Abstract: A method for compensating for acoustic influence of a listening room on an acoustic output from an audio system including at least a left and a right loudspeaker, the method comprising determining a left frequency response and a right frequency response, designing a left compensation filter FL, and a right compensation filter FR, and during playback applying the left and right filters to left and right channel inputs. According to the invention, a target response in the listening position is simulated, and the left and right compensation filters are designed to filter transfer functions based on the simulated target function multiplied by an inverse of the left/right frequency responses. By relying on a simulated target instead of relying on an empirical approach, the general impact of a room can be more accurately captured by the target functions.

Abstract: A method of operating a UI for controlling a virtual musical instrument can include receiving a first input corresponding to a selection of an array of cells within a cell matrix, each array of the cell matrix assigned to audio sample data stored in a computer-readable medium that, when triggered, causes the audio sample data to output corresponding audio, and each cell within its particular array of the cell matrix, in response to being selected for playback and upon being triggered to begin playback, causes the audio sample data corresponding to the cell's particular array to be played. The method can further include receiving a second input corresponding to a changing of a number of cells within the particular selected array; and changing the number of cells within the selected array based on the second input and maintaining a number of cells in other arrays within the cell matrix.

Abstract: Provided is an electronic device. The electronic device may include: a housing including a front plate, a rear plate facing away from the front plate, and a side member surrounding a first space between the front plate and the rear plate; a display panel exposed to an exterior of the electronic device through the front plate and configured to detect a pen input using a magnetic field; a speaker structure disposed between the display panel and the rear plate, and including a first surface facing the display panel, a second surface facing in a direction opposite the display panel, a side surface surrounding a second space between the first surface and the second surface, and a yoke facing the first surface; and a first shield structure made of a ferromagnetic material and disposed on the first surface.

Abstract: The present application discloses a speaker earphone. Each earphone unit includes a first speaker group and/or a second speaker group. The first speaker group includes a first speaker, a second speaker, and at least one capacitor. The second speaker group includes a first speaker, a second speaker, a third speaker, and at least one capacitor. In the present disclosure, at least two speakers are arranged on an enclosure of the earphone. The two speakers are in reverse connection according to polarity in order to achieve a better sound effect.

Abstract: A method, and system, of digital room correction for a device, such as a smart speaker, including a loudspeaker. The method comprises capturing audio from an environment local to the device, for example from one or more microphones of a smart speaker. The captured audio is then processed to recognize one or more categories of sound. A digital room correction procedure may then be controlled dependent upon recognition and/or analysis of at least one of the categories of sound.

Abstract: The present disclosure relates to near-end crosstalk (NEXT) cancellation. A transmit communication signal is transmitted over a first Digital Subscriber Line (DSL) connection using a first group of frequencies and a receive communication signal is received over a second DSL connection using a second group of frequencies that at least partially overlaps the first group of frequencies. A crosstalk correlation between the first and second communication signals is determined. Based on the crosstalk correlation, a crosstalk cancellation signal is generated. The crosstalk cancellation signal is subtracted from the second communication signal, with the intention of reducing NEXT.

Abstract: A system includes a circuitry that provides for psychoacoustic frequency range extension for a speaker. The circuitry generates quadrature components from an audio channel, and generates rotated spectral quadrature components by applying a forward transformation that rotates a spectrum of the quadrature components from a standard basis to a rotated basis. In the rotated basis, the circuitry isolates components of the rotated spectral quadrature components at target frequencies, and generates weighted phase-coherent harmonic spectral quadrature components by applying a scale-independent nonlinearity to the isolated components. The circuitry generates a harmonic spectral component by applying an inverse transformation that rotates a spectrum of the weighted phase-coherent harmonic spectral quadrature components from the rotated basis to the standard basis.

Abstract: Systems and methods are disclosed for audio enhancement. For example, methods may include accessing audio data; determining a window of audio samples based on the audio data; inputting the window of audio samples to a classifier to obtain a classification, in which the classifier includes a neural network and the classification takes a value from a set of multiple classes of audio; selecting, based on the classification, an audio enhancement network from a set of multiple audio enhancement networks; applying the selected audio enhancement network to the window of audio samples to obtain an enhanced audio segment, in which the selected audio enhancement network includes a neural network that has been trained using audio signals of a type associated with the classification; and storing, playing, or transmitting an enhanced audio signal based on the enhanced audio segment.

