s). | 06-05-2014 |
20140153745 | Equalizer - An equalizer ( | 06-05-2014 |
20140177872 | FREQUENCY CHARACTERISTICS DETERMINATION DEVICE - A frequency characteristics determination device that can prevent generation of a frequency response curve having an unnatural shape is provided. A CPU | 06-26-2014 |
20140185828 | AMBIENT AUDIO INJECTION - An exemplary system comprises a device including a memory with an audio injection application installed thereon. The application comprises an equalizer module that analyzes sound characteristics of individual digital audio samples including a discrete signal, a selector module that applies a selection heuristic to select the discrete signal from the individual digital audio samples based on the sound characteristics, and an audio module that supplies to an output an insert signal generated according to the discrete signal selected by the selection heuristic. | 07-03-2014 |
20140185829 | IN-LINE SIGNAL PROCESSOR - The present invention provides for a communications cable for processing an input signal. Specifically, the present invention includes a communication cable having an input connector structured to receive an input signal, an audio enhancement module configured to process the input signal using a plurality of processing components in order to create an output signal, and an output connector structured to transmit the output signal. | 07-03-2014 |
20140193002 | PORTED ENCLOSURE AND AUTOMATED EQUALIZATION OF FREQUENCY RESPONSE IN A MICRO-SPEAKER AUDIO SYSTEM - One embodiment of the present invention sets forth a system that includes a detection device and a processor. The detection device is configured to sense that a handheld device has not been placed on or near a surface. In response to sensing that the handheld device has not been placed on or near a surface, the detection device is configured to transmit an indicator to the processor. The processor is configured to receive a first audio signal, and determine that the handheld device has not been placed on or near a surface by receiving the indicator from the detection device. In response to determining that the handheld device has not been placed on or near a surface, the processor is further configured to apply a compensating function to the first audio signal to generate a second audio signal, and transmit the second audio signal to a speaker. | 07-10-2014 |
20140205111 | SOUND PROCESSING APPARATUS, METHOD, AND PROGRAM - The present technique relates to a sound processing apparatus, a method, and a program capable of alleviating degradation of the quality of sound in a case where the gain of a sound signal is amplified. | 07-24-2014 |
20140211968 | METHOD FOR DYNAMICALLY ADJUSTING GAIN OF PARAMETRIC EQUALIZER ACCORDING TO INPUT SIGNAL, DYNAMIC PARAMETRIC EQUALIZER AND DYNAMIC PARAMETRIC EQUALIZER SYSTEM EMPLOYING THE SAME - The present invention relates to a method for dynamically adjusting a gain of parametric equalizer according to an input signal, a dynamic parametric equalizer employing the same and a dynamic parametric equalizer system employing the same wherein a gain of parametric equalizer is dynamically adjusted according to a level of an input digital audio signal to prevent distortion of output signal. | 07-31-2014 |
20140219475 | APPARATUS AND METHOD OF MULTI-SENSOR SOUND RECORDING - An apparatus and a method for sound recording are provided. The apparatus stores an equalizer and includes a housing, a microphone, and a processor. The microphone includes a plurality of sensors and receives a sound signal through an acoustical resonator in the housing. The microphone generates a plurality of electronic signals in response to the sound signal. The equalizer generates a plurality of equalized signals according to the electronic signals. The processor generates an output signal according to the equalized signals. The equalizer compensates the gain margin and the phase margin caused by the sound signal passing through the acoustical resonator in order to prevent the output signal from being affected by resonance of the sound signal in the acoustical resonator. | 08-07-2014 |
20140219476 | SYSTEM AND METHOD OF FILTERING AN AUDIO SIGNAL PRIOR TO CONVERSION TO AN MU-LAW FORMAT - Disclosed is a method of processing a signal such as an audio signal. The method includes normalizing a received signal to a level of between −2 dB and −10 dB to yield a normalized signal, compressing the normalized signal exceeding a threshold (such as 70% of the maximum volume and above) to yield a compressed signal, performing frequency equalization on the compressed signal to yield an equalized signal; and normalizing the equalized signal. | 08-07-2014 |
