EP0691050B1 - Circuit for deriving audio signal masking signals - Google Patents

Abstract

In a circuit for deriving audio signal masking signals in a radio receiver, a signal which is substantially proportional to the intensity of the reception field is transmitted through a low pass filter then weighted with a predetermined function.

Description

The invention relates to a circuit arrangement for deriving signals for masking audio signals in a radio receiver. Due to drops in the received field strength, the reception quality can fluctuate significantly, particularly with car radios. In order to keep the resulting interference as low as possible, measures for masking these interference in audio signals are known. For example, with low reception field strength, it is possible to reduce the stereo channel separation or to temporarily attenuate the audio signals.

A noise reduction circuit is known from EP 0 449 199 A, by means of which interference at the output of a receiver can be reduced. For this purpose, the field strength of the received radio signal is determined. Furthermore, a noise value is tapped at the output of the radio receiver and both signals are fed to a fuzzy circuit. On the basis of the output signal of the fuzzy circuit, an attenuation circuit is now activated, which is intended to reduce the noise level at the output of the receiver. A further noise reduction circuit is known from EP 0 418 036 A, in which occurring noises are damped by a low-pass filter with a variable cut-off frequency in the differential signal after the stereo demodulator is switched on. The cut-off frequency is determined as a function of a control signal which is determined on the basis of the received RF level, the multipath propagation level and the spectral content of the audio signal. And finally, from WO 89/02198 an FM radio system is known in which the receiver also has a noise reduction circuit. To reduce noise, both the received field strength of the received signal and the spectrum at the output of the discriminator are evaluated, with frequencies above 3 kHz being considered. In the case of weak signals or relay faults, the output signal is only attenuated if the control signal for the attenuation rises above a predetermined signal-to-noise ratio.

The measure according to the invention further improves interference suppression. In particular, this makes it possible to reduce the stereo channel separation even in the case of relatively short field strength drops, while the signals are damped as a function of the presence of interference signals in the received signal when the field strength drops are more or less short.

Advantageous further developments and improvements result from the subclaims. Although the coefficients can also be stored permanently in the circuit arrangement according to the invention, a further development of the invention is particularly advantageous in that the coefficient or coefficients are stored in a non-volatile memory and with the aid of a microcomputer, a display device and an operating device and with the aid of a program are changeable for operator guidance. This further training means that individual copies can be adapted a larger series of radio receivers to different, for example, typical operating conditions possible. The coefficients can also be changed by a service workshop or by the user.

A further development of the invention consists in combining the weighted field strength signals to form masking signals with auxiliary signals which indicate the presence of interference signals. The combination with the auxiliary signals is preferably carried out by multiplication.

Figur 1

ein erstes Ausführungsbeispiel,

Fig. 2

ein Teil des Ausführungsbeispiels in detaillierterer Darstellung,

Fig. 3

ein weiteres Teil des Ausführungsbeispiels,

Fig. 4

ein Diagramm zur Abhängigkeit der Stereo-Kanaltrennung von der Empfangsfeldstärke,

Fig. 5

ein Diagramm zur Abhängigkeit der Dämpfung der Audiosignale von der Empfangsfeldstärke,

Fig. 6

ein zweites Ausführungsbeispiel und

Fig. 7

wesentliche Teile eines Rundfunkempfängers mit einer erfindungsgemäßen Schaltungsanordnung.

Exemplary embodiments of the invention are shown in the drawing using several figures and are explained in more detail in the following description. Show it:

Figure 1

a first embodiment,

Fig. 2

a part of the embodiment in more detail,

Fig. 3

another part of the embodiment,

Fig. 4

a diagram of the dependence of the stereo channel separation on the reception field strength,

Fig. 5

1 shows a diagram of the dependence of the attenuation of the audio signals on the reception field strength,

Fig. 6

a second embodiment and

Fig. 7

essential parts of a radio receiver with a circuit arrangement according to the invention.

Identical parts are provided with the same reference symbols in the figures. The circuit arrangement according to the invention can be implemented in various ways. For example, individual or groups of the blocks shown can be implemented using suitable circuits, in particular integrated circuits. With a very high degree of integration, it is also possible to implement the entire digital signal processing of the receiver in an integrated circuit, signal processing steps, such as filtering or nonlinear weighting, being carried out by arithmetic operations. To implement a receiver with the circuit arrangement according to the invention, digital signal processors and other digital circuits, such as shift registers, flip-flops etc., can also be arranged together within an integrated circuit.

In the exemplary embodiment according to FIG. 1, a signal H3 is fed to an input 1 which corresponds to the received field strength in is substantially proportional and is referred to below as auxiliary signal H3. This is averaged in two low-pass filters 2, 3 with different time constants. A changeover switch 4 forwards one of the output signals of the low-pass filters 2, 3 as a signal AMC depending on a signal DD2 to be explained later. This is weighted at 5 to generate a signal AFE indicating the noise attenuation and can be removed at an output 6. The signal WF with a smaller time constant is also weighted at 7 and can be taken from an output 8 as signal WF2.

