Abstract: An image sensor in a first earcup captures an image of a pinna. First sound is output by a first transducer in the first earcup located at the pinna and respective second sound is detected by each of one or more microphones in a second earcup located at the pinna. Based on the captured image and the respective second audio sound from each of the one or more microphones, a non-linear transfer function is determined which characterizes how sound is transformed by the pinna. A signal is generated indicative of one or more audio cues for spatializing third sound based on the determined non-linear transfer function.

Abstract: A magnetic sensor mounted on a headband of the personal audio delivery device may output a sensor signal indicative of an interaction between a magnetic field of a transducer of a personal audio delivery device and the magnetic sensor. A head size of a head on which the personal audio delivery device is worn is calculated based on the sensor signal from the magnetic sensor. Based on the head size, a non-linear transfer function is identified which characterizes how sound is transformed via the head with the calculated head size. An output signal is generated indicative of one or more audio cues to facilitate spatialization of sound associated with the output signal based on the identified non-linear transfer function. The sound associated with the output signal is output by the transducer of the personal audio delivery device.

Abstract: An image of a pinna is captured. Based on the image of the pinna, a non-linear transfer function is determined which characterizes how sound is transformed at the pinna. A signal is output indicative of one or more audio cues to facilitate spatial localization of sound via the pinna, where the one or more audio cues is based on the non-linear transfer function.

Abstract: A first sound is output by a first transducer in an earcup located at a pinna. Respective second sound based on the output of the first audio sound is detecting by each of one or more microphones in the earcup located at the pinna. Based on the respective detected second sound, a non-linear transfer function is determined which characterizes how sound is transformed via the pinna. A signal indicative of one or more audio cues for spatializing third sound is generated based on the determined non-linear transfer function.

Abstract: A signal indicative of sound detected by at least one sensor at an audio device is received. The audio device may at least partially covers a pinna and the detected sound may interact with at least a torso of a human body, but can also interact with the head and shoulder. The signal is modulated with a non-linear transfer function to generate a modulated signal indicative of one or more audio cues for spatializing the detected sound while the audio device at least partially covers a pinna. Sound is output by the audio device based on the modulated signal.

Abstract: A system including first and second gain modules, an operator module, and a priori and posteriori modules. The first gain module applies a non-linear function to generate a gain signal based on an amplitude of a first speech signal and an estimated a priori variance of noise included in the first speech signal. The operator module generates an operator based on the gain signal and the estimated a priori variance of noise. The a priori module determines an a priori signal-to-noise ratio based on the operator. The posteriori module determines a posteriori signal-to-noise ratio based on the amplitude of the first speech signal and (ii) the estimated a priori variance of noise.

Abstract: A method includes determining a first filtered signal based on an audio signal; determining a second filtered signal based on the audio signal; determining, based on the first filtered signal and the second filtered signal, a portion of the audio signal corresponding to a sharp noise; determining, based on the first filtered signal and the second filtered signal, a gain signal that, for the portion of the audio signal corresponding to the sharp noise, has a value that is smaller than a value of the gain signal for the remaining portion of the audio signal; and suppressing, based on the gain signal, the sharp noise from an amplifier input signal determined based on the audio signal.

Abstract: An audio processing circuit comprises an analyzer circuit that includes a plurality of energy detector units, and an equalizer circuit that includes a plurality of equalization filters. The equalizer circuit is coupled with the analyzer circuit. The analyzer circuit is configured to receive an audio signal, obtain sub-bands of the audio signal using the energy detector units, measure energy of each sub-band, compare the energy of each sub-band to a threshold energy value and, based on the comparison, determine parameters for an equalization filter for processing the sub-band. The equalizer circuit is configured to receive the audio signal concurrently with reception of the audio signal by the analyzer circuit, obtain the sub-bands using the equalization filters, receive the parameters for the equalization filters from the analyzer circuit, equalize each sub-band by applying the parameters corresponding to the sub-band, and generate an output audio signal that includes the equalized sub-bands.

Abstract: A digital signal processing (DSP) system includes an analog to digital converter, program random access memory (PRAM), N switching devices, and a control module. The analog to digital converter is configured to convert samples of an analog signal into digital samples. The PRAM includes: N PRAM blocks, where N is an integer greater than one; and code for M digital signal processing functions stored in the N PRAM blocks, where M is an integer greater than one. The N switching devices are configured to connect and disconnect the N PRAM blocks, respectively, to and from a power source. The control module is configured to: control the N switching devices; and execute selected ones of the M digital signal processing functions on the digital samples to produce an output.

Abstract: Aspects of the disclosure provide an amplification circuit. The amplification circuit includes an amplifier and a first variable resistive device. The amplifier includes first and second input nodes configured to receive the first and second input electrical signals and first and second output nodes configured to output first and second output electrical signals having amplified voltages relative to the first and second input electrical signals. The first variable resistive device is electrically coupled to the second input node of the amplifier. The first variable resistive device being configured to have a selected resistance value to compensate for a direct current (DC) voltage difference between the first and second input electrical signals based on a DC voltage difference between first and second output electrical signals that are output from the first and second output nodes of the amplifier.

