G10H2250/115

Mode selection for modal reverb
11043203 · 2021-06-22 ·

Methods and systems for performing modal reverb techniques for audio signals are described. The method may involve simplifying a reverb effect to be applied to the audio signal by receiving an IR, dividing the IR into a plurality of sub-bands, using a parametric estimation algorithm to determine respective parameters of the modes included in each sub-band, aggregating the respective modes of the sub-bands into a set; and truncating the set of aggregated modes into a subset of modes. Reverberation of the audio signal may be manipulated based on an IR that itself is based on the truncated subset of modes.

Mode Selection For Modal Reverb

Methods and systems for performing modal reverb techniques for audio signals are described. The method may involve simplifying a reverb effect to be applied to the audio signal by receiving an IR, dividing the IR into a plurality of sub-bands, using a parametric estimation algorithm to determine respective parameters of the modes included in each sub-band, aggregating the respective modes of the sub-bands into a set; and truncating the set of aggregated modes into a subset of modes. Reverberation of the audio signal may be manipulated based on an IR that itself is based on the truncated subset of modes.

Timbre fitting method and system based on time-varying multi-segment spectrum

The disclosure discloses a timbre fitting method and system based on time-varying multi-segment spectrum, the system includes an input device for obtaining audio signals of musical instruments and a segmented multi-model compensation module. The segmented multi-model compensation module learns a timbre of a source musical instrument and a target musical instrument, and establishes a multi-segment model of the sound feature of the source musical instrument and a multi-segment model of the sound feature of the target musical instrument. The sound feature is set to be based on maximum amplitude of the audio signal played the same sequence on the target musical instrument and the source musical instrument, and the audio signal of the sequence is divided into multiple segments according to the amplitude. The sound feature includes frequency spectrums of notes respectively within each amplitude range. The segmented multi-model compensation module establishes a multi-model structure with time-varying gain.

ENHANCING ARTIFICIAL REVERBERATION IN A NOISY ENVIRONMENT VIA NOISE-DEPENDENT COMPRESSION

A system for enhancing artificial reverberation in a noisy listening space includes a sensor configured to generate a signal indicating a current noise level in a listening space, a loudspeaker configured to output sound in the listening space based on an output signal, and at least one processor. The at least one processor is configured to execute instructions to: generate an artificial reverberation signal based on a source signal and a response of a listening space, wherein the artificial reverberation signal includes a first low output-level portion having output levels that are below a compression threshold level and a first high output-level portion having output levels that are above the compression threshold level; generate a compressed artificial reverberation signal based on the artificial reverberation signal by increasing the output levels of the low output-level portion by a first magnitude and increasing the output levels of the high output-level portion by less than the first magnitude; and combine the compressed artificial reverberation signal with the source signal to form an output signal for the listening space.

Timbre fitting method and system based on time-varying multi-segment spectrum

The disclosure discloses a timbre fitting method and system based on time-varying multi-segment spectrum, the system includes an input device for obtaining audio signals of musical instruments and a segmented multi-model compensation module. The segmented multi-model compensation module learns a timbre of a source musical instrument and a target musical instrument, and establishes a multi-segment model of the sound feature of the source musical instrument and a multi-segment model of the sound feature of the target musical instrument. The sound feature is set to be based on maximum amplitude of the audio signal played the same sequence on the target musical instrument and the source musical instrument, and the audio signal of the sequence is divided into multiple segments according to the amplitude. The sound feature includes frequency spectrums of notes respectively within each amplitude range. The segmented multi-model compensation module establishes a multi-model structure with time-varying gain.

INFORMATION PROCESSING DEVICE, TEMPO DETECTION DEVICE AND VIDEO PROCESSING SYSTEM
20200211517 · 2020-07-02 · ·

An information processing device, a tempo detection device and a video processing system are provided. A beat of a piece of performed music is detected from a musical viewpoint. The information processing device includes: an acquisition part that acquires samples of musical sound signals in a time series; an evaluation part that has an adaptive filter using the acquired samples of the musical sound signals as reference signals and using samples of musical sound signals acquired a predetermined time earlier than the samples of the musical sound signals as input signals; and a tempo determination part that sequentially inputs the samples of the musical sound signals to the adaptive filter and determines a tempo corresponding to a musical sound based on a filter coefficient when a value of the filter coefficient of the adaptive filter converges.

Signal processing apparatus
10339907 · 2019-07-02 · ·

A signal processing apparatus has a first memory in which plural pieces of FIR coefficient data used for implementing an FIR filter algorithm are stored, a second memory which stores plural pieces of input data to be subjected to the FIR filter algorithm, and a processor implements the FIR filter algorithm using the plural pieces of FIR coefficient data stored in the first memory and the plural pieces of input data stored in the second memory as many times as the number corresponding to a designated filter order, in which filter algorithm each piece of coefficient data and each piece of input data are multiplied together and resultant products are summed up. The signal processing apparatus is provided, which can implement plural sorts of FIR filter algorithms of filter order which can be changed flexibly.

Filter characteristics changing device
10311845 · 2019-06-04 · ·

When an instruction is provided for changing a characteristic of a set filter which includes a plurality of partial filters and forms a specified characteristic by combining a plurality of partial filters, a processor performs, as crossfading processing for a first filter and a second filter among the plurality of partial filters, fade-out processing of gradually decreasing a degree of contribution of the first filter to the characteristic and fade-in processing of gradually increasing a degree of contribution of the second filter to the characteristic. As a result, unnaturalness occurring at the time of changing filter characteristics is solved.

FILTER CHARACTERISTICS CHANGING DEVICE
20180268793 · 2018-09-20 · ·

When an instruction is provided for changing a characteristic of a set filter which includes a plurality of partial filters and forms a specified characteristic by combining a plurality of partial filters, a processor performs, as crossfading processing for a first filter and a second filter among the plurality of partial filters, fade-out processing of gradually decreasing a degree of contribution of the first filter to the characteristic and fade-in processing of gradually increasing a degree of contribution of the second filter to the characteristic. As a result, unnaturalness occurring at the time of changing filter characteristics is solved.

SIGNAL PROCESSING APPARATUS
20180268794 · 2018-09-20 · ·

A signal processing apparatus has a first memory in which plural pieces of FIR coefficient data used for implementing an FIR filter algorithm are stored, a second memory which stores plural pieces of input data to be subjected to the FIR filter algorithm, and a processor implements the FIR filter algorithm using the plural pieces of FIR coefficient data stored in the first memory and the plural pieces of input data stored in the second memory as many times as the number corresponding to a designated filter order, in which filter algorithm each piece of coefficient data and each piece of input data are multiplied together and resultant products are summed up. The signal processing apparatus is provided, which can implement plural sorts of FIR filter algorithms of filter order which can be changed flexibly.