METHOD AND APPARATUS FOR RANDOM FM NOISE CANCELLATION

Abstract: A method for reducing noise caused by fading in FM radio systems . The algorithm described operates on a sequence of complex samples received at the input of the discriminator (501) from the output of the IF filter of a digitally implemented radio. After determination of magnitude (502), which yields the raw envelope of the received signal, the raw envelope information is processed via parallel paths to determine characteristics indicative of shape, amplitude and time of occurrence of noise pulses. The output signal from the discriminator (501) is processed (508) to determine noise pulse direction. A pulse generator (510) produces counter pulses based upon the previously determined characteristcs, and the counter pulses are combined (513) with the discriminator output to substantially cancel the noise pulses.

Description: METHOD AND APPARATUS FOR RANDOM FM NOISE CANCELLATION Technical Field This invention relates generally to FM (frequency modulation) radio communication systems and in particular to FM radio systems based on digital signal processing (DSP) techniques, and is more particularly directed toward a method for cancelling random FM noise caused by undesired phase modulation of the transmitted signal induced by a fading transmission channel. Background of the Invention In order to more fully comprehend the problem with which the present invention is concerned, the concept of Rayleigh fading should be understood. Rayleigh fading refers to the rapid fluctuation in the magnitude and phase of a received signal resulting from multipath propagation. This Rayleigh-faded signal is most objectionable to a listener when the magnitude of the fade is large enough to cause a substantial momentary decrease in the received signal amplitude. This results in the listener hearing an objectionable burst of noise Rayleigh fading frequently occurs when a mobile radio user travels down a road, passing relatively close to nearby objects such as utility poles, buildings, and bridges, which results in field strength variations at the vehicle's mobile antenna. These changes in field strength cause corresponding fluctuations in the envelope of the received signal. Objectionable noise bursts and pops in the recovered audio that substantially coincide with envelope minima are characterized as random FM noise. Prior art approaches to improving response to Rayleigh-faded signals have been largely analog in nature, requiring precious space within mobile radio units and leading to increased costs. Further, most such approaches are double-ended in nature, requiring that some form of signal processing be added to both mobile units and base units. One approach to improving receiver performance in a Rayleigh fading environment, described in Application Serial No. 091 ,160, filed August 28, 1987, and assigned to the same assignee as the present invention, actually mutes receiver audio during fading events. Although this technique effectively eliminates noise bursts and pops, there is the potential for loss of information and potential unintelligibiiity. Additionally, the technique described in the above-cited Application involves a mixing operation (multiplication of two or more signals), which can result in sub-audible imbedded signalling, such as tone-coded squelch or low-speed trunkiπg data, being translated into the audible portion of the spectrum, causing an objectionable rumbling in receiver audio. Accordingly, a need arises for a method for cancelling random FM noise that takes advantage of modern DSP (digital signal processor) techniques in order to save space and eliminate the expense of analog components. In addition, a technique is needed that counteracts the effects of a fading event, rather than muting the audio. Furthermore, a need arises for a method for cancelling the effects of random FM noise that does not result in sub-audible signalling being heard in the receiver audio. Summary of the Invention According to the invention, a method is provided for cancelling random FM noise that substantially eliminates discriminator output noise pulses. Cancellation is provided by processing the envelope of an FM modulated signal to determine characteristics indicative of shape, amplitude, and time of occurrence of noise pulses. The discriminator output signal is processed to determine characteristics indicative of noise pulse direction. Counter-pulses are generated based upon the characteristics, and the counter-pulses and discriminator output are combined to substantially cancel the random FM noise. Brief Description of the Drawings Fig. 1 shows the real and imaginary parts, and the resultant magnitude, of a two ray fading parameter; Fig. 2 illustrates the real and imaginary parts, and the resultant