Abstract: Embodiments relate to an audio processing system that adjusts for unmatched speakers, such as by frequency response, output power, directionality, etc. For example, a mobile device may include an earpiece speaker and a micro-loudspeaker. The audio processing system may include a crossover network that separates an input audio signal into a low frequency signal and a high frequency signal. Subband spatial processing and b-chain processing are applied to the high frequency signal to spatially enhance the input signal, and adjust the input signal for the unmatched speakers. The low frequency signal is processed using a parametric band-pass filter and a first gain to generate a low frequency resonator signal, and a second gain to generate a low frequency passthrough signal. The processed low frequency signals are provided to the micro-loudspeaker by combination with one of a left channel or a right channel of the processed high frequency signal.

Abstract: An impulse response is computed between i) an audio signal that is being output as sound by a loudspeaker that is integrated in a loudspeaker enclosure, and ii) a microphone signal from a microphone that is recording the output by the loudspeaker and that is also integrated in the loudspeaker enclosure. A reverberation spectrum is extracted from the impulse response. Sound power spectrum at the listening distance is computed, based on the reverberation spectrum, and an equalization filter is determined based on i) the estimated sound power spectrum and ii) a desired frequency response at the listening distance. Other aspects are also described and claimed.

Abstract: Technologies for time-delayed augmented reality (AR) presentations includes determining a location of a plurality of user AR systems located within the presentation site and determining a time delay of an AR sensory stimulus event of an AR presentation to be presented in the presentation site for each user AR system based on the location of the corresponding user AR system within the presentation site. The AR sensory stimulus event is presented to each user AR system based on the determined time delay associated with the corresponding user AR system. Each user AR system generates the AR sensory stimulus event based on a timing parameter that defines the time delay for the corresponding user AR system such that the generation of the AR sensory stimulus event is time-delayed based on the location of the user AR system within the presentation site.

Abstract: 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.

Abstract: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: An apparatus comprising: an input configured to receive at least two audio signals, the at least two audio signals having a relative displacement between them; an orientation determiner configured to determine an audio capture orientation based on a device orientation and information concerning microphone configuration in the apparatus; and an audio output generator configured to output the at least two audio signals based on the capture orientation, such that playback of the at least two audio signals is performed based on the capture orientation.

Abstract: An MTM loudspeaker in which an array of smaller drivers replaces each of the larger drivers in a traditional MTM design. Doing so reduces the total moving mass of the drivers reproducing the lower-frequency sounds in the MTM design, while still allowing enough air to be moved for the loudspeaker to reproduce sounds in lower frequency ranges that would normally be produced by the larger drivers and could not be adequately produced by a single instance of the smaller driver. For example, a single tweeter or supertweeter, or a pair of tweeters or supertweeters still serves a traditional tweeter or tweeter function in the MTM design (handling high frequencies), but a plurality of tweeters or supertweeters arranged in two arrays placed on either side of the tweeter(s) or supertweeter(s) handling high frequencies is tasked with reproducing lower frequencies not traditionally handled in loudspeakers by tweeters or supertweeters.

Abstract: Example systems and methods represent audio using a sequence of two-dimensional (2D) Fourier transforms (2DFTs), and such a sequence may be used by a specially configured machine to perform audio identification, such as for automated cover song identification. Such systems and methods are robust to timbral changes, time skews, and pitch skews that occur in cover songs found in content repositories. The systems and methods allow copyright holders to search the content repositories for unlicensed cover song.

Abstract: A method for compensating for a change in back electric motive force in a transducer in a device, wherein the transducer is driven by an output signal output by an amplifier, and a back volume of the transducer is formed by an enclosure within the device. The method comprises: receiving an input signal; determining, based on a characteristic of the transducer that the transducer has moved from an equilibrium position within the device; and based on the determination, adjusting a gain applied to the input signal by the amplifier to generate the output signal.

Abstract: An effective and simple psychoacoustic bass generation system generates a harmonic signal having inter-modulation controllable to remain below a threshold level and includes a high-pass filter configured to pass harmonics which are reproducible with fidelity by the loudspeaker or other transducer and a loudness matching block configured to compensate the loudness of the desired harmonics to match the loudness of the original signal.

Abstract: Example systems and methods represent audio using a sequence of two-dimensional (2D) Fourier transforms (2DFTs), and such a sequence may be used by a specially configured machine to perform audio identification, such as for cover song identification. Such systems and methods are robust to timbral changes, time skews, and pitch skews. In particular, a special data structure provides a time-series representation of audio, and this time-series representation is robust to key changes, timbral changes, and small local tempo deviations. Accordingly, the systems and methods described herein analyze cross-similarity between these time-series representations. In some example embodiments, such systems and methods extract features from an audio fingerprint and calculate a distance measure that is robust and invariant to changes in musical structure.

