In various applications, the system provides a method for processing audio signals, including: receiving a request for audio content; receiving an identifier encoded in a personal audio device comprising a transducer for playing audio; retrieving at least one parameter associated with the identifier; and processing the audio content using at least the request, the identifier and the at least one parameter, wherein the processing is customized for the personal audio device based on the at least one parameter associated with the identifier. In various applications the parameter is, one or more of, associated with a specification of the personal audio device, acoustic metrics of the transducer, relates to control of equalization, relates to permission to enable proprietary sonic processing for enhanced acoustic reception of streaming content, relates to acoustic metrics of the transducer and wherein the identifier is associated with permission to enable proprietary sonic processing for enhanced acoustic reception of streaming content, and/or is stored in a chip on the personal audio device, is retrieved from a server in a network, among other things. In various applications, the personal audio device comprises ear buds and the identifier is stored in a non-volatile memory of the ear buds.

A method for flattening power of a musical sound signal, said method being characterized by comprising: determining second values corresponding to respective first values indicating power at a plurality of time points of a musical sound signal each on the basis of the result of a comparison between the present value of the first value and the present value of the second value; and flattening the plurality of first values using the second values corresponding to the plurality of first values, respectively, wherein the second value changes while drawing a predetermined trajectory when, in the result of the comparison, a state where the present value of the second value is larger than the present value of the first value continues.

A musical sound signal generation device continuously connects any one of a connected zeroth delay unit and a connected second delay unit to a fractional delay block and connects at least any one of a new zeroth delay unit and a new second delay unit to at least any one of the fractional delay block other than the fractional delay block connected to a new first delay unit in response to setting any one of the connected zeroth delay unit and the connected second delay unit as the new first delay unit, setting a delay unit in a preceding stage of the new first delay unit as the new zeroth delay unit, and setting a delay unit in a subsequent stage of the new first delay unit as the new second delay unit in accordance with a change in a designated tone pitch.

In various applications, the system provides a method for processing audio signals, including: receiving, by a processor, a digital audio signal from a recorded audio file; analyzing, by the processor, the digital audio signal to identify pitch distortion caused by changes in momentary sound level; determining, by the processor, an amount of compensation of the audio signal to correct the identified pitch distortion; dynamically adjusting, by the processor, the digital audio signal by the compensation amount to correct the identified pitch distortion; and outputting, by the processor, the digital audio signal to an audio transducer device of a listener to improve a listening experience for the listener of the recorded audio file.

A virtual instrument can manage separate static and dynamic samples for various notes that can be played by the virtual instrument. In some cases, the static samples correspond to resonance sounds recorded for an instrument and are the same for every note. However, the dynamic samples may correspond to isolated sounds that are recorded for each variation of a note that can be played. In response to a user's selection of a note on a user interface of the virtual instrument, the virtual instrument may determine a rule for layering the various static and dynamic samples for playback.

An environmental sound generating apparatus generates an environmental sound signal representing an environmental sound that forms sound environment by being emitted. The environmental sound has at least one chain of phonemes constituted of individual phonemes which sound-emission start timings as one of attributes thereof are sequentially shifted. A plurality of subgroups are prepared each formed by combining individual plural pitches from pitches constituting a primary pitch group that is a group of phonemes musically treated as consonances if sounded simultaneously. One of the plurality of subgroups selected at random is set to each of the sections of the chain of phonemes, and each of the individual phonemes of each section of the chain of phonemes is set to a pitch selected at random from the plural pitches constituting the selected subgroup to attain hypersonic effects.

A content control device includes: a plurality of controls to which a plurality of parameters for controlling properties of a content containing at least one of sound and video are respectively assigned, each of the plurality of controls outputting a first indicated value in accordance with an operation amount of the control; and a processor configured to previously create setting information used to determine respective values of the plurality of parameters in accordance with the second indicated value; determine the values of the plurality of parameters corresponding to the second indicated value respectively in accordance with the second indicated value and the setting information; and revise each of the values of the parameters to be determined in accordance with the first indicated value outputted for the control assigned to the parameter.

The invention is a process for the creation of an audio effect in the context of an audio editing software. The effect is created by applying a series or sequence of reversal instances across a sample or waveform in time.

A performance analysis method is realized by a computer and includes acquiring a time series of input data representing played pitch that is played, inputting the acquired time series of input data into an estimation model that has learned a relationship between a plurality of items of training input data representing pitch and a plurality of items of training output data representing an acoustic effect to be added to sound having the pitch, and generating a time series of output data for controlling an acoustic effect to be added to sound having the played pitch represented by the acquired time series of input data.

An acoustic device includes an audio recording playback unit, an analysis unit, and an acoustic effect imparting unit. The audio recording playback unit records and plays back string independent acoustic signals for each string independent acoustic signal. The string independent acoustic signals respectively correspond to different strings of a stringed instrument and being independent from each other. The analysis unit analyzes at least one string independent acoustic signal from among the recorded string independent acoustic signals. The acoustic effect imparting unit imparts an acoustic effect to the at least one string independent acoustic signal for each string independent acoustic signal, based on a result of the analysis by the analysis unit.