EP0748043B1 - Automatic volume control - Google Patents

Description

The invention relates to a method for automatic volume correction when playing sound signals through a speaker and a Circuit arrangement for performing the method.

The invention is based on the problem that the broadcasters sound signals for television or audio programs with different strokes send. In addition, it is often in the background music for films dynamic control of the sound provided to certain scenes to be particularly emphasized by a musical accompaniment. In particular it is annoying when jumping from one program to another Program the sound can be heard at different volumes. The television viewer usually makes do with the fact that after switching the volume control on its remote control to another program pressed to correct the playback volume of the sound so that that he can hear the sound performance, but on the other hand no excessive Volume is given. With the often given habits, after which continuously from one program to another program is jumped, it means that the remote control for the volume correction must be used, which is not only a continuous Operation leads to faster consumption of the Battery from which the remote control unit is powered.

From DE 36 02 643 A1 it is known in an electroacoustic conference system equip each microphone with a circuit that one Includes dynamic controls and facilities to soften the microphone from one Ground level quickly becomes a switching level for a tone bus switch through and after switching off slowly (2 seconds) to the basic level attributed. The sound signal becomes quick when switched on increasing and regulated at low level to a high level device and slowly returns to the basic level when switched off. It a threshold circuit is also provided, which is a switching device switches on when a preset threshold value is reached.

From US 5 046 107 is a semi-automatic method for volume correction known in which the desired average level of the signals of Channels can be saved at the push of a button. The user must do this for adjust the volume once for each sound channel.

The invention has for its object a method and a circuit to specify for a sound signal reproduction device of the generic type, which can be used for both monotone and stereo sound reproduction and a volume control so automatically depending on the Reception level changes that the selected volume during playback is not exceeded and nevertheless a distortion-free as possible Playback of the sound is given.

The object is achieved by the method specified in claim 1, that the implementation of any received analog audio signals first in digital audio signals requires before peak detection, filtering and weighting a correction factor is determined that the volume control is supplied to correct the setting value.

As an alternative, an analogous method is specified in the independent claim 8, that works equally. Appropriate embodiments of the process according to claim 1 are in dependent claims 2 to 7 and a circuit arrangement for performing the method according to a of claims 1 to 7 in claim 9 and expedient embodiments in claims 10 and 13 indicated while a circuit arrangement to carry out the specified in the secondary method claim 8 analogous method in claim 11 and further training in Claim 12 are specified. To those specified in the claims Design forms are referenced.

In both methods, namely those in connection with an analog or digital audio signal, it is possible to set the control delay. So it is e.g. when switching from one television program to another desires that the regulation be made immediately while it is ongoing Operation should not appear so suddenly, for example at the Play back one scene in a movie and one overemphasized Piece of music with high dynamic modulation to perceive no sound distortion. In this case, the same is done Way a correction of the volume setting, but with a time delay with smooth transition.

The method according to the invention can be used in a simple manner in connection with known sound processors in digital processing using a filter. Such processors are offered, for example, by ITT Semiconductors Intermetall, D-79108 Freiburg, with the type designations MSP 3400, 3410, 3400 C and 3410 C. Such sound processors have a volume and balance control that can be set via an I ² C bus. Furthermore, they have a peak value detector which detects the peak values for monotone transmission as well as for stereo sound transmission in both channels. The peak values are evaluated for other purposes. According to the invention, these determined values can now be read out in a simple manner and subjected to a weighting, as specified in the method.

The volume adjuster is automatically dependent on the invention controlled by the average peak value of the received audio signal so that the volume selected by the user is not exceeded, but neither is clearly undercut in the long run.

Advantageous embodiments of the invention for both analog and digital Sound signal processing is in the drawings in the form of block diagrams shown.

Fig. 1

das Grundprinzip der Regelung,

Fig. 2

eine analoge Regelungs- bzw. Korrekturschaltung zur Ermittlung eines Korrekturfaktors und

Fig. 3

eine digitale Lösung unter Verwendung einer Filterschaltung.

The drawings show:

Fig. 1

the basic principle of regulation,

Fig. 2

an analog control or correction circuit for determining a correction factor and

Fig. 3

a digital solution using a filter circuit.

The basic structure is shown in FIG. 1. There is an analog one at input 1 Sound signal on, which is amplified via volume control 33 according to the setting and the output 34 is supplied by a loudspeaker in a certain volume is played. According to the invention, it is now provided this setting value automatically depending on the level correct the input signal in the desired way so that none Overdrive or excessive volume is played. This lies the sound signal at a peak detector 6. The peak values are in an automatic volume correction circuit 32 evaluated and on Correction value calculated, the line 27 to the electronic setting element the volume control is fed. For example, by Multiply the correction to be made in the desired manner. The correction circuit can be constructed analog or digital.

