Entries |
Document | Title | Date |
20080205666 | Device For Processing Audio Data, A Method Of Processing Audio Data, A Program Element And A Computer-Readable Medium | 08-28-2008 |
20080212797 | BASS BOOST FILTERING TECHNIQUES - Bass frequencies of audio can be boosted using various techniques and tools. The described techniques and tools can be applied separately or in combination. For example, bass frequencies of audio can be boosted using an integer bass boost filter by receiving user-settable parameters, such as “c” and “s” coefficients, and implementing the integer bass boost filter using a coupled form structure implementation and the user-settable parameters. Bass frequencies of audio can also be boosted using an integer bass boost filter that is configured to use any of plural coupled form structure implementations. Bass frequencies of audio can be also be boosted using a linear combination of an input audio signal and output of a high-pass filter. | 09-04-2008 |
20080267426 | Device for and a Method of Audio Data Processing - An audio data processing device ( | 10-30-2008 |
20080273717 | Audio amplifier thermal management using low frequency limiting - A digital audio speaker system for a digital audio player that includes a housing and a control unit positioned in the housing. An amplifier is positioned in the housing and connected with the control unit and a speaker system. A frame assembly is positioned in the housing that includes a motorized drive assembly. A drawer assembly is movably connected with the motorized drive assembly. The motorized drive assembly is operable to move the drawer assembly between an open position and a closed position. | 11-06-2008 |
20080273718 | Bass enhancing method, signal processing device, and audio reproducing system - If the absolute value of the current sample is greater than or equal to an envelope value at the immediately preceding sample, an envelope value at the current sample is made greater than the envelope value at the immediately preceding sample. If the absolute value of the current sample is smaller than the envelope value at the immediately preceding sample and a count value C does not reach a predetermined number N, the count value C is incremented by one and the envelope value at the current sample is held at the envelope value at the immediately preceding sample. If the absolute value of the current sample is smaller than the envelope value at the immediately preceding sample and the count value C reaches the predetermined number N, the envelope value at the current sample is made smaller than the envelope value at the immediately preceding sample. | 11-06-2008 |
20080279395 | Feedback Compensation in a Sound Processing Device - There is disclosed a sound processing device ( | 11-13-2008 |
20080292113 | METHOD AND APPARATUS FOR AUDIO PATH FILTER TUNING - Methods and apparatus for tuning the audio path response of an audio device are described herein. The audio output of an audio device is captured and monitored over a predetermined frequency band. The frequency response of the captured audio output can be compared to one or more predetermined limits defining an acceptable frequency response. One or more dynamically configurable bands can be defined, and one or more parameters affecting the response within each configurable band can be adjusted. A simulated frequency response is produced and filter coefficients for a corresponding filter within the audio path are determined. The filter coefficients can be loaded to a digital filter within the audio path to modify the actual frequency response produced by the audio device. | 11-27-2008 |
20090041265 | SOUND SIGNAL PROCESSING DEVICE, SOUND SIGNAL PROCESSING METHOD, SOUND SIGNAL PROCESSING PROGRAM, STORAGE MEDIUM, AND DISPLAY DEVICE - A sound signal processing device causes an amplifier to amplify a low frequency signal and synthesizes the amplified low frequency signal with a middle or high frequency signal so as to output the resultant output signal. A gain control section controls a gain of the amplifier on the basis of an amplitude of the output signal. As a result, it is possible to amplify the low frequency signal without any clipping in a receiving end device and without any unstable volume of a middle or high frequency component. | 02-12-2009 |
20090041266 | Electrostatic loudspeaker driver - An electrostatic loudspeaker driver includes a class-D amplifier and a demodulator circuit. The class-D amplifier is operated with a PWM signal, creating an amplified digital signal according to an input signal. A low-pass filter in the demodulator circuit filters out the PWM carrier frequency in the digital signal and retrieves an audio signal therefrom. The efficiency is improved significantly and heat sink is no longer needed. | 02-12-2009 |
20090046871 | METHOD AND APPARATUS FOR AUDIO PROCESSING - A method and apparatus for introducing a time-varying time delay randomly into the individual reproduction channels of a sound recording, two in the case of binaural presentation. This emulates the temporal aspect of microphone and/or listener motion. The present invention may be applied as a unidirectional process. No preparation of the source material is required. It can be applied to any multichannel audio signal set. It can process analog or digital signals. The process may be used with headphones, loudspeakers, hearing aids or similar assistive hearing devices. | 02-19-2009 |
20090052693 | Sound Reproducing Apparatus - A sound reproducing apparatus is structured to use a first dividing circuit and a determining circuit to analyze a frequency characteristic of an inputted signal and to output a signal adjusted by a first adjusting circuit to have a prescribed frequency characteristic by a switching by a switch circuit. Thus, when signals having different frequency characteristics are inputted, frequency characteristics of the input signals can be automatically adjusted without using an external signal and channels having different frequency characteristics can be reproduced by a single system to provide good speaker-reproduced sound. | 02-26-2009 |
20090052694 | Pseudo deep bass generating device - A multiplier | 02-26-2009 |
20090067644 | Economical Loudness Measurement of Coded Audio - Measuring the loudness of audio encoded in a bitstream that includes data from which an approximation of the power spectrum of the audio can be derived without fully decoding the audio is performed by deriving the approximation of the power spectrum of the audio from said bitstream without fully decoding the audio, and determining an approximate loudness of the audio in response to the approximation of the power spectrum of the audio. The data may include coarse representations of the audio and associated finer representations of the audio, the approximation of the power spectrum of the audio being derived from the coarse representations of the audio. In the case of subband encoded audio, the coarse representations of the audio may comprise scale factors and the associated finer representations of the audio may comprise sample data associated with each scale factor. | 03-12-2009 |
20090074204 | Information processing apparatus, information processing method, and program - According to the present invention, a parameter adjustment section setting, in accordance with a first parameter indicating a variant factor for playback speed that is input, a second parameter and a third parameter, and a signal processing section adjusting at least one of playback speed and pitch of a sound of an audio signal based on the second parameter and the third parameter are provided, wherein the signal processing section adjusts the playback speed of the audio signal when the variant factor for playback speed that is input is less than a predetermined threshold and adjusts the playback speed and the pitch of a sound of the audio signal when the variant factor for playback speed that is input is above the predetermined threshold. | 03-19-2009 |
20090080675 | DYNAMIC BASS BOOST FILTER - Bass frequencies of audio can be dynamically boosted using various techniques and tools. The described techniques and tools can be applied separately or in combination. Bass frequencies of audio can be boosted using a linear combination of an input audio signal and output of a high-pass filter. For example, bass frequencies of audio can be boosted by applying a high-pass filter to an input audio signal to produce an output of the high-pass filter, determining a current level, determining a target gain amount, dynamically adjusting the input audio signal and the output of the high-pass filter, and combining the gain-adjusted signals to produce an output signal. A dynamic bass boost filter can comprise a high-pass filter and a dynamic boost module. | 03-26-2009 |
20090161887 | DATA PROCESSING APPARATUS AND METHOD OF CONTROLLING THE SAME - According to one embodiment, a data processing apparatus includes a control section which, when the performance of the memory device is at a predetermined threshold value or higher, stores the frequency while associating it to a volume of the sound generated and increase a frequency of sound generated next by a certain frequency band from the frequency, when the performance of the memory device is less than the predetermined threshold level, decreases the volume of the sound by a certain volume level from the maximum volume, when the volume corresponding to the highest frequency, which is the predetermined frequency, is stored, finishes the measurement, and generates the sound from the lowest frequency to the highest frequency from the loudspeaker in accordance with the volume stored and the frequency corresponding to the volume. | 06-25-2009 |
20090238380 | SPATIALLY ROBUST AUDIO PRECOMPENSATION - A discrete-time audio precompensation filter is designed based on a linear model that describes the dynamic response of a sound generating system at p>1 listening positions. The filter construction is based on providing information (S2) representative of n non-minimum phase zeros {z,} that are outside of the stability region |z|= | 09-24-2009 |
20090252346 | METHOD OF PROCESSING AUDIO FILES - A method of processing audio files includes the steps of providing an audio frequency adjusting section, via which one or more first audio files may be changed in the frequency property thereof to create one or more second audio files without the need of opening the first audio files. In this manner, a plurality of audio files may be easily and quickly modified at one time via the audio frequency adjusting section to shorten the time needed to optimize the tone quality of the audio files. | 10-08-2009 |
20090285412 | INTEGRATED CIRCUIT BIASING A MICROPHONE - The invention provides an integrated circuit. The integrated circuit receives a first signal from a microphone via a first node. In one embodiment, the integrated circuit comprises a biasing circuit and a buffering circuit. The biasing circuit is coupled between the first node and a second node, drives the microphone with a first voltage source, and filters the first signal to generate a second signal at the second node. In one embodiment, the biasing circuit comprises a first resistor, a first capacitor, and a load element. The first resistor is coupled between the first voltage source and the first node. The first capacitor is coupled between the first node and the second node. The load element is coupled between the second node and a second voltage source. The buffering circuit is coupled between the second node and a third node and buffers the second signal to generate a third signal at the third node. | 11-19-2009 |
20090304204 | Controlling reproduction of audio data - For controlling the acoustic reproduction of audio data containing audio elements that are periodically repeated, movement data regarding a movement process is detected. The movement process contains recurring events. Reproduction of the audio data is controlled using the movement data in such a way that at least within a certain period, one out of n audio elements that are periodically repeated is reproduced in synchrony with the moment one of the recurring events occurs (synchronization) or is reproduced temporally offset by a given amount of time from the moment one of the recurring events occurs (offset synchronization). The value n represents a positive integer. | 12-10-2009 |
20090310799 | INFORMATION PROCESSING APPARATUS AND METHOD, AND PROGRAM - An information processing apparatus includes a band spreading unit configured to perform a band spreading process for generating components in a specific frequency band and adding the components to audio data, and a control unit configured to control the band spreading unit to execute the band spreading process using a band spreading method determined among a plurality of different band spreading methods, the band spreading method being defined in advance for a musical class determined using a feature of the audio data. | 12-17-2009 |
20090323983 | Method and a system for reconstituting low frequencies in audio signal - The method comprises the steps of: filtering the audio signal by means of a lowpass filter ( | 12-31-2009 |
20090323984 | INFORMATION PROCESSING DEVICE, CONTROL METHOD THEREFOR, AND PROGRAM - An information processing device according to an embodiment of the present invention includes the following elements: a detection unit configured to detect an amount of change in position of an image data item that is displayed on a display screen; and a processing unit configured to perform image processing on the image data item in accordance with a detection result that is obtained by the detection unit, and to perform audio processing on an audio data item corresponding to the image data item in accordance with the detection result that is obtained by the detection unit. | 12-31-2009 |
20100008521 | VOLUME AND TONE CONTROL IN DIRECT DIGITAL SPEAKERS - A system that includes a direct digital speaker volume control device configured to be coupled to a direct digital speaker. The direct digital speaker includes many pressure producing elements being adapted to generate a sound at a sound pressure level (SPL) and at a given frequency in response to an input signal, without using digital to analog converter. The direct digital speaker inherently exhibits a frequency response throughout its entire frequency range. The direct digital speaker volume control device includes a module for providing few filters each having a distinct cutoff frequency such that each filter exhibits no attenuation below its cutoff frequency and an attenuation response above the filter's cutoff frequency. And a selector for selecting one of the filters according to a selection criterion that depends on a desired volume and frequency of generated sound, and applying the filter to the input signal for generating a filtered signal that fed to the speaker. | 01-14-2010 |
20100040244 | METHOD AND SYSTEM FOR AUDITORY ENHANCEMENT AND HEARING CONSERVATION - A digital and/or analog signal processing system and method for auditory enhancement and hearing conservation includes providing an audio signal with high intensity peaks, clipping the audio signal by limiting peak power to produce a clipped signal, and amplifying the clipped signal. | 02-18-2010 |
20100046771 | MULTIPLE-USE ACOUSTIC PORT - Two or more acoustic transducers share the same acoustic port in a device. The acoustic properties—such as acoustic impedance and frequency response—of the shared acoustic port are matched to each of the two or more acoustic transducers. To accomplish acoustic impedance matching, a separate back volume is provided for each of the acoustic transducers, matched to that transducer. Frequency response matching can be accomplished by the design of the transducer itself, but also by providing an adjacent element in the acoustic system of the transducer. One transducer may serve as an element in the acoustic system of another transducer. Frequency response adjustment of an individual element may also affect acoustic impedance of the entire port-transducer system. | 02-25-2010 |
