Abstract: A dynamic range control unit (210) configured to apply dynamic range control, referred to as DRC, to an audio signal (211). The DRC unit (210) is configured to downsample a subband signal (212) derived from the audio signal (211), to provide a downsampled subband signal (321), to determine a DRC gain (329) based on the downsampled subband signal (321), and to apply the DRC gain (329) to the subband signal (212), to provide a compressed subband signal (213) of a compressed audio signal (214).

Abstract: An output feedback control circuit includes, as a means for generating an error signal ERR between a feedback voltage FB commensurate with an output voltage and a reference voltage REF (means for substituting for a single error amplifier 140), a first signal processor (for example, an amplifier 140a) that forms a first path, and a second signal processor (for example, an amplifier 140b) that forms a second path. The amplifier 140a is more accurate than the amplifier 140b, and the amplifier 140b is faster than the amplifier 140a. To the output terminal of the amplifier 140a, preferably, a capacitor Cc other than a parasitic element is connected. To the output terminal of the amplifier 140b, preferably, no capacitor other than a parasitic element is connected, and a resistor Rc is connected. Preferably, the amplifier 140a has a transconductance gm1, and the amplifier 140b has a transconductance gm2 (?gm1).

Abstract: Disclosed is a method for increasing a perceived loudness of an audio data signal comprising the steps of obtaining a first digital audio data signal; determining at least one temporal amplitude peak in the first digital audio data signal; generating a second digital audio data signal by reducing the at least one temporal amplitude peak in the first digital audio data signal based on a predicted perceptual difference model representing a predicted perceptual difference between the first digital audio data signal and a peak reduced version of the first digital audio data signal; and generating a third digital audio data signal by amplifying the second digital audio data signal so that a peak of the second digital audio data signal has a predetermined signal value, wherein a perceived loudness of the third digital audio data signal is larger than a perceived loudness of the first digital audio data signal.

Abstract: The present disclosure generally relates to user interfaces and techniques for managing audio exposure using a computer system (e.g., an electronic device). In accordance with some embodiments, the electronic device displays a graphical indication of a noise exposure level over a first period of time with an area of the graphical indication that is colored to represent the noise exposure level, the color of the area transitioning from a first color to a second color when the noise exposure level exceeds a first threshold. In accordance with some embodiments, the electronic device displays noise exposure levels attributable to a first output device type and a second output device type and, in response to selecting a filtering affordance, visually distinguishes a set of noise exposure levels attributable to the second output device type.

Abstract: An audio player device includes a sound capturing unit, an audio speaker unit, a storage unit and a processing unit. The storage unit stores multiple environmental volume ranges, and multiple volume setting values respectively corresponding to the environmental volume ranges. The processing unit obtains an environmental volume based on environmental sound received by the sound capturing unit, and selects a volume setting value that corresponds to an environmental volume range which covers the environmental volume to be an automatically-determined volume setting value. Then, the processing unit controls an output volume of the audio speaker unit according to the automatically-determined volume setting value.

Abstract: The concepts and technologies disclosed herein are directed to venue seat assignment based upon hearing profiles. According to one aspect of the concepts and technologies disclosed herein, a device can include a processor and a memory. The device can receive a request to upload a hearing profile, and can upload the hearing profile to a seat assignment system. The device can receive, from the seat assignment system, a customized seating chart based, at least in part, upon the hearing profile. The customized seating chart can include a visual representation of at least a portion of seating in a venue. The device can select, from the customized seating chart, a seat from the portion of the seating in the venue. This selection can be made automatically or based upon user input.

Abstract: To provide an audio reproduction device including: a signal processing circuit that generates multi-channel audio data; an attenuation amount control unit that is connected to the signal processing circuit and controls an attenuation amount of audio data of each channel; and a speaker device that is driven by a multi-channel audio signal from the attenuation amount control unit and has multi-channel speakers including a woofer, in which the attenuation amount control unit estimates, from an audio data value of each channel, a total value of currents flowing through the speaker device, and controls the attenuation amount in such a way that the estimated total value of currents does not exceed a predetermined value that has been set in advance, and the attenuation amount is controlled in accordance with a priority that has been set in advance in such a way that relationships between the channels are maintained.

