Yamaha Corporation

Ryo Kamiya

Akira Sogo

Masahiro Mukojima

Tomoaki Ando

Masao Noro

Shigemitsu Yamaoka

Akitoshi Saito

Satoshi Tominaga

From a storage device storing waveform sample data sampled at a given sampling rate, the waveform sample data are read out asynchronously with the given sampling rate and temporarily stored in an input buffer. The thus-stored waveform sample data are read out from the input buffer and converted to an inner sampling rate selected from among a plurality of predetermined inner sampling rates. Predetermined arithmetic processing is performed on the waveform sample data converted to the selected inner sampling rate. Then, the waveform sample data having undergone the predetermined arithmetic processing are converted to a predetermined output sampling rate and written into an output buffer. The thus-written waveform sample data are then sequentially read out at an sampling frequency corresponding to the output sampling rate, to thereby generate tone data in accordance with the output sampling frequency.

Device and method for processing tone data by controlling sampling rate

A method of entering an audio signal into a data processing apparatus, in which analog audio signals of at least two channels as the audio signal are digitized by and transmitted from an audio signal output device on a side of an audio source to an audio signal input device on a side of the data processing apparatus. A clock signal is supplied from the audio signal input device to the audio signal output device. The audio signal output device performs delta-sigma (&Dgr;&Sgr;) modulation on the analog audio signals of at least two channels so as to convert each of the analog audio signals of at least two channels into one-bit digital audio data for each of the at least two channels, based on the clock signal supplied to the audio signal output device. The one-bit digital audio data for the each channel is alternately selected based on the clock signal, to perform time-division multiplexing on the one-bit digital audio data for the at least two channels, to thereby produce one-bit multiplex audio data. The one-bit multiplex audio data is supplied to the audio signal input device.

Method of entering audio signal, method of transmitting audio signal, audio signal transmitting apparatus, and audio signal receiving and reproducing apparatus

A series of digital data to be filtered is divided into a plurality of frames, and f samples of the digital data of each of the frames are burst-transmitted via a computer bus. In transferring the digital data of any one of the frames, n samples of the digital data belonging to a next frame is transferred, along with the digital data of the one frame, to thereby transfer a time series of the digital data consisting of (f&plus;n) samples that are more than the f samples contained in the one frame. Also, a set of k filter coefficients to be used for filtering arithmetic operations in the frame is burst-transmitted via the computer bus, where n&gE;k. Filtering arithmetic processor, such as a DSP, connected to the bus carries out filtering arithmetic processing using the transmitted f&plus;n samples of the digital data and k coefficients, to provide filtered data corresponding to at least the number of samples for a single frame. The filtering arithmetic processing may itself be conducted at high speed asynchronously with a predetermined sampling cycle. Thus, the filtering arithmetic operations are performed intermittently on a frame-by-frame basis. By buffering the filtered data and outputting them with the predetermined sampling cycle, there are provided filtered outputs synchronous with the predetermined sampling cycle. Cross-fade synthesis (interpolation) between filtered data of adjoining frames can smoothly interconnect the filtered data of the frames.

Digital filtering method and device and sound image localizing device

The present invention is directed to a method for managing memory in a computer whereby memory is divided into a first and a second section, and whereby the first and second sections are separately managed via a CPU. In accordance with the preferred embodiment of the present invention, data characteristics are used to determine which one of the first or second sections the data should be stored in. When the stored data are to be read out from the first section, single physical address information corresponding to a start position of the first section is received from the CPU so that a plurality of physical addresses corresponding to the individual read addresses are generated on the basis of the received physical address. However, when the stored data are to be read out from the second section, logical read addresses are generated and the physical address information is received from the CPU at each break between the pages so that the logical read addresses are converted into physical addresses using the received physical address. As a result, the frequency of interrupts made to the CPU for receiving the physical address can be considerably lowered.

Memory management method for use in computer system

An A/D conversion apparatus is provided, which is capable of securing a wide dynamic range of A/D conversion with a simple construction through suitably switching the input gain of the input analog signal between predetermined levels. An input gain control device controls gain of an input signal based on a control signal. A &Dgr;&Sgr; modulator carries out oversampling of the input signal having the gain thereof controlled by the input gain control device to convert the input signal to data of one bit. A detecting device detects a peak value of the input signal based on the data of one bit. A gain control device generates the control signal based on the peak value detected by the detecting device in a manner such that the input signal having the gain thereof controlled falls within a predetermined range. To effectively reduce noise of the output digital signal while securing a wide dynamic range of A/D conversion, the &Dgr;&Sgr; modulator may also control the gain of the input signal based on the control signal to a predetermined value (1/A) smaller than 1, and the decimation circuit may have a gain of a second predetermined value (A) for compensating for the gain of the input signal controlled to the predetermined value (1/A).

A/D conversion apparatus

A plurality of analog signals are subjected to a delta sigma modulation for conversion into respective bitstream data (PDM). The respective bitstream data are weighted for conversion into respective plural-bit PCM data, and then these plural-bit PCM data corresponding to the analog signals are added together. At that time, optional PCM to be mixed with the analog signals may also be added. PCM data resultant from the addition is then subjected to a delta sigma modulation for conversion into bitstream data. In this way, the mixed data is provided as one bitstream data. The mixed data can be coded as PCM data by performing a decimation process on the mixed data. The mixed data can also be provided as analog data by subjecting the mixed data to an analog low-pass filtering process.