20100086147 | HARMONICS GENERATION APPARATUS AND METHOD THEREOF - A harmonic generating apparatus and the method are provided, which are used to enhance the quality of the bass audio signals. The method includes the steps of: providing a frequency signal having a present level and a preceding level; comparing the present level with the preceding level to generate a compared result; and generating the plurality of harmonics based on the compared result. | 04-08-2010 |
20100086148 | APPARATUS AND METHOD FOR PROCESSING AUDIO SIGNAL - An apparatus for processing an audio input signal is provided and includes an audio processing circuit and an audio compressing circuit. The audio processing circuit receives the audio input signal, and enhances a first frequency part of the audio input signal to output a bass-enhancement signal. The audio compressing circuit is coupled to the audio processing circuit, and reduces a gain of a second frequency part of the bass-enhancement signal to output an audio output signal. | 04-08-2010 |
20100119081 | ELECTROACOUSTICAL TRANSDUCING WITH LOW FREQUENCY AUGMENTING DEVICES - A method for clipping and post-clipping processing an audio signal, includes clipping an audio signal to provide a clipped audio signal; filtering, by a first filter, the audio signal to provide a filtered unclipped audio signal; and filtering, by a second filter, the clipped audio signal to provide a filtered clipped audio signal. The method further includes differentially combining the filtered clipped audio signal and the clipped audio signal to provide a differentially combined audio signal; and combining the filtered unclipped audio signal and the differentially combined audio signal to provide an output signal. | 05-13-2010 |
20100128900 | ADAPTER ACCESSORY FOR ELECTRONIC DEVICE SHARING - Users with headsets may share an electronic device such as a portable computer or handheld device. The electronic device may have a connector such as an audio jack for receiving mating audio plugs on headsets. During normal operation with a single user, audio signals may be conveyed through the audio jack to the headset of the single user. When more than one user wishes to share the electronic device, an adapter accessory may be inserted into the audio jack of the electronic device. The headset of each user may be plugged into mating audio jacks in the adapter accessory. Circuitry in the adapter accessory may receive and process user input from each of the users. User input may be used to make local audio adjustments in the adapter accessory. User input may also be provided from the adapter accessory to the electronic device for processing. | 05-27-2010 |
20100128901 | WIND NOISE REJECTION APPARATUS - An apparatus for reduction of wind noise comprised of an electro-acoustic transducer arrangement with at least one transducer element. The exposed structure is covered with at least one thin layer of wind-resistive material, in the form of a mesh, and optionally includes one or more further layers of felt, foam and/or mesh in any combination. Where a plurality of elements is provided, the electrical outputs of the elements may be added together to provide an output signal with increased signal to wind noise ratio. The signal may be subject to additional signal processing such as filtering and/or level sensitive signal inhibition. | 05-27-2010 |
20100142727 | SOUND PROCESSING METHODS AND APPARATUS - A sound processing method includes transforming an input signal from a time domain to a frequency domain to produce a spectrum, detecting a peak of the spectrum, calculating a target attenuation amount based on one of the input signal and the spectrum, calculating attenuation amounts of respective frequency components of the spectrum based on the target attenuation amount and the detected peak, correcting levels of the spectrum by attenuating the spectrum in response to the calculated attenuation amounts of respective frequency components, and performing inverse frequency transform with respect to the level-corrected spectrum to produce an output signal. | 06-10-2010 |
20100150377 | SOUND OUTPUTTING APPARATUS TO CORRECT SOUND QUALITY AND METHOD OF CORRECTING SOUND QUALITY THEREOF - A sound outputting apparatus to correct sound quality and a method of correcting sound quality thereof includes a speaker to output a sound signal, a remaining vibration detector to detect a remaining vibration of the speaker, a signal processor to generate an offset signal for the detected remaining vibration and output a correction signal in which the input sound signal is mixed with the generated offset signal, and a power amplifier to amplify the correction signal and transmit the correction signal to the speaker. Accordingly, sound quality can be corrected without a mechanical damping device. | 06-17-2010 |
20100158272 | ASYMMETRIC POLYNOMIAL PSYCHOACOUSTIC BASS ENHANCEMENT - Psychoacoustic bass audio signal enhancement can be accomplished using a monotonic, asymmetric polynomial distortion. A non-linear process applies a monotonic, asymmetric polynomial distortion function that has continuous first and second derivatives to generate even and odd harmonics of missing fundamental frequencies. This polynomial distortion produces the desired psychoacoustic effect with a fairly rapid roll-off so as to avoid unpleasant aliasing. Moreover, the lack of first-order discontinuities prevents clicks or glitches. | 06-24-2010 |
20100166217 | METHOD OF AND CIRCUIT FOR ADJUSTING FREQUENCY OF A DRIVING SIGNAL OF ELECTRONIC HORN BY MEANS OF CAPACITOR - The present invention discloses a driving circuit for electronic horn and a driving method for the electronic horn. The driving circuit includes an oscillating circuit, which generates a signal having oscillating frequency. Based on the signal, said driving circuit generates a driving signal to drive the electronic horn to produce sound. It is characterized in that said oscillating circuit includes a variable capacitor, and the oscillating frequency is changed by adjusting capacitance of the variable capacitor so as for the frequency of the driving signal to be consistent with working frequency of the electronic horn. The present invention overcomes the problem of mismatch between the frequency of circuit's driving signal and the horn's sounding diaphragm and horn's tone inflexion due to resistance change caused by vehicle vibration. | 07-01-2010 |
20100166218 | Sound Processor, Sound Reproducer, and Sound Processing Method - According to one embodiment, a sound processor includes a creating module, a filter, and a combining module. The creating module creates a plurality of first acoustic models based on a frequency characteristic that represents an acoustic property of an object to be measured. The first acoustic models are modeled with respect to resonance properties that vary depending on frequency bands. The filter extracts frequency components in the frequency bands from the respective first acoustic models. The combining module combines the frequency components extracted from the first acoustic models to create a second acoustic model. | 07-01-2010 |
20100166219 | ELEVATED TOROID MICROPHONE APPARATUS - A video teleconferencing directional microphone includes three microphone elements arranged coincidentally on a vertical axis. The three microphone elements are placed on a supporting surface so that a first microphone element is on the surface, and the second and third microphone elements are elevated above the supporting surface. The directional microphone also includes three filters, a summing node, and an equalizer, which are used to shape the directivity pattern of the directional microphone into an elevated toroid sensitivity pattern. The elevated toroid sensitivity pattern increases sensitivity in the direction of a sound source of interest, but reduces sensitivity to any sound waves generated by noise sources at other locations. | 07-01-2010 |
20100183168 | SEMICONDUCTOR DEVICE - A semiconductor device is disclosed. The semiconductor device includes a digital audio circuit which converts an input digital signal into an analog audio signal, a DC-DC converter having a switching power source circuit, and an audible frequency determining circuit. In order that a difference between a frequency of a first clock signal for digital to analog conversion which is used in the digital audio circuit and a frequency of a second clock signal for switching control which is used in a DC-DC converter exceeds a maximum audible frequency, a frequency comparing circuit in the audible frequency determining circuit outputs a signal to a frequency changing circuit in the DC-DC converter. The frequency changing circuit causes a second oscillating circuit to change the second frequency. | 07-22-2010 |
20100202629 | SOUND REPRODUCTION SYSTEMS - A sound reproduction system includes an electro-acoustic transducer and a transducer driver for driving the electro-acoustic transducer. The transducer drive includes a filter which is configured to reproduce at a listener's location an approximation to the local sound field that would be present at the listener's ears in recording space, taking into account the characteristics and intended position of the electro-acoustic transducer relative to the listener's ears. The electro-acoustic transducer includes a first sound emitter which provides an intermediate sound emission channel, and second and third sound emitters providing respective left and right sound emission channels. The first sound emitter is located intermediate of second and third sound emitters. Higher frequencies from at least one of the second and third sound emitters are transmitted closer to the first sound emitter while lower frequencies are transmitted away from the first sound emitter. | 08-12-2010 |
20100215192 | METHOD AND DEVICE FOR EXTENSION OF LOW FREQUENCY OUTPUT FROM A LOUDSPEAKER - A method and device for enhancing low frequency content of an input signal (X), e.g. bass boosting of an audio signal. An overdriving (ODR) of a low frequency signal part (LS | 08-26-2010 |
20100239106 | Probabilistic Method of Loudspeaker Detection - A method and apparatus for enhancing cutoff detection of a loudspeaker. The method comprising retrieving a loudspeaker model cutoff and model error, generating a probability distribution of the cutoff frequency based on the retrieved models, and utilizing the generated probability distribution to enhance the detection of the cutoff of the loudspeaker. | 09-23-2010 |
20100278356 | AUDIO AMPLIFICATION APPARATUS - A method of adjusting frequency-dependent amplification in an audio amplification apparatus. The audio amplification apparatus includes a forward transfer path ( | 11-04-2010 |
20100322438 | METHOD AND CIRCUIT FOR CONTROLLING AN OUTPUT OF AN AUDIO SIGNAL OF A BATTERY-POWERED DEVICE - A method and a control circuit for controlling an output of an audio signal of a battery-powered device are described. | 12-23-2010 |
20110013783 | OVERTONE PRODUCTION DEVICE, ACOUSTIC DEVICE, AND OVERTONE PRODUCTION METHOD - When boosting treble, first a signal X(T) of a fundamental tone component which is to be a subject of overtone production is extracted by a fundamental tone extraction filter section | 01-20-2011 |
20110019838 | AUDIO PROCESSING IN A PORTABLE LISTENING DEVICE - The invention relates to a method of processing an audio signal in a portable listening device, the audio signal comprising a low frequency part having an LF-bandwidth Δf | 01-27-2011 |
20110044471 | GENERATION OF A DRIVE SIGNAL FOR SOUND TRANSDUCER - An apparatus for generating a drive signal for a sound transducer ( | 02-24-2011 |
20110058685 | METHOD OF SEPARATING SOUND SIGNAL - The present invention obtains a separated signal from an audio signal based on the anisotropy of smoothness of spectral elements in the time-frequency domain. A spectrogram of the audio signal is assumed to be a sum of a plurality of sub-spectrograms, and smoothness of spectral elements of each sub-spectrogram in the time-frequency domain has directionality on the time-frequency plane. The method comprises obtaining a distribution coefficient for distributing spectral elements of said audio signal in the time-frequency domain to at least one sub-spectrogram based on the directionality of the smoothness of each sub-spectrogram on the time-frequency plane, and separating at least one sub-spectrogram from said spectral elements of said audio signal using said distribution coefficient. | 03-10-2011 |
20110058686 | AUDIO PROCESSING DEVICE AND AUDIO PROCESSING METHOD - There is provided a sound processing apparatus and a sound processing method which are capable of reproducing discrete data with a high-quality sound matching users' preferences. In a sound processing means | 03-10-2011 |
20110064244 | Method and Arrangement for Processing Audio Data, and a Corresponding Computer Program and a Corresponding Computer-Readable Storage Medium - A method and an arrangement for processing audio data, and a corresponding computer program and a corresponding computer-readable storage medium, which can be used, in particular, in the field of audio software and sampling. At least a first spectrum with at least one first spectral property is removed from the spectrum of the audio data, the resulting spectrum of the audio data is transformed after removal of the at least one first spectrum, and the at least one first spectrum or at least one of the first spectrum and/or at least one second spectrum with at least one second spectral property are impressed on the transformed spectrum. | 03-17-2011 |
20110116655 | Apparatus and Method for Audio Conversion - An apparatus and method for audio conversion is provided to upgrade the resolution of transmission frequency of an FM (frequency modulation) transmitter and reduce the size of the FM transmitter by applying frequency coarse tune and fine tune. The apparatus comprises a digital FM modulator, a digital frequency synthesizer, a signal converter, and an analog frequency converter. The digital FM modulator modulates a digital audio input signal into a first digital audio signal. The digital frequency synthesizer converts the first digital audio signal into a second digital audio signal, whose frequency is determined according to a first frequency conversion parameter. The signal converter converts the second digital audio signal into an analog audio signal. The analog frequency converter generates an audio transmission signal with a predetermined frequency according to a second clock signal and the analog audio signal while the second clock signal is generated according to a first clock signal. | 05-19-2011 |
20110123046 | SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD, AND PROGRAM THEREFOR - A signal processing apparatus includes: a separation processing unit that generates observed signals in the time frequency domain by performing the short-time Fourier transform on mixed signals as outputs, which are acquired from a plurality of sound sources by a plurality of sensors, and generates sound source separation results corresponding to the sound sources by a linear filtering process on the observed signals. The separation processing unit has a linear filtering process section that performs the linear filtering process on the observed signals so as to generate separated signals corresponding to the respective sound sources, an all-null spatial filtering section that applies an all-null spatial filter to generate signals filtered with the all-null spatial filter (spatially filtered signals) in which the acquired sounds in null directions are removed, and a frequency filtering section that performs a filtering process by inputting the separated signals and the spatially filtered signals. | 05-26-2011 |
20110135112 | AUDIO SIGNAL PROCESSING DEVICE AND AUDIO SIGNAL PROCESSING METHOD - A technique is provided that can reduce any unnecessary distortion components of an input audio signal that arises as a result of varying the gain of the signal regardless of the type of the input audio signal. | 06-09-2011 |