20140254828 | System and Method for Personalization of an Audio Equalizer - The invention is directed to synthesizing a personalized audio equalization filter based on an individual's hearing profile when listening to audio signals using a preferred electronic means of transduction and production of the audio signals through headphones, earphones, or loudspeakers and is presented in two steps. In the first step, the deviation of the listener's hearing profile, when the preferred listening device is donned, is measured and recorded relative to predetermined equal-loudness contours (e.g., ISO standard equal-loudness contours). The hearing profile is used to compute an equalization filter such that when the audio signal to be reproduced is played through the filter, the equal-loudness contour for a specified phon level is restored. Thus, the invention compensates for deviations of the user's listening device, the user's own hearing profile from a standard hearing profile that represents natural hearing, and other factors. The equalization filter can be adapted to any phon level through adaptation of filter coefficients as a function of the volume of the listening level. | 09-11-2014 |
20140301569 | ACTIVE NOISE EQUALIZATION - An active noise equalization (ANE) system may be run on the existing audio/infotainment system as a software library. The ANE system may share components (e.g., microphones and sensors) with other audio applications. Some ANEs include a complex-domain formulation of a multiple-frequency multiple-channel ANE that requires less memory and processing requirements. The complex-domain system replaces the multiplication of multiple real gains with multiple real signals with a single complex multiplication operation. | 10-09-2014 |
20140321669 | SYSTEMS, METHODS, AND APPARATUS TO FILTER AUDIO - Systems, methods, and apparatus to filter audio are disclosed. An example method includes determining that an orientation of a playback device is one of (a) a first orientation and (b) a second orientation, the playback device including a first speaker and a second speaker and configuring the first speaker and the second speaker. In the example, configuring the first and second speakers includes, when the determined orientation is the first orientation, configuring a first audio filter to be applied to a first audio signal for the first speaker and configuring a second audio filter to be applied to a second audio signal for the second speaker. However, when the determined orientation is the second orientation, configuring the first and second speakers includes configuring a third audio filter to be applied to the first audio signal and the second audio signal. | 10-30-2014 |
20140369523 | PROCESS FOR IMPROVING AUDIO (API) - A computer implemented method for improving audio, comprising processing audio with a combination of wave adjustment tool WAT and a collection of preset settings for adjusting specific harmonic content to specific genres of music. The preset is selected from the group comprising different genres (styles or types) of music; auto-preset; and generic preset. The auto preset value is determined by the metadata which is commonly included in an audio file. In the event there is no match between the preset genre names and the one in the metadata, then the selected preset will be “Generic.” This particular genre preset is designed for optimal sound across many genres, making it generic. The wave adjustment tool (WAT) allows for low, mid, and hi tone controls. The process is carried out by installing software in an existing audio path immediately after initial input and prior to output. A host device provides for a buffer to allow for processing of audio. A program placed in libraries that work together is used improve audio quality. | 12-18-2014 |
20140369524 | WAVE ADJUSTMENT TOOL (WAT) - A Wave Adjustment Tool process and system for customizing sound is provided. An input audio sound is received. A tone adjusting circuit that comprises three sections, including a first section for adjusting a low frequency tone; a second section for adjusting a mid frequency tone; and a third section for adjusting a high frequency tone. Audio processed by the first, second and third sections are mixed to produce an output audio sound. According to an exemplary embodiment the low frequency tone has a frequency of 100 Hz and a bandwidth of 0.5.; the mid frequency tone has a frequency of 2500 Hz and an adjustable bandwidth; and the high frequency tone has a frequency of 10 KHz and an adjustable bandwidth. | 12-18-2014 |
20150036841 | FILTER WITH INDEPENDENTLY ADJUSTABLE BAND GAIN AND BREAK POINT SLOPES AND METHOD OF CONSTRUCTING SAME - A filter for equalizing the frequency response of loudspeaker systems includes at least one band filter section ( | 02-05-2015 |
20150036842 | LOUDNESS LEVEL CONTROL FOR AUDIO RECEPTION AND DECODING EQUIPMENT - The application discusses a computer implemented method and apparatus for performing audio equalisation in an audio receiver device, such as an integrated receiver/decoder or set top box, or integrated TV, connected to one or more audio playback devices, such as a television unit, computer screen and speakers, amplifier or home theatre equipment. The method and apparatus use an equalisation process which compares audio signals received in different audio formats (e.g. MPEG-1 Layer II, AC-3 2.0, AC-3 5.1 and HE-AAC) with one another, allowing a correction gain factor to be determined for equalising the perceived loudness of the signals when played-back at a connected playback device. The correction gain factor is then applied in the audio receiver device before output. | 02-05-2015 |