magnitude, of a Rayleigh fading parameter; Fig. 3 depicts the normalized derivative of a fade envelope around an envelope minimum; Fig. 4 illustrates the relationship between the normalized envelope derivative and the discriminator output for Rayleigh fading; Fig. 5 is a block diagram of the random FM noise cancellation algorithm of the present invention; and Fig. 6 shows counter pulses and noise pulses properly aligned for summation to result in noise pulse cancellation. Detailed Description of the Invention A faded FM signal can be modeled by the following equation: f(t) = r(t)s(t) where: s(t) - e' θ M is the desired FM signal, and r(t) ■ p(t)e jφ(t) is the fading parameter. Thus, the received signal can be expressed as: f(t) = pfDθfrw+ . ) ] If this signal is applied to an FM limiter/discriminator, the resulting output is: d(t) = θ'(t) + φ'(t) where: θ '(*) ■ J j θ (*)» is the desireα message signal, and 'W " ^(Φ^)) is the Random FM noise signal - P(t). During deep fades, P(t) contributes pulse-like noise to the desired signal. If these pulses are of a substantial amplitude with respect to the desired signal, and contain energy in the audio bandwidth (300 to 3000 Hz), they can cause significant degradation of audio quality. The random FM noise cancellation method of the present invention uses information contained in the fading envelope (ρ(t)) to effectively cancel random FM noise pulses at the discriminator output. The pulse amplitude and shape are obtained from the envelope, while the pulse direction (sign) is obtained by proper filtering of the discriminator output signal. The effectiveness of the instant invention hinges on the assumption that the real and imaginary parts of the fading parameter are approximately linear for the duration of the noise pulses. The fading parameter has previously been defined, in exponential form, as: r(t) = p(t)e jφ(t) In rectangular form, the fading parameter can be expressed as: r(t) = x(t) + jy(t) For the duration of the random FM noise pulse, P(t), the real and imaginary parts of r(t) are approximately linear. That is: x(t) = m x t + b x and y(t) = m y t + b y This assumption is entirely reasonable and defensible for two reasons. First, Rayleigh (or Rician, multipath, etc.) fading is a lowpass process. Thus, over time segments significantly smaller than the minimum time period of the process, the signal will be approximately linear. The objectionable noise pulses caused by fading generally have widths that are considerably less than the minimum period of the process. This linear assumption is especially valid during envelope minima, at which times the noise pulses are generated. Fig. 1 illustrates the real part (102) and imaginary part (103) of a two ray fading parameter, as well as the resultant magnitude (101). A two ray fade consists of the direct reception of the desired signal plus one additional reflected signal, with the amplitudes of both signals remaining constant. As the Figure indicates, the real part (102) and imaginary part (103) of the fading parameter are the most linear during envelope minima. Fig. 2 depicts the real part (202) and imaginary part (203) of a Rayleigh fading parameter and the resultant magnitude (201 ). Although the effect is less obvious with a Rayleigh fading parameter than with a two ray fade, the real part (202) and imaginary part (203) are still most likely to be linear around the envelope minima as the Figure shows. It can be shown that, given this linear assumption, the amplitude and shape of the random FM noise pulse at the discriminator output can be calculated using only information contained in the fade envelope. Although this result is true in general, the derivation shown here will be for the simplified linear case in order to minimize algebraic complexity. The fading parameter has been expressed previously, in rectangular form, as: r(t) = x(t) + jy(t) The magnitude and phase of the fading parameter are thus given by: p(t) = Vx 2 (t) + y 2 (t) Assume a fade of the form: x(t) - b (where b is a constant ≠ 0) y(t) = mt (where m is a constant slope ≠ 0) If r(t) is input to a limiter/discriminator for FM demodulation, then the output will be given by: p(t) m χ (t)y / (t) - x'(t)y(t) x 2 (t) + y 2 (t) = ______ P 2 (t) P(t) is maximum for t - t m - 0, or the minimum of ρ 2 (t). Thus, Pmax can be defined as follows: Pmax = -ωb- P 2 (tm) b The envelope of this fade is given by: Then the derivative of ρ(t) with respect to time, ρ'(t), is given by: P'(t) = _ m 2 t P(t) The normalized derivative of ρ (t) is defined as: P'(t) g(t) P(t) _ m 2 t ~ P 2 (t) _ m t m 2 t 2 + b 2 Fig. 3 is a plot of g(t) around the envelope minimum. It can be shown that the magnitude of the random FM noise signal fl P max |) is simply the peak-to-peak amplitude of the normalized derivative. That is: Pmaxl = 9max " 9min From Fig. 3 it can be seen that g m ax - g(tmaχ) and g m j n = g(tmin)- These maximum and minimum times can be found using elementary calculus, since the solutions of g'(t) = 0 will give t max and t m .n: g'(t) β m 4 t 2 + m 2 b 2 - 2m 4 t 2 m 0 P (t) Since p(t) has been defined such that it is never equal to zero, the solution to this equation is given by: m 4 t 2 = m b or: t - ± m Therefore: tmax m and mm - m Substituting these values back into g(t) gives g m ax and Finally: maxl = m - 9max " g mm b This relationship also holds for the general case: x(t) = m x t + b x y(t) - m y t + by For the case of Rayleigh fading, Fig. 4 shows noise pulses at the discriminator output by a dashed line, with the normalized envelope derivative depicted by a solid line. From an examination of the Figure, it is apparent that the noise pulse magnitude is equal to the peak-to- peak value of the normalized derivative. The signal at the output of an FM limiter/discriminator in response to the general linear fade expressed above is: bvin, b y m ; P(t) = P 2 (t) = k P 2 (t) (where k is a constant). It has already been demonstrated that: maxl = — - 9max " gmiπ P 2 (tm) Since ρ(t) is known, as well as | P max | > |k| can be found by: " [gmax - 9min]P 2 (tm) Thus, the absolute value of the entire discriminator output noise pulse can be reproduced both in amplitude and shape using only information contained in the fade envelope: | p t x | _ [9max - gminJP 2 (tm) P 2 (t) In order to effectively cancel the noise pulse at the discriminator output, only the direction of the noise pulse remains to be determined. Pulse direction can be detected by proper filtering of the discriminator output signal. If no sub-audible signaling is present, such as coded squelch or low-speed trunking data, the pulse direction can be detected via a low pass filter. In the case of sub-audible signaling, a properly designed low pass or band pass filter will serve to determine pulse direction. In some instances (i.e., where the bandwidth of the message signal is lowpass), a high pass filter may be employed to determine pulse direction. A block diagram of the preferred implementation of the method of the present invention is shown in Fig. 5, as generally depicted by the number 500. Since the method is designed for use in a digitally implemented radio, the noise cancellation algorithm is implemented using a DSP, preferably a DSP56000 family device manufactured by Motorola, Inc. The algorithm receives its input from a digital IF filter, in the form of a sequence of complex samples, at the input of the discriminator (501 ). The algorithm necessarily runs at the discriminator sampling rate. The magnitude of each complex sample at the discriminator is calculated in block 502. These magnitude samples constitute the raw envelope (p) of the received signal. This envelope signal is then filtered by the envelope low pass filter (LPF) (503). The envelope LPF (503) is required to reduce the effects of noise and IF filter-induced ripple on the envelope. Since a filter introduces a delay to the calculation performed by the algorithm, the envelope LPF (503) is preferably linear phase and symmetric in order that the delay introduced will be an integral number of sample periods. Restricting filter delay to an integral number of sample periods simplifies alignment of generated counter pulses with actual noise pulses, which will be discussed in more detail below. The envelope LPF is implemented in the DSP as a symmetric finite impulse response (FIR) filter with an odd number of taps. The output of the envelope LPF (503), designated in Fig. 5 as ρ 1 f is processed via two parallel paths. In the lower path, the derivative of p 1 is calculated in differentiation block 504. The expression appearing within block 504, 1 - z **1 , is one possible z-transform approximation of differentiation, as is known in the art, and introduces a delay of one-half a sample period. In order to compensate for this one-half sample period delay, the upper path includes a two-tap filter (505) that adds a one-half sample period delay. The expression [1 + z- 1 ]/2 is one possible z-transform implementation of an appropriate delay process. The delayed envelope is subjected to an inversion operation (506), then the envelope derivative and the inverted envelope are multiplied in block 507 to yield the normalized derivative. The sign of the counter pulse is estimated by a pulse sign filter (508) that uses, as input, the output of the discriminator (501). The filter must be designed such that its passband covers a frequency region which contains minimal energy from normal audio, and any subaudibie signaling, while maximizing noise pulse energy. Since random FM noise pulses are lowpass in nature, the sign filter passband must reside in