Abstract: Some of the embodiments of the present disclosure provide a device comprising: a first channel configured to receive a signal, wherein the signal comprises (i) a target signal and (ii) a background signal; a second channel configured to receive the signal a time t after the first channel receives the signal; a delay control circuit configured to iteratively determine a fractional delay to maximize a correlation coefficient between the signal on the first channel and the signal on the second channel; and an adaptive fractional delay filter in the first channel configured to adaptively align, in the digital domain, the signal on the first channel with the signal on the second channel based, at least in part, on the fractional delay.

Abstract: A device includes a combining circuitry that receives an incoming signal, and one or more delayed signals from a delay circuitry. The combining circuitry combines the incoming signal and the one or more delayed signals to generate a combined signal. The device includes a comparing circuitry that receives the combined signal from the combining circuitry, and compares a pulse width of the combined signal to a threshold pulse width. When the pulse width of the combined signal is greater than or equal to the threshold pulse width, the comparing circuitry provides the combined signal to an amplifier circuit and provides a null signal to the delay circuitry. The amplifier circuit generates a pulse width modulated (PWM) signal based on the combined signal.

Abstract: A method includes comparing a voltage level detected by a detection circuit to a set of threshold voltages to determine a type of a headphone, determining the headphone to be a first type when a first signal travels through the first path to ground, and determining the headphone to be a second type when a second signal travels through the second path. The detection circuit includes a first path, a second path, and a bias capacitor. The headphone of the second type includes a plug that has a microphone connection.

Abstract: A method of determining inbound roaming market share of a host network is disclosed. The network subscribers are identified by their subscriber identity (IMSI). Each IMSI is categorized based on its roaming status. The roaming categories are as follows: successfully roaming, roaming but steered to another network, not roaming due to errors related to steering, not roaming due to errors not related to steering, roaming below a threshold, roaming below a threshold duration, and roaming on rival networks. SRI-for-SM results of SMS activity are used to determine the count of IMSIs roaming on rival networks. Market share is determined by calculating a quotient of the count of IMSI's categorized as successfully roaming on the host network and the difference between the total count of the IMSIs and the count of IMSIs categorized as not roaming due to errors not related to steering.

Abstract: An audio processing circuit comprises an analyzer circuit that includes a plurality of energy detector units, and an equalizer circuit that includes a plurality of equalization filters. The equalizer circuit is coupled with the analyzer circuit. The analyzer circuit is configured to receive an audio signal, obtain sub-bands of the audio signal using the energy detector units, measure energy of each sub-band, compare the energy of each sub-band to a threshold energy value and, based on the comparison, determine parameters for an equalization filter for processing the sub-band. The equalizer circuit is configured to receive the audio signal concurrently with reception of the audio signal by the analyzer circuit, obtain the sub-bands using the equalization filters, receive the parameters for the equalization filters from the analyzer circuit, equalize each sub-band by applying the parameters corresponding to the sub-band, and generate an output audio signal that includes the equalized sub-bands.

Abstract: A method includes determining a first filtered signal based on an audio signal; determining a second filtered signal based on the audio signal; determining, based on the first filtered signal and the second filtered signal, a portion of the audio signal corresponding to a sharp noise; determining, based on the first filtered signal and the second filtered signal, a gain signal that, for the portion of the audio signal corresponding to the sharp noise, has a value that is smaller than a value of the gain signal for the remaining portion of the audio signal; and suppressing, based on the gain signal, the sharp noise from an amplifier input signal determined based on the audio signal.

Abstract: A method includes determining a preprocessed audio signal by removing some noise from an input audio signal. Here, portions of the preprocessed audio signal that include speech are separated by portions of the preprocessed audio signal that include residual noise. Additionally, the method includes determining an amplified signal by suppressing the preprocessed audio signal over the portions that include residual noise, and maintaining the preprocessed audio signal over the portions that include speech.

Abstract: In accordance with an implementation of the disclosure, systems and methods are provided for providing an estimate for noise in a speech signal. An instantaneous power value is received that corresponds to a frequency index of a portion of the speech signal. A first weighted power value is updated based on the instantaneous power value and a first weighting parameter. A second weighted power value is updated based on the first weighed power value and a second weighting parameter. An estimate of the noise is computed from the instantaneous power value and the second weighted power value.

Abstract: Systems, methods, and other embodiments associated with converting an input signal into an output signal with a different sampling rate. In one embodiment, an apparatus includes a feedforward circuit configured to receive the input signal comprised of discrete data samples with the first sampling rate and to generate a first intermediate value based, at least in part, on a feedforward coefficient and the input signal. The apparatus includes a feedback circuit configured to generate a second intermediate value that is based, at least in part, on a feedback coefficient and a predetermined number of previous samples of the output signal. The apparatus includes a signal combiner configured to combine the first intermediate value and the second intermediate value together to interpolate a data sample of the output signal at the second sampling rate. The output signal is a converted form of the input signal at the second sampling rate.

Abstract: In aspects of concurrent sound source localization of multiple speakers, audio signals from two or more microphones are upsampled, and then the upsampled audio signals are time-multiplexed to a plurality of beamformers. A first sound source received at the two or more microphones is localized at a first beamformer, and a second sound source received at the two or more microphones is localized at a second beamformer, where localizing the second sound source is constrained by the localization of the first sound source. The beamformers can filter the upsampled audio signals using beamformer coefficients from the localizations to produce beamformed audio signals.