20110142258 | Apparatus for Processing an Audio Signal - An apparatus for processing an audio signal to focus an acoustic signal by an arrangement of a plurality of loudspeakers comprises a frequency analyzer, a signal processor and a signal output interface. The acoustic signal is based on the audio signal. The frequency analyzer is configured to determine a fundamental frequency in a frequency spectrum of the audio signal depending on a geometry parameter of the arrangement of the plurality of loudspeakers. The signal processor is configured to adapt an overtone of the fundamental frequency to obtain the processed audio signal and the signal output interface is configured to output the processed audio signal to the plurality of loudspeakers. | 06-16-2011 |
20110150240 | MODULATION DEVICE AND DEMODULATION DEVICE - A modulation device includes: a spread code generation unit which generates a spread code having a predetermined cycle; an audio signal input unit to which an audio signal is input; a first modulation unit which phase-modulates the spread code in each cycle on the basis of a data code; and a combining unit which combines the audio signal with a modulation signal which has been generated on the basis of the phase-modulated spread code and distributed in a frequency range higher than a predetermined frequency to output a combined signal. | 06-23-2011 |
20110170710 | METHOD AND APPARATUS FOR ADJUSTING VOLUME - A method of adjusting volume removes an audio signal of a first frequency band from audio signals, and increases the volume of the audio signal from which the signal of the first frequency band is removed. | 07-14-2011 |
20110170711 | Audio Encoder, Audio Decoder, Methods for Encoding and Decoding an Audio Signal, and a Computer Program - An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. | 07-14-2011 |
20110170712 | SOUND REPRODUCING APPARATUS - In a sound reproducing apparatus, part of a frequency band where mode-coupled vibration can be excited is regarded as a carrier frequency. A frequency of mode coupling, with a low rate of change in vibration displacement with respect to the frequency, is regarded as a carrier signal so that a signal in an audible band which is outputted from an audible band signal source can be demodulated and reproduced with stable sound pressure in a broad frequency band. | 07-14-2011 |
20110188672 | ACOUSTIC REPRODUCTION DEVICE - A sound reproduction apparatus reproduces a sound wave at a listening position. The sound reproduction apparatus includes a compensation processing unit for compensating an audible band signal having an audible band frequency, a carrier signal oscillator for generating a carrier signal, a modulator for outputting a modulated signal obtained by modulating the carrier signal based on the audible band signal compensated by the compensation processing unit, and a sound emission unit for outputting a sound wave depending on the modulated signal output from the modulator. The compensation processing unit compensates the audible band signal based on a distance from the sound emission unit to the listening position. This sound reproduction apparatus can reproduce the original audible band signal with a high fidelity regardless of the listening position. | 08-04-2011 |
20110216918 | Apparatus and Method for Generating a Bandwidth Extended Signal - An apparatus for generating a bandwidth extended signal from an input signal includes a patch generator and a combiner. The input signal is represented for first and second bands by first and second resolution data, respectively, the second resolution being lower than the first. The patch generator generates first and second patches from the first band of the input signal according to first and second patching algorithms, respectively. A spectral density of the second patch generated using the second patching algorithm is higher than a spectral density of a first patch generated using the first patching algorithm. The combiner combines both patches and the first band of the input signal to obtain the bandwidth extended signal. The apparatus scales the input signal according to the first and second patching algorithms or scales the first and second patches, so that the bandwidth extended signal fulfills a spectral envelope criterion. | 09-08-2011 |
20110274292 | ACOUSTIC CHARACTERISTIC CORRECTION COEFFICIENT CALCULATION APPARATUS, ACOUSTIC CHARACTERISTIC CORRECTION COEFFICIENT CALCULATION METHOD AND ACOUSTIC CHARACTERISTIC CORRECTION APPARATUS - According to one embodiment, an acoustic characteristic correction coefficient calculation apparatus includes a frequency converter, a smoother, a frequency inverter, a cutter, and a calculator. The frequency converter is configured to convert a first impulse response corresponding to an input acoustic signal to a frequency domain. The smoother is configured to smooth amplitude and phase corresponding to the frequency domain converted by the frequency converter. The frequency inverter is configured to convert a frequency characteristic smoothed by the smoother to a time domain. The cutter is configured to cut out a second impulse response configured by the time domain obtained by converting the frequency characteristic by use of the frequency inverter by a preset tap number. The calculator is configured to calculate a correction coefficient used to correct an acoustic characteristic based on a result cut out by the cutter. | 11-10-2011 |
20110299704 | Frequency-tracked synthesizer employing selective harmonic amplification and/or frequency scaling - This invention relates to effects processing of a monophonic analog signal, meaning a signal whose frequency components are all integer multiples of a first fundamental frequency. For example, the signal could come from almost any musical instrument, voice included. However, for generality, the invention is not restricted to cases where the signal source is musical. The digital signal processing is simplified as a result of the DSP being clocked at a constant multiple of f | 12-08-2011 |
20110305352 | Cross Product Enhanced Harmonic Transposition - The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal. | 12-15-2011 |
20110317849 | SIGNAL CONDITIONER FOR SUBWOOFERS - A signal conditioner for subwoofers has a band pass filter which passes a lower frequency spectrum of an input signal. The band pass filter includes a high pass filter having a predetermined corner frequency, and a low pass filter having about the same corner frequency, and produces a filtered input to a peak voltage follower circuit. The peak voltage follower circuit traces and smoothes the upper boundary of the filtered input to produce a control voltage. An analog multiplier multiplies the input signal by the control voltage produced by the peak voltage follower to produce an enhanced signal where low frequency transients have been magnified. | 12-29-2011 |
20110317850 | GENERATOR AND GENERATION METHOD OF PSEUDO-BASS - An absolute-value circuit outputs the absolute value of a signal SIN′ that corresponds to an input signal SIN. A clipping circuit clips the signal SIN′ that corresponds to the input signal, to a positive limit value and to a negative limit value. A first multiplier multiplies the signal SIN′ that corresponds to the input signal, by a predetermined coefficient. A first adder subtracts the output signal of the first multiplier from the output signal of the clipping circuit. A second adder sums a signal that corresponds to the output signal of the first adder and a signal that corresponds to the output signal of the absolute-value circuit. A third adder sums the input signal SIN and a signal that corresponds to the output signal of the second adder. | 12-29-2011 |
20110317851 | AUDIO SIGNAL MIXING DEVICE - An audio signal mixer can be applied to a DJ device and the like, for example. The audio signal mixer acquires a plurality of audio signals and acquires mixing rate information being information on a mixing rate of a plurality of the audio signals. The audio signal mixer decides an adjusting amount of balance between levels at every band when a plurality of the audio signals are mixed based on the mixing rate. A plurality of the audio signals are mixed at every band, adjusts the levels of the mixed signals of the respective bands based on the adjusting amount, adds the adjusted signals of the respective bands and outputs it as a mixed audio signal. | 12-29-2011 |
20110317852 | FREQUENCY CHARACTERISTICS CONTROL DEVICE - A frequency characteristics control device of a mixer mixes a first audio signal and a second audio signal inputted to the mixer. In the frequency characteristics control device, a characteristics detection section detects a first frequency characteristic of the first audio signal and a second frequency characteristic of the second audio signal. Based on the first and second frequency characteristics, a removal band detection section detects a removal band in which a level of the first audio signal is higher than a level of the second audio signal. A filtering process section performs a filtering process on the second audio signal so as to attenuate a component of the second audio signal in the removal band. A output section mixes the first audio signal inputted to the mixer and the second audio signal on which the filtering process section has performed the filtering process. | 12-29-2011 |
20120002823 | ACOUSTIC CORRECTION APPARATUS, AUDIO OUTPUT APPARATUS, AND ACOUSTIC CORRECTION METHOD - According to one embodiment, an acoustic correction apparatus includes an input module, a calculator, a divider, a converter, an extractor, a synthesizer, and a generator. The input module receives an audio signal propagated through a sound field. The calculator calculates an impulse response from the audio signal. The divider divides the impulse response into first and second impulse responses. The converter converts the first and second impulse responses into first and second frequency spectrums. The extractor specifies an amplitude component of the first frequency spectrum with a peak relatively higher than that of the amplitude component of the first frequency spectrum, and extracts the peak as a resonance component. The synthesizer synthesizes a first property and a second property for attenuating the resonance component. The generator generates a correction filter for performing correction to obtain the synthesized property. | 01-05-2012 |
20120002824 | AUDIO EQUIPMENT AND A SIGNAL PROCESSING METHOD THEREOF - The present invention relates to an audio equipment ( | 01-05-2012 |
20120008798 | Method and Apparatus For Stereo Enhancement Of An Audio System - A processing apparatus which is suitable for signal communication with a system having at least one of an input module and an output module. The processing apparatus can be configured to receive input signals from the input module of the system. The processing apparatus includes a first channel processing portion to which the input signals are communicable. The first channel processing portion can be configurable to receive and process the input signals in a manner such that bass frequency audio signals are extracted from the input signals. The bass frequency audio signals can be further processed via at least one of linear dynamic range processing, manipulation of dynamic range via compression and manipulation of dynamic range via expansion. | 01-12-2012 |
20120008799 | APPARATUS AND METHOD FOR DETERMINING A PLURALITY OF LOCAL CENTER OF GRAVITY FREQUENCIES OF A SPECTRUM OF AN AUDIO SIGNAL - An apparatus for determining a plurality of local center of gravity frequencies of a spectrum of an audio signal includes an offset determiner, a frequency determiner and an iteration controller. The offset determiner determines an offset frequency for each iteration start frequency of a plurality of iteration start frequencies based on the spectrum of the audio signal, wherein a number of discrete sample values of the spectrum is larger than a number of iteration start frequencies. The frequency determiner determines a new plurality of iteration start frequencies by increasing or reducing each iteration start frequency of the plurality of iteration start frequencies by the corresponding determined offset frequency. The iteration controller provides the new plurality of iteration start frequencies to the offset determiner for further iteration or provides the plurality of local center of gravity frequencies, if a predefined termination condition is fulfilled. The plurality of local center of gravity frequencies can be utilized as a basis for generating a new plurality of iteration start frequencies. | 01-12-2012 |
20120020496 | Fast Acoustic Cancellation - A speech enhancement system improves the perceptual quality of an aural signal. A receiver detects and receives an unvoiced signal, a fully voiced signal, or a mixed voice remote signal. A coherence processor identifies the similarities or differences between a local signal and the remote signal. A cancellation processor or controller dampens reflected signals that may be part of the local signal. | 01-26-2012 |
20120033830 | AUDIO PROCESSING DEVICE AND AUDIO PROCESSING METHOD - There is provided a sound processing apparatus and a sound processing method which are capable of reproducing discrete data with a high-quality sound matching users' preferences. In a sound processing means | 02-09-2012 |
20120033831 | Microphone with Adjustable Characteristics - A microphone comprises a movable diaphragm | 02-09-2012 |
20120045075 | Energy saving PWM sound device - An energy saving PWM sound device forms a frequency-dependent impedance device implemented on the PWM output of the voice-ICs in series to lower the power consumption and to enhance sound quality of the voice-IC products. The device includes a hollow core body and an elongated conductive element. The core body has a top rim, a bottom rim, and two side rims to define a through channel therewithin. The conductive element winds around the top rim of the core body, wherein two ends of the conductive element are adapted for connecting with the PWM output, such that when the conductive element is electrically conducted, the core body acts as an inductor at lower frequency and a frequency-dependent resistor in series with the inductor at higher frequency for blocking most of PWM carrier frequency so as to reduce the power consumption of the voice-ICs product. | 02-23-2012 |
20120063614 | AUDIO SIGNAL DYNAMIC EQUALIZATION PROCESSING CONTROL - Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile. | 03-15-2012 |
20120063615 | EQUALIZATION PROFILES FOR DYNAMIC EQUALIZATION OF AUDIO DATA - Apparatuses for and methods of carrying out dynamic equalization processing of an audio signal, and apparatuses for and methods of controlling such equalization processing of the audio signal to dynamically adjust the time-varying spectrum of an audio signal to more closely match a user specified target time-invariant perceived audio signal spectrum while preserving the original dynamic range of the audio signal. The dynamic equalization is carried out according to a user-defined spectral profile specified by a control interface that allows a user to define, create, modify and/or apply the user-defined spectral profile. | 03-15-2012 |
20120076324 | SYSTEM AND METHODS FOR APPLYING BASS COMPENSATION IN AN AUTOMOBILE - Systems and methods for providing bass compensation to correct for uneven bass response are disclosed. An example bass compensation system includes a low pass filter configured to receive an audio signal from an audio source and provide a filtered audio signal, the low pass filter having a roll off of at least 18 dB per octave. The bass compensation system further includes a summing amplifier coupled to the low pass filter and configured to sum the audio signal from said audio source and the filtered audio signal to provide a summed audio signal, wherein the summed audio signal provided by the summing amplifier provides a bass boost at a first frequency and mid bass cut at a second frequency greater than the first frequency. | 03-29-2012 |