Abstract: This application discloses a method and an apparatus for processing an audio signal. The method includes obtaining a first speech signal acquired by a first device; performing frame blocking on the first speech signal, to obtain multiple speech signal frames; converting the multiple speech signal frames into multiple first frequency domain signal frames; performing aliasing processing on a first sub-frequency domain signal frame among the multiple first frequency domain signal frames with a frequency lower than or equal to a target frequency threshold, and retaining a second sub-frequency domain signal frame among the multiple first frequency domain signal frames with a frequency higher than the target frequency threshold, to obtain multiple second frequency domain signal frames, the target frequency threshold being related to a sampling frequency of a second device; and performing frame fusion on the multiple second frequency domain signal frames, to obtain a second speech signal.

Abstract: To enable a patron of a venue attending a performance to feel as if he or she is ‘front and center’ even if located much farther away or angled from a stage of the venue, system and methods that enhance audio quality of audio signals especially over the voice frequency spectrum to generate a processed audio signal including an increased voice frequency spectrum are provided. The processed audio signal may be communicated to a receiving system with a headset or other transducer (e.g., loudspeakers) that allows for ambient sounds to be heard by a patron and for the patron to control volume of the processed audio signal, thereby mixing the processed audio signal with the ambient sounds. The receiving system may be rented and an operator may control usage for both performance or long-term rentals using various control features of the processed audio signals and/or receiving system.

Abstract: Methods and apparatus are provided for automatically adjusting, by an audio device, the SPL of its audio output. As described herein, the SPL is adjusted based on detected ambient noise. According to aspects, audio device iteratively adjusts the SPL based on the ambient noise. According to aspects, the SPL is adjusted to be greater than the ambient noise by a threshold SPL amount. According to aspects, the audio device outputs sound in substantially a first direction and the microphone detects sound substantially outside of the first direction. The adjusted sounds are output by the audio device.

Abstract: Methods, systems, and apparatuses are described herein for improved processing audio in a video stream. A system may split audio in a frame of video content into multiple bands based on their audio levels. The system may then dynamically compress and dynamically normalize the audio level in each band. When dynamically compressing the bands, the system may determine, based on stored information, what audio level range is acceptable for an end user and may smooth and maintain the ranges of the audio to be within the acceptable range. The system may include the dynamically normalized and dynamically compressed frames as a second audio track in the video content. A computing device receiving the video content may select the second audio track during playback. If an end user selects the second audio track, the video is delivered with the modified sound of the second audio track.

Abstract: A system, method and computer product for training a neural network system. The method comprises applying an audio signal to the neural network system, the audio signal including a vocal component and a non-vocal component. The method also comprises comparing an output of the neural network system to a target signal, and adjusting at least one parameter of the neural network system to reduce a result of the comparing, for training the neural network system to estimate one of the vocal component and the non-vocal component. In one example embodiment, the system comprises a U-Net architecture. After training, the system can estimate vocal or instrumental components of an audio signal, depending on which type of component the system is trained to estimate.

Abstract: Methods and systems for aligning amplification gains in a plurality of interconnected devices are disclosed. The method includes receiving by each device limiter gain attenuations and brownout gain attenuations broadcasted by the plurality of devices and selecting the maximum brownout gain attenuation and the maximum limiter gain attenuation. The method includes determining a total attenuation as a sum of the maximum brownout attenuation gain and the maximum limiter attenuation gain. The method includes receiving a frame synchronization signal and adjusting the amplification gain by applying the total attenuation responsive to the frame synchronization signal.