Coefficients K1, K2 required for weighting are stored in a non-volatile memory 9 and are supplied to the circuits 5, 7 via a microcomputer 10. K1 and K2 can be individual coefficients or a group of coefficients. A display device 11 and an input device 12 are connected to the microcomputer 10. The microcomputer 10 is provided with a program which allows the setting of the coefficients in a menu-driven manner.

Fig. 2 shows details of the circuit 7 (Fig. 1). The signal WF can be fed to an input 15, while inputs 16, 17 are fed to coefficients K1.1 and K1.2. In a multiplier 18, the signal WF is multiplied by the coefficient K1.1. The product is then added to the coefficient K1.2 at 19.

So that the signal WF2 at the output 22 does not assume any negative values, the output signal of the adder 19 is compared with the value 0 at 20 and replaced with the value 0 in the case of negative values with the aid of a changeover switch 21.

Fig. 3 shows an example of a circuit 5 (Fig. 1), in which the signal AMC supplied at 23 with an am Input 24 applied coefficient K2 is multiplied by 25. The signal AFE can be taken from an output 26.

The dependence of the stereo channel separation SK shown in FIG. 4 on the reception field strength E can be set with the aid of the coefficients K1.1 and K1.2. A solid and a dashed curve are shown as examples. The coefficient K1.1 is essentially the slope and the coefficient K1.2 the shift on the field strength axis. The curve shown includes the dependency of the stereo channel separation on the signal WF2, which is given by characteristics within the stereo decoder.

Fig. 5 shows the attenuation L as a function of the received field strength E for two different values of the coefficient K2. By changing the coefficient, the slope and the beginning (0 dB point) of the attenuation or volume reduction can be adjusted as the reception field strength decreases.

Fig. 6 shows a second embodiment. The auxiliary signals H1, H2 and H3 are fed to inputs 45, 46, 27. The auxiliary signal H3 characterizing the reception field strength is averaged in two low-pass filters 28, 29 with different time constants. Depending on a signal DD2 to be explained later, a changeover switch 30 forwards one of the output signals of the low-pass filters 28, 29 as the signal AMC. This is weighted at 32 in the form of a noise curve to generate the noise attenuation AFE. The field strength signal with the smaller time constant is also weighted at 31 (signal WF2). This is multiplied at 33 by a signal AT1 to form the control signal D, which is available at the output 34.

Auxiliary signals H2 and H3 are used to generate the signal DD2, the generation of which is explained in more detail in connection with FIG. 7. For this purpose, the auxiliary signal H1 representing the spectral components above the useful range of the stereo multiplex signal is first squared at 35, thereby forming a measure of the energy content of these components. This is passed through a threshold value detector at 36, so that a signal AHD arises which indicates the presence of spectral components with an energy lying above a predetermined threshold. After squaring at 37, the auxiliary signal H2 formed from the symmetry signal SY (FIG. 1) is passed via a threshold value detector 37 ', the output signal ASD of which thus indicates asymmetries which exceed a predetermined threshold. Such asymmetries indicate, among other things, the presence of adjacent channel interference.

In many applications, the use of one of the signals AHD or ASD as signal DD2 already brings considerable advantages. In the exemplary embodiment shown, however, both detectors 36, 37 are provided, the output signals AHD and ASD of which are routed via a controllable logic network 38. On the one hand, this has the advantage that, in the case of pure mono broadcasts in which no carrier-frequency stereo signal is transmitted, the signal DD2 is derived from the auxiliary signal H1. It is also possible to derive the DD2 signal using stereo signal transmission methods that deviate from the European standard - for example, the FMX method in the USA.

The logical network 38 enables a selection or a logical combination of the two signals AHD and ASD to the signal DD1. The logical network 38 can be formed in a simple manner from a controllable four-way switch, the inputs of which are the signals AHD and ASD, an OR combination of these signals and an AND combination these signals can be fed. The signal DD1 is then available at the output of the controllable changeover switch and is fed to a pulse width discriminator 39. This ensures that the signal DD2 only indicates a fault when the signal DD1 is active for an adjustable minimum time.

In addition to controlling the changeover switch 30, the signal DD2 serves as a trigger signal for two asymmetrical integrators 40, 41. These essentially each contain a counter which jumps to 0 or another predetermined value at the moment of triggering and retains it as long as the signal DD2 is at 0. If the signal DD2 then assumes the logic level 1, the output signals AT1 and AMU of the asymmetrical integrators 40, 41 increase linearly to a maximum value with adjustable time constants. The signal AT1 is fed to a multiplier 33 together with the field strength signal WF2 weighted at 32.

The output signal AMU of the asymmetrical integrator 41 is multiplied at 42 by the signal AFE, which results in a signal AFE_AMU which effects an attenuation of the audio signals by means of the multipliers 9, 10 (FIG. 1) by a maximum of 33 dB. This signal can be found in the circuit at output 43.

