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33results about How to "Improve audio signal quality" patented technology
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Method and Arrangements for Audio Signal Encoding
ActiveUS20090024399A1Audio signal be reduceImprove reproduction qualitySpeech analysisAudio signalAudio frequency
To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.
Owner:UNIFY GMBH & CO KG
Audio signal system
InactiveUS20060029248A1Improve audio signal qualityImprove perceived qualityTransducer detailsTelephone set constructionsDigital signal processingInductor
An audio signal system is provided having, in one embodiment, a magnetostrictive core of varying shapes, sizes, and permeability. The core is wound with a wire, thereby forming an inductor for producing a magnetic field when current is passed through the wire. A speaker, or other audio output device, is positioned in parallel or series with the inductor. In operation, an audio signal is received and transmitted to the speaker or other audio output device, wherein a replication of the received signal is produced. An output of the magnetostrictive inductor couples with the replicated audio signal to modify and enhance the quality of the projected signal. In yet another embodiment, digital signal processing may be used to modify the received signal as well.
Owner:ABLE PLANET
System and method for digital signal processing in deep diving environment
ActiveUS9564146B2Facilitate operational safetyFacilitate communicationGain controlSpeech analysisDigital signal processingComputer module
The present invention relates to a system for processing of an audio signal relating to a diver in a deep diving environment. The system comprises an input device structured to receive the signal, a pitch changing module configured to change the pitch of the received signal, a plurality of processing modules collectively configured to process the pitch changed signal and produce a processed signal. The present invention is further directed to a method for processing of an audio signal relating to a diver in a deep diving environment.
Owner:BONGIOVI ACOUSTICS LLC
Optimized Calibration of a Multi-Loudspeaker Sound Playback System
ActiveUS20150223004A1Improve audio signal qualityTaking reflectionsSpeech analysisLoudspeaker spatial/constructional arrangementsVocal tractA domain
A method of calibrating a sound restitution assembly for a multichannel sound signal, which includes a plurality of loudspeakers. The method includes: obtaining multidirectional impulse responses of the loudspeakers to reproduction of a predetermined audio signal; analyzing the multidirectional impulse responses obtained, in a domain of spatio-temporal representation, over at least one time window encompassing the instants of arrival of the first reflections of the audio signal reproduced to determine a set of characteristics of the first reflections; comparing the amplitude of each of the reflections with a predetermined perceptibility threshold and identifying imperceptible reflections for which the amplitude is below the predetermined threshold; modifying the impulse responses obtained to obtain perceptive impulse responses, by deleting the reflections identified as imperceptible; and determining a filtering matrix on the basis of the perceptive impulse responses for an application of this filtering matrix to the multichannel audio signal before sound restitution.
Owner:ORANGE SA (FR)
Apparatus and method for processing an audio signal using a harmonic post-filter
ActiveUS20170140769A1Avoid problemsImprove robustnessSpeech analysisVacuum gauge using compression chambersHarmonicLag
An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG EV
System and method for digital signal processing in deep diving environment
ActiveUS20160036402A1Facilitate operational safetyClearer digital signalGain controlSpeech analysisDigital signal processingComputer module
The present invention relates to a system for processing of an audio signal relating to a diver in a deep diving environment. The system comprises an input device structured to receive the signal, a pitch changing module configured to change the pitch of the received signal, a plurality of processing modules collectively configured to process the pitch changed signal and produce a processed signal. The present invention is further directed to a method for processing of an audio signal relating to a diver in a deep diving environment.
Owner:BONGIOVI ACOUSTICS LLC
Audio source position detection and audio adjustment
InactiveUS6952672B2Improve audio signal qualityImprove signal-to-noise ratioTwo-way loud-speaking telephone systemsSubstation equipmentLocation detectionBackground noise
A method for adjusting an operational characteristic of an audio device can include a series of steps. The method can include receiving a user spoken utterance from an audio speech source and detecting a position of the audio speech source relative to the audio device. The method further can include generating proximity data corresponding to the detected position and processing the received user spoken utterance with a selected signal processing technique based upon the proximity data. The signal processing technique can distinguish the user spoken utterance from background noise.