Mixing, coding and decoding devices and methods

Tone generator device, which is applied to a computer, has no waveform memory of its own and is connected via an extended bus to the computer having a main memory where waveform sample data are prestored. When a tone of predetermined pitch is to be reproduced, the tone generator device designates a block of a specific quantity of the waveform sample data in accordance with the pitch to be reproduced and requests that the block be read out from the main memory and burst-transmitted via the extended bus. The burst-transmitted waveform sample data are temporarily stored in an input buffer and then read out from the buffer in accordance with pitch information to generate tone waveform data of the desired pitch. The burst transmission of the data block and subsequent tone waveform data generation are carried out at high speed asynchronously with a predetermined reproduction sampling cycle. The generated tone waveform data are output via an output data in the predetermined reproduction sampling cycle. To raise the upper limit on the reproducible pitch without a need to increase the capacity of the input buffer, a thinned version of the original waveform sample data may also be prestored in the main memory so that the thinned waveform sample data can be selected to permit the raised upper limit. Alternatively, such thinned waveform sample data may be generated by thinning the original waveform sample data, rather than being prestored in the main memory.

Tone generator device using waveform data memory provided separately

A method of entering an audio signal into a data processing apparatus, in which analog audio signals of at least two channels as the audio signal are digitized by and transmitted from an audio signal output device on a side of an audio source to an audio signal input device on a side of the data processing apparatus. A clock signal is supplied from the audio signal input device to the audio signal output device. The audio signal output device performs delta-sigma (.DELTA..SIGMA.) modulation on the analog audio signals of at least two channels so as to convert each of the analog audio signals of at least two channels into one-bit digital audio data for each of the at least two channels, based on the clock signal supplied to the audio signal output device. The one-bit digital audio data for the each channel is alternately selected based on the clock signal, to perform time-division multiplexing on the one-bit digital audio data for the at least two channels, to thereby produce one-bit multiplex audio data. The one-bit multiplex audio data is supplied to the audio signal input device.

Method of entering audio signal, method of transmitting audio signal,

In a case where tone waveform sample data are to be arithmetically formed by software, there are installed, within a virtual device driver routine of an operating system, a MIDI processing routine for converting a received MIDI event into tone generator control data and a waveform forming processing routine for arithmetically forming tone waveform sample data for one frame. The MIDI processing routine is triggered by a software interrupt signal that is generated in response to a MIDI event produced from an application software program such as a sequencer software program, and the waveform forming processing routine is triggered by a hardware interrupt signal that is generated upon completion of tone reproduction for one frame.

Method and device for generating a tone

Each of microcodes 1 has an interrupt prohibit bit 10 that specifies acceptance or nonacceptance of an interrupt request. Upon occurrence of an interrupt request, a processor refers to a value set in the interrupt prohibit bit 10 of the microcode 1 being currently executed. When '0' is set to the interrupt prohibit bit 10, the processor accepts the interrupt request, and when '1' is set to the interrupt prohibit bit 10, the processor rejects the interrupt request and starts executing a succeeding microcode. This allows the processor to safely execute certain instructions without interruption, as indicated by the interrupt prohibit bit 10 contained in the microcode 1 being currently executed.

Method and processor for selectively marking instructions as

Method and apparatus for generating a tone based on tone generating

A digital signal processor has an arithmetic operation device that carries out arithmetic operations. The arithmetic operation device has a plurality of elementary arithmetic operation units. A signal path-forming device forms signal paths for inputting and outputting signals to and from the elementary arithmetic operations units, according to a predetermined program. The arithmetic operation device carries out processing of a digital signal input to the digital signal processor after the signal paths have been formed by the signal path-forming device.

Digital signal processor architecture using signal paths to carry out

A musical tone waveform sample is generated at each sampling time by the use of a processing unit carrying out an arithmetic operation every repetition period that is longer than the sampling time. An envelope characteristic of a waveform of a musical tone to be generated is approximated by a polygonal line formed by a plurality of segments each of which has a length corresponding to the repetition period. An envelope value of a first waveform sample of waveform data of the musical tone to be generated for the repetition period is determined. The envelope value of the first waveform sample is multiplied by a predetermined attenuation factor to obtain an envelope value of a last waveform sample of the waveform data for the repetition period. Each of the segments of the polygonal line is determined by the envelope value of the first waveform sample and the envelope value of the last waveform sample. A difference between the envelope values of the first and last waveform samples is calculated. A decremental or incremental value per the each sampling time is obtained based on the difference. The envelope value of the waveform at the each sampling time is determined by adding the decremental or incremental value to an immediately preceding value of the envelope value of the waveform.

Musical tone-generating method and apparatus capable of approximating

A musical tone-generating method and apparatus therefor are provided. Musical tones are generated by reading waveform data from a waveform table at a predetermined sampling frequency and carrying out arithmetic operations on the waveform data read from the waveform table to generate musical tone waveform samples, by the use of a processing unit. A first sampling frequency or a second sampling frequency lower than the first sampling frequency is selected as the predetermined sampling frequency. Interpolation of the waveform data read from the waveform table is effected by a second interpolation method which is lower in accuracy than a first interpolation method employed when the first sampling frequency is used, or alternatively the interpolation is inhibited, when the second sampling frequency is selected as the predetermined sampling frequency.

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Hamamatsu, Japan

Ryo Kamiya