20120114142 | ACOUSTIC SIGNAL PROCESSING APPARATUS, PROCESSING METHOD THEREFOR, AND PROGRAM - The present invention relates to an acoustic signal processing apparatus that suppresses auditory noise caused in a difference signal generated by acoustic signals of a plurality of channels, a processing method therefor, and a program. | 05-10-2012 |
20120114143 | SOUND OUTPUT APPARATUS - A sound output apparatus includes a mixing unit configured to mix an L (left) channel voice signal and an R (right) channel voice signal to generate an L+WF signal and an R−WF signal containing a mix of a WF(woofer) channel voice signal, and power amplifiers which amplify the L+WF signal and R−WF signal. The outputs of the power amplifiers are BTL-connected to a WF channel speaker. Thus, the mixing unit provided in a previous stage of the power amplifiers generates the WF channel voice signal, which eliminates the necessity for a passive low pass filter in a subsequent power amplifier stage. | 05-10-2012 |
20120121107 | Method for Signal Processing of Solid-Borne Sound Signals, in Particular in Motor Vehicles, and an Occupant Protection System with Corresponding Signal Processing Unit - The invention relates to a method for signal processing of solid-borne sound signals, in particular in motor vehicles, and a corresponding occupant protection system. A first-order high-pass filter is provided as a filter, wherein the −3 dB cut-off frequency thereof lies between the upper and lower operating frequency. | 05-17-2012 |
20120128178 | SOUND REPRODUCING APPARATUS, SOUND REPRODUCING METHOD, AND PROGRAM - Sound reproduction is controlled so as to be heard in the optimal state for the hearing function specific to elderly people. A frequency characteristic setting portion for setting the frequency characteristics of an inputted sound signal, and a sound volume setting portion for variable controlling the volume is disclosed. The frequency characteristic setting portion changes a frequency characteristic in which a sound band including a human voice band is emphasized to a frequency characteristic in which the characteristics of a gain in accordance with a frequency gradually becomes flat with an increase in the volume set by the sound volume setting portion. As a result, the frequency band of the human voice is strongly emphasized so as to help elderly people hear it at a low volume, and, as the volume grows higher, a frequency characteristic is changed to a flatter frequency characteristic, thereby making it possible to output easy-to-hear sound while reducing inconvenience caused by the emphasis of a specific frequency band. | 05-24-2012 |
20120195441 | METHOD OF OUTPUTTING AUDIO SIGNAL AND AUDIO SIGNAL OUTPUT APPARATUS USING THE METHOD - An audio signal output apparatus includes a modulation signal generator for generating a first modulation signal by pulse-modulating an input audio signal of one channel using a first carrier signal or a first sampling clock, which has a first frequency; a vacuum tube filter unit comprising a vacuum tube and for generating a vacuum tube signal by allowing the first modulation signal to pass through the vacuum tube; a frequency modulation unit for generating a second modulation signal by pulse-modulating the vacuum tube signal; and a power switching amplifier for outputting an amplification signal corresponding to the second modulation signal. | 08-02-2012 |
20120195442 | OVERSAMPLING IN A COMBINED TRANSPOSER FILTER BANK - The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank ( | 08-02-2012 |
20120201399 | SOUND SIGNAL PROCESSING APPARATUS, SOUND SIGNAL PROCESSING METHOD, AND PROGRAM - A sound signal processing apparatus includes a frequency analysis unit which executes frequency analysis of an input sound signal; a low-frequency envelope calculating unit which calculates low-frequency envelope information as envelope information of a low-frequency band based on a result of the frequency analysis; a high-frequency envelope information estimating unit which applies learned data generated in advance based on a sound signal for learning and generates estimated high-frequency envelope information corresponding to an input signal from the low-frequency envelope information corresponding to the input sound signal; and a frequency synthesizing unit which synthesizes a high-frequency band signal corresponding to the estimated high-frequency envelope information generated by the high-frequency envelope information estimating unit with the input sound signal and generates an output sound signal in which a frequency band is expanded. | 08-09-2012 |
20120219164 | PRODUCING SOUNDS IN A VIRTUAL WORLD AND RELATED SOUND CARD - Disclosed are a method and apparatus for producing sounds in a virtual world, as well as a sound card. The method comprises the steps of determining a spring mass model of an object in the virtual world based on a 3D model of the object; analyzing force components produced on the 3D model by a collision in the virtual world; and generating sounds produced by the collision according to the spring mass model and the force components. By considering sound material of the object in the virtual world, the method, apparatus, and sound card may produce sounds in the virtual world more vividly and in real time. | 08-30-2012 |
20120224719 | VIBRATION CONTROL - A device may include a speaker, a vibration device and logic. The speaker may be configured to output audio signals. The logic may be configured to perform audio spectrum analysis associated with the audio signals, and synchronize output of the vibration device with the audio signals output by the speaker based on the audio spectrum analysis. The vibration device may also be configured to vibrate at a number of different frequencies based on the audio spectrum analysis. | 09-06-2012 |
20120230515 | BANDWIDTH EXTENSION OF A LOW BAND AUDIO SIGNAL - Estimation of a high band extension of a low band audio signal includes the following steps: extracting (S | 09-13-2012 |
20120243706 | Method and Arrangement for Processing of Audio Signals - Method and arrangement in an audio handling entity, for damping of dominant frequencies in a time segment of an audio signal. A time segment of an audio signal is obtained, and an estimate of the spectral density or “spectrum” of the time segment is derived. An approximation of the estimate is derived by smoothing the estimate, and a frequency mask is derived by inverting the approximation. Frequencies comprised in the audio time segment are then damped based on the frequency mask. The method and arrangement involves no multi-band filtering or selection of attack and release times. | 09-27-2012 |
20120243707 | SYSTEM AND METHOD FOR PROCESSING SOUND SIGNALS IMPLEMENTING A SPECTRAL MOTION TRANSFORM - A system and method are provided for processing sound signals. The processing may include identifying individual harmonic sounds represented in sound signals, determining sound parameters of harmonic sounds, classifying harmonic sounds according to source, and/or other processing. The processing may include transforming the sound signals (or portions thereof) into a space which expresses a transform coefficient as a function of frequency and chirp rate. This may facilitate leveraging of the fact that the individual harmonics of a single harmonic sound may have a common pitch velocity (which is related to the chirp rate) across all of its harmonics in order to distinguish an the harmonic sound from other sounds (harmonic and/or non-harmonic) and/or noise. | 09-27-2012 |
20120243708 | Modulation of Audio Signals in a Parametric Speaker - Methods and systems for amplitude modulation in a parametric speaker system are provided that perform truncated double sideband (TDSB) frequency modulation of audio signal in which most of the processing is performed in the frequency domain, thus permitting use of fast processing techniques for amplitude modulation (AM) and filtering and reducing computation cost over time domain processing. A maximum envelope value of the time domain audio signal may be to the carrier signal in the frequency domain that avoids emitting the carrier signal when the input signal level is low or mute. The application of the envelope value may be smoothed to reduce discontinuity at input block boundaries. | 09-27-2012 |
20120250886 | CHARACTERISTIC CORRECTING DEVICE AND CHARACTERISTIC CORRECTING METHOD - According to one embodiment, a characteristic correcting device includes: a correction filter configured to correct sound quality characteristics of a plurality of bands in a frequency range of an input signal based on a frequency characteristic which is set in advance to generate an output signal; an input module configured to input a surrounding sound signal of sound around an output device outputting the output signal; and an adjusting module configured to reduce the number of the bands of which the sound quality characteristics are to be corrected of the bands in the frequency range of the input signal in accordance with an increase in amplitude of the surrounding sound signal which is input. | 10-04-2012 |
20120257769 | METHOD, SYSTEM AND APPARATUS FOR IMPROVING THE SONIC QUALITY OF AN AUDIO SIGNAL - A device, system and method of playing back a digital audio stream wherein large amounts of pre-emphasis of the high frequencies is applied before the digital to analog conversion and before an interpolation or digital filter, followed by de-emphasis in the analog domain in order to yield better audio fidelity. | 10-11-2012 |
20120281859 | APPARATUS AND METHOD FOR GENERATING A HIGH FREQUENCY AUDIO SIGNAL USING ADAPTIVE OVERSAMPLING - An apparatus for generating a high frequency audio signal that includes an analyzer for analyzing an input signal to determine a transient information adaptively. Additionally a spectral converter is provided for converting the input signal into an input spectral representation. A spectral processor processes the input spectral representation to generate a processed spectral representation including values for higher frequencies than the input spectral representation. A time converter is configured for converting the processed spectral representation to a time representation, wherein the spectral converter or the time converter are controllable to perform a frequency domain oversampling for the first portion of the input signal having the transient information associated and to not perform the frequency domain oversampling for the second portion of the input signal not having the associated transient information. | 11-08-2012 |
20120288118 | CONTROL OF A LOUDSPEAKER OUTPUT - A method of modeling the frequency-dependent input-voltage-to-excursion transfer function of a loudspeaker, comprises, for a plurality of measurement frequencies, measuring a voltage and current and deriving an impedance at the measurement frequency. A frequency-dependent impedance function is derived. | 11-15-2012 |
20120294457 | Audio System and Method of Using Adaptive Intelligence to Distinguish Information Content of Audio Signals and Control Signal Processing Function - An audio system has a signal processor coupled for receiving an audio signal from a musical instrument or vocals. A time domain processor receives the audio signal and generates time domain parameters of the audio signal. A frequency domain processor receives the audio signal and generates frequency domain parameters of the audio signal. The audio signal is sampled and the time domain processor and frequency domain processor operate on a plurality of frames of the sampled audio signal. The time domain processor detects onset of a note of the sampled audio signal. A signature database has signature records each having time domain parameters and frequency domain parameters and control parameters. A recognition detector matches the time domain parameters and frequency domain parameters of the audio signal to a signature record of the signature database. The control parameters of the matching signature record control operation of the signal processor. | 11-22-2012 |
20120294458 | SEMICONDUCTOR INTEGRATED CIRCUIT OF CAR NAVIGATION SYSTEM AND MULTIMEDIA PROCESSING METHOD APPLIED TO CAR NAVIGATION SYSTEM INTEGRATED WITH FM/AM BROADCAST RECEIVING FUNCTION - A semiconductor IC includes an analog audio input circuit, a selecting unit, and an audio processing circuit, which maybe a car navigation chip. The analog audio input circuit includes an RF module and at least one analog audio input module, respectively for providing a first analog audio input signal and at least one second analog audio input signal. The RF module includes an FM/AM broadcast receiving function. The selecting unit is coupled to the RF module and analog audio input module, and outputs a target analog audio input signal according to the first and second analog audio input signals. The audio processing circuit is coupled to the selecting unit, and performs an audio signal process upon the target analog audio signal to generate an audio output signal. | 11-22-2012 |
20120294459 | Audio System and Method of Using Adaptive Intelligence to Distinguish Information Content of Audio Signals in Consumer Audio and Control Signal Processing Function - A consumer audio system has a signal processor coupled for receiving an audio signal. The audio signal is sampled into a plurality of frames. The sampled audio frames are separated into sub-frames according to the type or frequency content of the sound generating source. A time domain processor generates time domain parameters from the separated sub-frames. A frequency domain processor generates frequency domain parameters from the separated sub-frames. The time domain processor or frequency domain processor can detects onset of a note of the audio signal. A signature database has signature records each having time domain parameters and frequency domain parameters and control parameters. A recognition detector matches the time domain parameters and frequency domain parameters of the separated sub-frames to a signature record of the signature database. The control parameters of the matching signature record control operation of the signal processor. | 11-22-2012 |
20120294460 | SOUND PROCESSING APPARATUS AND PARAMETER SETTING METHOD - A sound processing apparatus includes a processing unit that is configured to acquire an audio signal, perform a correction processing on the acquired audio signal and output the correction-processed audio signal to a sound emitting unit. The correction processing includes an indirect sound adjusting processing in which a given signal processing is performed on an audio signal so as to adjust an influence of an indirect sound to be heard at a sound receiving point, and a frequency characteristic adjusting processing in which a frequency characteristic of an audio signal is adjusted. In the correction processing, a frequency characteristic for the frequency characteristic adjusting processing is determined based on a frequency characteristic of the indirect sound adjusting processing. | 11-22-2012 |
20120308042 | Subwoofer Volume Level Control - An automatic adjustment of subwoofer volume levels relative to the main audio system volume and more specifically, to an improved algorithm control method for sub varying volume which allows adjustment of only the low frequency tones from the subwoofer relative to the main system volume. | 12-06-2012 |
20120308043 | Acoustic Manipulator Element - According to an exemplary embodiment of the present invention, an acoustic manipulator element is provided. The acoustic manipulator element is arrangable relatively to an acoustic source in a manner that the acoustic manipulator element splits frequency selectively sound waves originating from the acoustic source in a reflected and a through component, wherein at least a portion of the acoustic waves of the through component is attenuated by at most 15 dB for acoustic frequencies having a wavelength between 200 Hz and 16000 Hz compared to the sound waves of the acoustic source. | 12-06-2012 |
