A system and method for digital processing including a gain element to process an input audio signal, a high pass filter to then filter the signal and create a high pass signal, a first filter module to filter the high pass signal and create a first filtered signal and a splitter to split the high pass signal into two high pass signals. The first filter module filters one high pass signals before a first compressor modulates the signal or a high pass signal to create a modulated signal. A second filter module filters the modulated signal to create a second filtered signal that is processed by a first processing module including a band splitter that splits the signal into low and high band signals that are then modulated by compressors. A second processing module processes the modulated low and high band signals to create an output signal.

In various embodiments, a sound modification application selectively modifies one or more sounds included in one or more audio signals. In operation, the sound modification application determines classifications associated with multiple sounds included in one or more audio signals. The sound modification application selects a first frequency sub-band of a first sound included in the multiple sounds based on a first classification associated with the first sound. The sound modification application then modifies the first frequency sub-band of the first sound, without modifying at least a second frequency sub-band of the first sound, to generate a modified audio signal.

A sound system includes a sound source having an analogue audio signal output and a sound volume control; signal amplifier apparatus (10) comprising an analogue signal input (12) configured to be coupled to the analogue audio signal output of the sound source, an analogue signal output (18) and an amplifier module (62) coupled between the signal input and the signal output. The amplifier module (62) has a fixed amplification gain within a range of 10 to 12 decibels. Control of the audio signal is effected solely by the sound volume control of the sound source. The amplifier module (64) also provides a fixed gain bass boost of 6 decibels. The system provides linear amplification and allows a user to reduce the volume of the sound source, which significantly reduces signal distortion. Volume control is at the sound source and not at the amplifier module, which optimises amplification quality and reduces the power requirement of the amplifier apparatus (10).

A parametric equalizer includes a first parametric equalizer circuit, a second parametric equalizer circuit, a first multiplication circuit, a second multiplication circuit, an addition circuit, and a weighting control circuit. The first parametric equalizer circuit processes an input signal to output a first output signal. The second parametric equalizer circuit processes the input signal to output a second output signal. The first multiplication circuit multiplies the first output signal and a first weighting value to generate a first adjusted output signal. The second multiplication circuit multiplies the second output signal and a second weighting value to generate a second adjusted output signal. The addition circuit combines the first adjusted output signal and the second adjusted output signal to generate an equalizer output signal. The weighting control circuit dynamically adjusts the first weighting value and the second weighting value according to the equalizer output signal.

For detecting and resolving bad audio during conferencing, methods, apparatus, and systems are disclosed. One apparatus includes a processor and a memory that stores code executable by the processor. The processor detects bad audio for a conference call, the conference call involving a plurality of participants. The processor switches a first input stream to an analysis mode, where the bad audio corresponds to a first one of a plurality of input streams, the first input stream associated with a first participant. The processor sends a conference output channel to the first participant while in the analysis mode and concurrently analyzes the first input stream using a plurality of audio tools while in the analysis mode. The processor returns the first input stream to a conferencing mode in response to resolving the bad audio.

A method for adjusting sound playback of a portable device for constancy notwithstanding different environments outputs from the portable device detectable audio signals inaudible to user and the device receives reflected audio before the portable device is actually commanded to play an audio file. A list of volume weightings for reflected audio is calculated. Before commencing playback of the audio file, the portable device obtains reference volume weightings from a list according to the current volume setting, and calculates adjustment coefficients for different frequency bands based on weightings of the reference volume list and of the reflected audio list. The audio signals of the audio file are output after adjustment. A portable device is also disclosed.

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.

Systems and methods utilize a modified genetic algorithm for adapting an off-the-shelf audio system, such as in a high-end television, to a given, particular room or other physical location presenting a specific or unique auditory environment with a set of acoustic properties. An audio system is adapted to a given room by determining an IIR based EQ solution via iterative techniques, including an iterative technique based upon a genetic algorithm adapted for an audio frequency response equalization application. In a variant, an audio system is adapted to a particular room, adjust the EQ across a microphone's bandwidth while preserving the factory-calibrated EQ response across the remaining bandwidth.

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.

Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, while a playback device is a part of a first zone group that includes the playback device and at least one first playback device, the playback device applies a first audio setting. The embodiment also includes the playback device joining a second zone group that includes the playback device and at least one second playback device. The embodiment further includes the playback device applying a second audio setting based on an audio content profile corresponding to the second zone group.