JPS61196624A - Impulse noise removing processor - Google Patents

Abstract

PURPOSE:To supply the natural sound without a sense of incompatibility, to remove the impulse noise between discretes by sampling the sound signal and to obtain excellent noise removing processing effects by removing the pulse noise section by using a sampling frequency conversion. CONSTITUTION:A sound signal outputted by a distinguishing device 18 enters through a low-pass filter 34 of the invented circuit 30 to a sampled and hold circuit 36, and sampled at a frequency f1. Respective samples are converted to the digital value at an A/D converter 38, and by a noise detecting output from an impulse noise detecting device 32, the sample for the noise is removed. At a signal processor 40, respective samples which the A/D converter 38 outputs are sampled at a frequency fp again, low-pass filtering is executed for the resampled sample, and the result is resampled at a frequency f2 (here, fp=Mf1= Nf2, and M and N are integers and M, N is obtained). The output of the signal processor 40 is converted to an analog by a D/A converter 50, and through a low-pass filter 52, the analog sound signal, from which the noise is removed, is obtained.

Description

DETAILED DESCRIPTION OF THE INVENTION [Field of Industrial Application] The present invention relates to a processor for removing impulsive noise caused by ignition noise, motor noise, etc. contained in an audio signal of a vehicle-mounted radio receiver.

[Conventional technology]

A method for removing impulsive noise includes a method of detecting the start of pulse noise, holding the signal at a value before the start of pulse noise for a certain period of time, and smoothing the signal.

To explain with Figure 9, Tal indicates a signal with noise,
S is a signal, and N I" N 3 is a noise. These noises are detected, and from the time of detection they have a constant width to (
A gate pulse such as c+ is generated and the amplitude of the signal is held at the previous value during the pulse width of this gate pulse (a signal from which noise has been removed such as bl is obtained).

[Problem that the invention seeks to solve]

However, in this method, since the gate pulse width to is constant, there will be a portion where a wide noise such as N2 cannot be completely removed, and to avoid this, if the gate pulse width to is increased, the noise can be removed. The flat area that appears in the area becomes larger,
The distortion of the signal waveform becomes large, resulting in the possibility that noise removal becomes noise addition. Also, the gate pulse has noise Nl,
N? .

It is necessary that the timing of occurrence of this happens correctly, and if this happens, incomplete noise removal will occur frequently.
Also, non-noise parts of the signal are removed, increasing distortion.

The present invention utilizes digital signal processing technology to eliminate or reduce impulsive noise caused by igniter operation and motor commutator brushes specific to automotive receivers, thereby achieving desired noise suppression. It is an attempt to obtain characteristics.

[Means for solving problems]

The present invention provides a processor for removing impulsive noise contained in an audio signal, which includes a detector for detecting noise contained in the audio signal, a detector that samples the audio signal at a frequency f1, and digitizes each obtained sample. , a first circuit that removes samples for noise, resamples each sample output by the circuit at a frequency fp, performs digital low-pass filtering on the resampled sample, and resamples the result at a frequency f2. (here fp=M'f
1=Nf 2, M, N are integers and M>N) This is characterized by comprising a signal processor, and a second circuit that converts the output of the processor into analog and further performs low-pass filtering.

[Effect]

In the present invention, impulsive noise is removed by interpolating noise intervals in the process of converting the sampling frequency system, such as by resampling the audio signal or thinning out the output of a digital low-pass filter. To explain with Fig. 4, (al indicates each sample obtained by sampling the audio signal at frequency f1, 0, 1, 2
．．・・・・・・ is the O1 1st. Second.・・・・・・
Each sample. In this example, the third sample is abnormal, and is a sample of noise N. In the present invention, such samples are resampled at frequency fp. f
p is synchronized with fl, and fp=Ef+. Here, E is an integer, and in this example, E=3. When resampling at such a frequency (as shown in bl, the samples at fl are 0.1, 2... (in (b), 0, 3,
6゜...) and two samples 1 and 2 between each
, 4° 5, ... (all of these are 0) are obtained. Separately, the audio signal is passed through a bypass filter to detect noise N, and the sample at the time when noise N is detected is not taken into the sample at rp, and the sample in this part is separately processed as the average value of the samples before and after it. Occur.

Sample 9 of fb) shows this average value. Furthermore, this is
If the signal level is low, it may be possible to simply set it to 0 instead of using the average value.