20120321103 | IN-EAR HEADPHONE - A headphone device includes a housing having a leakage hole to reduce pressure between a user's ear and the housing, a speaker positioned within the housing, and an audio processing module. The audio processing module is configured to receive an audio signal from an audio device, determine whether the audio signal includes at least a predetermined level of audio having a frequency in a first range of frequencies, transmit a first leakage control signal to a leakage hole valve when it is determined that the audio includes at least the predetermined level of low frequency audio; and transmit a second leakage control signal to the leakage hole valve when it is determined that the audio does not include at least the predetermined level of low frequency audio. The leakage hole valve is configured to close the leakage hole upon receipt of the first leakage control signal and open the leakage hole upon receipt of the second leakage control signal. | 12-20-2012 |
20120328123 | SIGNAL PROCESSING APPARATUS, SIGNAL PROCESSING METHOD, AND PROGRAM - Provided is a signal processing apparatus, including a filter unit that filters an audio signal created by decimating a portion of frequency components by an all-pass filter and outputs a filtering result thereof as improvement components to improve sound quality of the audio signal, and an adder that generates an improved sound in which the sound quality of the audio signal is improved by adding the improvement components to the audio signal. | 12-27-2012 |
20120328124 | Processing of Audio Signals During High Frequency Reconstruction - The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described. The system comprises means for receiving the plurality of low frequency subband signals; means for receiving a set of target energies, each target energy covering a different target interval within the high frequency interval and being indicative of the desired energy of one or more high frequency subband signals lying within the target interval; means for generating the plurality of high frequency subband signals from the plurality of low frequency subband signals and from a plurality of spectral gain coefficients associated with the plurality of low frequency subband signals, respectively; and means for adjusting the energy of the plurality of high frequency subband signals using the set of target energies. | 12-27-2012 |
20130016856 | AUDIO DEVICEAANM Koike; HiroyukiAACI KanagawaAACO JPAAGP Koike; Hiroyuki Kanagawa JP - Disclosed is an audio device that plays back compressed audio signals that can be sufficiently comfortable to be audible even in a play back environment with improved acoustic quality. In order to correct a weak signal component near a frequency with a high output level of a compressed audio, weak-signal component adding unit ( | 01-17-2013 |
20130028442 | Loudspeaker system - A loudspeaker that can be bi-wired or not according to a user's preference comprises a case in which is provided a plurality of audio drivers supplied by a respective plurality of audio networks, the networks being supplied (in turn) by a respective plurality of input terminal pairs, further comprising at least one switch bridging the terminals of different pairs, and adapted to selectively connect the terminals to each other. | 01-31-2013 |
20130044896 | Virtual Bass Synthesis Using Harmonic Transposition - In some embodiments, a virtual bass generation method including steps of: performing harmonic transposition on low frequency components of an input audio signal (typically, bass frequency components expected to be inaudible during playback of the input audio signal using an expected speaker or speaker set) to generate transposed data indicative of harmonics (which are expected to be audible during playback, using the expected speaker(s), of an enhanced version of the input audio which includes the harmonics); generating an enhancement signal in response to the transposed data; and generating an enhanced audio signal by combining (e.g., mixing) the enhancement signal with the input audio signal. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method. | 02-21-2013 |
20130051581 | AUDIO SIGNAL PROCESSING CIRCUIT - An audio signal processing circuit includes: a first low-pass filter configured to pass a component whose frequency is in a band lower than a lowest reproducible frequency of a speaker out of an audio signal inputted for reproduction by the speaker; a first high-pass filter substantially similar in phase characteristics to the first low-pass filter configured to pass a component whose frequency is in a band higher than the lowest reproducible frequency of the speaker out of the audio signal inputted for reproduction by the speaker; a harmonic generation unit configured to generate a harmonic from the audio signal having passed through the first low-pass filter; and a first addition unit configured to add the audio signal according to an output of the harmonic generation unit to the audio signal according to an output of the first high-pass filter. | 02-28-2013 |
20130058499 | INFORMATION PROCESSING APPARATUS AND INFORMATION PROCESSING METHOD - An information processing apparatus includes: a sound source configured to generate sound waves; and a housing configured to incorporate the sound source and include a first main surface and a second main surface opposite to each other, the first main surface including a groove-like concave portion configured to communicate in one axial direction along the second main surface, and an opening for transmitting sound waves of the sound source, the opening being provided in an area of the concave portion and at a position offset from the middle in the one axial direction of the first main surface. | 03-07-2013 |
20130058500 | FREQUENCY BAND EXTENDING APPARATUS, FREQUENCY BAND EXTENDING METHOD, PLAYER APPARATUS, PLAYING METHOD, PROGRAM AND RECORDING MEDIUM - A player apparatus for playing an input signal after band-extending the input signal includes: an extension controller to determine an extension start band for the input signal in accordance with information relating to the input signal; and a band divider to divide the input signal into a plurality of sub-band signals. The frequency band is extended on the basis of a plurality of the sub-band signals on a side lower than the extension start band, among the plurality of sub-band signals into which the input signal is band-divided by the band divider. | 03-07-2013 |
20130083945 | NOVEL EFFICIENT DIGITAL MICROPHONE DECIMATION FILTER ARCHITECTURE - A new and more efficient filtering system (e.g., digital microphone decimation filter architecture system) is described. A key to this architecture is the use of two parallel filter paths. Each path operates at the output sample rate, and comprises a shorter FIR filter followed by a series of allpass stages (e.g., implementing IIR filters). The FIR filter is designed to remove all but the last octave of out-of-band noise. The allpass stages are designed such that when the two paths are summed together, the out-of-band noise for the final octave cancels out, leaving only the desired sign | 04-04-2013 |
20130094665 | DEVICE AND METHOD FOR REPRODUCING AN AUDIO SIGNAL - A device and method for controlling reproduction of an audio signal is provided, wherein the device is operated by means of an energy storage device. The method comprises the steps of deactivating a normal mode and activating an energy saving mode. Power consumption from the energy storage device for reproduction of the audio signal is reduced in the energy saving mode when compared to the normal mode. The method comprises reducing in the energy saving mode, a bass frequency component of a frequency spectrum of the audio signal and outputting the audio signal with reduced bass frequency component. The method further comprises ascertaining a charge state of the energy storage device and controlling the reduction in the bass frequency component based on a decrease in the charge state of the energy storage device. | 04-18-2013 |
20130108076 | Compensating for Different Audio Clocks Between Devices Using Ultrasonic Beacon | 05-02-2013 |
20130108077 | Device and Method for Processing a Real Subband Signal for Reducing Aliasing Effects | 05-02-2013 |
20130114829 | Recursive audio modulation system using nested inductor arrays - Nested inductor arrays magnetically modulate an analog audio input signal recursively, so that the overall amplitude envelope of the output signal replicates the wave pattern of the input signal. The nested inductor arrays produce multiple levels of recursive modulation, so that the output signal incorporates multiple integrated self-similar harmonic layers, such that the phasing of the various layers are locked in by the analog waveform of the output signal itself. As a result, the spatial “depth” and temporal “immediacy” of the original analog recording is restored and can be encoded in digital format. | 05-09-2013 |
20130114830 | DETERMINING A CONFIGURATION FOR AN AUDIO PROCESSING OPERATION - A computer-implemented method of determining a configuration for an audio processing operation, wherein the audio processing operation comprises a predetermined set of one or more audio processing sub-operations, each audio processing sub-operation being configurable with one or more respective configuration parameters, the method comprising: specifying the predetermined set of one or more audio processing sub-operations; specifying a target frequency response; and performing a convergent optimization process to determine a configuration for the audio processing operation that reduces a difference between the frequency response of the audio processing operation and the target frequency response, wherein the configuration comprises a respective value for each configuration parameter of each audio processing sub-operation. | 05-09-2013 |
20130114831 | METHODS AND APPARATUS TO PERFORM AUDIO WATERMARKING AND WATERMARK DETECTION AND EXTRACTION - Encoding and decoding methods and apparatus as described. An example method of obtaining auxiliary information in an audio signal using a plurality of frequency components residing in a plurality of code bands comprises transforming an audio signal into a frequency domain representation; determining characteristic of frequencies of the frequency domain representation that may contain the auxiliary information; normalizing across the code bands the characteristics of frequencies of the frequency domain representation in a respective one of the code bands that may contain the auxiliary information, wherein the normalization is carried out against a characteristic of a frequency in that code band; summing the normalized characteristics of the frequencies representative of auxiliary information to determine a sum for a frequency representative of auxiliary information; and determining that the sum is representative of the auxiliary information. | 05-09-2013 |
20130121507 | SYSTEM AND METHOD FOR DIGITAL SIGNAL PROCESSING - The present invention provides methods and systems for digitally processing audio signals. Some embodiments receive an audio signal and converting it to a digital signal. The gain of the digital signal may be adjusted a first time, using a digital processing device located between a receiver and a driver circuit. The adjusted signal can be filtered with a first low shelf filter. The systems and methods may compress the filtered signal with a first compressor, process the signal with a graphic equalizer, and compress the processed signal with a second compressor. The gain of the compressed signal can be adjusted a second time. These may be done using the digital processing device. The signal may then be output through an amplifier and driver circuit to drive a personal audio listening device. In some embodiments, the systems and methods described herein may be part of the personal audio listening device. | 05-16-2013 |
20130121508 | Non-Speech Content for Low Rate CELP Decoder - A method and device for modifying a synthesis of a time-domain excitation decoded by a time-domain decoder, wherein the synthesis of the decoded time-domain excitation is classified into one of a number of categories. The decoded time-domain excitation is converted into a frequency-domain excitation, and the frequency-domain excitation is modified as a function of the category in which the synthesis of the decoded time-domain excitation is classified. The modified frequency-domain excitation is converted into a modified time-domain excitation, and a synthesis filter is supplied with the modified time-domain excitation to produce a modified synthesis of the decoded time-domain excitation. | 05-16-2013 |
20130129114 | CLOCK GENERATOR - A clock generator receives first and second clock signals, and input representing a desired frequency ratio. A comparison is made between frequencies of an output clock signal and the first clock signal, and a first error signal represents the difference between the desired frequency ratio and this comparison result. The first error signal is filtered. A comparison is made between frequencies of the output clock signal and the second clock signal, and a second error signal represents the difference between the filtered first error signal and this comparison result. The second error signal is filtered. A numerically controlled oscillator receives the filtered second error signal and generates an output clock signal. As a result, the output clock signal has the jitter characteristics of the first input clock signal over a useful range of jitter frequencies and the frequency accuracy of the second input clock signal. | 05-23-2013 |
20130142359 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD - A signal processing apparatus includes: a control section; a signal processing section connected with a plurality of signal processing elements and configured to perform signal processing for enhancing or attenuating an input signal in a specific frequency band; and a crossfade signal section including a crossfade signal processing element capable of replacing at least one of the signal processing elements, wherein the control section is configured to control any one of the signal processing elements among the plurality of signal processing elements, and the crossfade signal processing element, to crossfade to the crossfade signal processing element having the signal processing element as a new characteristic, to perform processing for replacing any one of the signal processing elements by the crossfade signal processing element, and to perform the processing on remaining signal processing elements of the plurality of signal processing elements in the signal processing section. | 06-06-2013 |
20130142360 | METHOD AND SYSTEM FOR CONTROLLING DISTORTION IN A CRITICAL FREQUENCY BAND OF AN AUDIO SIGNAL - In some embodiments, a method and system for controlling distortion of the output of a miniature speaker by attenuating critical frequency band of the input signal to be reproduced, using tuning parameters that have been predetermined where the critical frequency band is a frequency range of the speaker's frequency response in which Total Harmonic Distortion (THD) peaks. The distortion control is performed in a manner which allows an increase in the average loudness of the speaker's output without significantly increasing distortion. The tuning parameters include a center frequency and a bandwidth of the critical frequency band, and a power threshold value. In some embodiments, the system is a loudness maximizer configured to limit distortion of a speaker's output by limiting distortion in a critical frequency band using predetermined control parameters, and limit the dynamic range of the output signal and increase its perceived overall average loudness level. | 06-06-2013 |
20130156225 | SYSTEM AND METHODS FOR APPLYING BASS COMPENSATION - Systems and methods for providing bass compensation to correct for uneven bass response are disclosed. An example bass compensation system includes a low pass filter configured to receive an audio signal from an audio source and provide a filtered audio signal, the low pass filter having a roll off of at least 18 dB per octave. The bass compensation system further includes a summing amplifier coupled to the low pass filter and configured to sum the audio signal from said audio source and the filtered audio signal to provide a summed audio signal, wherein the summed audio signal provided by the summing amplifier provides a bass boost at a first frequency and mid bass cut at a second frequency greater than the first frequency. | 06-20-2013 |
