摘要:
A method is provided for designing an acoustic correction filter applicable to a stringed instrument, which is composed of a string member operable to undergo a vibration, a support member for supporting the string member, a body member responsive to the vibration transmitted through the support member for generating a natural sound and a mute attachment for muting the natural sound. The acoustic correction filter is operable when the natural sound is muted by the mute attachment for filtering a signal derived from the vibration so as to create an artificial sound instead of the muted natural sound. The method is carried out by the steps of acquiring a first sample signal from the vibration under a mute state, acquiring a second sample signal from the vibration under a free state, extracting a difference between the acquired first sample signal and the acquired second sample signal, and determining a correction characteristic of the acoustic correction filter based on the extracted difference such that the acoustic correction filter can filter the signal in accordance with the determined correction characteristic so as to create the artificial sound comparable to the natural sound.
摘要:
A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory.
摘要:
A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory.
摘要:
A digital sound generating system is provided which is capable of, in addition to generating a tone (first waveform data), recording or reproducing PCM sound data representing a human voice or sound of a music piece lasting for a relatively long time (second waveform data). To this end, in the tone generating process, tone waveform sample data is read out from a waveform RAM on the basis of address data given from an address calculating section and is then output as an analog signal via an interpolating section and D/A converter. In reproduction of PCM sound data, the PCM sound data supplied via a CPU interface is transferred using the waveform RAM as a buffer and is then supplied outside of the system through a same channel as in the tone generating process. In recording of PCM sound data, a high-frequency-component removing process is performed, by a filtering calculating section, on PCM sound data received via an A/D converter so as to prevent unwanted aliasing noise. The resultant filtered PCM sound data is then transferred using the waveform RAM as a buffer and output via the CPU interface.
摘要:
A reverberation imparting device for electro-acoustically enhancing reverberation in acoustic space comprises a microphone disposed in the acoustic space, a loudspeaker disposed in the acoustic space for diffusing the sound picked up by the microphone, and feedback means comprising a signal processing circuit for electrically processing an electric signal corresponding to the sound picked up by the microphone, an output of the signal processing circuit being supplied to the loudspeaker. The microphone, feedback means and loudspeaker form a feedback loop. The signal processing circuit comprises a circuit for subjecting impulse responses of finite length to a convolution operation. Time axis of reflected sounds is extended and extension of reverberation time thereby is realized without depending upon loop gain. Density of reflected sounds is increased by subjecting impulse responses of finite length to a convolution operation whereby separation of reflected sounds is prevented.
摘要:
The present invention is directed to creating sound reverberation. In accordance with a preferred embodiment, an apparatus estimates an impulse response for use in reproduction of a sound in a desired acoustic space. In particular, the apparatus collects space information concerning an acoustic space and point information indicating positions of a sound generation and reception points in the acoustic space, estimates a set of acoustic ray paths of the sound traveling from the sound generation point to the sound reception point, acquires directivity information indicating an acoustic directivity of the sound generation point and the sound reception point, estimates an acoustic intensity of each acoustic ray path and weights each acoustic intensity by the acquired directivity information, and determines the impulse response based on directions of the respective acoustic ray paths toward the sound reception point and the weighed acoustic intensities of the respective acoustic ray paths.
摘要:
A method and apparatus models with a digital computer the impulse response characteristics of an acoustic system (“room” for convenience), based upon a measured impulse response signal from the acoustic system. The steps of the method include: processing the measured impulse response signal by band-limited decimation in at least one band to obtain a band-limited, decimated signal (BLD signal); calculating with a digital computer, based on said BLD signal, a model impulse response of a rational polynomial form; and finding poles and zeros of said rational polynomial. In a preferred embodiment, the rational polynomial form is further simplified by removing a set of pole/zero pairs that meet a specified set of pair criteria.
摘要:
A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory. The present invention includes a plurality of interpolator circuits utilizing a single waveform memory where each of the interpolator circuits produces a unique bus request signal which is responsive to a unique bus acknowledge signal to determine which of the interpolator circuits has control of the waveform memory at any given waveform memory cycle.
摘要:
A digital sampling instrument for multi-channel interpolatative playback of digital audio data stored in a waveform memory provides improved interpolation of musical sounds by use of a cache memory. The present invention includes a plurality of interpolator circuits utilizing a single waveform memory where each of the interpolator circuits produces a unique bus request signal which is responsive to a unique bus acknowledge signal to determine which of the interpolator circuits has control of the waveform memory at any given waveform memory cycle.
摘要:
An electronic sound processor creates reverberation effects using a simulated impulse function. The simulated impulse response is generated by combining frequency bands of white noise. The power of each band decays exponentially in time. The time constants are generated from the characteristics of a real or imaginary listening space. A reverberated sound signal is generated from an original sound signal by convolution of the original signal and the simulated impulse response. The method can adapted to generated stereophonic signals and to generate a forward and inverse reverb effect using only one set of multiplications.