Abstract: A system for applying a filtering scheme to an audio signal in which one or more predefined audio filtering schemes may be applied. A user interface is provided for receiving input from a user and displaying status information to the user, and may include one of a variety of adjustment controls such as a physical knob, switch, dial, slide, button, digital touchscreen, or remote control, among others. Varying linear and nonlinear filtering schemes may be provided to impact various frequency ranges in the audio signal to generate an output signal.

Abstract: An example implementation may involve a control device displaying a prompt to initiate a calibration sequence that involves calibration of a playback device for a given environment in which the playback device is located. The example implementation may also involve the control device displaying (i) a prompt to prepare a playback device for calibration within a given environment, (ii) a prompt to prepare the given environment for calibration of the playback device, and/or (iii) a prompt to prepare the control device for calibration of the playback device. The example implementation may also involve a control device displaying a selectable control, that, when selected, initiates calibration of the playback device. The example implementation may further involve a control device initiating calibration of the playback device.

Abstract: A method, computer program product, and computing system for acoustic speech localization, comprising receiving, via a plurality of microphones, a plurality of audio signals. Modulation properties of the plurality of audio signals may be analyzed. Speech sounds may be localized from the plurality of audio signals based upon, at least in part, the modulation properties of the plurality of audio signals.

Abstract: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: An electronic device is provided. The electronic device includes a communication circuit, a speaker, a microphone, a memory, and a processor. The processor is configured to determine whether there is auditory data stored in the memory of the electronic device when an external electronic device is connected thereto through the communication circuit, send a request for auditory data to the external electronic device when there is no auditory data stored in the memory, receive the auditory data from the external electronic device, correct data, at the electronic device, about the sound received through the microphone using the auditory data, receive corrected audio data from the external electronic device, synthesize the corrected data about the sound and the corrected audio data received from the external device to obtain first synthesized data, and output a first new sound corresponding to the first synthesized data through the speaker.

Abstract: A signal processing apparatus includes a control section, a signal processing section connected with a plurality of signal processing elements and performs 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 controls 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.

Abstract: An input audio signal is separated into input audio signal components (X(k,b)). A set of two or more band branches provides output audio signal components (Y(k,b)). The set of band branches comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal and the summed audio signal(y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: A sound source separation apparatus includes: a separation-matrix processor that transforms a plurality of observation signals corresponding to sounds being propagated from a plurality of sound sources into a frequency-domain signal group the separation-matrix processor updating a separation matrix based on the frequency-domain signal group and transforming the updated separation matrix into time-series filter coefficients to output; a filter-coefficient transformer that partially removes non-causal components from the filter coefficients to transform the filter coefficients, and a separator that supplies the filter coefficients to a filter group, the separator generating a plurality of separation signals separated from the plurality of observation signals corresponding to the separation matrix.

Abstract: Example techniques may involve calibration with multiple recording devices. An implementation may include detecting, via a microphone, one or more calibration sounds as emitted by one or more playback devices of one or more zones during a calibration sequence. The implementation may further include determining a first response, the first response representing a response of a given environment to the one or more calibration sounds as detected by the first recording device and receiving data indicating a second response, the second response representing a response of the given environment to the one or more calibration sounds as detected by a second recording device. The implementation may also include determining a calibration for the one or more playback devices based on the first response and the second response and sending, to the one or more zones, an instruction that applies the calibration to playback by the one or more playback devices.

Abstract: A remote speaker microphone with touch volume control is provided. The remote speaker microphone includes a housing; a microphone; a speaker; and an interface configured to connect the speaker and the microphone. The remote speaker microphone further includes a touch sensor configured to, in response to receiving touch input, control a volume level of the speaker from a first volume level to a lower volume level, the first volume level being one of a plurality of discrete volume levels.

Abstract: Methods and systems are provided for adjusting a crossover frequency between a plurality of audio speakers rendering audio content. In one example, a first subset of a plurality of audio speakers may be rendering a first sub-range of a range of audio frequencies of an audio content, and a second subset of speakers of the plurality of audio speakers may be rendering a second sub-range of the range of audio frequencies. In this example, the first sub-range and the second sub-range may be substantially separated at the crossover frequency. In one case, a characteristic of the audio content may be determined, and the crossover frequency may be adjusted based on the determined characteristic to help improve the audio content rendering quality by the respective subsets of audio speakers in the plurality of audio speakers.

