Electrical audio signal processing systems and devices – Noise or distortion suppression
Reexamination Certificate
1999-06-28
2003-11-25
Harvey, Minsun Oh (Department: 2747)
Electrical audio signal processing systems and devices
Noise or distortion suppression
C341S122000, C381S094400
Reexamination Certificate
active
06654469
ABSTRACT:
BACKGROUND OF THE INVENTION
The advent of specialized, electronic components has made it possible to design and build sophisticated audio and video devices which are capable of producing high-quality reproductions of sounds, pictures and the like. One type of component often used is a digital signal processor (“DSP”). Generating audio or video signals requires other electronic components in addition to DSPs. One common component is a coder-decoder, otherwise known in the art as a “codec”.
Typically, original audio signals (e.g., sounds) and video signals (e.g., pictures) are made up of “analog” signals. The function of a codec is to convert or “code” an analog signal into a “digital” one so that the signal can be processed and the like. Likewise, it is sometimes desirable to convert or “decode” a stored digital signal into an analog signal for output to another device such as an audio speaker or video display. The differences between analog and digital signals are well known in the art and need not be discussed here in great detail.
“Single channel” codecs perform the coding/decoding just discussed on one audio/video “channel” or signal. One advantage of using a DSP in combination with a codec is the ability to “multiplex” or combine a number of signals into one signal. Multiplexing may be envisioned as the interleaving of data from separate signals to form one signal. It is possible, therefore, for a codec to output a multiplexed signal (or as the case may be, output a “demultiplexed” signal). Codecs which use “time-multiplexing” operate by designating certain time slots for each signal. Interleaving is accomplished by taking data from each signal only during predetermined time slots. Again, in general, multiplexing is well known in the art and need not be discussed here in detail.
There are, however, problems associated with time-multiplexing multiple channels using a codec/DSP combination. One problem relates to “sampling noise”. An analog signal can be divided into two portions; a “signal” portion which consists of useful data and a “noise” portion which consists of interfering, non-useful data. At some point, a digitized signal is converted back into an analog signal by the digital-to-analog conversion section of a codec. The analog signal must then be output from the codec to an external device, such as a speaker or display. Digital signals are typically “clocked” out of a digital-to-analog converter and fed to a single “sample and hold” circuit. During the process of sampling and holding the signal, sampling noise is introduced.
Attempts have been made to reduce sampling noise by increasing the rate at which the digital signal is fed to the digital-to-analog converter and sampled/held. It is sometimes not feasible to do so, however, because in some instances the sampling rate and/or clock frequency is fixed. As a last resort, another electronic device, called a filter, is connected to the output of the codec. The filter effectively removes the noise but, it is an expensive solution.
Accordingly, it is an object of the present invention to provide devices and methods which reduce sampling noise in sample and hold circuits.
It is a further object of the present invention to provide devices and methods which reduce sampling noise in an analog signal output from a digital-to-analog converter.
It is yet another object of the present invention to provide devices and methods which reduce sampling noise in an analog signal output from a codec, such as a speech codec.
Other objectives, features and advantages of the present invention will become apparent to those skilled in the art from the following description taken in conjunction with the accompanying drawings.
SUMMARY OF THE INVENTION
The present invention provides novel devices and methods which reduce sampling noise in an analog signal without the need to increase the sampling rate or use additional external filters.
The novel devices and methods apply linear interpolation to sampled analog signals. In one illustrative embodiment, a novel device comprises two sample and hold circuits which alternatively transfer voltages derived from portions or samples of an analog, input voltage signal to an output capacitance. For each sample input, the novel device and/or method outputs two signals. One of the sample and hold circuits transfers a voltage which creates an output or “interpolated” voltage midway between the input voltage and an historical output voltage, while the second sample and hold circuit transfers a voltage sufficient to create an output voltage approximately equal to an input voltage. By generating two output voltage signals for every input voltage signal, sampling noise is reduced.
REFERENCES:
patent: 5744985 (1998-04-01), Nishida
patent: 6510313 (2003-01-01), Rapeli
Nelson Dale H.
Tsai Tseng-Nan
Harvey Minsun Oh
Lucent Technologies - Inc.
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