Abstract: At least one exemplary embodiment is directed to a method of testing the earphone for proper sealing then generating a self-administered hearing test.

Abstract: Disclosed herein is a method of constructing and utilizing a sound engineering evaluation and comparison process to allow for improved finished results. Such a method entails the utilization of a high-pass filter for listening evaluation of recorded music or sounds including consistency with low-frequency mixing to allow for a tool to implement changes in relation to the filtered results in order to accommodate sensitivities of the human ear (with the optional inclusion of a comparison method to provide possible further enhanced results and the avoidance of biases). In such a manner, a facilitating method for sound engineering mixing adjustments that provide such accommodations are provided for improved sound recordings for distribution within on-line or recording product frameworks.

Abstract: A method is provided that includes obtaining two or more microphone audio signals; analysing the two or more microphone audio signals for a defined noise type; and processing the two or more microphone audio signals based on the analysis to generate at least one audio signal suitable for automatic speech recognition. A corresponding apparatus is also provided.

Abstract: An audio normalization gain value is applied to an audio signal to produce a normalized signal. The normalized signal is processed to compute dynamic range control (DRC) gain values in accordance with a selected one of several pre-defined DRC characteristics. The audio signal is encoded, and the DRC gain values are provided as metadata associated with the encoded audio signal. Several other embodiments are also described and claimed.

Abstract: At least one exemplary embodiment is directed to a method of generating preferred dynamic range function to process audio reproduced by an earphone device. The function includes processing the audio to improve speech intelligibility. The function is acquired with a self-administered hearing test.

Abstract: An audio signal processing apparatus is provided. The audio signal processing apparatus includes an input terminal receiving an input audio signal, a processor obtaining a difference between a playback loudness of the input audio signal and a desired loudness thereof and correcting a frequency band spectrum of an output audio signal for each of a plurality of frequency bands based on the difference between the playback loudness and the desired loudness of the input audio signal and a relationship between a loudness and a sound pressure for each of the plurality of frequency bands, and an output terminal outputting the output audio signal. The playback loudness is a loudness of the output audio signal when the input audio signal is output without the correction.

Abstract: 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.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: A media system and a method of using the media system to accommodate hearing loss of a user, are described. The method includes selecting a personal level-and-frequency dependent audio filter that corresponds to a hearing loss profile of the user. The personal level-and-frequency dependent audio filter can be one of several level-and-frequency-dependent audio filters having respective average gain levels and respective gain contours. An accommodative audio output signal can be generated by applying the personal level-and-frequency dependent audio filter to an audio input signal to enhance the audio input signal based on an input level and an input frequency of the audio input signal. The audio output signal can be played by an audio output device to deliver speech or music that the user perceives clearly, despite the hearing loss of the user. Other aspects are also described and claimed.

Abstract: The present application relates to a method, system, and computer program product of dynamically adjusting thresholds of a compressor responsive to an input audio signal. A scene switch analyzer receives an input audio signal having a plurality of frequency band components. The scene switch analyzer determines whether a scene switch has occurred in the input audio signal. The frequency band components of the input audio signal are processed. In response to determine that scene switch has not occurred, a distortion audibility system applies slow smoothing to compressor thresholds of the frequency band components. In response to determine that scene switch has occurred, the distortion audibility system applies fast smoothing or no smoothing to the compressor thresholds of the frequency band components.

Abstract: Disclosed are systems and methods for automatic control of gain in audio and video conferencing applications to maintain a predetermined and stable audio level. In one embodiment, a first stage applies a first stage gain based on a long-term estimate of signal level, while a second stage gain, based on a short-term estimate of signal level assists the first stage gain to achieve a target level. Some embodiments of long-term level estimation utilize statistical analysis of a buffer to validate or arrive at a more accurate long-term signal level estimate.

