RU2267867C2 - Method and device for controlling reproduction of audio-signal bass components in electro-acoustic transformers - Google Patents

Abstract

FIELD: acoustics, in particular, technology and device for controlling reproduction of audio-signals in electro-acoustic transformers.

SUBSTANCE: method, realized by means of device for controlling reproduction of audio-signals in electro-acoustic transformers, includes selecting frequency components of audio-signal and amplifying them with amplification coefficient, calculated on basis of audio-signal, joining them with audio-signal in a way to produce modified audio-signal, which is sent to electro-acoustic transformer. By means of filter with terminal pulse response or filter with infinite pulse response, or single band filter, band filtering of audio-signal is performed for selection and amplification of first frequency component, low-frequency filtering and/or band filtering of audio-signal is performed for selection of second frequency component, contouring line and/or energy of unfiltered, subjected to low-frequency filtering and/or band filtering, audio-signal, is calculated.

EFFECT: improved reproduction of audio-signals bass components.

2 cl, 7 dwg

Description

The invention relates to a method for controlling the reproduction of audio signals in electro-acoustic transducers according to the generic concept of paragraph 1 of the claims and to a device for controlling reproduction of audio signals in electro-acoustic transducers according to the generic concept of paragraph 9 of the claims.

The reproduction of audio signals in an electro-acoustic transducer, especially in a loudspeaker or earphone, is due to the dimensions of the electro-acoustic transducer, in particular a loudspeaker or earphone. The smaller the loudspeaker membrane and its maximum displacement, the higher the lower resonant frequency.

Figure 1 shows a typical frequency response of a small speaker. Electronic audio devices in which such small electro-acoustic transducers are used and in which, therefore, bass reproduction is unsatisfactory, are primarily audio devices (devices for issuing or reproducing audio signals) communication and data transmission technologies, as well as consumer and consumer electronics, such as handsets of mobile phones and cordless phones, laptops, personal digital assistants (portable computers), miniature radios, radio wake-up call niki, portable playing devices.

In order to improve bass reproduction in a small speaker, the well-known psychoacoustic principle can be used. This principle is defined as “residual audibility” (the audibility of missing base tones) or as “virtual pitch”.

According to this principle, the perception of the fundamental frequency can be imitated by a combination of overtones. Therefore, the perception of low frequency can be simulated by the appropriate combination of its overtones.

A detailed description of the virtual pitch principle is found in the publication Psychoakustik E.Zwicker; H. Fastl; Springer Verlag, 2 ^nd . Edition, 1999.

Methods based on the psychoacoustic principle are also known from patents 6111960 and 5930373, which, using audio signals, generate an appropriate sequence of overtones to simulate frequencies below the cutoff frequency.

From the international publication WO 00/15003, a method based on the psychoacoustic principle is known in which the overtones contained in an audio signal are amplified. In order to improve the bass reproduction of audio signals in electro-acoustic transducers, the low-frequency components of the audio signal are extracted into the low-frequency audio signal, the selected low-frequency components are filtered out using a variety of band-pass filters, the frequency components filtered in the bands are amplified in the amplifier with an adjustable gain, and the gain is obtained from the envelopes from the filter frequency bands, and simulated low-frequency audio nal produce the original audio signal by combining with enhanced frequency components.

The task underlying the invention is to control the bass reproduction of audio signals in electro-acoustic transducers on the basis of the psychoacoustic principle, defined as "virtual pitch" or "residual audibility (audibility of missing base tones)" so that the perception of virtual bass reproduction The audio signals have been improved over the prior art.

This problem is solved on the basis of the method defined by the generic term in paragraph 1 of the claims, through the features specified in the characterizing part of paragraph 1, and also on the basis of the device defined by the generic term in paragraph 9 of the claims, by the features indicated in the distinctive part of 9.

The idea underlying the invention is to control the reproduction of the bass or bass supplied to the electro-acoustic transducer by amplifying the harmonic overtones already contained in the audio signal in the sense of imitation so that the listener perceives improved bass reproduction. In this case, control or imitation can be carried out both in digital form (paragraph 1), by means of a software module in a digital signal processor (DSP) of an electronic device, for issuing and / or reproducing audio signals by an electro-acoustic transducer, and in analog form (paragraph 9), by hardware circuitry included between the digital-to-analog converter and the terminal amplifier of an electronic device for issuing and / or reproducing audio signals through an electro-acoustic converter.