Owner:WISTRON CORP
System and method for improving hearing
ActiveUS20160249141A1Improve hearingImprove audio signal qualityMicrophonesSignal processingHearing perceptionComputer science
A system and method for improving hearing is disclosed. The system includes a microphone array that includes an enclosure, a plurality of beamformer microphones and an electronic processing circuitry to provide enhanced audio signals to a user by using information obtained on the position and orientation of the user. The system is preferably in the form of a smartphone having a retractable piece having the beamformer microphones mounted thereon. The system may also include a corresponding method for improving hearing and a non-transitory computer storage media having instructions stored thereon which, when executed, execute the method.
Owner:NOOPL INC
Portable physiology monitor configured to measure tympanic temperature
ActiveUS20190212198A1Improve audio signal qualityAvoid imbalanceEar treatmentBody temperature measurementThermopileDistal portion
Wearable devices (100) capable of measuring a core body temperature and other vital signs of a user in a range of situations are described herein. The wearable device is arranged to be retained within the ear canal of the ear, in order to prevent the wearable device from inadvertently removing itself from the ear. Providing an infrared thermopile (101) at the innermost end of the ear insert ensures that the infrared thermopile is provided as close as possible to the tympanic membrane which will be used to provide an indication of the core body temperature. The device has an audio conduction channel (111) at least partly defined within an ear canal extending member (114), the audio conduction channel configured as a waveguide to conduct sound through a blocking member (212) to a distal portion of the ear insert.
Owner:INOVA DESIGN SOLUTIONS
Audio signal processing method and device
ActiveCN106356070AAchieve noise reductionAccurate positioning of noiseSpeech analysisBroadcastingAudio frequency
An embodiment of the invention discloses an audio signal processing method and device. The method comprises steps as follows: to-be-processed audio signals generated in a direct broadcasting process are acquired, and audio frames are extracted from the to-be-processed audio signals; a first probability, a second probability and a third probability are determined according to a transcendental audio model; the first probability is the probability that the audio frame belongs to voice, the second probability is the probability that the audio frame belongs to music, and the third is the probability that the audio frame belongs to noise; if the first probability is smaller than a first threshold or the second probability is smaller than a second threshold and the third probability is larger than a third threshold, the fact that the audio frame contains noise is determined; after the fact that the to-be-processed audio signals contain noise is determined, the audio frame belonging to noise is denoised. The method and the device can be suitable for the application scene of direct broadcasting, denoising of the audio signals is realized, and the quality of the audio signals is improved.
Owner:GUANGZHOU BAIGUOYUAN NETWORK TECH
Coding of audio signals
InactiveCN1942928AChange encoding propertiesImprove audio signal qualitySpeech analysisSignal onLow frequency band
An encoder comprises an input for inputting frames of an audio signal in a frequency band, an analysis filter dividing the frequency band into lower and higher frequency bands, a first encoding block for encoding the audio signals of the lower frequency band, a second encoding block for encoding the audio signals of the higher frequency band, and a mode selector for selecting an operating mode for the encoder among at least a first mode where signals only on the lower frequency band are encoded, and a second mode where signals on both the lower and higher frequency band are encoded. The encoder has a scaler to gradually change the encoding properties of the second encoding block in connection with a change in the operating mode of the encoder. The invention also relates to a device, a decoder, a method, a module, a computer program product, and a signal.
Owner:NOKIA CORP
Method and apparatus for restoring audio signal
ActiveUS20170337926A1Improve audio signal qualityImprove sound qualitySpeech analysisComputer scienceAudio frequency
Provided is a method of reconstructing an audio signal, the method including detecting a lossy frequency band, based on an energy value of each of frequencies of the audio signal; obtaining a cut-off frequency, based on the lossy frequency band; and reconstructing the audio signal of the lossy frequency band, based on the cut-off frequency.
Owner:SAMSUNG ELECTRONICS CO LTD
Audio signal system
InactiveUS7978867B2Improve perceived qualityImprove audio signal qualityTelephone set constructionsDeaf-aid setsDigital signal processingPower flow
Owner:ABLE PLANET
Arrangement and a method for handling an audio signal
InactiveUS20050169245A1Without deteriorationLess-delayHybrid switching systemsTelephonic communicationTime delaysAnalog signal
The present invention relates to a sound device (SD1), connected to a computer (P1), for handling of asynchronously transferred digital audio packets (5) on a network (LAN1). The computer has an interface (3) connected to a telephony application (1), a driver (D3) and a bus (4). The sound device (SD1) is connected (9) via the bus (4) and includes a software frame buffer (B2), codecs (C2) and an A / D-D / A converter (AD2), which is connected to in / out devices (10, 11, 12). The sound packets (5) are transferred asynchronously through the computer (P1), are buffered in the sound device frame buffer (B2), decoded in the codec (C2) and D / A converted into an analog signal for the in / out devices. Speech to the in devices (11, 12) is processed in a corresponding manner. Having the buffer (B2) close to the codec (C2) enables processing of the sound packets, e.g. with respect to the varying time delay in the computer (P1), restoring lost packets and producing replacement frames. The sound device (SD1) relieves the computer (P1) of the heavy workload of processing the sound packets (5).