Next, (sample rows 0, 1, 2, 3, etc. shown in bl)
... is passed through a digital low-pass filter to obtain the sample sequence (C). That is, when passing through this filter, samples 1, 2, 4, 5, etc., which were O before passing through this filter. . . . are samples 0, 3, 6, . . . as shown in the figure. It is reproduced to the value at the relevant timing in the low frequency determined by . In addition, for a sample such as sample 9, which is not a sample of an audio signal, the shape is such that the low frequency signal determined from the sample rows on the left and right sides of the sample are matched, and the envelope waveform is discontinuous. Such a sequence of samples is then resampled at frequency f2. Frequency f2 is synchronized with fp, and there is a relationship f'p=Ff2.

Here, F is an integer, and in this example, F=2. How can I set the outputs of the digital low-pass filter to 0, 1,
2, 3. In this example, take out every other one and write f
Sampling can be done in 2 steps, and since only a portion is taken out and the rest is discarded (in this example, half is discarded)
If a sample corresponding to noise is included in the remaining portion, it is possible to prevent the sample 9 from being included in this example. In this way, by obtaining a sample sequence of (dl) and applying it to an analog low-pass filter, it is possible to obtain an audio signal from which noise N has been removed.

FIG. 5 shows a block diagram of this process. Sample the audio signal at frequency fI, resample it at frequency rp,
The sheep output is resampled at frequency f2 through a digital low-pass filter, and noise is removed in this process. This
So rp is f p=Ef + =F f 2 (E>F)
・・・・・・+11, that is, the least common multiple of fl and f2. In this way, the sample point based on f2 can be matched with the point interpolated by the digital low-pass filter, and each sample in the later stage can be matched with the point interpolated by the digital low-pass filter. Samples at each stage can be captured more reliably. For example, f+ is 40KHz
,, f2 is 60 KHz, and fp is 120 KHz.

If there is a sample value of 60KHz, f+/2=L of 20KHz
PF can restore signals up to 20KHz.

Next, to explain the filter, the frequency characteristic H(e'') of a digital filter is generally expressed by the following formula.

T((ej”) −Σ h(nl e −””
--(21 unit impulse response is Sin (ωcn) ωc 5in (ωcn)
π n π ωcn. These are shown in FIG. As shown < (4) formula % formula %
) Approximate to fnl.

h f fn) −h fnl x ω(n)
++ ++ (51 Let us consider configuring an FIR digital filter using Equation (5) to have multiplication coefficients of □ discrete values on the left and right sides centered at n=Q in FIG. 7 fbl.

Since the cutoff frequency of the filter for restoring the discrete signal of the system with frequency fI to an analog signal is f1/2, the cutoff frequency ωC at al is ωC; π...(7)
, and the system of frequency f1 is sampled ωC− at frequency fp.
π/E (8). h d (n) in the above case is h d (
nl-earth
E nπ/E , which is shown in FIG. The FIR digital low-pass filter in which the value of n in equation (9) is set within the range of equation (6) h d (nl is the multiplication coefficient is as shown in Fig. 8. Now E = 3.F = 2, that is, fp =3f+=2f
2. If ωc-rt/3 and N-13, then FIG. 3 is obtained. As the digital low-pass filter, the one shown in FIG. 3(C) may be used.

〔Example〕

FIG. 2 shows the configuration of an on-vehicle radio receiver to which the present invention is applied, and FIG. 1 shows the main part thereof, that is, an embodiment of the present invention. An AM or FM radio receiver, as shown in Figure 2,
It has a high frequency amplifier 10, a mixer 12, a local oscillator 14, an intermediate frequency amplifier 16, a discriminator (or detector) 18, and an audio amplifier 20, and amplifies, selects, detects, and amplifies the broadcast waves received by the antenna 24. The speaker 22 is made to sound. The inventive circuit 30 is inserted between the discriminator 18 and the audio amplifier 20 to eliminate impulsive noise. The detector 32 detects this noise. This detector 32 includes a bypass filter and a shaping circuit that converts the output into a rectangular wave, and generates a rectangular wave that is generated simultaneously with the pulse noise and whose pulse width is the duration of the noise.

FIG. 1 shows details of the circuit 30 of the present invention, including an analog low-pass filter 34 for band limiting, and a sample-hold circuit 3.
6, A/D converter 38, signal processor 40, D/
A converter 50, analog low-pass filter 52 for interpolation
Consisting of The audio signal output from the discriminator 18 passes through a low-pass filter 34 and enters a sample-and-hold circuit 36.
Here, it is sampled at frequency f1. Each sample enters an A/D converter 38, where it is converted to a digital value, and the noise detection output from the pulse noise detector 32 removes the sample for noise (sets it to O).