Using a software module and a hardware circuit, harmonic overtones are amplified that are above the resonant frequency of the electro-acoustic transducer, in particular a loudspeaker, to simulate the perception of the fundamental frequency. The selection of harmonic overtones when using a software module is performed by band-pass filtering, and when using a hardware circuit, using band-pass filters, while the gain of the overtones is controlled by a gain based on software in the software module, and in the hardware circuit, by for this amplifier with adjustable gain. The gain is preferably controlled by the components of the audio signal below the resonant frequency or the cutoff frequency of the electro-acoustic transducer.

The advantage of the method according to paragraph 1 is that the amplification of the harmonic source overtones contained in the audio signal provides a clearly better quality of the modified audio signal generated in the digital signal processor. This makes it possible to avoid distortion of the audio signal. In addition, the method according to the invention places lower demands on the processing power and the amount of memory in a digital signal processor.

Additional embodiments of the invention are reflected in the dependent claims.

So, according to clause 2 and clause 4, it is preferable if, when applying a filter with a finite impulse response (CIO filter), in contrast to the use of an filter with an infinite impulse response (BIO filter) according to clause 3, the audio signal is combined with amplified frequency components, in order to compensate for the phase shifts between the amplified frequency components and the audio signal for a combination based on the use of a KIO filter.

According to paragraphs 7 and 10, it is preferable if, to improve the quality of the modified audio signal output from the electro-acoustic transducer, this modified audio signal is filtered out to amplify the selected frequencies.

Two embodiments of the invention are explained below with reference to figures 2-7, which represent the following:

Figure 2 is a digital implementation of the method according to the invention in the form of a software module in a digital signal processor of an electronic radio communication device for issuing and / or reproducing audio signals,

Figure 3 is an analogue implementation of a method according to the invention in the form of hardware of an electronic radio communication device for issuing and / or reproducing audio signals,

FIG. 4 is a first embodiment of the software module of FIG. 2,

FIG. 5 is a second embodiment of the software module of FIG. 2,

6 is a third embodiment of the software module of figure 2,

Fig.7 is an embodiment of the control device of Fig.3.

Figure 2 presents an example of execution in the form of a functional diagram or block diagram of the speech signal processing path in the radio communication device UR for issuing and reproducing audio signals, in particular speech signals, the invention being implemented in the PM software module of a digital DSP signal processor (digital implementation). The US radio communication device receives an analogue radio signal PC, which is modulated by encoded speech information, through the antenna ANT. In the PR receiver, using the MP microprocessor and the analog-to-digital converter of the ADC, a digital demodulated DDS signal is generated from the modulated encoded analog radio signal PC. This digital demodulated DDS signal is then supplied to the speech decoder RD of the digital DSP signal processor. In the speech decoder DC from the digital demodulated signal DDS generated speech signal or in a generalized form - the audio signal AC. This AC audio signal is supplied to the PM software module for controlling the bass reproduction of audio signals in the electro-acoustic transducers of the digital DSP signal converter. In the PM software module of the digital DSP signal converter, a modified MAC audio signal is generated from the AC audio signal, which is then filtered in the PL filter of the digital DSP signal processor. The filtered modified MAC audio signal is then fed to the digital-to-analog DAC converter and amplified in the op amp amplifier, after which the speech information contained in the modified MAC audio signal is output by the EAC electro-acoustic transducer, which is preferably made in the form of a loudspeaker.

FIG. 3 shows a second embodiment in the form of a functional diagram or block diagram of a voice signal processing path in a radio communication device UR, in which the invention is implemented, in contrast to the digital DSP signal processor shown in FIG. 2, in the analog part of the UE radio communication device in a device for controlling the playback of bass audio signals in electroacoustic transducers EAP (analog implementation). The processing of the speech signal in the radio communication device US again begins with the fact that the analogue radio signal PC, which is modulated by coded speech information, is received by the ANT antenna and fed to the PR receiver. In the PC receiver, using the MP microprocessor and the analog-to-digital converter of the ADC, a digital demodulated DDS signal is generated from the modulated encoded analog radio signal PC. This digital demodulated DDS signal is then supplied to the RD speech decoder in the digital DSP signal processor. In the speech decoder DC from the digital demodulated signal DDS generated speech signal or in a generalized form - the audio signal AC. This AC audio signal is supplied to the FL filter of the digital signal processor of the CAC. The filtered modified MAC audio signal is then converted to a digital-to-analog D / A converter. The transformed AC audio signal is supplied to the control unit for the bass reproduction of audio signals in electro-acoustic transducers, where a modified MAC audio signal is generated from the AC audio signal. The modified MAC audio signal is amplified in the OA terminal amplifier, after which the speech information contained in the modified MAC audio signal is output through the EAC electro-acoustic transducer, which is preferably made in the form of a loudspeaker.

