Electrical audio signal processing systems and devices – Hearing aids – electrical
Reexamination Certificate
1998-01-30
2001-07-03
Le, Huyen (Department: 2643)
Electrical audio signal processing systems and devices
Hearing aids, electrical
C381S320000, C381S321000
Reexamination Certificate
active
06256395
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to output clipping apparatus. More specifically, the present invention relates to output clipping apparatus for hearing aids.
2. Description of the Prior Art
FIG. 1
(prior art) shows an analog hearing aid having microphone
11
connected to sound processing
12
, incorporating clipping
14
and connected to power amplifier
16
and a speaker
18
. Analog hearing aids clip occasionally, as it is impossible to get sufficient maximum signal level in a low power device like a hearing aid without clipping. The amplifier itself may perform the clipping function. In an analog device, the distortion caused by output clipping is acceptable, because the distortion is mostly odd order harmonics and some inter-modulation products. In a digital hearing aid output, such as that shown in
FIG. 2
, the effects of clipping are much worse. In a typical digital hearing aid, the circuit of
FIG. 2
replaces blocks
14
,
16
, and
18
of FIG.
1
. The clipping in such a system will create distortion products which are not harmonics or inter-modulation components, but are instead entirely unrelated to the signal, and are thus acoustically very undesirable.
One possible solution to the problem of clipping in a digital hearing aid is to convert the signal to analog, and then amplify and clip in the analog domain. This would remove the offending distortions, but at the cost of requiring greater precision in the D/A converter. As there is gain after the converter, noise will be amplified, so the noise floor would have to be better. This approach would also eliminate the possibility of a class D output stage directly in the D/A converter.
A typical digital hearing aid such as that shown in
FIG. 8
includes an output digital to analog converter as one component.
FIG. 2
(prior art) shows an oversampling digital to analog (D/A) converter, which utilizes a second order delta sigma quantizer
70
and a one-bit D/A converter
71
as the demodulator
69
, and a low pass filter
73
to remove the noise from the one-bit signal. In one specific example of the oversampling D/A converter of
FIG. 2
, the input signal xi,
60
, consists of data encoded into 16 bit words at 16 kHz. In a conventional D/A converter, signal
60
is clipped by clipper
61
, and then placed into a register
63
from which it is fed into a low pass filter
64
at 32 kHz, with each word repeated two times. The low pass filter would typically be of the finite impulse response type. The linear interpolator
66
, which is also a type of low pass filter, inserts three new words between each pair of words from low pass filter
64
, which raises the data rate to 128 kHz. These words are fed into a second register
67
, which feeds each word into the demodulator
69
, repeating each word eight times, resulting in a data rate of 1 MHz. The 1 MHz sample rate is a sufficiently high data rate so that the quantization noise which will be introduced into the signal is small, and the requirements of the analog smoothing filter are easily met. Output yi,
61
, is an analog signal.
Techniques for increasing the sample rate, generally called interpolation, are well understood by those versed in the art. Most designs will utilize several stages of increase, with each successive stage being simpler in structure, and running at a faster rate.
This sort of structure is frequently used in audio applications. The output of demodulator
69
can sometimes be driven directly into amplifier
75
and speaker
77
, because the speaker can act as a low pass filter. This configuration uses what is called class D output. Power dissipation in a class D stage has the potential for being very low, as the output transistors are always in either a fully shorted or open position, removing most resistive power consumption. The remaining power is dissipated by the switching of capacitance, which is equal to C*V
2
*F. C, the capacitance being switched, is typically set by the parasitic capacitance of the output transducer and of the driver transistors. V, the voltage being switched, is set by the available supplies, and the required audio output. F, the average frequency of the output, can be varied by the designer. As F is made larger, the quality of the signal improves, but the power also increases.
An over-sampling digital to analog (D/A) converter like that of
FIG. 2
, which includes clipping prior to the interpolating and up sampling blocks and utilizes a second order delta sigma quantizer
70
, and a low pass filter
71
to convert the data from the delta sigma quantizer
70
to analog signal yi,
61
, is a very effective device. However, clipping the digital signal prior to interpolating and up sampling results in a large amount of unpleasant distortion.
FIG. 3
shows a common second order delta sigma quantizer, which might be used as delta sigma quantizer
70
in FIG.
2
. Delta sigma modulation incorporates a noise-shaping technique whereby the noise of a quantizer (often one-bit) operating at a frequency much greater than the bandwidth is moved to frequencies not of interest in the output signal. A filter after the quantizer removes the out of band noise. The resulting system synthesizes a high resolution data converter, but is constructed from low resolution building blocks. A good overview of the theory of delta sigma modulation is given in Oversampling Delta-Sigma Data Converters, by Candy and Temes, IEEE Press, 1992.
In practice, delta sigma modulators are generally at least second order, because higher order modulators better reduce noise in the signal band, due to improved prediction of the in-band quantization error. Thus, the resulting signal to noise ratio is better. Second order delta sigma modulators are still relatively stable, and easy to design.
Input xi,
35
, is added to feedback signal
54
by adder
38
. The signal from adder
38
is fed into first accumulator
40
, comprising delay
42
and adder
41
. The output of accumulator
40
is added to feedback signal
54
and fed into second accumulator
44
, comprising delay
47
and adder
45
. The output of accumulator
44
goes into quantizer
50
, modeled as error signal ei,
52
, added to the input by adder
51
. Quantized output
36
also feeds back as feedback signal
54
. Quantizer
50
may quantize the signal into ones and zeroes (one-bit format) or into multiple levels.
A need remains in the art for clipping apparatus for use with a digital hearing aid which reduces distortion.
SUMMARY OF THE INVENTION
An object of the present invention is to provide clipping apparatus for use with a digital hearing aid which reduces distortion. The present invention improves distortion from clipping by moving the clipping step after the interpolation steps.
A digital hearing aid according to the present invention comprises a microphone for receiving an audio analog signal, an A/D converter for converting the analog signal into a digital signal, a digital signal processing stage for processing the digital signal, an interpolation stage for increasing the ample rate of the processed digital signal, a clipper for clipping the increased sample rate signal, a demodulation stage for converting the clipped digital signal into an analog signal, and a speaker.
Alternatively, the clipper could be incorporated into the demodulation stage. Such a demodulation stage includes a clipping delta sigma quantizer including a quantizer and at least one accumulator having an accumulator arithmetic element for adding a delayed output signal from the accumulator arithmetic element to an input signal provided to the accumulator arithmetic element, wherein the accumulator arithmetic element includes clipping means. The accumulator provides a signal to the quantizer which provides an output signal and a feedback signal to the accumulator, and the delta sigma modulator further includes a feedback arithmetic element for adding the feedback signal to the accumulator input signal. The demodulation stage also includes a digital to analog converter for con
Bales Jennifer L.
GN ReSound as
Harvey Dionne N.
Le Huyen
Macheledt Bales & Johnson LLP
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