Multiplex communications – Communication techniques for information carried in plural... – Combining or distributing information via time channels
Reexamination Certificate
1998-07-13
2002-02-26
Cangialosi, Salvatore (Department: 2661)
Multiplex communications
Communication techniques for information carried in plural...
Combining or distributing information via time channels
C370S537000
Reexamination Certificate
active
06351475
ABSTRACT:
BACKGROUND OF THE INVENTION
The present invention generally relates to a mixing apparatus suitable for use in imparting effects to voice signals of plural channels generated by time-divisional multiplexing.
Recently, digital signal processors (DSPs) have been made available for performing various numerical computations on an inputted digital signal by executing plural control programs in various fields with advances in semiconductor fabrication technologies. For example, the DSP is used in electronic musical instruments. In this application, the DSP is used to formulate effect blocks for imparting sound effects to music tones or voices (hereinafter, voice refers to any type of sounds treated in audio technology). Namely, the DSP is adapted to impart different sound effects to voices in a parallel manner by means of desired combination of the effect blocks.
Generally, electronic musical instruments are equipped with a tone generator for generating a music tone or voice. The tone generator operates in a time-division multiplex manner to generate multiple channels of voice signals simultaneously. In such a case, a predetermined sound effect is created by the DSP as follows. Namely, the tone generator accumulates plural channels of voice signals at every sample period, and supplies results of this accumulation to the DSP through the plural channels. The DSP executes a predetermined effect algorithm, imparts predetermined sound effects to the received accumulation results in a parallel manner, and accumulates the voice signals imparted with the sound effects at every sample period. The digital voice signals resulted from the accumulation by the DSP are converted by a D/A converter into analog voice signals, which are then sounded.
As described, the prior-art constitution requires a mixing circuit for accumulating the voice signals for each of the channels in the tone generator and another mixing circuit for accumulating the voice signals for each of the channels in the DSP at every sample period. Each mixing circuit is composed of a multiplier and an accumulator, presenting a problem of a large circuit scale. It should be noted here that sharing the accumulator between the tone generator and the DSP is difficult because the tone generator operates in the time-division multiplex manner.
Meanwhile, in effect imparting by the DSP, it should be noted that the computation capability of the DSP is obviously limited. Therefore, installation of only one DSP limits the effects to be imparted. To overcome this problem, several measures are possible, such as installation of the DSP of higher performance and installation of plural DSPs. These measures are suitable for only professional-use of electronic musical instruments because of the resulting high-level specifications and tradeoff of increased fabrication cost. This problem may be circumvented by providing hardware extensibility such that an extra DSP satisfying professional users can be added as required to a main DSP that has performance enough for general users. However, this additive method requires considering compatibility of the extra DSP to the output of the tone generator and the input/output of the DSP installed as standard. Namely, extra DSPs cannot be added simply.
Recently, there are demands for imparting effects by the electronic musical instrument also to external inputs such as a voice signal generated by another electronic musical instrument and a voice signal inputted from a microphone. However, it is difficult to handle these external voice signals the same way as the internal voice signals generated by the tone generator operating in the time-division multiplex manner.
SUMMARY OF THE INVENTION
It is therefore an object of the present invention to provide a mixing apparatus that is small in entire circuit scale, easy in functional expansion, and easy in handling of external input voice signals.
The inventive sound mixing apparatus is constructed for synthesizing M channels of output voice signals from m channels of input voice signals. In the apparatus, first providing means provides n channels of input voice signals at a sample period or frame period that is arranged with n number of timeslots to accommodate the n channels of the input voice signals. Rearranging means rearranges the timeslots so as to increase the number of the timeslots from n to m within the frame period, and then distributes the n channels of the input voice signals to the rearranged timeslots so as to create m-n number of free timeslots within the frame period. Second providing means is disposed separately from the first providing means for providing at most m-n channels of input voice signals in addition to the n channels of the input voice signals. Allocating means allocates at most the m-n channels of the input voice signals to the m-n number of the free timeslots so as to accommodate within the frame period the m channels of the input voice signals provided from both of the first providing means and the second providing means. Mixing means mixes the m channels of the input voice signals with one another by time-division manner of the m channels to produce the M channels of the output voice signals.
According to the present invention, for the sake of time conversion, the rearranging means compressively increases the n timeslots corresponding to the n channels of the voice signals inputted by the first providing means to the m timeslots, which are greater than the n timeslots. The allocating means allocates or assigns the voice signal inputted by the second providing means to the free or space timeslots produced by the time conversion. The mixing means outputs the voice signals assigned to each timeslot to the output channels according to data indicative of the output destination. In the mixing means, accumulating means accumulates the input voice signals for every output channel at one frame period. Consequently, the voice signals inputted by the first and second providing means are accumulated in each output channel, thereby allowing the accumulating means to be shared. This arrangement contributes to a simplified overall constitution of the mixing apparatus. Further, according to the invention, the internal voice signals inputted by the first providing means and the external voice signals inputted by the second providing means are processed mutually equally. This holds true for the output channels. This arrangement allows the user to facilitate setting of the mixing. Because the input and output channels are compatible to the first and second providing means, the expansion of capabilities may be made with ease and other external inputs may be handled with ease. In addition, according to the invention, the input voice signals can be processed on a stereo basis, so that handling of the mixing apparatus is simplified for both the user and the electronic musical instrument associated with the present invention. Lastly, according to the invention, overflow is detected for each output channel during the accumulation of the input voice signals, thereby preventing the distortion due to the overflow from occurring on all of the output channels.
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Derwent Abstract Accession No. 96-192639/20, JP 08063157, Yamaha Corporation, Mar. 8, 1996.
Cangialosi Salvatore
Morrison & Foerster / LLP
Yamaha Corporation
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