A method for controlling an FIR filter includes generating, based on operation information, a first and second control data, generating a second amplitude characteristic, and setting filter coefficients of the FIR filter based on the second amplitude characteristic. The first control data indicates an amount of expansion/compression in a frequency axis direction of a first amplitude characteristic that corresponds to a predetermined transfer function that is expressed as a function of an angular frequency, and the amount of expansion/compression is an integer value or a non-integer value. The second control data indicates an amount of shift in the frequency axis direction of the first amplitude characteristic. The second amplitude characteristic is generated by expanding/compressing the first amplitude characteristic in the frequency axis direction in accordance with the first control data and shifting the first amplitude characteristic in the frequency axis direction in accordance with the second control data.

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 of enhancing dialog intelligibility in an audio signal, comprising determining a speech confidence score that the audio content includes speech content, determining a music confidence score that the audio content includes music correlated content, in response to the speech confidence score, and applying a user selected gain of selected frequency bands of the audio signal to obtain a dialogue enhanced audio signal. The user selected gain is smoothed by an adaptive smoothing algorithm, an impact of past frames in said smoothing algorithm being determined by a smoothing factor, the smoothing factor being calculated in response to the music confidence score, and having a relatively higher value for content having a relatively higher music confidence score and a relatively lower value for speech content having a relatively lower music confidence score, so as to increase the impact of past frames on the dialogue enhancement of music correlated content.

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 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.

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.

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 controlling an FIR filter includes generating, based on operation information, a first and second control data, generating a second amplitude characteristic, and setting filter coefficients of the FIR filter based on the second amplitude characteristic. The first control data indicates an amount of expansion/compression in a frequency axis direction of a first amplitude characteristic that corresponds to a predetermined transfer function that is expressed as a function of an angular frequency, and the amount of expansion/compression is an integer value or a non-integer value. The second control data indicates an amount of shift in the frequency axis direction of the first amplitude characteristic. The second amplitude characteristic is generated by expanding/compressing the first amplitude characteristic in the frequency axis direction in accordance with the first control data and shifting the first amplitude characteristic in the frequency axis direction in accordance with the second control data.

A filter determination system, a filter determination device, a filter determination method and a program with which a filter can be appropriately determined are provided. A processing device according to this embodiment includes a frequency response acquisition unit that acquires a frequency response of a filter for an audio signal, a smoothing unit that smoothes the frequency response and obtains a smoothed response, a candidate point determination unit that determines candidate split points based on a bottom position of the smoothed response, and a split point determination unit that determines one or more band split points from the candidate split points.