In some embodiments, a window circuit for an audio amplification system can include a pulse train generator configured to generate a train of rectangular pulses having M amplitude values, with the quantity M being an integer greater than 1, and M-1 accumulators arranged in series to transform the train of rectangular pulses into an output that is representative of an M-th order window. In some embodiments, such a window circuit can be utilized for a calibration circuit that includes a gain adjustment circuit configured to generate a correction signal to compensate for a gain variation of an audio amplifier based at least in part on a window of frequency at or about a frequency of a calibration tone applied to the audio amplifier.

An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image.

A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker. The values obtained by multiplying these gains by the gain of the virtual speaker are set as the gains of the lower right and lower left speakers for fixing a sound image at the target sound image position. The present technology can be applied to sound processing apparatuses.

A method for processing a digital audio broadcast signal includes: separating an analog audio portion and a digital audio portion of the digital audio broadcast signal; determining the loudness of the analog audio portion and the digital audio portion over a first short time interval; using the loudness of the analog and digital audio portions to calculate a short term average gain; determining a long term average gain; converting one of the long term average gain or the short term average gain to dB; if an output has been blended to digital, adjusting a digital gain parameter by a preselected increment to produce a digital gain parameter; if an output has not been blended to digital, setting the digital gain parameter to the short term average gain; providing the digital gain parameter to an audio processor; and repeating the above steps using a second short time interval.

Methods, systems, and computer program products of automatic de-essing are disclosed. An automatic de-esser can be used without manually setting parameters and can perform reliable sibilance detection and reduction regardless of absolute signal level, singer gender and other extraneous factors. An audio processing device divides input audio signals into buffers each containing a number of samples, the buffers overlapping one another. The audio processing device transforms each buffer from the time domain into the frequency domain and implements de-essing as a multi-band compressor that only acts on a designated sibilance band. The audio processing device determines an amount of attenuation in the sibilance band based on comparison of energy level in sibilance band of a buffer to broadband energy level in a previous buffer. The amount of attenuation is also determined based on a zero-crossing rate, as well as a slope and onset of a compression curve.

Workplace safety is a principal concern in many environments. Protecting user ears from damage due to extended exposure to unacceptably high sound volume serves as an important component to workplace safety. Monitoring a device, such as a phone, utilized by a user often provides an incomplete picture of the sound level presented to the user. As provided herein, monitoring a user's sound exposure on one device may cause the sound level presented to the user from a second device to become limited. Additionally, over time the sound level limits may be adjusted based on the cumulative historic sound exposure. As a result, the user may avoid exposure to unacceptably high sound levels originating from more than one source and/or over an extended period of time.

Exemplary multipath digital microphone described herein can comprise exemplary embodiments of adaptive ADC range multipath digital microphones, which allow low power to be achieved for amplifiers or gain stages, as well as for exemplary adaptive ADCs in exemplary multipath digital microphone arrangements described herein, while still providing a high DR digital microphone systems. Further non-limiting embodiments can comprise an exemplary glitch removal component configured to minimize audible artifacts associated with the change in the gain of the exemplary adaptive ADCs.

In one aspect, an example method includes (i) determining, by a playback device, a first loudness level of a first portion of first media content from a first source while the playback device presents the first media content, with the first portion having a first length; (ii) switching, by the playback device, from presenting the first media content from the first source to presenting second media content from a second source; (iii) based on the switching, determining, by the playback device, second loudness levels of second portions of the first media content while the playback device presents the second media content, with the second portions having a second length that is shorter than the first length; and (iv) while the playback device presents the second media content, adjusting, by the playback device, a volume of the playback device based on one or more of the second loudness levels.

The present invention provides a deep learning based method and system for processing sound quality characteristics. The method comprises: obtaining data characteristics of an audio data to be processed by extracting features from user preference data including the audio data to be processed; based on the data characteristics, generating a sound quality processing result of the audio to be processed by using a trained baseline model; wherein the baseline model is a neural network model trained by using audio data behavioral data, and other relevant data from multiple users or a single user.

The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.