摘要:
Techniques have been developed to facilitate (1) the capture and pitch correction of vocal performances on handheld or other portable computing devices and (2) the mixing of such pitch-corrected vocal performances with backing tracks for audible rendering on targets that include such portable computing devices and as well as desktops, workstations, gaming stations, even telephony targets. Implementations of the described techniques employ signal processing techniques and allocations of system functionality that are suitable given the generally limited capabilities of such handheld or portable computing devices and that facilitate efficient encoding and communication of the pitch-corrected vocal performances (or precursors or derivatives thereof) via wireless and/or wired bandwidth-limited networks for rendering on portable computing devices or other targets.
摘要:
Techniques have been developed to facilitate (1) the capture and pitch correction of vocal performances on handheld or other portable computing devices and (2) the mixing of such pitch-corrected vocal performances with backing tracks for audible rendering on targets that include such portable computing devices and as well as desktops, workstations, gaming stations, even telephony targets. Implementations of the described techniques employ signal processing techniques and allocations of system functionality that are suitable given the generally limited capabilities of such handheld or portable computing devices and that facilitate efficient encoding and communication of the pitch-corrected vocal performances (or precursors or derivatives thereof) via wireless and/or wired bandwidth-limited networks for rendering on portable computing devices or other targets.
摘要:
In an audio signal processing apparatus, a generation section generates an audio signal representing a voice. A distribution section distributes the audio signal generated by the generation section to a first channel and a second channel, respectively. A delay section delays the audio signal of the first channel relative to the audio signal of the second channel for creating a phase difference between the audio signal of the first channel and the audio signal of the second channel such that the created phase difference has a duration corresponding to either an added value of a first duration which is approximately one half of a period of the audio signal generated by the generation section and a second duration which is set shorter than the first duration, or a difference value of the first duration and the second duration. An addition section adds the audio signal of the first channel and the audio signal of the second channel with one another, between which the phase difference is created by the delay section, and outputs the added audio signal which represents natural voice with various characteristics.
摘要:
There are provided a tone source for generating tone signals in plural channels independently and effect impartment sections, provided in corresponding relations to the channels, for imparting individual effects to the tone signals of the channels generated by the tone source. Each of the effect impartment sections, in accordance with tone control information unique to the tone signal of the corresponding channel, controls a parameter of the effect to be imparted. In a case where any of the effect impartment sections uses a signal delaying memory to impart an effect, when a damping (or truncating process) is performed in any of the channels, the storage area to be used for the channel is switched to another unused storage area so as to prevent any preceding tone's is delayed signal from being undesirably mixed into the signal of a new tone. In an application where a DSP is used to perform effect impartment, tone synthesis or the like, the size and position of a storage area for storing processed data are set variably to achieve efficient use of data memory. Further, a DSP program of different algorithms is automatically selected in accordance with performance information, to provide processing variety.
摘要:
An audio signal processing system including an input circuit for inputting musical instrument digital interface (MIDI) commands in real time over a plurality of channels, a computer including a central processing unit (CPU) supplied with the MIDI commands for simultaneously synthesizing one or more voices for each of the channels in response to the MIDI commands, each of the voices being generated by one or more of a plurality of predefined audio synthesis algorithms executed in software, a random access memory (RAM) for storing digital voice data representative of each of the voices generated by the CPU, an output circuit for audibly reproducing the voices from the digital voice data stored in the RAM, and wherein the CPU, in generating the voices selects the one or more audio synthesis algorithms based on one or more of the following criteria: the external processing demands placed upon the CPU by other operations being performed by the personal computer, a best match, according to predetermined criteria, between the type of voice required and audio synthesis algorithms available to the CPU, and the availability of wavetable voice data to be buffered into the RAM.
摘要:
An effect adding system for use in Karaoke performance applications is provided which, when one person sings, singing with different height from that of the actual singing, or singing with different timing from that of the actual singing, is automatically carried out to yield, in part, the same effect as if a chorus, duet, round, or the like is performed by a plurality of persons. In one embodiment, the effect adding system accomplishes this by means of pitch conversion and/or delay of the aural input signal in response to pitch conversion and/or delay information deriving from stored performance information for a particular musical composition.
摘要:
A karaoke apparatus is comprised of a data supply, a tone generator, an ADPCM decoder and a pitch shifter for sounding a requested karaoke song containing an instrumental accompaniment and a back chorus. The data supply supplies an ADPCM data representative of a phrase back chorus and being compressed by a variable compression condition, and supplies a song data containing accompaniment information prescriptive of an instrumental accompaniment, additional information prescriptive of a common back chorus and decoding information indicative of the variable compression condition of the ADPCM data, and further supplies key information which determines a pitch shift of the karaoke song. The tone generator processes the accompaniment information and the additional information to generate a first audio signal effective to sound synthesized tones of the instrumental accompaniment and the common back chorus. The ADPCM decoder operates in synchronization with the tone generator to decode the ADPCM data according to the decoding information for reproducing a second audio signal effective to sound the phrase back chorus concurrently with the instrumental accompaniment. The pitch shifter processes the second audio signal according to the key information to carry out the pitch shift of the phrase back chorus.
摘要:
A sound effect-creating device amplitude-modulates a monaural original sound signal obtained from stereophonic sound signals, delays the resulting modulated signal by delay amounts different from each other, selects and adds up a combination of a plurality of delayed modulated signals to form signals to be added to the stereophonic sound signals, and outputs the resulting stereophonic sound signals.
摘要:
Disclosed are a method and apparatus for analyzing an input vocal signal to produce a plurality of harmony signals that are combined with the input vocal signal to produce a multivoice signal. The method makes a current estimate of the fundamental frequency of the input vocal signal and determines if the current estimate is the correct estimate of the fundamental frequency. If the current estimate is correct, a reference note is assigned to correspond to the current estimate and a plurality of harmony notes are selected to correspond to the reference note. The method then generates a plurality of harmony signals by scaling the input vocal signal with a piecewise linear approximation of a Hanning window to extract a portion of the input vocal signal and by replicating the extracted portion at a plurality of rates equal to the fundamental frequencies of each of the harmony notes. The plurality of harmony signals and the input vocal signal are combined to produce the multivoice signal. The steps of the method are carried out with a microprocessor and a signal processing circuit.
摘要:
A music entertaining device which reproduces music instrumental sound and back chorus so that an entertainer can sing a song to the accompaniment of the reproduced music instrumental sound and the back chorus. The music instrumental data and back chorus data are separately stored in a memory device. Upon detection of a code instructing to insert back chorus during the reproduction of the music instrumental data, a computer base controller accesses the memory device in which the back chorus is stored and reproduces the back chorus identified by the code detected.