2 is an analog embodiment in the form of a block diagram specified. There is either a mono or at inputs 1 and / or 2 Stereo signal on, which is rectified via diodes 3 and 4. The rectified signal is the one input of an operational amplifier 6 supplied, at whose second input a signal is present for comparison, that corresponds to the breastfeeding function, so that in the event of a breastfeeding signal at the output no signal is present and in the case of a sound signal to be reproduced there is a signal at the output which controls a switch 5. In case that no signal is present, the switch 5 is ineffective, so that the rectified Peak values are not fed to the sequential circuit. In the case of In contrast, sound signals to be reproduced are the peak values the signals an RC element from a resistor 30 and one Capacitor 29 supplied, to which a further capacitor 31 is connected in parallel is switched. The two capacitors can be switched on and off are used to determine an average level from the detected received signals with channel switching and normal operation. The two capacitors are each connected to ground via the switch 28 switched. Control takes place via the operating processor 14, which is operated by an operating unit 15 receives its command. For example, at television reception switched from one channel to another, so takes each in the switched state of the switch 28, the position shown, so that due to the smaller time delay there is a quick regulation, while in steady state by switching the switch 28 on the capacitor 31 a slower regulation during the Operation, i.e. playback. The middle level at the output is the volume control shown in Fig. 1, whereby the adjusted volume is corrected accordingly, which leads to an increase, but in particular also to lower the volume to the desired level Way leads.

In Fig. 3, a digital version is shown in which the digitized Sound signals from the left and right channels are present at inputs 1 and 2, the left or right signals via the peak value detectors 3 and 4 are fed to a maximum detector 5, so that always the Peak value of both signals is determined. The measured peak value is in a downstream breastfeeding detector 6 compared with a reference value. For example, the peak value is below -42 dB based on the Tax limit, so this is identified as "silent" and the scheme will ineffective. For example, if the value is above - 42 dB, the The detector detects the received audio signal or the determined peak values a filter in which the average level is determined. The filter 7, represented by blocks 8, adder 9, block 10 and Delay stage 11, provides a medium level when considered a time constant for the transfer function depending on the Output signal from the breastfeeding detector. Due to the time delay in the Software loop over block 11 is dependent on one predetermined setting by a switching logic 14 that e.g. to Program change the control runs fast and slower in operation. The Constant for the transfer function is dependent on the corresponding control command changed. This information is needed symbolically indicated by line 16. In practice, this is done using known software solutions to implement one digital PT1 filter. Switching takes place via case inquiry and Table. The filtered average level is at y (k + 1) via output 12 then converted into a target correction level. This is done via a table as a line / log converter and a subsequent comparison with a reference level in the subtractor 20, the reference level above the Input 19 is present. The reference level can e.g. -24 dB, referred to the tax limit.

Furthermore, the correction value determined in this way runs through a subsequent one Limiter 21, which limits the correction to, for example, ± 12 dB. The way determined target correction level is fed to an I controller with dead zone 18, consisting of a subtraction element 22, via which the limited Target correction level compared with the actual correction level at output 27 will continue to consist of the dead zone 23, continue to consist of an integrator, formed from an adder 26 and a delay 25, at the output of the actual correction level as a control signal for the volume control is available.

anliegt. Die Zeitkonstante wird dabei für auf- und abklingenden Pegel sehr unterschiedlich gewählt. Bei T = 30 msec erscheinen Werte von q1 = 0..2, q2 = 8..10. Diese Werte werden als sinnvoll erachtet. Das Ausgangssignal des Filters bildet nach linear-logarithmischer Wandlung den mittleren Signalpegel in dB mit 1 dB Auflösung. Die Differenz aus dem mittleren Signalpegel und einem vorgegebenen Differenzpegel ergibt nach der Begrenzung den Sollkorrekturwert, wie vorher angegeben. Der Sollkorrekturwert geht auf den I-Regler wie beschrieben, der ein Akkumulator ist, mit einer toten Zone von ±1 dB, dessen Ausgangssignal den tatsächlichen Ist-Korrekturwert bildet. Dieser Korrekturwert wird zur aktuellen vom Benutzer gewählten Lautstärke hinzuaddiert. Dabei bewirkt der I-Regler mit toner Zone, daß die Verstellung in Einzelschritten von 1 dB erfolgt. Parameter mit tvpischen Werten sind: Referenzpegel -20dB (bezogen auf die Aussteuergrenze) Stillsignalpegel -38dB Zeitkonstante q1 für aufklingende Signale 1 Zeitkonstante q2 für abklingende Signale 9 To determine the correction value, the instantaneous signal level, which exceeds a certain still level, is fed to the digital PT 1 filter, at whose output the signal