Abstract: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: The reliability and resilience of a speaker in a portable device may be improved by enabling the device to determine a status of an enclosure of a transducer. The impedance profiles of transducers vary depending on the enclosure status, whether the enclosure is leaky, ported, or sealed. Determination of parameters related to an impedance of a transducer may aid in a determination of the status of the enclosure. Once an enclosure status is determined, that information may be used to determine an excursion model for the transducer. Adapting the excursion model or other audio processing parameters can improve protection and performance of the transducer.

Abstract: A microphone assembly includes an acoustic transducer element configured to convert sound into a microphone signal in accordance with a transducer frequency response including a first highpass cut-off frequency. The microphone assembly additionally includes a processing circuit including a signal amplification path configured to receive, sample and digitize the microphone signal to provide a digital microphone signal. A frequency response of the signal amplification path includes a second highpass cut-off frequency which is higher than the first highpass cut-off frequency of the acoustic transducer element.

Abstract: A voltage regulator system, comprising: a switched capacitor (SC) regulator that operates at a switching frequency and receives an input voltage; and a controller configured to control an operation of the SC regulator by adjusting the switching frequency of the SC regulator based on efficiency. In some embodiments, the switching frequency is swept to determine a best efficiency. In some embodiments, the switching frequency is swept at each of a plurality of values for the input voltage. In some embodiments, the system includes further one or more switches in series with the SC regulator. In some embodiments, the SC regulator includes an output terminal that is coupled to a battery.

Abstract: A wearable includes an article of clothing, an intelligent control and a plurality of sets of vibration motors operatively connected to the intelligent control and positioned on the article of clothing to provide tactile feedback wherein each of the sets of vibration motors is associated with a different frequency range of audio which may be communicated to the wearable via Bluetooth or otherwise.

Abstract: A group of devices connected on a network operate in conjunction with one another to present content. Individual devices may be wirelessly connected to presentation devices such as speakers. Skew is a delay between when content is sent to the presentation device and then output. Drift is a difference in timing of output between two or more presentation devices not synchronized to a common clock. Non-zero values of skew and drift may result in poor content presentation by the group. An actual rate at which data is transferred to a presentation device is measured and compared to an expected rate. This difference is analyzed to determine a drift value. The content to be delivered to the presentation device is resampled. During presentation the combination of the drift and the resampled content reduces skew. The drift value may also be used to lookup a skew value used to delay initial presentation.

Abstract: Method for determining distance to target using a multitone nonlinear radar system comprising providing a transmitter that transmits a signal comprising at least two predetermined frequency components; receiving transmitted signal upon reflection from target; determining the phase relationships of the frequency components when signal strikes target; determining distance the signal has travelled to target based upon the phase relationship of the frequency signal components at the time of reflection from target; computing the distance to target.

Abstract: A computerized method for filtering a digital audio file to generate an output audio file that induces optimal health and cognitive ability in a listener of a playback of the output audio file is described herein. The method includes the steps of identifying a plurality of target frequencies that span within an octave, identifying a plurality of mid-point frequencies that are situated at mid-points between any two adjacent target frequencies, applying a peaking filter to the digital audio file centered around the plurality of mid-point frequencies to produce highest frequency attenuation at the plurality of mid-point frequencies, and generating the output audio file.

Abstract: A system and method automates the tuning of one or more multiband compressors or multiband limiters to minimize loudspeaker distortion. The system and method render one or more test loudspeaker signals that vary in frequency and in amplitude out of a loudspeaker and records the responses of the loudspeaker at a microphone. The system and method measure a distortion of the loudspeaker, relative to a frequency and an amplitude, with respect to the one or more test loudspeaker signals and a microphone signal and calculate tunable parameters of the multiband compressor that includes cutoff frequencies in response to the measured distortion.

Abstract: 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.

Abstract: A method for characterizing a loudspeaker comprises: filtering an input test signal by applying the inverse of a linear model describing an excursion of the loudspeaker in response to a given input signal, the linear model containing only linear terms; applying the filtered input test signal to the loudspeaker; measuring the excursion of the loudspeaker in response to the filtered input test signal; and, based on the measured excursion, determining one or more non-linear parameters for a non-linear model describing the excursion of the loudspeaker in response to a given input signal.