Each digitized sample is processed by a signal processor 40.
Then, the process is performed as explained in FIG.

Processor 40 is RAM (Random Access Memory)
42, an arithmetic circuit 44, a control circuit 46, a ROM (read-only memo 48), and coefficients h d (nl) necessary for constructing a digital low-pass filter are stored in the ROM 48. From the A/D converter 38 Each sample is sequentially written to the RAM 42, and the sum of products with the coefficients of the ROM 48 is obtained in the operation circuit vIr44.To explain with reference to FIG. 4, the output of the A/D converter 38 is as shown in FIG. Each sample is 0, 1°2, ... (but digitized), and these are sequentially written into the RAM 42.The RAM write frequency is fp, and each two in between is Data is written as 0, and the noise corresponding sample set to 0 is newly generated as the average value of the left and right samples, and thus the data fbl in Figure 4 is stored in the RAM.The product-sum operation is clear from Figure 3. Multiply 13 consecutive samples (N-13 in this example) by coefficients h(-6), h(-5), h(6), and calculate the sum of these products. is obtained, multiplied by 3, and the above process is repeated while shifting the entire sample by pinching one sample.

That is, z −j in FIG. 3 is each input sample X (n)
It is a circuit that provides a delay corresponding to the time between phase tiles, and is 12
Since there are 13 consecutive samples, the coefficient is...
, 6), h(-5), ...... Multiplier M of h(6)
simultaneously, and after one sample time has passed, the rightmost sample disappears, a new sample enters the leftmost sample, and the whole becomes 1.
The new No. 13 sample group is simultaneously input to the multiplier M while shifting the sample to the right, and this process is repeated thereafter. In order for the arithmetic circuit 44 to perform such processing, 13 consecutive pieces of data are read out from the RAM 42, and the combination of these and the coefficients J-6), h(-sl+ . . . h(6)) of the ROM 4B is performed. The product is multiplied, new 13 pieces of data shifted by one sample are read out from the RAM 42, and the same process is repeated, and this process is repeated thereafter.

As a result of this processing, each output of tel in Fig. 4 is 0, 1°2,
. . . are obtained, and these are written to the RAM 42.

Then, if we take out every other sum of products written to this RAM 42 (resampling with f2), we get +dl in Figure 4,
This becomes the output of the signal processor 40.

The output of the signal processor is converted to analog by the D/A converter 50 to obtain (el) in Figure 4.This looks the same as (dl), but (dl is a digital signal and (11) is a discrete signal. Yes, this figure 4 (e) immediately D/A
When the output of the converter 50 is passed through a low pass filter 52, a noise-free analog audio signal is obtained.

The delay circuit 54 delays the timing of signal processing to ensure that the samples at the output of the detector 32 are processed against noise. The control circuit 46 performs various controls.

〔Effect of the invention〕

As described above, according to the present invention, since the pulse noise section is removed using sampling frequency conversion, it is possible to provide a more natural and natural sound. Further, by sampling the audio signal, pulse noise of discrete values is also removed at this processing stage, and an excellent noise removal processing effect can be expected.

[Brief explanation of drawings]

FIG. 1 is a block diagram showing an embodiment of the present invention, FIG. 2 is a block diagram showing the configuration of a radio receiver to which the present invention is applied,
FIG. 3 is an explanatory diagram of the digital filter, FIG. 4 is a waveform diagram for explaining the operation, FIG. 5 is an explanatory diagram of the processing procedure of the present invention, and FIG.
8 are explanatory diagrams of digital filters, and FIG. 9 is an explanatory diagram of conventional noise removal. In the drawing, 32 is a pulse noise detector, 36.38 is a first circuit, 40 is a signal processor, and 50.degree. 52 is a second circuit.

Claims (1)

[Claims] A processor for removing impulsive noise contained in an audio signal includes: a detector for detecting noise contained in the audio signal; and a detector for sampling the audio signal at a frequency f_1 and each obtained sample a first circuit for digitizing and removing samples for noise; resampling each sample output by the circuit at a frequency f_p; digitally low-pass filtering the resampled samples; and resampling the result at a frequency f_2. A signal processor performs sampling (here f_p=Mf_1=Nf_2, M and N are integers and M>N), and a second signal processor converts the output of the processor into analog and further performs low-pass filtering.
A pulse noise removal processor comprising a circuit.