Abstract: According to the present invention, there is provided an electronic musical instrument that allows a player to learn a certain range covering a key of correct pitch and to feel as if he or she were playing the music. The instrument includes a controller to perform a pitches determination process of, based upon first timing and first pitch included in music data, determining pitches within a fixed range from the first pitch, which is allowed to be designated in accordance with the first timing, a display process of displaying an identifier to identify the pitches determined, and an automatic playing process of advancing automatic playing of the music data by producing sound corresponding to the first pitch from a sound producing unit when one of the pitches identified by the identifier displayed by the display process is designated.

Abstract: The present invention receives sound wave data from a sound inputted into a microphone and samples the sound wave data received using a CPU to obtain sampled data as a digitized tone data, which is then stored in a sampling memory. The CPU performs auto-play of a sound using the digitized tone data sampled by the sampling and stored in the sampling memory. Thus, a result of the sampling is automatically provided to the user after the sampling takes place and the user can intuitively understand what can be done through sampling.

Abstract: A recording and playback device according to the present invention stores in a memory section sound data for an amount of time corresponding to a beat which is acquired from an external source, and creates loop sound data in bar units by connecting a plurality of stored sound data. Then, the recording and playback device performs an operation to overlap newly acquired sound data onto the loop sound data while the loop sound data is being repeatedly replayed, and store the overlapping sound data in the memory section.

Abstract: The present invention receives sound wave data from a sound inputted into a microphone and samples the sound wave data received using a CPU to obtain sampled data as a digitized tone data, which is then stored in a sampling memory. The CPU performs auto-play of a sound using the digitized tone data sampled by the sampling and stored in the sampling memory. Thus, a result of the sampling is automatically provided to the user after the sampling takes place and the user can intuitively understand what can be done through sampling.

Abstract: In a recording and playback device of the present invention, when input data exceeding a threshold value is supplied, the CPU records input data for an amount of time corresponding to a single beat in an area specified by syllable number SPLIT in an input buffer IB of the RAM, and after incrementing the syllable number SPLIT, waits until the recorded data becomes silent. The CPU repeats this series of processing until the value of the incremented syllable number SPLIT reaches “4”, and thereby stores input data recorded for an amount of time corresponding to a single beat in each input buffer IB(1) to IB(4) corresponding to syllable numbers SPLIT1 to SPLIT4. Then, the CPU copies the input data to the recording area of the RAM such that these input data are sequentially connected and formed into loop data for an amount of time corresponding to a single bar.

Abstract: A sampling device includes: an obtainer that obtains a streaming audio waveform; a detector that detects a tap by a user; a designator that designates a sampling start point on the obtained audio waveform when at least a single tap has been detected by the detector, the designating being performed on the basis of one tap among a plurality of the taps when the plurality of taps are detected; a calculator that calculates a sampling duration on the basis of time intervals between the respective taps when the plurality of taps are detected, the calculating being performed from the sampling start point to a subsequent tap that is performed after the one tap; and a waveform sampler that samples the obtained audio waveform, the sampling starting from the designated sampling start point and ending in accordance with the calculated sampling duration.

Abstract: A sampling device includes: an obtainer that obtains a streaming audio waveform; a detector that detects a tap by a user; a designator that designates a sampling start point on the obtained audio waveform when at least a single tap has been detected by the detector, the designating being performed on the basis of one tap among a plurality of the taps when the plurality of taps are detected; a calculator that calculates a sampling duration on the basis of time intervals between the respective taps when the plurality of taps are detected, the calculating being performed from the sampling start point to a subsequent tap that is performed after the one tap; and a waveform sampler that samples the obtained audio waveform, the sampling starting from the designated sampling start point and ending in accordance with the calculated sampling duration.

Abstract: An input frame data producing unit produces from data stored in an input buffer input frames each including a predetermined number of sub-frames of a first hopsize determined based on the first frame size and the overlapping rate. A frame processing unit executes a window function on the input frames and shifts the windowed input frames by the first hopsize and overlaps the shifted input frames, storing the overlapped frames in an output frame. An output buffer data producing frame unit stores data from the output frame to an output buffer including a predetermined number of sub-frames of a second hopsize. A CPU sets the first hopsize and overlapping rate in a slow-speed reproduction when the reproducing speed ratio is set lower than 1 different from in a high-speed reproduction when the reproducing speed ratio is set larger than 1.

Abstract: A recording and playback device according to the present invention stores in a memory section sound data for an amount of time corresponding to a beat which is acquired from an external source, and creates loop sound data in bar units by connecting a plurality of stored sound data. Then, the recording and playback device performs an operation to overlap newly acquired sound data onto the loop sound data while the loop sound data is being repeatedly replayed, and store the overlapping sound data in the memory section.

Abstract: An input frame data producing unit 26 produces from data stored in an input buffer 25 input frames 27 each including a predetermined number of sub-frames of a first hopsize determined based on the first frame size and the overlapping rate. A frame processing unit 30 execute a window function on the input frames 29 and shifts the windowed input frames by the first hopsize and overlaps the shifted input frames, storing the overlapped frames in an output frame 29. An output buffer data producing frame unit 28 stores data from the output frame 29 to an output buffer 31 including a predetermined number of sub-frames of a second hopsize. CPU 11 sets the first hopsize and overlapping rate in a slow-speed reproduction when the reproducing speed ratio is set lower than 1 different from in a high-speed reproduction when the reproducing speed ratio is set larger than 1.