some frequency region below the normal audio cutoff frequency of 300 Hz. The impulse response of the sign filter should be short, with only one dominant lobe, in order to minimize inter noise pulse interference. Noise pulses that are bad enough to necessitate cancellation look essentially like impulses to the sign filter. Thus, the sign estimation delay from the detected pulse center is equal to the delay to the maximum amplitude excursion of the filter's impulse response. The normalized derivative computed in block 507 of Fig. 5 triggers the algorithm to begin collecting data for the creation of the counter pulse when it crosses the pulse detection threshold indicated in Fig. 3. Although data collection is triggered by the normalized derivative crossing a minimum threshold in the preferred embodiment, data collection could also be initiated through the normalized derivative crossing a maximum threshold, or the absolute value of the normalized derivative exceeding a predetermined threshold value. The pulse detection and measurement block (509) detects this threshold crossing, and begins searching for the local minimum of g(t) (g m in) between the threshold crossing and the zero crossing. The local maximum (gmax) is then located, and the absolute value of the pulse amplitude is calculated as described above. The output of the pulse sign filter (508) yields a sign estimate. All that remains is the pulse shape. As discussed above with reference to the derivation for the general case, the pulse shape is determined by the square of the ratio of the envelope value at the pulse center (p(t m )) to the envelope amplitude around the fade event (ρ(t)). The envelope amplitude value at the pulse center is determined from the delayed envelope passed to the pulse detection and measurement block (509) from delay process 505. The final pulse shape determination is made in the pulse generator block (510) through multiplication by properly delayed (by τ, ) samples of the inverse envelope provided by a delay line (511). The pulse detection and measurement block (509) provides the necessary information regarding pulse amplitude and sign to the pulse generator (510). Since the algorithm is implemented in a DSP, the delays introduced by each individual process are known exactly, so it is a simple matter to compute v to ensure that inverse envelope information will be delayed by the appropriate amount such that counter pulses will coincide properly with noise pulses at the discriminator output. The output of the discriminator (501 ) is also input to a delay line (512) of length χ c , where χ c denotes the cancellation delay. The cancellation delay represents the minimum audio delay necessary for the algorithm to operate correctly, and thus must be known exactly as well. The noise pulses at the output of the delay line (512) and the properly aligned counter pulses from the pulse generator (510) are added together in a summer (513) before being processed by a simple de-emphasis filter (514) to yield the final audio output. Fig. 6 depicts counter pulses (solid line) and noise pulses (dashed line) properly aligned for noise cancellation. Because proper operation of the noise cancellation algorithm of the present invention is highly dependent upon an accurate characterization of the envelope of the received signal, some modifications of the algorithm are necessary under operating conditions that tend to cause corruption of envelope data. These adverse operating conditions include low signal-to-noise (S/N) ratio, adjacent channel or co-channel interference, and frequency selective fading. Under good operating conditions, the normalized derivative, as depicted in Fig. 3, will have only one zero crossing during the collection period. However, at low S/N, the normalized derivative becomes a function of noise in addition to the desired envelope signal. Under these conditions, the normalized derivative will have significantly more zero crossings. Additional extraneous zero crossing events are also caused by high frequency envelope ripple due to interference or frequency selective fading. Accordingly, whenever a second zero crossing occurs unexpectedly within a predetermined time after the initial zero crossing, data collection is disabled and the algorithm is reset. Computer simulation of Rayleigh fading has shown that random FM noise pulses virtually never achieve amplitudes greater than 2 radians. Thus, whenever !g max ; or |g mjn | exceeds the value of 1.0, something is probably wrong. In response to this error condition, the pulse amplitude estimate produced by the algorithm is set to zero, so that a counter pulse of amplitude zero is added to the discriminator output. A third possible error condition may occur during computation of the shape of the counter pulse. As described above, the shape is the square of the ratio of the envelope minimum to the amplitude of envelope samples around the noise pulse event. This ratio should always be less than or equal to one, since the envelope minimum itself is the local minimum of envelope sample amplitudes around the noise pulse event. If this ratio is greater than one, an error has occurred, and the pulse amplitude estimate is set to zero to prevent a non-zero counter pulse from being added to the discriminator output. What is claimed is:

Claims: Claims 1. For an FM modulated RF signal having an envelope, a method for cancellation of random FM noise that substantially eliminates discriminator output noise pulses, the method comprising the steps of: (a) processing the envelope of the FM modulated signal to determine characteristics indicative of shape, amplitude, and time of occurrence of noise pulses; (b) processing the discriminator output signal to determine at least one characteristic indicative of direction of the noise pulses; (c) generating counter-pulses based upon the characteristics determined in steps (a) and (b); and (d) combining the counter-pulses and the discriminator output to substantially cancel the random FM noise pulses. 2. The method in accordance with claim 1 , wherein the step (a) of processing the envelope of the FM modulated signal further comprises the steps of: (a1 ) differentiating the envelope to provide an envelope derivative; (a2) dividing the envelope derivative by the envelope to yield a normalized derivative; (a3) computing peak-to-peak amplitude of the normalized derivative to provide noise pulse amplitude; (a4) computing time of occurrence of envelope minima to provide time of occurrence of noise pulses; and (a5) computing the square of the ratio of envelope amplitude at envelope minima to envelope amplitude around envelope minima to provide noise pulse shape. 3. The method in accordance with claim 2, wherein processing steps (a3) through (a5) are executed only when the amplitude of the normalized derivative crosses a predetermined threshold. 4. The method in accordance with claim 2, wherein processing steps (a3) through (a5) are terminated when the normalized derivative has more than one zero crossing within a predetermined time. 5. The method in accordance with claim 1 , wherein the step (b) of processing the discriminator output signal further comprises the steps of: (b1 ) filtering the discriminator output signal to provide a filtered signal; and (b2) sampling the filtered signal to provide noise pulse direction. 6. The method in accordance with claim 5, wherein the filtered signal is sampled at a time substantially coincident with the maximum amplitude excursion of the filter's impulse response. 7. The method in accordance with claim 1 , wherein the step (d) of combining the counter pulses and the discriminator output comprises adding the counter pulses and the discriminator output. 8. For an FM modulated RF signal having an envelope, an apparatus for cancellation of random FM noise that substantially eliminates discriminator output noise pulses, the apparatus comprising: means for processing the envelope of the FM modulated signal to determine first characteristics indicative of shape, amplitude, and time of occurrence of noise pulses; means for processing the discriminator output signal to determine at least one second characteristic indicative of direction of the noise pulses; means for generating counter-pulses based upon the first and second characteristics; means for combining the counter-pulses and the discriminator output to substantially cancel the random FM noise pulses. 9. The apparatus in accordance with claim 8, wherein the means for processing the envelope of the FM modulated signal further comprises: means for differentiating the envelope to provide an envelope derivative; means for dividing the envelope derivative by the envelope to yield a normalized derivative; means for computing peak-to-peak amplitude of the normalized derivative to provide noise pulse amplitude; means for computing time of occurrence of envelope minima to provide time of occurrence of noise pulses; means for computing the square of the ratio of envelope amplitude at envelope minima to envelope amplitude around envelope minima to provide noise pulse shape. 10. The apparatus in accordance with claim 8, wherein the means for processing the discriminator output signal further comprises: means for filtering the discriminator output signal to provide a filtered signal; means for sampling the filtered signal to provide noise pulse direction.

Inventors: BIRCHLER MARK ALLEN, JASPER STEVEN CHARLES, BIRCHLER, MARK, ALLEN, JASPER, STEVEN, CHARLES
Applicants: MOTOROLA INC, MOTOROLA, INC.
Application year: 1990
Main classification: H04B15
IPC codes: H03G3, H04B1