Owner:TELEFON AB LM ERICSSON (PUBL)
Method and apparatus for encoding and decoding excitation patterns from which the masking levels for an audio signal encoding and decoding are determined
InactiveUS20110238424A1Improve audio signal qualityImprove calculationElectrophonic musical instrumentsSpeech analysisBit planeExcitation pattern
For the quantisation of spectral data in an audio transform encoder psycho-acoustic information is required, i.e. an approximation of the true masking threshold. According to the invention, for each spectrum to be quantised in the audio signal encoding, an excitation pattern is computed and coded for both long and short window / transform lengths. The excitation patterns are grouped together in a variable-size matrix. A pre-determined sorting order with a fixed number of values only is applied to the excitation pattern data matrix values, and by that re-ordering a quadratic matrix is formed to which matrix' bit planes a SPECK encoding is applied.
Owner:THOMSON LICENSING SA
Apparatus and method for processing an audio signal using a harmonic post-filter
ActiveUS10242688B2Improve audio signal qualityImprove subjective qualitySpeech analysisVacuum gauge using compression chambersHarmonicLag
An apparatus for processing an audio signal having associated therewith a pitch lag information and a gain information, includes a domain converter for converting a first domain representation of the audio signal into a second domain representation of the audio signal; and a harmonic post-filter for filtering the second domain representation of the audio signal, wherein the post-filter is based on a transfer function including a numerator and a denominator, wherein the numerator includes a gain value indicated by the gain information, and wherein the denominator includes an integer part of a pitch lag indicated by the pitch lag information and a multi-tap filter depending on a fractional part of the pitch lag.
Owner:FRAUNHOFER GESELLSCHAFT ZUR FOERDERUNG DER ANGEWANDTEN FORSCHUNG EV
Apparatus and method for calculating filter coefficients for a predefined loudspeaker arrangement
InactiveUS20130136281A1Easy to calculateReduce effortStereophonic systemsLoudspeaker spatial/constructional arrangementsEngineeringSelf adaptive
An apparatus for calculating filter coefficients for a predefined loudspeaker arrangement has a multi-channel renderer. The multi-channel renderer calculates a filter coefficient for each loudspeaker of a virtual loudspeaker arrangement, being different from the predefined loudspeaker arrangement, based on a property (e.g. position or type) of a virtual source of an audio object to be reproduced by the predefined loudspeaker arrangement. Further, the multi-channel renderer determines an adapted filter coefficient for a loudspeaker of the predefined loudspeaker arrangement based on one or more calculated filter coefficients of one or more loudspeakers of the different virtual loudspeaker arrangement.
Owner:IOSONO
Audio signal processing method and electronic device supporting the same
InactiveUS20130294621A1Provide accuratelyImprove audio signal qualityGain controlSpeech analysisComputer hardwareAudio frequency
A method of processing an audio signal is provided. The method includes receiving at least one digital audio stream from at least one audio source, applying at least one gain value to each of the at least one digital audio stream by a gain adjustment unit, mixing the at least one digital audio stream to which the at least one gain value has been applied, calculating a representative gain value from the at least one gain value, calculating an extra valid headroom based on the representative gain value for correction of the mixed at least one digital audio stream, and post-processing the mixed at least one digital audio stream using the extra valid headroom.
Owner:SAMSUNG ELECTRONICS CO LTD
Method and apparatus for encoding and decoding excitation patterns from which the masking levels for an audio signal encoding and decoding are determined
InactiveUS8515770B2Improve audio signal qualityEasy to calculateElectrophonic musical instrumentsSpeech analysisFrequency spectrumExcitation pattern
A method and apparatus for approximating a true masking threshold for the quantization of spectral data in an audio transform encoder. According to the invention, for each spectrum to be quantised in the audio signal encoding, an excitation pattern is computed and coded for both long and short window / transform lengths. The excitation patterns are grouped together in a variable-size matrix. A pre-determined sorting order with a fixed number of values only is applied to the excitation pattern data matrix values, and by that re-ordering a quadratic matrix is formed to which matrix' bit planes a SPECK encoding is applied.