The exemplary embodiments explained with reference to FIGS. 1 to 6 are parts of a radio receiver with digital signal processing, for which an exemplary embodiment is shown in FIG. 7. The signal received via an antenna 51 is amplified, selected and demodulated in a receiving part (tuner) 52 in a manner known per se. A stereo multiplex signal MPX1 with a sampling rate of 456 kHz is available at an output 53 of the receiving part 52. For a subsequent reduction in the sampling rate - also called decimation To achieve 228 kHz without alias interference, a low-pass filter 55 is provided before the sampling rate reduction 54. A low-pass filter with a flat frequency response in the pass band is required for proper further processing of the stereo multiplex signal. In order to save the effort required for this, in particular at the high sampling rate of 456 kHz, a simpler low-pass filter with a decreasing frequency response is provided in the exemplary embodiment. However, the drop in frequency response is compensated in a subsequent compensation filter 56.

The stereo multiplex signal MPX2 is then routed via a circuit 57 for automatic interference suppression, which repeats the sample values before the start of the interference until the end of the interference, in particular when spark interference occurs. This circuit is followed by a stereo decoder 58, which generates two audio signals L, R, which are passed to outputs 61, 62 via multipliers 59, 60. From there, the audio signals are fed to the loudspeakers via NF amplifiers.

A signal is generated from the stereo multiplex signal MPX1 with the aid of a high pass 63 and a decimation circuit 64 which contains signal components above the useful frequency range of the stereo multiplex signal, but which are folded into a lower frequency range by the decimation. This signal MPX3 indicates various faults, for example the faults caused by spark from vehicles. It is used on the one hand to control the circuit 57 for automatic interference suppression and on the other hand to form the auxiliary signal H1 by decimation of the sampling rate to 9.5 kHz at 65.

The auxiliary signal H2, whose sampling rate is also 9.5 kHz is formed by low-pass filtering at 66 and decimation at 67 from a symmetry signal SY. This in turn is shaped in the stereo decoder 58. It is known that the stereo subcarrier is amplitude-demodulated to form the differential signal LR. This is done by multiplying the subcarrier by a subcarrier of the same phase position regenerated in the radio receiver. In the stereo decoder 58, the stereo subcarrier is additionally multiplied by a carrier rotated by 90 ° with respect to the reference carrier, thereby producing a signal which is 0 for symmetrical sidebands of the stereo subcarrier and deviates from 0 accordingly for asymmetries. The further auxiliary signal H2 is formed from this signal by low-pass filtering at 66 and decimation at 67.

At an output 68, the receiving part 52 emits a signal AM, which is produced by amplitude demodulation of the FM intermediate frequency signal. In the exemplary embodiment shown, this likewise has a sampling rate of 456 kHz and is decimated by a factor of 48 after a low-pass filtering 69 at 70, so that the resulting third auxiliary signal H3 has a sampling rate of 9.5 kHz.

In the circuit 71 (see FIG. 6 for details), the auxiliary signals H1, H2 and H3 are combined with one another to form control signals D and AFE_AMU, the sampling rate of which is initially 9.5 kHz, but is increased to 228 kHz at 72 and 73. This is done by interpolating 24 samples each, which in the simplest case consists in repeating each sample 24 times. The control signal D is fed to a control input of the stereo decoder 58 and is used there to switch over to mono operation in the event of a disturbed reception. The signal AFE_AMU is fed to the multipliers 59 and 60, as a result of which the volume (masking) is reduced when there are faults.

Claims (7)

1. Circuit arrangement for deriving signals for masking audio signals in a broadcast radio receiver, a signal (H3) which is substantially proportional to the received field strength being supplied to a first low-pass filter (2), characterized in that the signal (H3) which is substantially proportional to the received field strength is supplied to a second low-pass filter (3), in that the output signal of the first low-pass filter (2) is weighted in a first circuit (7) with predetermined first coefficients and can be used to form a masking signal for reducing the stereo channel isolation (7), in that the output signal of the first low-pass filter (2) is likewise weighted in a second circuit (5) with predetermined second coefficients and can be used to form a masking signal for attenuation of the audio signals, and in that, in the event of interference in the audio signal, the output signal of the second low-pass filter (3) is supplied, instead of the output signal of the first low-pass filter (2), to form the masking signal for damping the audio signals of the second circuit (5).
2. Circuit arrangement according to Claim 1, characterized in that the coefficients are stored in a nonvolatile memory (9) and can be varied with the aid of a microcomputer (10), a display device (11), a control device (12) and a programme for operator guidance.
3. Circuit arrangement according to Claims 1 or 2, characterized in that the weighted output signals of the first low-pass filter and of the second low-pass filter are combined with auxiliary signals to form the masking signals, which auxiliary signals are derived from interference in the audio signal.
4. Circuit arrangement according to Claim 3, characterized in that the combination of the weighted output signals of the first low-pass filter and of the second low-pass filter with the auxiliary signals is carried out by means of multiplication.
5. Circuit arrangement according to one of Claims 1 to 4, characterized in that interference in the audio signal is confirmed when spectral components of the audio signal which are above the useful range of the stereo multiplex signal exceed a predetermined threshold over a predetermined time period.
6. Circuit arrangement according to one of Claims 1 to 5, characterized in that the weighted signal which is proportional to the received field strength is limited to a maximum value.
7. Circuit arrangement according to one of Claims 1 to 6, characterized in that the masking signal for reducing the stereo channel isolation is limited to non-negative values.

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