Embodiments relate generally to hearing protection devices which incorporate a calibration mode which allows the user to perform a calibration test to determine an individualized hearing threshold. Embodiments of the device may also comprise a normal mode to prevent input signals greater than the individualized hearing threshold from being transmitted to the user's ear canal. In addition, embodiments may comprise a normal mode configured to limit input signals to less than or equal to a standard industry threshold in the case no calibration test has been completed by the user. This may increase user comfort and prevent hearing damage.

Embodiments of the present technology may include a method of making a thermoplastic composite strand. The method may include melting a reactive thermoplastic resin to form a molten reactive resin. The method may also include fully impregnating a plurality of continuous fibers with the molten reactive resin in an impregnation device. The method may further include polymerizing the molten reactive resin to form a thermoplastic resin matrix. In addition, the method may include cooling the thermoplastic resin matrix to form a thermoplastic composite strand.

Disclosed are an audio ADC for supporting voice wake-up and an electronic device. The audio ADC includes a programmable gain amplifier (PGA) having an input terminal for receiving an audio signal; a bypass switch having an input terminal for receiving an analog audio signal; and a successive approximation ADC having input terminals respectively connected to output terminals of the PGA and the bypass switch; the PGA gains and amplifies the audio signal, the bypass switch bypasses the PGA, and outputs the analog audio signal; the successive approximation performs analog-to-digital conversion with noise shaping on the analog audio signal after gain amplification at a first sampling rate/oversampling rate when the audio ADC is normal working, and turns off noise shaping when the audio ADC is sleep, performs analog-to-digital conversion on the analog audio signal output by the bypass switch at a second sampling rate/oversampling rate, and outputs to a DSP.

An audio hub is disclosed including a plurality of daughterboards and a mainboard communicatively coupled to each of the plurality of daughterboards. Each of the plurality of daughterboards is associated with an audio channel and each audio channel is independent from the other of the audio channels associated with the other ones of the plurality of daughter boards. Further, each daughterboard includes at least one wireless transceiver operably configured to wirelessly transmit an audio signal associated with the corresponding audio channel from a musical instrument over a network to the audio playback device. The mainboard is configured so that, for each of the plurality of daughterboards, the mainboard is operable to independently modify at least one characteristic of the audio signal from the musical instrument before the audio signal is transmitted to the audio playback device.

Systems, devices, and methods are described for providing loudspeaker protection. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model, and associated parameters, of the loudspeaker as well as a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the voltage and current sense data, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

An example method includes receiving data indicating a configuration of one or more playback devices. The one or more playback devices may include one or more transducers. The method further includes, based on the received data, associating each of one or more audio streams respectively with at least one transducer of the one or more transducers. The method further includes generating the one or more audio streams and sending at least one of the generated one or more audio streams to each of the one or more playback devices. An example non-transitory computer readable medium and an example computing device related to the example method are also disclosed herein.

Approaches are provided for an apparatus that includes an input buffer, an analog-to-digital converter coupled to the input buffer, a decompress module coupled to the analog to digital converter, and a gain control module coupled to the input buffer and the decompress module. The input buffer has a first adjustable gain and operating in the analog domain. The analog-to-digital converter converts the input analog data received from the input buffer into digital data. The decompress module operates in the digital domain, and is configured to decompress the digital data received from the analog-to-digital converter. The decompress module has a second adjustable gain and produces an output digital signal. The gain control module determines when to compensate for changes in characteristics the input analog data by selectively controlling the first gain of the input buffer in the analog domain and the second gain of the decompress module in the digital domain.

An electrolarynx includes a case housing tone-producing circuitry, a power switch to turn on the circuitry, a control button (i.e., a pushbutton) to actuate the power switch, and a pressure-sensitive-resistor (PSR) that is physically coupled to the pushbutton. The tone-producing circuitry is configured to respond to variations in PSR resistance that are caused by a user depressing the pushbutton. User-selected modes, that are selectable in one embodiment with a mode switch, include multiple frequency-varying modes (FVMs) in which the frequency of the electrolarynx tone is varied with different sensitivities to variations in PSR resistance, and multiple volume-varying modes (VVMs) for varying the volume of the electrolarynx tone with different sensitivities. A preferred embodiment provides a nonlinear frequency characteristic that facilitates operation at the low end of the pushbutton-controlled (i.e., PSR-controlled) frequency range, accomplishing same by applying suitable shape factors to frequencies of a linear frequency characteristic.

An audio playback path of an audio apparatus includes a digital modulator, a digital-to-analog converter (DAC), and a power amplifier. The digital modulator receives a playback signal corresponding to playback audio content and generates a digital input signal in accordance with the playback signal. The DAC receives the audio input signal and generates an analog preamplifier signal. The power amplifier generates an audio output signal in accordance with the preamplifier signal and an analog attenuation determined by the analog attenuation signal. The apparatus may include a volume control input to receive a volume control signal and a playback controller configured to perform operations including generating an analog attenuation signal in accordance with the volume control signal, monitoring a playback state indicated by the playback parameters, and responsive to detecting the playback state satisfying the playback criterion, modifying a selected playback parameter to improve a performance parameter of the playback path.

Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, determining how much to attenuate the audio signal when a power level of the audio signal exceeds a threshold power level, combining the audio signal with an auxiliary attenuation signal from an auxiliary attenuation source and a compressed attenuation signal from the feedback compressor circuit to create a combination signal, and generating an audio output signal of the feedback compressor circuit based on the combination signal.