Audio A/D converter using frequency modulation

Electrical audio signal processing systems and devices – Circuitry combined with specific type microphone or loudspeaker – With piezoelectric microphone

Reexamination Certificate

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Details

C381S113000, C341S155000, C341S157000

Reexamination Certificate

active

06731763

ABSTRACT:

BACKGROUND
The present invention relates generally to reducing the cost of electronic systems that involve voice processing, such as telephones or cellular phones, by integrating various electronics into a single silicon chip. More particularly, the present invention relates to analog-to-digital convertors that minimize a number of analog components used therein and which furthermore is relatively immune to noise pick-up from digital circuits operating in the same chip.
Many forms of conventional analog-to-digital convertors exist, for example those known by the types of techniques used therein such as Successive Approximation, Delta-Sigma Modulation, and Continuously Variable Slope Delta Modulation (CVSD). The purpose of these devices is to produce a stream of numbers representing samples of the instantaneous signal value at a desired sample rate. The desired sampling rate is usually higher than the minimum Nyquist rate of twice the maximum frequency of the analog signal to be numerically represented. The drawbacks of these prior art techniques concerns the very small signal level output from the microphone, which results in the connection between the microphone and the analog-to-digital convertor being sensitive to noise pick-up.
SUMMARY
According to the exemplary embodiments, a variable frequency oscillator is controlled in frequency by a variable electrical parameter in a microphone, typically a variable capacitance. The frequency modulated signal is applied to a direct digital discriminator that produces a digital representation of the instantaneous frequency at the desired speech sampling rate. The digital discriminator may be formed, for example, by applying the oscillator signal to a direct phase digitizing circuit along with a reference frequency and calculating a sequence of instantaneous phases of the oscillator relative to the reference frequency. The phase sequence is then applied to a digital phase locked loop (or otherwise numerically differentiated) to generate a sequence of binary words representative of instantaneous frequency and therefore representative of the speech waveform. Since the low-level speech waveform substantially does not enter the integrated circuit except as a high-level frequency-modulated carrier, the technique is substantially immune to noise caused by high speed random logic circuits such as microprocessors and DSPs operating on the chip.
According to other exemplary embodiments of the present invention, microphone circuitry still based on the variable electrical parameter of voltage or current can be provided which is less susceptible to noise than conventional microphone circuitry. For example, conventional FET preamplifiers can be omitted according to the invention so that noise typically created by bias supplies is avoided.


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