20130163782 | Sound Processing Apparatus and Sound Processing Method - A sound processing apparatus includes an information acquisition unit which acquires control information including at least one of mode information for designating a reproduction mode of the sound processing apparatus and attribute information for designating an attribute of an audio content represented by a sound signal, a frequency band expansion unit which performs frequency band expansion processing for adding an expanded component generated from the sound signal to the sound signal, and a control unit which changes parameters to be applied to the frequency band expansion processing in accordance with the control information acquired by the information acquisition unit. | 06-27-2013 |
20130170667 | CLOCK REGENERATION CIRCUIT AND DIGITAL AUDIO REPRODUCTION DEVICE - A frequency detection circuit of a clock regeneration circuit measures time which an input clock takes to change a predetermined number of times, and outputs a count value proportional to the time. A division ratio generation circuit truncates bits of the output of the frequency detection circuit by using a quantizer, and outputs the obtained value as a division ratio. A variable frequency divider divides a master clock by the division ratio output from the division ratio generation circuit, and outputs the obtained clock as a new clock. A high-quality clock having reduced jitter is regenerated, so that audio reproduction with high-quality sound is possible. | 07-04-2013 |
20130177170 | INTELLIGENT METHOD AND APPARATUS FOR SPECTRAL EXPANSION OF AN INPUT SIGNAL - A method, and a corresponding apparatus, for processing an input signal comprise filtering the input signal to separate a passband frequency component of the input signal from a stopband frequency component of the input signal, and adjusting relative signal power values of the passband frequency component and the stopband frequency component of the input signal based at least in part on signal values of a number of samples associated with the input signal. In the case of audio signals, for example, such processing is used for spectral expansion of the input signal by enhancing the power of the stopband, or low and high frequencies, component with respect to the power of the passband component of the input signal. As a result, a better audio quality is achieved. | 07-11-2013 |
20130177171 | PSEUDO BASS GENERATING APPARATUS - A pseudo bass generating apparatus includes a first 4 | 07-11-2013 |
20130177172 | CABLE CONNECTOR HAVING HIGH RESOLUTION ADJUSTABLE CAPACITANCE AND METHOD FOR USING THE SAME - A cable connector configured to receive a sound signal and alter the tonality of the sound by switching among a plurality of capacitors. The cable connector comprises a switch coupled to a binary coded plurality of capacitors. The cable connector receives an input signal corresponding to a sound through a conductor, which is also coupled to the switch. The switch may be used to select one of a plurality of capacitances, affecting the tonality of the sound. A cable implementing the cable connector and a method for using the cable connector are also disclosed. | 07-11-2013 |
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 |
20130208915 | SYSTEM AND METHOD FOR A PCM INTERFACE FOR A CAPACITIVE SIGNAL SOURCE - In accordance with embodiment, a method includes amplifying signal provided by a microphone to form an amplified signal. The method also includes converting the amplified signal into a frequency-based signal having a frequency dependent on an amplitude of the amplified signal. The frequency-based signal is converted into a pulse code modulated bitstream. | 08-15-2013 |
20130208916 | MULTICHANNEL SPEAKER ENCLOSURE - The invention relates to a speaker enclosure comprising at least two channels, wherein a first and second channel are respectively dedicated to separate and adjacent first and second frequency bands, each of said channels including a filtering stage supplied by the control signal and a speaker unit, wherein said speaker enclosure is characterized in that said channels also include means for simultaneously: disconnecting the first channel from the control signal; and modifying the filtering stage of the second channel so as to modify the frequency response thereof. | 08-15-2013 |
20130223647 | ELECTRO-ACOUSTIC TRANSDUCER - An electro-acoustic transducer includes a needle electrode, an opposite electrode, a discharge region between the needle electrode and the opposite electrode, a high-frequency oscillating circuit in the discharge region causing a high-frequency discharge and modulating and extracting an audio signal in accordance with a sound wave introduced to the discharge region, or converting the discharge in the discharge region into a sound wave, the discharge being performed in accordance with a high-frequency signal modulated by an audio signal, and an inert gas supply channel that supplies inert gas toward the peripheral surface of the needle electrode. The electro-acoustic transducer includes a needle-electrode cover as a part of the inert gas supply channel. The needle-electrode cover extends beyond the tip of the needle electrode toward the opposite electrode and has a gas flow outlet disposed beyond the tip of the needle electrode toward the opposite electrode. | 08-29-2013 |
20130223648 | AUDIO SIGNAL PROCESSING FOR SEPARATING MULTIPLE SOURCE SIGNALS FROM AT LEAST ONE SOURCE SIGNAL - An audio signal processing device is provided whereby, from two systems of audio signals in which audio signals of multiple audio sources are included, the audio signals of the multiple audio sources can be suitably separated. The audio signal processing device transforms the two systems of audio signals into frequency region signals, calculates a level ratio or a level difference between corresponding frequency division spectrums and extracts and outputs frequency band components of and nearby values regarding the level ratio or the level difference. Predetermined transfer coefficients for each of two output channels are multiplied by the frequency region signals and respective output signals are added together. The resulting summed signals are inverse transformed to generate time-sequence signals. | 08-29-2013 |
20130236031 | SPEAKER SYSTEM WHICH COMPRISES SPEAKER DRIVER GROUPS - A speaker system ( | 09-12-2013 |
20130287226 | REDUCED-DELAY SUBBAND SIGNAL PROCESSING SYSTEM AND METHOD - A method for signal processing, receiving a time domain signal having a sample-rate Fs and generating N time domain signal bands, each having a bandwidth equal to Fs/N. Receiving the N signal bands and transforming a first time domain signal band to a frequency domain at a first resolution and a second time domain signal band to the frequency domain at a second resolution, where the first resolution may be different from the second resolution. Determining one or more first filter coefficients using the frequency domain components from the first signal band and one or more second filter coefficients using the frequency domain components from the second signal band. Transforming the first and second filter coefficients from the frequency domain to a time domain. Applying the first and second time domain filter coefficients to the first and second time domain signals, respectively. | 10-31-2013 |
20130308792 | SPECTRAL ENVELOPE CODING OF ENERGY ATTACK SIGNAL - MDCT or FFT-based audio coding algorithms often have the problem named here spectral pre-echoes when coding an energy attack signal. This invention presents several possibilities to avoid the spectral pre-echoes existing in decoded signal segment before the energy attack point. The spectral envelope before the attack point can be improved by performing spectrum smoothing, replacing the segment of having spectral pre-echoes or filtering the segment with a combined filter obtained by doing LPC analysis. | 11-21-2013 |
20130308793 | Device For Adding Harmonics To Sound Signal - In a harmonic adding device, an input/output function part converts a digital sound signal having an input signal level into a digital sound signal having an output signal level determined according to a input/output function and the input signal level. A parameter storage unit stores at least one inflection point. A controller sets the inflection point to the input/output function to enable the input/output function part to add harmonics to the digital sound signal. The input/output function is defined between a positive maximum point and a negative maximum point. The inflection point is set between the positive maximum point and the negative maximum point. The positive maximum point, the inflection point and the negative maximum point are sequentially connected by linear interpolation to formulate the input/output function. | 11-21-2013 |
20130315419 | METHOD FOR CONTROLLING VOLUME OF ELECTRONIC DEVICE AND ELECTRONIC DEVICE USING THE SAME - A method for controlling volume of an electronic device is proposed along with the electronic device using the same. The electronic device has a speaker module, a display and a touch sensor disposed on the display. The method includes the following steps. The touch sensor is driven when the display is disabled from displaying an image. A sensing signal is received from the touch sensor. Inputted information is determined based on the sensing signal. An audio signal of the electronic device is changed according to the inputted information. The changed audio signal is outputted. | 11-28-2013 |
20140003623 | Smart Audio Settings | 01-02-2014 |
20140003624 | ACOUSTIC ELEMENT | 01-02-2014 |
20140064516 | SOUND TO HAPTIC EFFECT CONVERSION SYSTEM USING MAPPING - A haptic conversion system is provided that intercepts audio data, such as a digital audio signal, analyzes the audio data in frequency, and divides the analyzed audio data into one or more audio frequency regions, where each audio frequency region includes one or more audio sub-signals. The haptic conversion system further maps the one or more audio frequency regions to one or more haptic frequency regions, where each haptic frequency region includes one or more haptic signals. The haptic conversion system further maps the one or more haptic effects to one or more actuators. The haptic conversion system further sends the one or more haptic signals to one or more actuators, in order to generate one or more haptic effects. | 03-06-2014 |
20140064517 | MULTIMEDIA PROCESSING SYSTEM AND AUDIO SIGNAL PROCESSING METHOD - A multimedia processing system is provided. The system comprises: a depth analyzing unit configured to receive an input image and retrieve a depth image according to the input image; and a audio processing unit configured to receive an input audio signal and the depth image, detect an audio object and position information corresponding to the audio object from the depth image, and retrieve an acoustic frequency range corresponding to the audio object from the input audio signal; wherein when the position information exceeds a predetermined range, the audio processing unit adjusts the acoustic frequency range of the input audio signal according to the position information to generate an output audio signal. | 03-06-2014 |
20140086433 | Microphone with Programmable Frequency Response - Methods and apparatus automatically cancel or attenuate an unwanted signal (such as low frequencies from wind buffets) from, and/or control frequency response of, a condenser microphone, or control the effective condenser microphone sensitivity before the signal reaches an ASIC or other processing circuit. As a result, the maximum amplitude signal seen by the processing circuit is limited, thereby preventing overloading the input of the processing circuit. Remaining (wanted) frequencies can be appropriately amplified to reduce the noise burden on further processing circuits. A corrective signal is applied to a bias terminal of the condenser microphone to cancel the unwanted signal. Optionally or alternatively, a controllable impedance is connected to a line that carries the signal generated by the MEMS microphone, so as to attenuate unwanted portions of the signal. | 03-27-2014 |
20140086434 | METHOD AND APPARATUS FOR CUSTOMIZING AUDIO SIGNAL PROCESSING FOR A USER - A method and apparatus for customizing audio signal processing for a user by a mobile device is provided. The method includes identifying hearing characteristics of the user by testing hearing abilities of the user, by the mobile device, at a plurality of frequencies; adjusting a dynamic range of each of the plurality of frequencies based on the hearing characteristics; processing a decoded audio signal based on the adjusted dynamic range of each of the plurality of frequencies; and outputting the processed audio signal. | 03-27-2014 |
20140105418 | FREQUENCY DOMAIN MULTIBAND DYNAMICS COMPRESSOR WITH SPECTRAL BALANCE COMPENSATION - A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band. | 04-17-2014 |
20140112497 | SYSTEM AND METHOD FOR DIGITAL SIGNAL PROCESSING - The present invention provides for methods and systems for digitally processing an audio signal to reproduce high quality sounds on various materials. In various embodiments, a method comprises filtering the signal with a low shelf filter and/or high shelf filter, passing the signal through a first compressor that, filtering the signal again with a low shelf filter and/or high shelf filter, processing the signal with a graphic equalizer based on a selected material profile, passing the signal through a second compressor, and outputting the signal to a transducer. | 04-24-2014 |
20140140535 | Dynamic Speaker Management for Multichannel Audio Systems - A multiband limiter with selective sideband linking includes first and second frequency band splitters, a first and second plurality of limiters, first and second summers, and a plurality of selectable links coupling the first plurality of limiters to the second plurality of limiters. The first plurality of limiters each have a band input coupled to one of the first plurality of band outputs, a link port and a limiter output, and the first summer is receptive to the limiter outputs of the first plurality of limiters and has a first channel output. The second plurality of limiters each have a band input coupled to one of the second plurality of band outputs, a link port and a limiter output, and the second summer is receptive to the limiter outputs of the second plurality of limiters and has a second channel output. | 05-22-2014 |
20140140536 | SYSTEM AND METHOD FOR ENHANCING AUDIO - A system and method for enhancing audio, the method including receiving audio input tracks, with at least one audio input track including a restriction parameter, determining a restricted audio input track, where the restricted audio input track is the audio input track including the restriction parameter, manipulating another audio input track based on musical properties of the restricted audio input track, and combining the restricted audio input track and the manipulated audio input track into a single output audio track. | 05-22-2014 |
20140161280 | AUDIO SIGNAL CORRECTION AND CALIBRATION FOR A ROOM ENVIRONMENT - Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal. | 06-12-2014 |
20140161281 | AUDIO SIGNAL CORRECTION AND CALIBRATION FOR A ROOM ENVIRONMENT - Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include identifying a target sub-region of the frequency response which has a predetermined area percentage of a total area under a curve generated by the frequency response, determining whether the target sub-region is a narrow energy region, creating a filter to adjust the frequency response, and applying the filter to the audio signal. | 06-12-2014 |