FIG. 4 shows a first embodiment of the software module of FIG. 2. The audio signal AC to isolate the first frequency component of the emergency is filtered by the band in the bandpass filter PF, implemented using software, and to extract the second frequency component of the emergency 'is subjected to low-pass filtering in the low-pass filter of the low-pass filter, implemented using software. While the first frequency component of the emergency situation is amplified, with the help of the second frequency component of the emergency situation, a gain is generated that determines the gain of the first frequency component of the emergency situation.

Instead of a low-pass filter, the low-pass filter can be used additional, also implemented using software band-pass filter or even a band-pass filter PF, generating the first frequency component of the emergency. In the latter case, both frequency components of the emergency, emergency 'would be the same (emergency = emergency').

The PF bandpass filter is preferably made in the form of a filter with a finite impulse response KIO-F or, alternatively, in the form of a filter with an infinite impulse response BIO-F. If the bandpass filter PF is made in the form of a filter with a finite impulse response KIO-F, then the software module PM for buffering the audio signal AC contains an intermediate memory PP. This intermediate memory PP in the case if the bandpass filter is made in the form of a filter with an infinite impulse response BIO-F, is not required. To illustrate this in FIG. 4, the intermediate memory of the software is presented as a dotted block.

The AC audio signal filtered in the band or the frequency component of the EM extracted by the PF bandpass filter is fed to the input of the US amplifier with an adjustable gain KU realized using software for amplification. To determine the gain of KU in the PM software module, there is software-implemented means for calculating the signal envelope and / or signal energy (SRES), which from the low-pass filtered audio signal PS 'generates an input value for the means for calculating the coefficient realized with the software gain SRKU software module PM. The means for calculating the gain of the SRKU gives the gain of the KU used to regulate the amplifier US. At the output of the amplifier, there is, therefore, an amplifier signal, amplified with a gain of KU, that is filtered in a band. This amplifier amplified with a gain factor KU, the UHFSC audio signal and the AU audio signal, which, if necessary, was stored in the intermediate memory, are then combined or summed by means of the SS summation means of the PM software module, made preferably as an adder implemented in software. As a result of this operation, a modified MAC audio signal is generated, which is preferably filtered out using a UKF filter implemented with software to improve signal quality. But it is also possible that the modified MAC audio signal, as explained in the description of FIG. 2, is supplied to the PL filter without additional filtering by a signal quality-improving filter UKF.

Figure 5 presents, based on the variant of figure 4, the second embodiment of the software module PM in figure 2. The audio signal AC to isolate the first frequency component of the emergency is filtered by the band in the bandpass filter PF, and to extract the second frequency component of the emergency 'is subjected to low-pass filtering in the low-pass filter of the low-pass filter. While the first frequency component of the emergency situation is amplified, with the help of the second frequency component of the emergency situation, a gain is generated that determines the gain of the first frequency component of the emergency situation.

Instead of the low-pass filter of the low-pass filter, an additional band-pass filter or even a PF band-pass filter can be used, which produces the first frequency component of the emergency. In the latter case, both frequency components of the emergency, emergency 'would be the same (emergency = emergency').

The PF bandpass filter is preferably made in the form of a filter with a finite impulse response KIO-F or, alternatively, in the form of a filter with an infinite impulse response BIO-F. If the bandpass filter PF is made in the form of a filter with a finite impulse response KIO-F, then the software module PM for buffering the audio signal AC contains an intermediate memory PP. This intermediate memory PP in the case if the bandpass filter is made in the form of a filter with an infinite impulse response BIO-F, is not required. To show this in figure 5, the intermediate memory PP is presented in the form of a dotted block.

The AC audio signal filtered in the band or the frequency component of the EM extracted by the PF bandpass filter, as in FIG. 4, is fed to the input of the US amplifier with an adjustable gain KU for amplification. To determine the gain of KU in the PM software module, there is a tool for calculating the envelope of the signal and / or energy of the SRES signal, which from the low-pass filtered audio signal ES 'generates an input value for means for calculating the gain of the SRKU of the PM software module.

In the embodiment of the PM software module of FIG. 5, in contrast to the embodiment of FIG. 4, another input quantity is supplied to the means for calculating the gain of the SRKU, which comes from additional means for calculating the envelope of the signal and / or energy of the SRES signal. This additional input quantity is calculated by the indicated means for calculating the SRES from the unfiltered audio signal of the speaker.