Owner:THOMSON LICENSING SA
Method and arrangements for audio signal encoding
ActiveUS8612216B2Improve audio signal qualityImprove audio qualitySpeech analysisAudio synthesisPulse shaping
To form an audio signal, frequency components of the audio signal which are allotted to a first subband are formed by means of a subband decoder using supplied fundamental period values which respectively indicate a fundamental period for the audio signal. Frequency components of the audio signal which are allotted to a second subband are formed by exciting an audio synthesis filter using an excitation signal which is specific to the second subband. To produce this excitation signal, an excitation signal generator derives a fundamental period parameter from the fundamental period values. The fundamental period parameter is used by the excitation signal generator to form pulses with a pulse shape which is dependent on the fundamental period parameter at an interval of time which is determined by the fundamental period parameter and to mix them with a noise signal.
Owner:UNIFY PATENTE GMBH & CO KG
Anti-ground loop noise circuit, system and unbalanced audio equipment
ActiveCN113132860AImprove stabilitySimple designTransducer acoustic reaction preventionTelecommunicationsNoise
The embodiment of the invention provides an anti-grounding loop noise circuit, an anti-grounding loop noise system and unbalanced audio equipment, which are applied to the unbalanced audio equipment and comprise a differential amplification circuit, the positive input end of the differential amplification circuit is connected with the signal end in an audio interface, the negative input end of the differential amplification circuit is connected with the grounding end in the audio interface, and the negative input end of the differential amplification circuit is connected with the signal end in the audio interface. The audio interface is an input interface of the audio equipment, and the differential amplification circuit is used for amplifying a differential audio signal sent by source end audio equipment and inhibiting ground loop noise. According to the embodiment of the invention, ground isolation between the two audio devices is realized from the perspective of hardware design, and balance and non-balance output ports of the audio device at the source end are compatible, so that additional adaptation is not needed. By adopting the technical scheme, the design of field wiring is greatly simplified, and the audio signal quality and the stability of an audio system are improved.
Owner:HANGZHOU HIKVISION DIGITAL TECH
Electronic accessory apparatus and audible signal transmission method thereof
ActiveUS9807507B1Improve qualityImprove audio signal qualitySignal processingTwo-part coupling devicesVIT signalsUSB
An electronic accessory apparatus and audible signal transmission method thereof are provided. The electronic accessory apparatus includes a type C USB connector, a first switch circuit, a second switch circuit, a grounding switch circuit and a first channel selector. The type C USB connector has a plurality of data pin sets, a first ground pin, and a second ground pin. The first switch circuit transports a data signal set or an audible output signal set to a first selected data pin set according to a first control signal. The second switch circuit transports the audible output signal set to the first selected data pin set according to the first control signal. The grounding switch circuit is turned on or cut off according to a second control signal. The first channel selector sets a signal transmission relation between the first audible input signal set and the first and second ground pins.
Owner:HTC CORP
Audio processing unit
InactiveUS20180367228A1Improve audio signal qualityEnhanced and processedSignal processingVolume compression/expansion having semiconductor devicesAudio signal flowAudio signal
The present invention provides an audio processing unit having a microchip for processing an inputted audio signal. The microchip operates by duplicating the inputted audio signal, splitting the duplicated signal into at least two exact duplicates of the original inputted signal, layering the signals, processing the layers, and recombining the layers to provide an enhanced or processed audio signal. Thereafter, the processed audio file may be further processed by a user by utilizing any number of knobs, buttons, and the like disposed on the unit. Preferably, the unit is mountable on an open-frame rack.
Owner:AFTERMASTER INC
Apparatus and method for calculating filter coefficients for a predefined loudspeaker arrangement
InactiveUS8462966B2Easy to calculateReduce effortLoudspeaker spatial/constructional arrangementsStereophonic systemsEngineeringSelf adaptive
An apparatus for calculating filter coefficients for a predefined loudspeaker arrangement has a multi-channel renderer. The multi-channel renderer calculates a filter coefficient for each loudspeaker of a virtual loudspeaker arrangement, being different from the predefined loudspeaker arrangement, based on a property (e.g. position or type) of a virtual source of an audio object to be reproduced by the predefined loudspeaker arrangement. Further, the multi-channel renderer determines an adapted filter coefficient for a loudspeaker of the predefined loudspeaker arrangement based on one or more calculated filter coefficients of one or more loudspeakers of the different virtual loudspeaker arrangement.
