An electronic device and method that automatically adjusts an audio output volume level based on a live environmental acoustic scenario input via a microphone using a machine learning algorithm trained with Human Activity Recognition (HAR). Equipped with such an intelligence the electronic device classifies ambient sounds occurring in the environment of the listening area in which the device is situated into different acoustic scenario mappings such a voice or conversation, for an ambient human conversation detected event, and noise, such as for example a vacuum cleaner or dish washer noise detected event, and automatically adjust the audio output volume accordingly.

Apparatus, systems, articles of manufacture, and methods for volume adjustment are disclosed herein. An example method includes collecting data corresponding to a volume of an audio signal as the audio signal is output through a device, when an average volume of the audio signal does not satisfy a volume threshold for a specified timespan, determining a difference between the average volume and a desired volume, and applying a gain to the audio signal to adjust the volume of the audio signal to the desired volume, the gain determined based on the difference between the average volume and the desired volume.

An apparatus for providing an output signal to an audio transducer comprises: one or more signal paths for receiving respective digital audio input signals, applying respective digital gains, and outputting respective amplified digital audio input signals; one or more inputs for receiving one or more volume parameters associated with the digital audio input signals; converter circuitry, coupled to the one or more signal paths, for converting the one or more amplified digital audio input signals into the analogue domain, and outputting an analogue audio input signal; an analogue gain element, for applying an analogue gain to the analogue audio input signal and outputting the output signal; and a control circuit, coupled to the one or more signal paths, operative to select the analogue gain based on a comparison of the volume parameters and the one or more digital audio input signals as multiplied by the volume parameters, and to select the respective digital gains for each digital audio input signal so that an overall gain in the respective signal path corresponds to a volume parameter associated with the respective digital audio input signal.

An audio system has an amplifier for driving an audio actuator and includes a switched-mode power supply that draws power from a power source (e.g., battery) to supply power to the amplifier, a capacitor charged by the switched-mode power supply to supply power to the amplifier, and a feed-forward compressor that performs dynamic range compression of an audio input to provide an audio output for amplification by the amplifier. The compressor includes a sidechain frequency-biasing filter that generates a frequency-biased version of the audio input that is attenuated as frequency increases which causes the compressor to decrease the compression as frequency increases. A control block limits current drawn from the battery by the switched-mode power supply independent of audio input frequency, but the frequency-biasing filter enables the amplifier to service audio power transients greater than the current-limited power supply can supply by advantageously concurrently sourcing extra power from the capacitor.

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.

Provided are a method and device for audio mixing, and non-transitory computer-readable medium. The method comprises: acquiring a current system volume parameter of a terminal apparatus; adjusting an audio mixing parameter for audio material according to the system volume parameter and a frequency range of the audio material; performing audio mixing processing on the audio material according to the audio mixing parameter; and playing the processed audio material. In the invention, an audio mixing parameter of audio material can be adjusted according to an acquired current system volume parameter of a terminal apparatus and a frequency of the audio material, thereby applying different audio mixing parameters to the audio material according to different system volume parameters, and accordingly playing the audio material according to the audio mixing parameters.

Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, receiving an auxiliary attenuation signal from an auxiliary attenuation source, determining a threshold power level based on a value of the auxiliary attenuation signal, determining an output power level of the audio signal exceeds the threshold power level, combining the audio signal with the auxiliary attenuation signal from the 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.

An apparatus for processing an audio signal includes an audio signal analyzer and a filter. The audio signal analyzer is configured to analyze an audio signal to determine a plurality of noise suppression filter values for a plurality of bands of the audio signal, wherein the analyzer is configured to determine a noise suppression filter value so that a noise suppression filter value is greater than or equal to a minimum noise suppression filter value and so that the minimum noise suppression value depends on a characteristic of the audio signal. The filter is configured for filtering the audio signal, wherein the filter is adjusted based on the noise suppression filter values.

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.

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.