20140161282 | SUBSTANTIALLY PLANATE PARAMETRIC EMITTER AND ASSOCIATED METHODS - A parametric speaker comprises a generally planate radiating element, suitable for radiating ultrasonic vibrations into a fluid medium, and an emitter, having an ultrasonic output and/or resonant frequency, the emitter being intimately coupled to the radiating element. The radiating element is physically configured to have a mechanical resonance that substantially matches the output and/or resonant frequency of the emitter. | 06-12-2014 |
20140177870 | 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. | 06-26-2014 |
20140198930 | PROCESS AND DEVICE FOR SYNTHESIS OF AN AUDIO SIGNAL ACCORDING TO THE PLAYING OF AN INSTRUMENTALIST THAT IS CARRIED OUT ON A VIBRATING BODY - Process for synthesis of a synthesized audio signal, in which at least one audio signal of contact is produced for each excitation contact of a sequence of contacts carried out on a vibrating body ( | 07-17-2014 |
20140219473 | SIGNAL FILTERING APPARATUS AND SIGNAL FILTERING METHOD - The present invention provides a signal filtering apparatus, which comprises a control circuit and a filter for receiving a transmitted input signal and generating an output signal. The filter comprises multiple filter taps for processing the transmitted input signal corresponding to different timings with different filter coefficients, respectively, to generate the output signal. The control circuit is configured to shrink at least part of the low-frequency-response filter taps having filter coefficients less than a predetermined value. | 08-07-2014 |
20140219474 | METHOD OF REDUCING UN-CORRELATED NOISE IN AN AUDIO PROCESSING DEVICE - An audio processing device comprises a multitude of electric input signals, each electric input signal being provided in a digitized form, and a control unit receiving said digitized electric input signals and providing a resulting enhanced signal. The control unit is configured to determine the resulting enhanced signal from said digitized electric input signals, or signals derived therefrom, according to a predefined scheme. | 08-07-2014 |
20140254826 | VIRTUAL PRE-AMPLIFIER AND EFFECTS SYSTEM AND METHODS FOR CUSTOMIZING AND USING THE SAME IN LIVE PERFORMANCES - The technology in one embodiment includes an audio-processing unit, for processing an input audio signal using a pre-established audio feature. The unit includes a processor, an audio-device input in operative communication with the processor, and a system-user interface being in operative communication with the processor. The unit also includes a computer-readable medium being in operative communication with the processor and comprising a tone-manipulation component and computer-executable instructions that, when executed by the processor, cause the processor to perform various operations. The operations include presenting, using the system-user interface, a graphical-user-interface (GUI) display including an identifier associated with the tone-manipulation component, receiving, from the system-user interface, a user signal indicating a user selection of the tone-manipulation component from the GUI display, receiving the input audio signal from the audio-device input, and manipulating, using the tone-manipulation component, the input audio signal, yielding a manipulated signal. | 09-11-2014 |
20140254827 | Method and Circuitry for Processing Audio Signals - An audio signal processing method and circuitry that processes an input audio signal by filtering the input audio signal with a high pass filter to produce a filtered audio signal, which is input to a compressor. A first intermediate audio signal is produced based on the compressor output signal. The filtered audio signal is also input to a harmonics generator that produces harmonics of the filtered audio signal. A second intermediate audio signal is produced based on such harmonics. A third intermediate signal is produced based upon the input audio signal. An output audio signal is produced by combining the first intermediate audio signal, the second intermediate audio signal and the third intermediate audio signal. The compressor can be configured to reduce the dynamic range of components of the filtered audio signal that contribute to the first intermediate audio signal relative to the dynamic range of the harmonics that contribute to the second intermediate audio signal, thus enhancing the input audio signal. | 09-11-2014 |
20140270253 | Flexible Clocking for Audio Sample Rate Converter in a USB System - A processor according to embodiments comprises an on-board sample rate converter for converting a source audio signal that is sampled at a first sampling rate to an output audio signal that is sampled at a second sampling rate. The sample rate converter utilizes a master clock signal in converting the audio signal. The sample rate converter selects the master clock signal from available reference clock signals, such as an on-chip system clock or a bus interface clock, and scales the frequency of the selected clock signal to generate the master clock signal with the frequency of the second sampling rate. | 09-18-2014 |
20140270254 | CUSTOMIZING AUDIO REPRODUCTION DEVICES - The disclosure includes a system and method for sonically customizing an audio reproduction device. The system includes a processor and a memory storing instructions that when executed cause the system to: determine an application environment associated with an audio reproduction device associated with a user; determine one or more sound profiles based on the application environment; provide the one or more sound profiles to the user; receive a selection of a first sound profile from the one or more sound profiles; and generate tuning data based on the first sound profile, the tuning data configured to sonically customize the audio reproduction device. | 09-18-2014 |
20140307895 | ACOUSTIC SET COMPRISING A SPEAKER WITH CONTROLLED AND VARIABLE DIRECTIVITY - An acoustic chamber includes a loudspeaker, which includes at least two membranes that each reproduce a different frequency band, and a filter that makes it possible to generate a plurality of activation signals from an audio signal source. The activation signals are each applied to an actuator of one of the membranes. The acoustic chamber has an operating range having a variable and controlled directivity, each frequency of which belongs to at least two frequency bands reproduced by the membranes. The acoustic chamber obtains a directivity control signal, and the filter makes it possible to dose, for each frequency of the operating range and depending on the directivity control signal, the contribution of each one of the at least two membranes reproducing the frequency. | 10-16-2014 |
20140321667 | METHODS AND APPARATUS FOR PROCESSING AUDIO SIGNALS - Various methods and apparatus for processing audio signals are disclosed herein. The assembly may be attached, adhered, or otherwise embedded into or upon a removable oral appliance to form a hearing aid assembly. Such an oral appliance may be a custom-made device which can enhance and/or optimize received audio signals for vibrational conduction to the user. Received audio signals may be processed to cancel acoustic echo such that undesired sounds received by one or more intra-buccal and/or extra-buccal microphones are eliminated or mitigated. Additionally, a multiband actuation system may be used where two or more transducers each deliver sounds within certain frequencies. Also, the assembly may also utilize the sensation of directionality via the conducted vibrations to emulate directional perception of audio signals received by the user. Another feature may include the ability to vibrationally conduct ancillary audio signals to the user along with primary audio signals. | 10-30-2014 |
20140328498 | Generating Sound for a Rotating Machine of a Device - The invention relates to a method for generating sound for a rotating machine, including a step (E | 11-06-2014 |
20140334640 | AUDIO SIGNAL CONTROL OF ELECTRICAL OUTLET STRIP - An electrical outlet strip is controlled by sound or an audio signal. The electrical outlet strip can include a plurality of outlets having a selector switch for either each outlet or one selector switch that controls all the outlets. The selector switch can control whether the outlets are powered directly be the input power, or if the outlets are controlled by the audio signal or sound. When the outlets are controlled by the audio signal or sound, the audio signal or sound can be filtered into a plurality of bands, one for each outlet, for example. There audio signal or sound can be input to the outlet strip by various mechanisms, including WiFi, Bluetooth®, FM wireless receiver, via an audio jack, via a microphone, or the like. | 11-13-2014 |
20140334641 | SPECTRAL SHAPING FOR AUDIO MIXING - Techniques are described herein that are capable of spectrally shaping audio signal(s) for audio mixing. Spectrally shaping an audio signal means modifying a frequency spectrum of the audio signal. A frequency spectrum of an audio signal is a representation of the audio signal in the frequency domain. For instance, a frequency spectrum may be represented using multiple frequency bands. The frequency spectrum may be modified by modifying characteristic(s) (e.g., magnitude, phase, etc.) one or more of the frequency bands. | 11-13-2014 |
20140341393 | ELECTRONIC DEVICE AND METHOD FOR REPRODUCING AUDIO SIGNALS - In a method of reproducing audio signals using an electronic device, the electronic device decodes an audio file to generate original digital audio signals, and attenuates the original digital audio signals having frequencies that are lower than a preset cutoff frequency. The electronic device further amplifies each of attenuated digital audio signals and the original digital audio signals having frequencies that are higher than the preset cutoff frequency with a preset gain. Once the electronic device converts the amplified digital audio signals into analog audio signals, and outputs the analog audio signals to an audio amplifier, the audio amplifier amplifies the analog audio signals to drive a speaker to output the analog audio signals. | 11-20-2014 |
20140348344 | COMMUNICATION SYSTEM AND TRANSFER METHOD THEREOF - A communication system is provided. A receiver receives a plurality of audio signals, wherein a frequency of each of the audio signals is selected from a frequency group formed by at least three frequencies. A signal detector coupled to the receiver is configured to obtain the frequency of each of the audio signals. A processor coupled to the signal detector is configured to convert the frequency of each of the audio signals into a digital signal having a first logic level or a second logic level. Two adjacent audio signals of the audio signals have different frequencies, and at least one frequency of the frequency group is used to dynamically represent the first logic level or the second logic level. | 11-27-2014 |
20140355786 | SOUND SYNTHESIS WITH FIXED PARTITION SIZE CONVOLUTION OF AUDIO SIGNALS - A method for convolving an input signal with an impulse response function, the impulse response function being partitioned into a plurality of time segments of equal size, the method including transforming a segment of an input signal into the frequency domain to generate a frequency spectrum of the segment of the input signal; multiplying the frequency spectrum of the segment of the input signal with a frequency spectrum of each of the segments of the impulse response function; scaling the results from the multiplication of frequency spectra; accumulating the scaled results; and performing an inverse transform on the accumulated signals to generate a desired convolved signal in the time domain. The scaling includes performing a bitwise shift operation on the multiplication results, and performing the bitwise shift operation includes adding a bit to the multiplication results before the bitwise shift operation. Fast convolution of uniformly partitioned impulse response functions can be achieved by performing scaling of input signals, multiplication, and accumulation using fixed-point arithmetic. | 12-04-2014 |
20140355787 | ACOUSTIC RECEIVER WITH INTERNAL SCREEN - An acoustic apparatus includes a high frequency driver that has a first front volume and a low frequency driver that has a second front volume. The first front volume and the second front volume communicate with each other to form a common front volume. At least one acoustic resistance is placed between the first front volume and the second front volume. The acoustic resistance acts as a low pass filter. | 12-04-2014 |
20140363020 | SOUND CORRECTING APPARATUS AND SOUND CORRECTING METHOD - A sound correcting apparatus includes an air-conduction microphone, a bone-conduction microphone, a processor, and a memory. The air-conduction microphone picks up an air conduction sound using aerial vibrations. The bone-conduction microphone picks up a bone conduction sound using bone vibrations of a user. The processor calculates a ratio of a voice of the user for the air conduction sound to a noise. The memory stores a correction coefficient for making a frequency spectrum of the bone conduction sound identical with a frequency spectrum of the air conduction sound which corresponds to the ratio that is equal to or greater than a first threshold. The processor corrects the bone conduction sound using the correction coefficient, and generates an output signal from the corrected bone conduction sound when the ratio is less than a second threshold. | 12-11-2014 |
20140363021 | Collar/headset microphone cable control device - A collar/headset microphone cable control device is provided with a sound receiving part and a radio frequency emitting part. The sound receiving part includes a microphone sound receiving end for picking up the sound signal inputs, and a microphone cable control end electrically connected to the microphone sound receiving end for controlling the sound signal transmission state. The radio frequency emitting part includes a substrate, a microphone signal receiving element, a built-in sound receiving end, an external sound signal receiving element, a signal emitting element, buttons, indicator lights, a display element and a bus element that all are provided on the substrate and are electrically interconnected. The microphone signal receiving element picks up sound signals from the microphone cable control end. The built-in sound receiving end receives additional sound signals. A user can easily control the sound volume level and muteness on the microphone sound source cable. | 12-11-2014 |
20140369521 | SYSTEM AND METHOD FOR NARROW BANDWIDTH DIGITAL SIGNAL PROCESSING - The present invention provides methods and systems for narrow bandwidth digital processing of an input audio signal. Particularly, the present invention includes a high pass filter configured to filter the input audio signal. A first compressor then modulates the filtered signal in order to create a partially processed signal. In some embodiments, a clipping module further limits the gain of the partially processed signal. A splitter is configured to split the partially processed signal into a first signal and a second signal. A low pass filter is configured to filter the first signal. A pass through module is configured to adjust the gain of the second signal. A mixer then combines the filtered first signal and the gain-adjusted second signal in order to output a combined signal. In some embodiments, a tone control module further processes the combined signal, and a second compressor further modulates the processed signal. | 12-18-2014 |