Abstract: A dynamics processing method including receiving an input audio signal determining whether a level of the input audio signal exceeds a threshold value, determining whether a first amount of time, that is not specified to a unique value by a first speed at which a gain applied to the input audio signal is changed, has elapsed from when the level of the input audio signal exceeded the threshold value, and changing the gain applied to the input audio signal at the first speed.

Abstract: Various embodiments of the present disclosure are directed to modifying an input signal. In one example of a process for modifying an input signal, the process includes splitting the input signal into at least a first input part and a second input part, the amplification of at least the first input part with a linear gain to create a first output part, the nonlinear amplification of at least the second input part of the input signal to create a second output part, and summing the first output part and the second output part in order to provide an output signal.

Abstract: A nanomaterial-based gas sensor system comprising a low voltage circuitry which includes a transducer to detect changes in electrical properties of a multi-channel gas sensor array, analog signal conditioning, and an A/D conversion to provide a signal to a digital back-end.

Abstract: A hearing aid and a hearing aid output voice adjustment method thereof are disclosed. The hearing aid includes a microphone, a voice output adjustment module, a voice player, and a voice processing module. The microphone is used for receiving an input voice. The voice player is used for outputting an output voice. The voice processing module is used for receiving the input voice and modifying to the output voice, wherein the voice processing module is used for adjusting sound output energies of at least N steps, wherein the sound output energy of the nth step is greater than that of the n?1th step, 2?n?N, N?3. When the voice processing module adjusts the sound output energy of the Nth step, the voice output adjustment module also simultaneously reduces at least a portion of frequencies of the input voice.

Abstract: A computer, including a processor and a memory, the memory including instructions to be executed by the processor to determine a first root-mean-square signal level of an audio data stream with a first delay, apply a bass shelf filter to the audio data stream based on the first root-mean-square signal level and an audio volume level, determine a second root-mean-square signal level of an audio data stream with a second delay and apply a treble shelf filter to the audio data stream based on the second root-mean-square signal level and the audio volume level. The instructions include further instructions to output the audio data stream to an audio amplifier and one or more speakers.

Abstract: A hearing device system includes at least three devices configured to communicate wirelessly via near field magnetic induction wireless link(s) based on a predetermined communication protocol; wherein the predetermined communication protocol defines a frame comprising at least a first time slot and a second time slot; wherein at least a first pair of devices of the at least three devices is configured to communicate in the first time slot, and at least a second pair of devices of the at least three devices is configured to communicate in the second time slot; and wherein the first pair of devices is configured to transmit in the first time slot with a first transmit power level, and the second pair of devices is configured to transmit in the second time slot with a second transmit power level that is different from the first transmit power level.

Abstract: A method (300) of operating a hearing aid system wherein the acoustical output signal intensity levels are confined to a range that primarily high-spontaneous rate auditory nerve fibres respond to, hereby providing sound processing that may benefit individuals with an auditory neurodegeneration, a computer-readable storage medium having computer-executable instructions, which when executed carries out the method, a hearing aid system (100, 200) adapted to carry out the method and a method of fitting a hearing aid system.

Abstract: A hearing protection device includes: a sub-band splitting module configured to divide a first microphone input signal into sub-band signals having a first sub-band signal and a second sub-band signal; an estimator module configured to estimate signal strength parameters of respective sub-band signals, the signal strength parameters having a first signal strength parameter of the first sub-band signal, and a second signal strength parameter of the second sub-band signal; a multiband limiter; and a limiter controller; wherein the limiter controller is configured to determine gain reductions for the sub-band signals of the first microphone input signal, the gain reductions having a first gain reduction and a second gain reduction, and wherein the limiter controller is configured to control the multiband limiter to apply the second gain reduction to the second sub-band signal, wherein the second gain reduction for the second sub-band signal is based on the first signal strength parameter.

Abstract: At least one exemplary embodiment is directed to a method of generating preferred dynamic range function to process audio reproduced by an earphone device. The function includes processing the audio to improve speech intelligibility. The function is acquired with a self-administered hearing test.

