Pulse or digital communications – Pulse code modulation
Reexamination Certificate
1998-03-26
2001-08-07
Le, Amanda T. (Department: 2634)
Pulse or digital communications
Pulse code modulation
C375S377000, C341S141000, C341S155000, C348S572000, C370S537000
Reexamination Certificate
active
06272181
ABSTRACT:
TECHNICAL FIELD
The invention concerns a method and a device for aggregating N>1 band-limited time signals with a bandwidth of ≦B each, which are present as analog and/or digital sampling values and have a respective sampling frequency of f
A
>2B. Such a method is known from DE 32 00 934 A1.
BACKGROUND OF THE INVENTION
An aggregation of analog signals by means of adders is described for example in the textbook “Semiconductor circuit technology” by Tietze and Schenk, 8th. edition, 1986, pages 299 and 300 as well as 579 to 581.
The aggregation of digital signals from analog input signals with an intermediate analog-digital converter (ADC) via a digital signal processor (DSP), and a reanalogation of the processed signals by means of a digital-analog converter (DAC), particularly in the area of video signals, is described for example in EP 0 695 066 A2.
A linear aggregation of several band-limited time signals into a new composite signal takes place among other things in audio technology, where audio signals are superimposed by mixing the sounds from several different sources, or in video technology where video signals are combined into a new video signal by cross-fading the images from two different sources. The areas of application for sound mixing are for example in radio, in the disk recording industry and in the production of other sound carriers. Furthermore, sound mixing is required for audio conference circuits, i.e. for the aggregation of several sound signals from different sources in the area of telecommunications. A mixing of images by cross-fading several video signals is usual for example in television, in the production of video disks and video displays on other video carriers, video recorders, camcorders and such. Although no video mixing takes place in video conferences, windows are faded into a joint video for the different participants in the conference system.
With the method for mixing low frequency signals known from the DE 32 00 934 A1 cited in the beginning, which are present in the form of digital scanning samples, the pulses intended for the common terminal, which must be rendered jointly audible in the respective terminal, are aggregated by an analog adder and are transmitted within one time frame in the form of an aggregate pulse which controls the terminal during the entire time frame.
A disadvantage of the known methods is the relatively long calculation time for the aggregation of the individual signals by a digital computer, or by a hardware circuit of adding units. In addition there is considerable damping of the signals and thus a loss of information when converting from analog to digital signals and vice versa during the reanalogation of the added signals in the case of a digital aggregation.
SUMMARY OF THE INVENTION
The object of the present invention is therefore to improve a method of the kind cited in the beginning in a way so that the aggregation can be carried out in a considerably shorter calculation time, so that possibly a number of slow and expensive adder elements can be saved, and minimizing the damping of the signals during the processing and thus the corresponding loss of information.
The invention achieves this object in as surprising as well as an effective a manner in that the sampling values of all N time signals are offset in time and superimposed on each other, and are jointly input to a low-pass filter with a bandwidth of B′>B, and that a composite signal is tapped off from the output of the low-pass filter.
In contrast to the known methods, in which signals from different sources that are present as analog or digital sampling values are converted separately for each signal into analog signals, which are aggregated by means of one or several analog adders or by a digital processor, in the method of the invention the sampling values of different time signals are offset in time and superimposed on each other, and converted to analog by means of a passive low-pass filter. The sampling values of different signals, each of which was sampled at a frequency f
A
, are combined by means of a time-division multiplex method into a superimposed signal with the frequency of N·f
A
. During the subsequent filtration with a low-pass filter of the bandwidth B′=f
A
/2, an analog composite signal is generated, which can be sampled for further processing at the frequency of f
A
.
On the one hand this results in a qualitatively better and faster aggregation, on the other the aggregation can be achieved in a more cost-effective manner due to the saving of an adder unit or a corresponding processor for the digital aggregation of the input signals. Another advantage is that a device which is suitable for carrying out the method of the invention can be integrated in a simple manner into an integrated switching circuit, for example a VLSI chip. On the other hand the method of the invention can easily be built into a DSP software, with the corresponding gain in calculation time. The method of the invention is suitable for adding both digital as well as analog input values.
Particularly preferred is a configuration of the method of the invention in which the sampling values of the N time signals are offset from each other equidistantly in time. This allows from the outset to establish a rigid and always known time-relation of the signals from different sources, which remains the same.
Another preferred configuration of the invention provides for the sampling values, which are offset in time with respect to each other, to be input into the low-pass filter at a clock frequency of N·f
A
.
When analog sampling values are input in another advantageous configuration of the invention, an analog low-pass filter can be used, whose output has a time-continuous composite signal and causes the formation of a perfect aggregation of the partial signals.
Preferably this method is developed further in that analog sampling values are obtained by sampling the time-continuous composite signal.
As an alternative, other configurations into which digital sampling values are input provide for the use of a digital low-pass filter which operates at the clock frequency of n·f
A
, whose output has a composite signal with n·f
A
sampling values per unit, i.e. in oversampled form. This allows to utilize all the advantages of an oversampling method.
A further development of this configuration provides for the oversampled composite signal to be input into a digital-analog (D/A) converter which operates at the clock frequency of N·f
A
, and whose output signal produces the time-continuous composite signal via subsequent filtration, preferably by means of a resistor-capacitor (RC) element. Instead of the expensive filter installation, a very simple cost-effective RC filter element can be used which, because of the oversampling, ensures sufficient suppression of the mirror signals that periodically occur in the frequency space in accordance with a Fourier transformation.
A further development is particularly advantageous, whereby the oversampled composite signal is transferred to a lower sampling frequency i·f
A
<N·f
A
, where preferably i=1, by periodically omitting sampling values (=decimation). Inversely, a higher sampling frequency can also be achieved by means of sampling rate conversion, by introducing fictitious sampling values “0” in intermediate areas, where low-pass filtration produces a perfect total signal at the end.
The method of the invention can be carried out in a particularly simple and inexpensive manner with analog input values, if the aggregation and the low-pass filtration are performed with a digital signal processor.
The framework of the present invention also includes a device for aggregating N>1 band-limited time signals, each with a bandwidth ≦B, which are present as analog and/or digital sampling values, where the respective sampling frequency is f
A
>2B, and a time-division multiplex unit is provided in which the sampling values of all N time signals can be offset in time and superimposed on each othe
Alcatel
Le Amanda T.
Ware Fressola Van der Sluys & Adolphson LLP
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