Abstract: Embodiments of the invention provide methods and apparatus for processing audio input signals, for example, so as to linearize the output of a given loudspeaker. An audio processor is provided, for modifying an audio signal to be provided to a loudspeaker, the audio processor comprising: a first filter stage, for applying to an audio signal a linear model describing an excursion of the loudspeaker in response to a given input signal, the linear model containing only linear terms, and for generating one or more excursion signals; a plurality of second filters for receiving the one or more excursion signals, each of the second filters configured to apply to a respective one of a plurality of frequency bands in the one or more excursion signals the inverse of a model describing an excursion of the loudspeaker in response to a given input signal; and a combiner for combining the outputs of each of the plurality of second filters.

Abstract: Among other things, a sound processing device system is disclosed to assist a hearing-impaired human listener recognize speech sounds or phonemes. The device system may be configured at least to generate an output audio signal at least by transposing and causing a negative rank ordering of frequency of at least a portion of the input audio signal. Compression also may be performed on the at least the portion of the input audio signal as part of generating the output audio signal. The negative rank ordering may be performed on a high-frequency portion of the input audio signal that becomes a low-frequency portion of the output audio signal by the transposing. The low-frequency portion of the output audio signal may represent an inverted ordering of frequencies or frequency segments present in the high-frequency portion of the input audio signal.

Abstract: 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.

Abstract: An embodiment of the invention provides a method for monitoring for and detecting a leak in a line where a mobile detector having an audio sensor and an odor sensor transiting along the line. A sample audio reading is obtained by the audio sensor; and, a sample odor reading is obtained by the odor sensor. A processor compares the sample audio reading to a baseline audio reading to determine whether the line is leaking. The processor compares the sample odor reading to a baseline odor reading to determine whether the line is leaking. The location of the mobile detector is identified by a GPS device on the mobile detector when the processor determines that the line is leaking. The location of the mobile detector, the sample audio reading, and the sample odor reading is sent to a central monitoring center by a communications device on the mobile detector.

Abstract: Example embodiments disclosed herein relate to modelling a frequency response characteristic of an electro-acoustic transducer. A method includes obtaining at least one measurement of the frequency response characteristic for at least one electro-acoustic transducer of the category. A model of a frequency response characteristic specific to a category of electro-acoustic transducers is generated at least in part based on perceptual importance of a frequency band, an averaged, normalized or microphone compensated measurement such that the distortion of the model is optimized. A further method for estimating a frequency response characteristic of an electro-acoustic transducer is based on the generated model and the sensitivity of the electro-acoustic transducer or headphone. Corresponding system and computer program product are also disclosed.

Abstract: A presentation moderator assembly for facilitating nonverbal and nonphysical communication between a moderator and a presenter includes a base unit that is positioned in view of a presenter. The base unit has a first light emitter that is selectively illuminated to communicate to the presenter that the presenter may begin speaking. The base unit has a second light emitter that is selectively illuminated to communicate to the presenter that the presenter should begin finishing speaking. The base unit has a third light emitter that is selectively illuminated to communicate to the presenter to stop speaking. A remote unit is carried by a moderator and the remote unit is in electrical communication with the base unit. The remote unit selectively turns on each of the first, second and third light emitters.

Abstract: An active sound management system comprises a transducer that senses an actual sound or vibration from a sound generating source and generates a transducer signal in response to sensing the actual sound or vibration; a harmonic extractor device that extracts a plurality of harmonics from the transducer signal; and a harmonic modifier device that adjusts a feature of the extracted harmonic to be within a predetermined threshold with respect to a target harmonic corresponding to a desired sound.

Abstract: This invention describes methods of Audio Signal Personalized Enhancement for optimal listening to music. A combination of hearing test, audio signal compensation, and signal processing algorithms are implemented jointly in an audio Device. An audio signal is processed and tuned within the entire range of audible frequencies to actively compensate for both the characteristics of the listening device, and the specific hearing characteristics of the listener, resulting in a much improved music listening experience. A randomized self-test algorithm is used to analyze user's hearing ability in each ear within a preselected dynamic volume range at multiple frequencies spanning the audible spectrum.

Abstract: Systems and methods for handling silence in audio streams are disclosed. In one aspect, a transmitter detects a halt in an audio stream. After detection of the halt in the audio stream, the transmitter embeds a silence signal into the audio stream and transmits the silence signal to associated receivers. The associated receivers may respond to the embedded silence signal by “playing” silence or by using the silence signal to activate a silence protocol. In either event, the associated receivers do not receive the original audio halt and do not produce an unwanted audio artifact.