Abstract: An apparatus and methods for timing mismatch in a power amplifier includes a segmented PA with two-path timing mismatch calibration to improve ACLR performance over different signal transitions, process, voltage and temperature (PVT) variations and device aging; a fast and efficient algorithm for measuring and calibrating the delay of two paths (signal path and control path); a signal magnitude variation detection circuit, such as flash ADC, with improved comparator's performance for RF signal processing and minimum delay. A method for choosing the threshold voltage of the magnitude variation detection circuit, according to status of the signals and orthogonal frequency-division multiplexing (OFDM) related standards; other critical blocks.

Abstract: A context-aware voice guidance method is provided that interacts with other voice services of a user device. The voice guidance does not provide audible guidance while the user is making a verbal request to any of the voice-activated services. Instead, the voice guidance transcribes its output on the screen while the verbal requests from the user are received. In some embodiments, the voice guidance only provides a short warning sound to get the user's attention while the user is speaking on a phone call or another voice-activated service is providing audible response to the user's inquires. The voice guidance in some embodiments distinguishes between music that can be ducked and spoken words, for example from an audiobook, that the user wants to pause instead of being skipped. The voice guidance ducks music but pauses spoken words of an audio book in order to provide voice guidance to the user.

Abstract: An input-output apparatus is provided. The apparatus includes a lower-level layer interface connected to a field device, a higher-layer interface connected to a controller, a maintenance interface connected to a setting apparatus, a switch, and a control device. The switch switches between a first connection and a second connection. The control device, in an engineering mode, permits the switch to transition freely between the first connection and the second connection, and in an operation mode, controls the switch to switch to the first connection and prohibits communication through the maintenance interface.

Abstract: Disclosed are systems and methods for audio processing and sound personalization signal processing. Information suitable to derive an indication of a user's hearing ability is received, and a representative indication of hearing ability is determined for the user. A set of audio processing parameters are determined for the representative indication of hearing ability and a sound personalization signal processing algorithm is configured based on the set of audio processing parameters. An audio signal is received on an entertainment system and is processed using the configured sound personalization signal processing algorithm. The processed audio signal is then outputted from the entertainment system.

Abstract: A method operates a hearing aid where the hearing aid generates an input signal from acoustic signals from the environment. The hearing aid has a signal processor which is configured to modify the input signal and thereby generate an output signal. The signal processor has an automatic gain control for modifying the input signal, and has a compressor that can be operated with a compression scheme. The environment is divided into a plurality of directions of which one is selected by a direction determination unit as a relevant direction. The input signal is modified in a direction-dependent manner by the compressor being operated with a compression scheme, which is set dependent on the relevant direction, so that acoustic signals from the relevant direction are emphasized compared to acoustic signals from other directions.

Abstract: A compressor, an adder circuit, and an operation method thereof are provided. The compressor includes a first adder circuit and a second adder circuit. The first adder circuit receives a plurality of input values. The first adder circuit outputs a first inverted sum value (an inverted signal of a sum value) and a first inverted carry value (an inverted signal of a carry value). One of a plurality of input terminals of the second adder circuit is coupled to the first adder circuit to receive one of the first inverted sum value and the first inverted carry value. The second adder circuit outputs a second inverted sum value and a second inverted carry value.

Abstract: Methods and systems for cell lysis are disclosed. Particular embodiments relate to applying acoustic energy to a biological sample located in a sample chamber.

Abstract: Apparatus, systems, articles of manufacture, and methods for volume adjustment are disclosed herein. An example method includes identifying media represented in an audio signal, accessing metadata associated with the media in response to identify the media in the audio signal, determining, based on the metadata, an average volume for the media, and adjusting an output volume of the audio signal based on an average gain value, the average gain value determined based on the average volume for the media.