20150043750 | ELECTRONIC DEVICE AND METHOD FOR CONTROLLING ELECTRONIC DEVICE - An echo caused by sound leakage from a housing that vibrates due to a vibrating body is reduced. An electronic device ( | 02-12-2015 |
20150063596 | System and Method for Processing Signals to Enhance Audibility in an MRI Environment - A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the M RI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor. | 03-05-2015 |
20150086041 | Acoustic Signatures - Embodiments described herein provide for acoustic signatures in a playback system. In some embodiments, a playback device includes an audio processing component configured to receive an audio input signal, receive an acoustic signature signal, generate an audio output signal based on the audio input signal and the acoustic signature signal, and output the generated audio output signal. In some embodiments, the acoustic signature identifies at least one of the playback device or a group of playback devices. Some embodiments additionally include the playback device amplifying the generated audio output signal with one or more amplifiers, and playing the generated audio output signal via one or more speakers. | 03-26-2015 |
20150110293 | PLAYBACK BASED ON RECEIVED SOUND WAVES - Apparatus and methods are disclosed for acoustic optimization. An example playback device includes a first transducer to at least one of output sound waves and receive sound waves, a second transducer to at least one of output sound waves and receive sound waves, and an acoustic grille positioned in relation to the first transducer, where the acoustic grille is to reflect sound waves received at a first angle of incidence. | 04-23-2015 |
20150125000 | Method and device for storing equalization settings in active loudspeaker - The invention relates to a method and a loudspeaker for storing equalization settings in an active loudspeaker including a loudspeaker cabinet, at least one driver connected to the cabinet and an amplifier electronics including an equalizer, in which method the loudspeaker is calibrated in controlled conditions for forming factory calibration parameters for controlling the equalizer, the factory calibration parameters controlling the equalizer are stored in the amplifier electronics. In accordance with the invention the factory calibration parameters are also stored in a memory fixedly mounted to the loudspeaker. | 05-07-2015 |
20150341008 | VARIABLE EQUALIZATION - An equalizer that linearly interpolates between two equalization states when transitioning from one equalization state to the other equalization state is described. The equalizer includes a transfer function generator and an equalization module. Each equalization state is defined or determined based on a set of parameters. The transfer function generator generates a set of interpolated transfer functions by performing linear interpolation on a first equalization state and a second equalization state based on the set of parameters. The linear interpolation is performed on corresponding Z-domain poles and zeros of the transfer functions of the first and second equalization states. The equalization module applies the set of interpolated transfer functions generated by the transfer function generator to an input audio signal. | 11-26-2015 |
20150350786 | GROUP DELAY CORRECTION IN ACOUSTIC TRANSDUCER SYSTEMS - Methods are provided for equalising the group delay of a sound reproduction system, in particular a system comprising acoustic transducers with at least one crossover between a lower-frequency and a higher-frequency range. A correction is applied to a signal in the lower-frequency range, including the crossover region, to substantially equalise the group delay for the lower-frequency range, and a signal delay is applied to a signal in the higher-frequency range to bring it into closer alignment with the equalised lower-frequency range signal. The methods may be implemented in the design of an acoustic transducer system and also via a computer program product, which can be implemented as an update or enhancement to an existing digital signal processor loudspeaker system. | 12-03-2015 |
20150365061 | SYSTEM AND METHOD FOR MODIFYING AN AUDIO SIGNAL - A method and system for modifying an audio signal are provided. An audio subsystem can receive a control input signal including a predefined set of desired audio characteristics for an acoustic output of the audio subsystem. The audio subsystem can further receive an input audio electrical signal, and based on the control input signal, the audio subsystem can equalize, compress, limit and amplify the input audio electrical signal for output to at least one output device. | 12-17-2015 |