In this case, the means for calculating the gain of the SRKU from both of these input values produces the gain of the KU used to regulate the amplifier US. At the output of the amplifier, there is, therefore, an amplifier signal, amplified with a gain of KU, that is filtered in a band. This amplifier amplified with a gain factor KU, the PSD audio signal and the AS audio signal, which, if necessary, was stored in the intermediate memory, are then combined or summed using the SS program module PM, which is preferably used as an adder. As a result of this operation, a modified MAC audio signal is generated, which is preferably filtered out using a UKF filter implemented with software to improve signal quality. But it is also possible that the modified MAC audio signal, as explained in the description of FIG. 2, is supplied to the PL filter without additional filtering by a signal quality-improving filter UKF.

Figure 6 presents, based on the variant of figure 4, the third embodiment of the software module PM in figure 2. The audio signal AC to isolate the first frequency component of the emergency again is filtered by the band in the bandpass filter PF, and to extract the second frequency component of the emergency 'is again subjected to low-pass filtering in the low-pass filter of the low-pass filter. While the first frequency component of the emergency is amplified, with the help of the second frequency component of the emergency ', a gain is again generated that determines the amplification of the first frequency component of the emergency.

Instead of a low-pass filter, the low-pass filter can again use an additional band-pass filter or even a PF band-pass filter that produces the first frequency component of the emergency. In the latter case, both frequency components of the emergency, emergency 'would be the same (emergency = emergency').

The PF bandpass filter is preferably again made in the form of a filter with a finite impulse response KIO-F or, alternatively, in the form of a filter with an infinite impulse response BIO-F. If the bandpass filter PF is made in the form of a filter with a finite impulse response KIO-F, then the program module PM for buffering the audio signal AC again contains the intermediate memory PP. This intermediate memory PP in the case if the bandpass filter is made in the form of a filter with an infinite impulse response BIO-F, is not required. To show this in Fig.6, the intermediate memory PP is presented in the form of a dotted block.

The AC audio signal filtered in the band or the frequency component of the EM extracted by the PF bandpass filter, as in FIGS. 4 and 5, is fed to the input of the US amplifier with an adjustable gain KU for amplification. To determine the gain of KU in the PM software module, there is again a tool for calculating the envelope of the signal and / or energy of the SRES signal, which from the low-pass filtered audio signal PS 'generates an input value for means for calculating the gain of the SRKU of the PM software module.

In the embodiment of the PM software module of FIG. 6, in contrast to the embodiment of FIG. 4, another input quantity is supplied to the means for calculating the gain of the SRKU, which comes from additional means for calculating the envelope of the signal and / or energy of the SRES signal. This additional input quantity, in contrast to the variant of FIG. 5, is calculated by the indicated means for calculating the SRES from the speaker filtered out in the band of the audio signal.

In this case, the means for calculating the gain of the SRKU from both of these input values produces the gain of the KU used to regulate the amplifier US. At the output of the amplifier, there is, therefore, an amplifier signal, amplified with a gain of KU, that is filtered in a band. This amplifier amplified with a gain factor KU, the PSD audio signal and the AS audio signal, which, if necessary, was stored in the intermediate memory, are then combined or summed using the SS program module PM, which is preferably used as an adder. As a result of this operation, a modified MAC audio signal reappears, which is preferably filtered again using a UKF filter to improve signal quality. But it is also possible that the modified MAC audio signal, as explained in the description of FIG. 2, is supplied to the PL filter without additional filtering by a signal quality-improving filter UKF.

Figure 7 presents an embodiment of the control device of the UU of figure 3. The audio signal AC to isolate the first frequency component of the emergency using the hardware-implemented bandpass filter PF1 is filtered in the band, and to extract the second frequency component of the emergency 'is subjected to low-pass filtering using the low-pass filter LPF1 implemented by the hardware. While the first frequency component of the emergency is amplified, with the help of the second frequency component of the emergency ', a gain is again generated that determines the amplification of the first frequency component of the emergency.

Instead of the low-pass filter, the low-pass filter can again use an additional hardware-implemented bandpass filter or even a PF1 bandpass filter that produces the first frequency component of the emergency. In the latter case, both frequency components of the emergency, emergency 'would be the same (emergency = emergency').