Owner:IOSONO
Ground loop noise immunity circuits, systems and unbalanced audio equipment
ActiveCN113132860BImprove stabilitySimple designTransducer acoustic reaction preventionTelecommunicationsNoise
Owner:HANGZHOU HIKVISION DIGITAL TECH
Audio signal processing method and electronic device supporting the same
InactiveUS9966914B2Provide accuratelyImprove audio signal qualityGain controlSpeech analysisComputer hardwareVIT signals
A method of processing an audio signal is provided. The method includes receiving at least one digital audio stream from at least one audio source, applying at least one gain value to each of the at least one digital audio stream by a gain adjustment unit, mixing the at least one digital audio stream to which the at least one gain value has been applied, calculating a representative gain value from the at least one gain value, calculating an extra valid headroom based on the representative gain value for correction of the mixed at least one digital audio stream, and post-processing the mixed at least one digital audio stream using the extra valid headroom.
Owner:SAMSUNG ELECTRONICS CO LTD
An audio signal processing method and device
ActiveCN106356070BAchieve noise reductionAccurate positioning of noiseSpeech analysisBroadcastingAudio frequency
An embodiment of the invention discloses an audio signal processing method and device. The method comprises steps as follows: to-be-processed audio signals generated in a direct broadcasting process are acquired, and audio frames are extracted from the to-be-processed audio signals; a first probability, a second probability and a third probability are determined according to a transcendental audio model; the first probability is the probability that the audio frame belongs to voice, the second probability is the probability that the audio frame belongs to music, and the third is the probability that the audio frame belongs to noise; if the first probability is smaller than a first threshold or the second probability is smaller than a second threshold and the third probability is larger than a third threshold, the fact that the audio frame contains noise is determined; after the fact that the to-be-processed audio signals contain noise is determined, the audio frame belonging to noise is denoised. The method and the device can be suitable for the application scene of direct broadcasting, denoising of the audio signals is realized, and the quality of the audio signals is improved.
Owner:BIGO TECH PTE LTD
Optimized calibration of a multi-loudspeaker sound playback system
ActiveUS9584947B2Improve audio signal qualityTaking reflectionsSpeech analysisLoudspeaker spatial/constructional arrangementsVocal tractA domain
A method of calibrating a sound restitution assembly for a multichannel sound signal, which includes a plurality of loudspeakers. The method includes: obtaining multidirectional impulse responses of the loudspeakers to reproduction of a predetermined audio signal; analyzing the multidirectional impulse responses obtained, in a domain of spatio-temporal representation, over at least one time window encompassing the instants of arrival of the first reflections of the audio signal reproduced to determine a set of characteristics of the first reflections; comparing the amplitude of each of the reflections with a predetermined perceptibility threshold and identifying imperceptible reflections for which the amplitude is below the predetermined threshold; modifying the impulse responses obtained to obtain perceptive impulse responses, by deleting the reflections identified as imperceptible; and determining a filtering matrix on the basis of the perceptive impulse responses for an application of this filtering matrix to the multichannel audio signal before sound restitution.
Owner:ORANGE SA (FR)
Loudspeaker control method and device
InactiveCN110536214AEnsure safe workImprove output qualitySignal processingTransducer circuitsState parameterParameter control
Owner:SPREADTRUM COMM (SHANGHAI) CO LTD
Encoding method and system
InactiveUS20100305954A1Improve audio signal qualityImprove sound qualitySpeech analysisTransmissionData differencingComputer science
An encoding system includes a sampling unit, a computing unit, a comparing unit, a quantifying unit, and an encoding unit. The sampling unit obtains first sample data of a current sampling point and second sample data of a previous sampling point. The computing unit computes a data difference between the first sample data and the second sample data. The data difference includes a numeral and a sign. The comparing unit determines whether the data difference is more than or equal to 0 and outputs a determining result. The quantifying unit quantifies the numeral of the data difference. The encoding unit encodes the numeral of the data difference with or without the sign according to the determining result.
Owner:HON HAI PRECISION IND CO LTD
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