is present. The time constant is chosen very differently for rising and decaying levels. At T = 30 msec, values of q1 = 0..2, q2 = 8..10 appear. These values are considered useful. The output signal of the filter forms the mean signal level in dB with 1 dB resolution after linear-logarithmic conversion. After the limitation, the difference between the mean signal level and a predefined difference level gives the target correction value, as previously stated. The setpoint correction value goes to the I controller as described, which is an accumulator with a dead zone of ± 1 dB, the output signal of which forms the actual actual correction value. This correction value is added to the current volume selected by the user. The I-controller with toner zone ensures that the adjustment is made in single steps of 1 dB. Parameters with typical values are: Reference level -20dB (based on the tax limit) Still signal level -38dB Time constant q1 for ringing signals 1 Time constant q2 for decaying signals 9

On the different use of the term "delay" in control engineering and digital signal processing is pointed out. Here with "delay" is the "dead time" mentioned in the control technology Function block meant.

Claims (13)

1. Method of automatic correction of sound volume in the reproduction of sound signals by way of a loudspeaker, with the following features:

   a) the demodulated sound signals are, insofar as they are not present as digital signals, converted into digital sound signals;

   b) the instantaneous level of the digital sound signals is ascertained by a peak value measuring device;

   c) the ascertained maximum is checked in a silence detector with respect to whether it concerns a silence signal or a sound signal to be reproduced;

   d) in the case of a sound signal to be reproduced the mean level of the signal is determined by a filter; and

   e) the filtered mean level is subsequently recalculated into a target correction level which for correction of the set sound volume changes a sound volume setter, which is arranged in the sound signal branch, by addition in such a manner that a maximum correction value is not exceeded, wherein in the case of detection of an applied silence signal the regulation is switched to be ineffective.
2. Method according to claim 1, characterised in that the mean level is compared with the values of a correction table and the difference value is fed to a downstream limiter, from the output of which the target correction level can be derived.
3. Method according to claim 2, characterised in that the target correction level is fed to a downstream integral regulator with a dead zone, which establishes the actual correction value level which is fed to the sound volume setter.
4. Method according to claim 1, characterised in that the correction value is fed to the sound volume setter as a multiplication factor.
5. Method according to claim 1, characterised in that the mean level is ascertained by low-pass filtering with a switchable time constant.
6. Method according to claim 5, characterised in that the time constant of the filter is, for delay of the regulation, switchable over to at least two values by way of a control circuit, wherein the regulation runs rapidly at least during program change and more slowly in operation.
7. Method according to one of the preceding claims, characterised in that in the case of reception of stereo signals the sound signal level is ascertained by peak value measurements of the left and right sound signals and the maximum values are evaluated for determination of the correction level.
8. Method of automatic correction of sound volume in the reproduction of sound signals by way of a loudspeaker, characterised in that on reception of an analog sound signal the peak value of the input level is ascertained and this is compared with a silence signal level and the ascertained signal is fed by way of a switch of an integration circuit with a switchable time constant, which ascertains the target correction level, which is fed to the sound volume regulator, wherein in the case of detection of an applied silence signal the regulation is switched to be ineffective.
9. Circuit arrangement for carrying out the method according to one of claims 1 to 7, characterised in that a present integrated circuit with a peak value detector is used, which reads the values by way of a control bus, which values are subjected to weighting and filtering by a microprocessor with written-in program and that the microprocessor circuit controls the sound value setter present on the basis of the measured peak values, wherein the case of detection of an applied silence signal the regulation is switched to be ineffective.
10. Circuit arrangement according to claim 9, characterised in that a sound signal processing processor is used, which carries out peak value measurements of the sound signals and comprises a controllable sound volume setter and/or is readable and settable by way of a control bus.
11. Circuit arrangement according to claim 8, characterised in that rectifier diodes are provided, which rectify the analog signal of at least one channel, that the thus-ascertained peak value is applied to one input of a comparator, to the second input of which the silence signal is applied, that the comparator delivers a switching signal on exceeding of the silence signal level to a switch by way of which the rectified signal can be fed to a resistance-capacitance element, from the output of which the mean level can be derived, from which the target correction level, which regulates the sound volume regulator, can be ascertained by analog difference formation.
12. Circuit according to claim 11, characterised in that the resistance-capacitance element comprises two parallel capacitors which can be selectably switched to by way of a control circuit, wherein the capacitor for formation of a low time constant is switched on when program change takes place and the capacitor for formation of a higher time constant is switched on when continuous operation takes place.
13. Circuit according to one of claims 8 to 12, characterised in that the switching over from program change to continuous operation or conversely takes place in dependence on a control command for switching over to another reception program.

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