Abstract: An electronic device determines a first audio event of a first media content item and modifies the first media content item by superimposing a first set of data that corresponds to the first media content item over the first audio event. The first audio event has a first audio profile configured to be presented over a first channel for playback. The first set of data has a second audio profile configured to be presented over the first channel for playback. Playback of the second audio profile is configured to be masked by the first audio profile during playback of the first media content item. The electronic device transmits, to a second electronic device, the modified first media content item.

Abstract: A system may include a digital modulator configured to modulate an input signal received at an input of the digital modulator to generate a modulated input signal at an output of the digital modulator, a digital gain element having a digital gain and coupled to the digital modulator, an open-loop Class-D amplifier coupled to an output of the digital modulator and configured to amplify the modulated input signal, wherein the open-loop Class-D amplifier is powered from a variable power supply having a variable supply voltage which is variable in response to one or more characteristics of the input signal, and a control circuit configured to control the digital gain to approximately cancel changes in an analog gain of the open-loop Class-D amplifier due to variation in the variable supply voltage in response to the one or more characteristics of the input signal.

Abstract: A system may include an input configured to receive a first signal representative of a second signal to be driven to an amplifier input of an amplifier, processing circuitry configured to process the first signal in order to generate the second signal from the first signal such that the processing circuitry limits a current driven by the amplifier to an output load of the amplifier, and an output configured to drive the second signal to the amplifier input.

Abstract: A method for identifying active resource units in a high-efficiency wireless system includes quantizing phase angles of samples received in a high-efficiency short training field of a transmission, performing a transform operation on quantized phase angle samples to derive transmitted power, comparing transmitted power of an individual resource unit as determined from transformed quantized phase angle samples to transmitted power of other resource units as determined from transformed quantized phase angle samples, and identifying, as active, resource units whose transmitted power, as determined from transformed quantized phase angle samples, bears a first predetermined relationship to transmitted power of the other resource units. Transmitted power of a resource unit may be compared to the total transmitted power of all resource units, or to the transmitted power of one of the other resource units whose transmitted power is a maximum transmitted power of all resource units.

Abstract: A system and method for dynamically adjusting the quantity of volume levels and/or the loudness delta are disclosed. The quantity of volume levels may the number of increments between a minimum and maximum volume. The loudness delta may be the difference between adjacent volume levels. The device may one or more microphones to determine a background noise level. Based on the determined background noise level, and after receiving a user input to adjust the volume, the device may adjust the quantity of volume levels and/or the loudness delta depending on if the background noise level is increased or low.

Abstract: 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.

Abstract: 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.

Abstract: A computer-implemented method for restoring a wrapped audio signal comprising a plurality of digitised signal samples at respective sample times, the method comprising: estimating a sequence of corrections comprising a sequence of numerical values the estimating comprising, for each signal sample: applying, at the sample time, a numerical filter to each of a set of potential corrections to determine a filtered value associated with each set of potential integer corrections wherein the filter enhances the filtered value at sample times when a change in a degree of wrapping occurs relative to sample times when a change in degree of wrapping does not occur; determining a cumulative objective over a plurality of signal samples by accumulating objective values and determining a sequence by selecting for each sample time a correction from the set of potential corrections wherein the correction for each sample time are selected to optimise the cumulative objective.

Abstract: The present disclosure provides a multi-channel morphing digital audio filter, including methods and systems for real-time interpolation between a plurality of fixed frequency responses. Real-time interpolation may be performed simultaneously along three separate axes. Various embodiments of the present technology include receiving, in real-time via a digital filter module, at least one acoustic signal and one or more control input signals indicative of one or more corresponding interpolator values. The methods and systems may further include determining one or more filter coefficients based on a linear interpolation in an encoded space of a plurality of fixed frequency responses, generating a filter from the one or more determined filter coefficients, and applying the generated filter to the at least one received acoustic signal. In some embodiments, frequency and amplitude (i.e., angle and radius) of poles and zeros of the filters may be processed independently, and poles may be unconditionally stable.