20140376745 | HYBRID AUDIO DELIVERY SYSTEM AND METHOD THEREFOR - Methods and systems to produce audio output signals from audio input signals. In one embodiment, a first portion of the audio input signals can be pre-processed, with the output used to modulate ultrasonic carrier signals, thereby producing modulated ultrasonic signals. The modulated ultrasonic signals can be transformed into a first portion of the audio output signals, which is directional. Based on a second portion of the audio input signals, a standard audio speaker can output a second portion of the audio output signals. Another embodiment further produces distortion compensated signals based on the pre-processed signals. The distortion compensated signals can be subtracted from the second portion of the audio input signals to generate inputs for the standard audio speaker to output the second portion of the audio output signals. In yet another embodiment, noise can be added during pre-processing of the first portion of the audio input signals. | 12-25-2014 |
20150016631 | DYNAMIC TAIL SHORTENING - Systems and methods for reducing the length of music data files having decaying sound patterns are provided. A system and method can include analyzing a music data file including an attack portion and a tail portion. A fadeout range can be associated with the music data file, and a cut threshold of the tail portion can be identified. A modified version of the music data file can be played back. The playback can include reducing sound levels of the music data file in accordance with the fadeout range and ending the playback when the cut threshold of the tail portion is reached. | 01-15-2015 |
20150016632 | SYSTEMS AND METHODS FOR REMAPPING AN AUDIO RANGE TO A HUMAN PERCEIVABLE RANGE - A system for remapping an audio range to a human perceivable range includes an audio transducer configured to output audio and a processing circuit. The processing circuit is configured to receive the audio from an audio input, analyze the audio to determine a first audio range, a second audio range, and a third audio range. The processing circuit is further configured to use frequency compression on the first audio range based on the second audio range and third audio range to create a first open frequency range, move the second audio range into the first open frequency range to create a second open frequency range, move the third audio range into the second open frequency range, and provide audio output including the compressed first audio range, the moved second audio range, and the moved third audio range. | 01-15-2015 |
20150043747 | Microphone System and Method - A microphone system ( | 02-12-2015 |
20150049879 | METHOD OF AUDIO PROCESSING AND AUDIO-PLAYING DEVICE - A method of audio processing lowers the frequency of a high frequency audio area of an input audio in order to generate a lowered frequency audio area. The lowered frequency audio area is combined with the input audio to generate an output audio such that the output audio comprises the high frequency audio area, a low frequency audio area, and a lowered frequency audio area. | 02-19-2015 |
20150049880 | Low Delay Real-to-Complex Conversion in Overlapping Filter Banks for Partially Complex Processing - An arrangement of overlapping filter banks comprises a synthesis stage and an analysis stage. The synthesis stage receives a first signal segmented into time blocks and outputs, based thereon, an intermediate signal to be received by the analysis stage forming the basis for the computation of a second signal segmented into time frames. In an embodiment, the synthesis stage is operable to release an approximate value of the intermediate signal in a time block located L−1 time blocks ahead of its output block, which approximate value is computed on the basis of any available time blocks of the first signal, so that the approximate value contributes, in the analysis stage, to the second signal. The delay is typically reduced by L−1 blocks. Applications include audio signal processing in general and real-to-complex conversion in particular. | 02-19-2015 |
20150049881 | Audio Signal Processing Device - An audio signal processing device includes: an input section to which an audio signal is input; a bias processing section configured to add a bias signal to the audio signal; a calculation section configured to perform a power calculation on the audio signal to which the bias signal has been added by the bias processing section; and an adding unit configured to add the audio signal on which the power calculation is performed by the calculation section to the audio signal having been input to the input section. | 02-19-2015 |
20150063594 | SLEW RATE CONTROL APPARATUS FOR DIGITAL MICROPHONES - A driver, includes a driver block, a controller block, and a comparison block. The driver block includes an adjustable current source configured to produce a digital output stream. The controller block is coupled to the driver block. The comparison block is coupled to the driver block and the controller block. The comparison block is configured to compare the digital output stream to a reference value at a time delayed with respect to a master clock and based upon the comparison cause the controller block to adjust a strength of the driver block. | 03-05-2015 |
20150078583 | AUTOMATIC AUDIO HARMONIZATION BASED ON PITCH DISTRIBUTIONS - Two audio samples and/or sets of audio samples are identified. The pitch distributions of the audio samples and/or sets of audio samples are identified, the pitch distribution of an audio sample or set of audio samples referring to how much of each of multiple pitches of notes is present in the audio sample or set of audio samples. Based on the pitch distributions of the audio samples and/or sets of audio samples, at least one pitch of one of the audio sample and/or set of audio samples can be automatically adjusted (but need not be, depending on the pitch distributions) to increase harmonic coherence of the audio samples and/or sets of audio samples. | 03-19-2015 |
20150104040 | HIGH FREQUENCY ENERGY CONVERTER - A high frequency energy converter which has application as an acoustic actuator for converting incoming high frequency energy into outgoing harmonized high frequency mechanical (e.g., sound) and electromagnetic waves. The energy converter is adapted to improve the quality of sound heard by a listener by reducing random and spurious harmonics that are introduced by the environment in which the listener is located. The energy converter includes an outer body and a reactive crystalline material (e.g., quartz) lying at the bottom of the outer body that is responsive to the incoming high frequency energy. A dispersion horn is located at the top of the outer body to be seated upon the crystalline material. The dispersion horn has a throat extending therethrough so that both incoming high frequency energy and outgoing high frequency mechanical and electromagnetic waves are transmitted through the throat of the horn in opposite directions. | 04-16-2015 |
20150110289 | SYSTEM AND METHOD FOR DIGITAL SIGNAL PROCESSING - The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal. | 04-23-2015 |
20150110290 | Apparatus And Method For Frequency Detection - An application specific integrated circuit (ASIC) is used with an acoustic device. An input clock signal is received. The frequency of the input clock signal is determined, and the frequency is indicative of one of a plurality of operational modes of the ASIC. Based upon the determined frequency, an amount current provided to one or more operational blocks of the ASIC is changed. | 04-23-2015 |
20150110291 | Differential High Impedance Apparatus - A differential high impedance circuit for use in an acoustic apparatus includes a first set of transistor devices and a second set of transistor devices. The first set of transistor devices includes a first transistor ( | 04-23-2015 |
20150110292 | DEVICE, METHOD AND COMPUTER PROGRAM FOR FREELY SELECTABLE FREQUENCY SHIFTS IN THE SUBBAND DOMAIN - A device for producing a frequency-shifted audio signal based on an audio input signal is provided. The device has an interface and a frequency-shifting unit. The interface is configured for receiving the audio input signal. The frequency-shifting unit is configured for producing the frequency-shifted audio signal. The frequency-shifting unit is additionally configured to produce one of the second subband values based on one of the first subband values such that the second phase angle of this second subband value differs from the first phase angle of this first subband value by a phase angle difference, the phase angle difference being dependent on frequency information indicating by which frequency difference the audio input signal is to be shifted in order to obtain the frequency-shifted audio signal, and the phase angle difference being dependent on a frequency bandwidth of one of the first subbands. | 04-23-2015 |
20150124999 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR SYNTHESIZING SOUNDS USING ESTIMATED MATERIAL PARAMETERS - The subject matter described herein includes methods, systems, and computer readable media for synthesizing sounds using estimated material parameters. According to one aspect, a method for sound synthesis using estimated material parameters is provided. The method includes at a computing platform including a memory and a processor, estimating, by the processor and based on a recorded audio sample of an impact on a physical object, one or more material parameters of a physical object for use in synthesizing a sound associated with a virtual object having material properties similar to the physical object. The method further includes storing the one or more material parameters in memory. The method further includes using the estimated material parameters to generate a synthesize sound for the virtual object | 05-07-2015 |
20150139447 | AUTOMATED CONSTRUCTION OF INFINITE IMPULSE RESPONSE FILTERS - Systems and methods can support constructing an infinite impulse response (IIR) filter. A desired frequency response may be received. An initial filter model may be constructed comprising complimentary pairs of component IIR filters based upon the desired frequency response. The filter model may be converged according to stepwise refinement of individual terms within the filter model. A global error for the converged filter model may be computed. An additional complimentary pair of component IIR filters may be incorporated into the filter model in response to the global error exceeding a maximum acceptable error. In response to incorporation of additional component IIR filters, convergence and evaluation of the filter model may be iterated. Upon final convergence of the filter model, an aggregate IIR filter may be generated by combing the component IIR filters. | 05-21-2015 |
20150146885 | SYSTEMS AND METHODS FOR PROVIDING A WIDEBAND FREQUENCY RESPONSE - Electronic circuitry is described. The electronic circuitry includes a first microelectromechanical system (MEMS) structure that exhibits a first frequency response in a voice frequency range and that captures a first signal. The electronic circuitry also includes a second MEMS structure coupled to the first MEMS structure. The second MEMS structure exhibits a second frequency response in an ultrasound frequency range and captures a second signal. A combination of the first frequency response and the second frequency response achieves a target frequency response in a combined frequency range | 05-28-2015 |
20150311879 | MICROPHONE BIAS CIRCUIT - A bias circuit supplies a bias voltage V | 10-29-2015 |
20150312676 | SYSTEM AND METHOD FOR REDUCING LATENCY IN TRANSPOSER-BASED VIRTUAL BASS SYSTEMS - A latency reduction system in a virtual bass processing system performs harmonic transposition on low frequency components of an audio signal to generate transposed data indicative of harmonics of the audio signal. The system uses a base transposition factor greater than two, and generates the harmonics in response to frequency-domain values determined by forward and inverse transform stages that use asymmetric analysis and synthesis windows. The system combines a virtual bass signal with the delayed wide band audio signal through analysis filter banks having filter coefficient truncated Nyquist filters. The virtual bass signal may lag the delayed wide band audio signal when combining with the audio signal to further reduce the latency caused by the harmonic transposition. The virtual bass input signal may be directly routed from a CQMF analysis filter bank of a preceding Hybrid filter bank stage, in order to avoid the delay associated with a Nyquist filter bank. | 10-29-2015 |
20150326972 | COINCIDING LOW AND HIGH FREQUENCY LOCALIZATION PANNING - A method of panning includes panning an input signal or a low frequency portion of the input signal to split the input signal or the low frequency portion of the input signal into first and second channels, panning the input signal or a high frequency portion of the input signal to split the input signal or the high frequency portion of the input signal into third and fourth channels such that localization of the high frequency portion of the input signal coincides with localization of the low frequency portion of the input signal. | 11-12-2015 |
20150373454 | Sound-Emitting Device and Sound-Emitting Method - A sound-emitting device includes a high-frequency extractor, adapted to accept input of a sound signal, extract high-frequency components of sound and output a high-frequency sound signal, a low-frequency extractor, adapted to accept input of the sound signal, extract low-frequency components of sound and output a low-frequency sound signal, a delay processor, adapted to delay low-frequency components of the low-frequency sound signal within a time range not causing an echo, relative to the high-frequency sound signal, to thereby output a delayed low-frequency sound signal, and a sound emitter, adapted to emit sound based on the high-frequency sound signal and the delayed low-frequency sound signal. | 12-24-2015 |
20160012829 | ENCODING DEVICE AND METHOD, DECODING DEVICE AND METHOD, AND PROGRAM | 01-14-2016 |
20160014501 | MINIATURE SPEAKER MODULE, METHOD FOR ENHANCING FREQUENCY RESPONSE THEREOF AND ELECTRONIC DEVICE | 01-14-2016 |
20160014510 | Hybrid Test Tone for Space-Averaged Room Audio Calibration Using A Moving Microphone | 01-14-2016 |
20160014511 | Concurrent Multi-Loudspeaker Calibration with a Single Measurement | 01-14-2016 |
20160014512 | Playback Device Calibration User Interfaces | 01-14-2016 |
20160021465 | SPEAKER SYSTEM - The present invention provides a speaker system comprising: an electroacoustic converter film composed of a polymeric composite piezoelectric body in which piezoelectric body particles are dispersed in a viscoelastic matrix formed of a polymer material that exhibits viscoelasticity at normal temperature, and thin film electrodes formed on both surfaces of the polymeric composite piezoelectric body; and a driving circuit that attenuates signal intensity of an input signal from a signal source at a rate of 5 dB to 7 dB per octave and supplies the attenuated input signal to the electroacoustic converter film. | 01-21-2016 |
20160065161 | FREQUENCY DOMAIN MULTIBAND DYNAMICS COMPRESSOR WITH SPECTRAL BALANCE COMPENSATION - A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band. | 03-03-2016 |
20160149581 | CLOCK GENERATOR - A clock generator receives first and second clock signals, and input representing a desired frequency ratio. A comparison is made between frequencies of an output clock signal and the first clock signal, and a first error signal represents the difference between the desired frequency ratio and this comparison result. The first error signal is filtered. A comparison is made between frequencies of the output clock signal and the second clock signal, and a second error signal represents the difference between the filtered first error signal and this comparison result. The second error signal is filtered. A numerically controlled oscillator receives the filtered second error signal and generates an output clock signal. As a result, the output clock signal has the jitter characteristics of the first input clock signal over a useful range of jitter frequencies and the frequency accuracy of the second input clock signal. | 05-26-2016 |
20160157016 | Multi-Channel Playback of Audio Content | 06-02-2016 |
20160205464 | LOUDSPEAKER APPARATUS | 07-14-2016 |