Certain embodiments provide methods and systems for managing a sound profile. An example playback device includes a network interface and a non-transitory computer readable storage medium having stored therein instructions executable by the processor. When executed by the processor, the instructions are to configure the playback device to receive, via the network interface over a local area network (LAN) from a controller device, an instruction. The example playback device is to obtain, based on the instruction, via the network interface from a location outside of the LAN, data comprising a sound profile. The example playback device is to update one or more parameters at the playback device based on the sound profile. The example playback device is to play back an audio signal according to the sound profile.

A computer-implemented method comprises identifying an audio playback device configured to reproduce audio content based on an audio input signal, wherein the audio playback device has a previously measured characteristic frequency response; retrieving the previously measured characteristic frequency response associated with the audio playback device; generating, based on the previously measured characteristic frequency response and a preferred frequency response curve for a plurality of users, a frequency response filter that modifies the previously measured characteristic frequency response of the audio playback device to a modified frequency response; processing the audio input signal with the frequency response filter to generate a filtered audio signal that differs in relative amplitude from the audio input signal in one or more frequency bands; and causing the audio playback device to play back the filtered audio signal.

A parametric equalizer includes an equalizer circuit, a first protection circuit, a second protection circuit, and a first addition circuit. The equalizer circuit is arranged to receive an input signal, and process the input signal to generate an output signal. The first protection circuit is arranged to generate a first protection signal according to the output signal, the input signal, and a first processed signal. The second protection circuit is arranged to generate a second protection signal according to the input signal and a second processed signal. The first addition circuit is coupled to the first protection circuit and the second protection circuit, and is arranged to combine the first protection signal and the second protection signal to generate an equalizer output signal.

A multi-path audio amplification system that provides an output drive signal to electromechanical output transducers provides improved undistorted headroom, reduced path switching noise, and/or improved frequency response performance. Multiple signal amplification paths receive an audio input signal and have corresponding multiple output stages that have differing output impedances. A mode selector selects an active one of the multiple signal amplification paths is selected to supply the output drive signal. Outputs of the multiple output stages are coupled to the electromechanical transducer to provide the output drive signal and at least one of the multiple signal amplification paths includes an equalization filter for filtering the audio input signal to compensate for phase or gain differences referenced from the input to the outputs of the multiple output stages due to interaction between the differing output impedances and an impedance of the electromechanical transducer.

Some embodiments of the invention are directed to enabling a user to easily identify the frequency range(s) at which sound masking occurs, and addressing the masking, if desired. In this respect, the extent to which a first stem is masked by one or more second stems in a frequency range may depend not only on the absolute value of the energy of the second stem(s) in the frequency range, but also on the relative energy of the first stem with respect to the second stem(s) in the frequency range. Accordingly, some embodiments are directed to modeling sound masking as a function of the energy of the stem being masked and of the relative energy of the masked stem with respect to the masking stem(s) in the frequency range, such as by modeling sound masking as loudness loss, a value indicative of the reduction in loudness of a stem of interest caused by the presence of one or more other stems in a frequency range.

A method for processing an audio signal includes setting target loudness; receiving metadata of an audio signal from a server; and adjusting the loudness of the audio signal by using the received metadata, so that the loudness of the audio signal corresponds to the set target loudness.

Example electronic devices that may be implemented to generate an audio output via a resonance are disclosed. A received resonance sample of an audio output is categorized with a neural network to obtain an inference profile. The received resonance sample corresponding with an equalization setting. The equalization setting for the audio output is adjusted based on the inference profile.

An adaptive analog parallel combiner circuit for receiver data recovery from a communication signal is provided. The circuit includes a summer that sums outputs of a plurality of filter taps in parallel, including zeroth and first through Nth filter taps, each filter tap having as input the communication signal or a version thereof, wherein N is a finite integer greater than or equal to two. The zeroth filter tap has an amplifier with gain controlled by a zeroth adaptive gain control coefficient, and each of the first through Nth filter taps having an all pass filter and gain controlled amplification, with gain controlled by a corresponding one of a first through Nth adaptive gain control coefficients and the all pass filter implementing a transfer function having a zero and a pole equaling each other and at a base frequency divided by a corresponding integer from one through N.

A circuit and method is disclosed for filtering an audio signal. The circuit has a first quadrature source and multipliers for multiplying the input signal by the I and Q outputs of the quadrature source. The multiplied inputs are then passed through a pair of low pass filters, which may have an adjustable Q factor. The outputs of the low pass filters are then multiplied in a second pair of multipliers by the I and Q outputs, respectively, of a second quadrature source, which will typically be of the same frequency, but different amplitude and phase, of the first quadrature source. The twice-multiplied signals are then summed by an adder to provide an output signal. The circuit may be modified to include a companding circuit between the low pass filters and the second pair of multipliers that determines the amplitude of the input signal, filters it, and compands the signal in a compandor. The compandor may have adjustable parameters. The circuit thus allows for far greater flexibility and control of the processing of the input signal than prior art circuits.

A system, a method for interleaving a digital audio signal including a PWM signal, and a digital audio processing device employing the same in accordance with the present invention are disclosed. The system, the method and the digital audio processing device is capable of minimizing the interference with an RF signal such as an AM signal or an FM signal by interleaving the digital audio signal including the PWM signal.