H03G5/165

EQUALIZATION IN A MULTI-PATH AUDIO AMPLIFIER FOR CANCELING VARIATIONS DUE TO MULTI-PATH OUTPUT IMPEDANCE DIFFERENCES

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.

SYNCHRONIZED CONTROLLER
20230164489 · 2023-05-25 · ·

A system and method are described herein for configuring an audio distribution system, comprising a Redis server, the Redis server adapted to store Redis data to be used in configuring the audio distribution system; a plurality of audio devices, the plurality of audio devices and Redis server interconnected to form the audio distribution system, wherein each of the plurality of audio devices comprises—at least one processor; an electronic communications interface operatively connected to the at least one processor and adapted to receive data from a user and transfer the data to the at least one processor; and a memory operatively connected with the at least one processor, wherein the memory stores computer-executable instructions that, when executed by the at least one processor, causes the at least one processor in a first audio device to execute a method for configuring the audio distribution system that comprises: establishing communications using the electronic communications interface between the user and the at least one processor of the first audio device, such that data input by the user is received by the at least one processor of the first audio device; establishing communications to each of the remaining plurality of audio devices and Redis server in the audio distribution system; obtaining information from each of the remaining plurality of audio devices with which communications have been established, such information including one or more of an audio device name, part number, serial number, internet protocol address number, and physical location; receiving configuration information from the user that pertains to a specific audio device of the plurality of audio devices in the audio distribution system that, when installed on a specific audio device, causes the specific audio device to operate in a known manner; and copying that configuration information to others of the same specific type of audio device in the audio distribution system.

DEEP ENCODER FOR PERFORMING AUDIO PROCESSING

Embodiments are disclosed for determining an answer to a query associated with a graphical representation of data. In particular, in one or more embodiments, the disclosed systems and methods comprise receiving an input including an unprocessed audio sequence and a request to perform an audio signal processing effect on the unprocessed audio sequence. The one or more embodiments further include analyzing, by a deep encoder, the unprocessed audio sequence to determine parameters for processing the unprocessed audio sequence. The one or more embodiments further include sending the unprocessed audio sequence and the parameters to one or more audio signal processing effects plugins to perform the requested audio signal processing effect using the parameters and outputting a processed audio sequence after processing of the unprocessed audio sequence using the parameters of the one or more audio signal processing effects plugins.

Systems and methods for identifying and remediating sound masking

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.

Microphone system
11627414 · 2023-04-11 · ·

A microphone system, comprises a first transducer, for generating a first acoustic signal, and a second transducer, for generating a second acoustic signal. A high-pass filter receives the first signal and generates a first filtered signal, and a low-pass filter receives the second signal and generates a second filtered signal. An adder forms an output signal of the microphone system as a sum of the first filtered signal and the second filtered signal.

Circuits and methods for maintaining gain for a continuous-time linear equalizer
11469730 · 2022-10-11 · ·

A bias structure includes a reference voltage node connected to gate structures of a first NMOS transistor and a second NMOS transistor, a bias voltage node comprising a bias voltage, and a first op amp having a first input connected to the reference voltage, a second input connected to a drain of the first NMOS transistor, and an output connected to gate structures of a first PMOS transistor and a second PMOS transistor. The bias structure further includes a second op amp having a first input connected to the reference voltage, a second input connected to a drain of the second NMOS transistor, and an output connected to a gate structure of a third NMOS transistor and the bias voltage node. The first NMOS transistor matches a transistor of a differential pair of an integrated circuit device.

Automated tuning by measuring and equalizing speaker output in an audio environment

An example method of operation may include identifying speakers and microphones connected to a network controlled by a controller, assigning a preliminary output gain to the speakers used to apply test signals, measuring ambient noise detected from the microphones, recording chirp responses from all microphones simultaneously based on the test signals, deconvolving all chirp responses to determine a corresponding number of impulse responses, and measuring average sound pressure levels (SPLs) of each of the microphones to obtain a SPL level based on an average of the SPLs.

AUDIO OUTPUT ADJUSTMENT

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.

Digital signal generator for audio artefact reduction

A digital signal generator apparatus and method is described. The digital signal generator includes a counter, an integrator and a comparator. The counter counts up or down from an initial counter value dependent on a counter control input. The comparator has a first input coupled to the counter output, a threshold input and a comparator output coupled to the counter control input. The integrator has an input coupled to the counter output and an output coupled to the digital signal generator output. The digital signal generator determines the count direction after the initial direction dependent on the comparison between a threshold value applied to the threshold input and the counter output value. The digital signal generator may implement the generation of a waveform having an approximation to a raised cosine function. The generated waveform may be used for audio artefact reduction in an audio amplifier during mute or unmute operations or during power up power down operations.

Multi-band limiter system and method for avoiding clipping distortion of active speaker

A limiter system for an active speaker may include at least one lowpass filter configured to receive an input signal and output a signal lower than a crossover frequency, at least one highpass filter, configured to receive an input signal and output a signal higher than the crossover frequency, a first allpass filter configured to adjust the phase of the signal lower than the crossover frequency, a second allpass filter configured to adjust the phase of the signal higher than the crossover frequency, a first limiter, configured to receive and limit the signal from the first allpass filter, a second limiter, configured to receive and limit the signal from the second allpass filter, and a mixer, configured to mix the signal lower from the first limiter and the signal from the second limiter.