20160014509 | COMMUNICATION DEVICE AND METHOD FOR ADAPTING TO AUDIO ACCESSORIES | 01-14-2016 |
20160036404 | EQUALIZATION FILTER COEFFICIENT DETERMINATOR, APPARATUS, EQUALIZATION FILTER COEFFICIENT PROCESSOR, SYSTEM AND METHODS - An equalization filter coefficient determinator continuously or quasi-continuously fades between a plurality of different equalizer settings in dependence on one or more setting parameters, to obtain a current set of equalization filter target coefficients describing a current equalizer setting. A number of setting parameters is smaller than a number of equalization filter target coefficients of current set of equalization filter target coefficients. | 02-04-2016 |
20160037252 | AUDIO FILTERS UTILIZING SINE FUNCTIONS - A method of digitally filtering an audio signal using an adjusted audio filter. The adjusted audio filter is represented by an impulse response including a waveform in its time domain represented by a sine function of absolute values. A composite audio filter is derived from two adjusted audio filters although any number of filters may be used. The composite audio filter generally includes a bank of the filters which together define a frequency bandwidth representative of the audio signal or spectrum to be filtered. Also a bandpass filter is constructed by combining frequency responses for sine components of absolute values integrated from 0 to bpf and sine components of absolute values integrated from 1/bpf to 0. The frequency response may be the sum of the frequency responses for each of the filters used to create the composite bandpass filter. | 02-04-2016 |
20160126916 | ADAPTIVE EQUALIZATION FOR AN ULTRASONIC AUDIO SYSTEM - An ultrasonic audio system includes adaptive equalization techniques configured to adjust the overall output of the ultrasonic audio system. This can be done by adjusting the equalization of the electronic audio signal representing the audio content being reproduced by the ultrasonic audio system. For example, in various embodiments, systems and methods can be implemented to increase the gain of higher frequency portions of the electronic audio signal while making a corresponding decrease in the amount of gain at the lower frequency portions of the same signal. | 05-05-2016 |
20160149550 | METHOD AND APPARATUS TO EQUALIZE ACOUSTIC RESPONSE OF A SPEAKER SYSTEM USING MULTI-RATE FIR AND ALL-PASS IIR FILTERS - A speaker system includes an electronic signal processing unit configured to split an input audio signal into a low-pass audio component and a high-pass audio component. The electrical signal processing unit includes a multi-rate finite impulse response (FIR) filter configured to downsample the low-pass audio component and sample the downsampled low-pass audio component at a first sampling rate lower than a second sampling rate used to sample the high-pass audio component. The speaker system further includes an electronic adaptive equalizer coefficient (EAEC) module configured to dynamically determine at least one equalizer coefficient based on a signal characteristic of the input audio signal. An electronic filter coefficient update unit (FCUU) is included and is configured to control a frequency response of the speaker system based on the at least one equalizer coefficient. | 05-26-2016 |
20160150343 | Adaptive Audio Content Generation - Embodiments of the present invention relate to adaptive audio content generation. Specifically, a method for generating adaptive audio content is provided. The method comprises extracting at least one audio object from channel-based source audio content, and generating the adaptive audio content at least partially based on the at least one audio object. Corresponding system and computer program product are also disclosed. | 05-26-2016 |
20160173984 | Automatic Audio System Equalizing | 06-16-2016 |
20160192100 | Automatic Audio System Equalizing - An automated process for equalizing an audio system and an apparatus for implementing the process. An audio system includes a microphone unit, for receiving the sound waves radiated from a plurality of speakers, acoustic measuring circuitry, for calculating frequency response measurements; a memory, for storing characteristic data of the loudspeaker units and further for storing the frequency response measurements; and equalization calculation circuitry, for calculating an equalization pattern responsive to the digital data and responsive to the characteristic data of the plurality of loudspeaker units. Also described is an automated equalizing system including a acoustic measuring circuitry including a microphone for measuring frequency response at a plurality of locations; a memory, for storing the frequency responses at the plurality of locations; and equalization calculation circuitry, for calculating, from the frequency responses, an optimized equalization pattern. | 06-30-2016 |
20160197591 | Genetic EQ-Setup | 07-07-2016 |
20180026599 | Compressor System with EQ | 01-25-2018 |