Abstract: An FFT unit performs an FFT process on high-frequency-eliminated, pitch-shifted voice data for one frame. A time scaling unit calculates a frequency amplitude, a phase, a phase difference between the present and immediately preceding frames, and an unwrapped version of the phase difference for each channel from which the frequency component was obtained by the FFT, detects a reference channel based on a peak one of the frequency amplitudes, and calculates the phase of each channel in a synthesized voice based on the reference channel, using results of the calculation. An IFFT unit processes each frequency component in accordance with the calculated phase, performs an IFFT process on the resulting frequency component, and produces synthesized voice data for one frame.

Abstract: The present invention provides a technique for shifting pitch to target pitch without detecting the original pitch directly, and for extracting the pitch of the audio waveform exactly. A phase compensator extracts 2 or more frequency channels each having frequency components of a harmonic overtone whose frequency is 1 or more times as higher than frequency of a fundamental tone of the original sound, from the frequency channels from which the frequency components are extracted by fast Fourier transform. The phase compensator calculates a scaling value to be used for converting the fundamental tone to another target fundamental tone, and performs phase compensation in accordance with the scaling value. A pitch shifter performs pitch scaling in accordance with the scaling value onto the audio data resultant from inverse fast Fourier transform onto the phase compensated frequency components. Thus, audio data representing the target fundamental tone are generated.

Abstract: The present invention provides a technique for shifting pitch to target pitch without detecting the original pitch directly, and for extracting the pitch of the audio waveform exactly. A phase compensator extracts 2 or more frequency channels each having frequency components of a harmonic overtone whose frequency is 1 or more times as higher than frequency of a fundamental tone of the original sound, from the frequency channels from which the frequency components are extracted by fast Fourier transform. The phase compensator calculates a scaling value to be used for converting the fundamental tone to another target fundamental tone, and performs phase compensation in accordance with the scaling value. A pitch shifter performs pitch scaling in accordance with the scaling value onto the audio data resultant from inverse fast Fourier transform-onto the phase compensated frequency components. Thus, audio data representing the target fundamental tone are generated.

Abstract: An FFT unit performs an FFT process on high-frequency-eliminated, pitch-shifted voice data for one frame. A time scaling unit calculates a frequency amplitude, a phase, a phase difference between the present and immediately preceding frames, and an unwrapped version of the phase difference for each channel from which the frequency component was obtained by the FFT, detects a reference channel based on a peak one of the frequency amplitudes, and calculates the phase of each channel in a synthesized voice based on the reference channel, using results of the calculation. An IFFT unit processes each frequency component in accordance with the calculated phase, performs an IFFT process on the resulting frequency component, and produces synthesized voice data for one frame.

Abstract: A tone generating apparatus has an internal storage in which defining information (taking the form of table information, for instance) on relationship between operation conditions of the apparatus and applications of operation members, such as slider, switch and encoder, is stored in advance. CPU of the apparatus controls the apparatus in such a manner that applications of operation members in a current operation condition of the apparatus are effected according to the defining information. With this arrangement, the work for changing specifications of the apparatus is simplified since it is done primarily by changing contents of the defining information.

Abstract: Measures each contain plural event data, which are memorized in an arrangement of SNG-SONG. 1024 measures (the first measure to 1023-th measure) in each music or in each track of the music are grouped into 8 measure blocks. Leading event data of 8 measure blocks each containing 128 measures are memorized in an arrangement of SNG-BAR-PTR. Element numbers of leading event data of respective 128 measures contained in each measure block are memorized in an arrangement of SNG-BAR-TBL. For example, increment or decrement of a point value of the measure block in the arrangement of SNG-BAR-TBL or the measure in the arrangement of SNG-BAR-PTR allows a fast forward reproduction and/or a fast rewind reproduction.

Abstract: The lower significant n bits of a first differential value between adjacent sampled values of a sampled received signal are cut off to provide a second differential value, which is then subjected to adaptive quantization to thereby provide data compression. When this data is expanded, the second differential value decoded from the adaptive quantized value is sequentially accumulated by 2.sup.n times at a period corresponding to 1/2.sup.n of the sampling period of the received signal. As a result, a quantizing error (quantization noise) per sampling of the received signal is dispersed in 2.sup.n periods to thereby provide an improved acoustic S/N ratio. When the first differential value of the received signal is quantized, when the lower significant n bits of the first differential value are cut off, and when the resulting signal is subjected to adaptive quantization, the resulting respective quantizing errors are fed to corresponding values processed next.

Abstract: The lower significant n bits of a first differential value between adjacent sampled values of a sampled received signal are cut off to provide a second differential value, which is then subjected to adaptive quantization to thereby provide data compression. When this data is expanded, the second differential value decoded from the adaptive quantized value is sequentially accumulated 2.sup.n times at a period corresponding to 1/2.sup. of the sampling period of the received signal. As a result, a quantizing error (quantization noise) per sampling of the received signal is dispersed in 2.sup.n periods to thereby provide an improved acoustic S/N ratio. When the first differential value of the received signal is quantized, when the lower significant n bits of the first differential value are cut off, and when the resulting signal is subjected to adaptive quantization, the resulting respective quantizing errors are fed to corresponding values processed next. Thus, accumulation of the quantizing errors is prevented.