摘要:
A digital audio spatialization system that incorporates accurate synthesis of three-dimensional audio spatialization cues responsive to a desired simulated location and/or velocity of one or more emitters relative to a sound receiver. Cue synthesis may also simulate the location of one or more reflective surfaces in the receiver's simulated acoustic environment. The cue synthesis techniques are suitable for economical implementation in a personal computer add-on card.
摘要:
An electronic music system which imitates acoustic instruments addresses the problem wherein the audio spectrum of a a recorded note is entirely shifted in pitch by transposition. The consequence of this is that unnatural formant shifts occur, resulting in the phenomenon known in the industry as "munchkinization." The present invention eliminates munchkinization, thus allowing a substantially wider transposition range for a single recording. Also, the present invention allows even shorter recordings to be used for still further memory improvements. An analysis stage separates and stores the formant and excitation components of sounds from an instrument. On playback, either the formant component or the excitation component may be manipulated.
摘要:
An electronic music system which imitates acoustic instruments addresses the problem wherein the audio spectrum of a a recorded note is entirely shifted in pitch by transposition. The consequence of this is that unnatural formant shifts occur, resulting in the phenomenon known in the industry as "munchkinization." The present invention eliminates munchkinization, thus allowing a substantially wider transposition range for a single recording. Also, the present invention allows even shorter recordings to be used for still further memory improvements. An analysis stage separates and stores the formant and excitation components of sounds from an instrument. On playback, either the formant component or the excitation component may be manipulated.
摘要:
Systems and methods for sample rate tracking are provided. An example method includes computing an actual latency associated with an output sample from an output sample stream. The actual latency is calculated using a phase and a phase increment (conversion rate ratio). A measured latency is determined using an internal clock using a presentation time of the output sample, or an input sample from an input sample stream, or both. The measured latency is compared to the actual latency to generate a latency error. A successive phase increment can be determined based on the latency error by using a low-pass or adaptive filter to adjust the latency error.
摘要:
A DMA device which can quickly access main memory over a system bus without requiring an allocation of a contiguous block of memory on start-up. This is accomplished by providing a copy of the host microprocessor's page table to the DMA device for the portion of memory allocated to it. The DMA device preferably stores at least a portion of the page table internally, with any remainder of the page table being stored in main memory at an address stored in said DMA device.
摘要:
A digital sampling instrument is disclosed. The instrument provides the capability of accessing and outputting stored digital data within a single clock cycle. The instrument also provides improved volume scaling for sound generation and further eliminates redundant loading of a particular sound into a sound memory.
摘要:
Digital methods and systems for signal processing and filtering are provided. The methods and corresponding systems provide asynchronous conversion of sampling rate frequencies and utilize advanced multistage phasor filters for converting an input signal having a first sampling rate into an output signal sampled in an arbitrary sequence of sampling times. The conversion process provides a sequence of sets of complex numbers representing a filtered version of the input signal. More specifically, the conversion process includes the calculation of values of the output signal by multiplying (e.g., scaling) the sets of complex numbers by a corresponding set of complex phasors, the complex phasors corresponding to the timing of the arbitrary time sequence to obtain a corresponding set of real results with the value of the output signal being the sum of the real results.
摘要:
A cache memory is updated with audio samples in a manner which minimizes system bus bandwidth and cache size requirements. The end of a loop is used to truncate a normal cache request to exactly what is needed. A channel with a loopEnd in a request will be given higher priority in a two-stage priority scheme. The requested data is conformed by trimming to the minimum data block size of the bus, such a doubleword for a PCI bus. The audio data written into the cache can be shifted on a byte-wise basis, and unneeded bytes can be blocked and not written. Request data for which a bus request has been issued can be preempted by request data attaining a higher priority before a bus grant is received.
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.