The AC audio signal filtered in the band or the frequency component of the emergency signal selected by the PF bandpass filter for amplification is fed to the input of a US1 amplifier with adjustable gain KU, which is implemented in hardware. To determine the gain of the control unit, the control unit of the control unit has hardware-made means for calculating the envelope of the signal and / or energy of the SRES signal, which is preferably made in the form of a series connection of a VLF rectifier and another low-pass filter LPF2 and which is subjected to low-frequency filtering of the audio signal PS ' generates an input value for also made in the form of hardware for calculating the gain SRKU1 control device UU1. At the output of the amplifier, there is, therefore, an amplifier signal, amplified with a gain of KU, that is filtered in a band. This amplified with a gain KU band-filtered audio signal USHF and audio AC are then combined or summed using a preferably made in the form of an adder, implemented as hardware for summing the CC control device UU. As a result of this operation, a modified MAC audio signal is generated, which is preferably filtered using a UKF filter to improve signal quality. But it is also possible that the modified MAC audio signal, as explained in the description of FIG. 3, is supplied to the PL filter without additional filtering by a signal quality-improving filter UKF.

Claims (11)

1. The method of controlling the reproduction of bass audio signals in electro-acoustic transducers, namely, that a) isolate the frequency components (ES, ES ') of the audio signal (AS) and amplify (US, US1) with a gain (KU) calculated on the basis of the audio signal (AS), b) combine (CC, CC1) the amplified frequency components (PSD) of the audio signal (AC) and the audio signal (AC) so as to obtain a modified audio signal (MAC), c) apply the modified audio signal (MAC) to the electro-acoustic transducer (EAP), characterized in that d) carry out band-pass filtering (PF, PF1) of the audio signal (AC) to extract and amplify the first frequency component (ES), e1) carry out low-pass filtering and / or band-pass filtering (PF, PF1, low-pass filter, low-pass filter 1) of the audio signal (AC) to highlight the second frequency component (ES '), e2) calculate (SRES, SRES1) the envelope and / or energy of unfiltered, subjected to low-pass filtering and / or band-pass filtering of the audio signal (AC, ES '). 2. The method according to claim 1, characterized in that the band-pass filtering is carried out using a filter with a finite impulse response (KIO-F). 3. The method according to claim 1, characterized in that the band-pass filtering is carried out using a filter with an infinite impulse response (BIO-F). 4. The method according to claim 2, characterized in that the audio signal (AC) to be combined with the amplified frequency component (USHF) is buffered (PP). 5. The method according to any one of claims 1 to 4, characterized in that the band-pass filtering for isolating and amplifying the frequency components and for calculating the gain is performed using one band-pass filter (PF, PF1). 6. The method according to any one of claims 1 to 4, characterized in that the band-pass filtering for separation and amplification of the frequency components is performed using one band-pass filter (PF, PF1), and the band-pass filtering for calculating the gain is performed using another band-pass filter. 7. The method according to claim 1, characterized in that the filtered (PF, PF1) modified audio signal (MAC) to amplify the selected frequencies. 8. The method according to claim 1, characterized in that the said method is performed in an electronic device for issuing and / or reproducing audio signals. 9. A device for controlling the reproduction of bass audio signals in electro-acoustic transducers, containing a) extraction means (PF, PF1, LPF, LPF1), to the input of which an audio signal (AS) is supplied and which provide the allocation of frequency components (ES, ES ') of the audio signal (AS), b) means for calculating (SRKU, SRKU1), which on the basis of the audio signal (AC) calculate the gain (KU), c) an amplifier (US), which is associated with the means of selection and with the means of calculation, so that the frequency components (emergency, emergency ') of the audio signal (AC) are amplified with the calculated gain (KU), d) combining means (SS, CC1), to the input of which an audio signal (AS) and amplified frequency components (USCH) of an audio signal (AU) are supplied and which combine an audio signal (AS) and amplified frequency components (AES) of an audio signal (AS) in such a way that at the output of the combining means (CC, CC1) there is a modified audio signal (MAC) intended for supply to an electro-acoustic transducer (EAP), characterized in that e) contains at least one band-pass filter (PF, PF1) or, respectively, at least one band-pass filter (PF, PF1) and a low-pass filter (low-pass filter, low-pass filter 1) to select the first frequency component (PS) and the second frequency component (ES) of the audio signal (AC). f) from the said band-pass filters (PF, PF1), the band-pass filter for isolating the first frequency component (ES) from the output side is connected to an amplifier (US, US1), g) contains means for calculating the envelope of the signal and / or energy of the signal (SRES, SRES1), the input of which is unfiltered, subjected to low-pass filtering and / or band-pass filtering of the audio signal (AC, ES '), h) the mentioned calculation tools (SRKU, SRKU1) for calculating the gain (KU) on the input side are connected to means for calculating the signal envelope and / or signal energy (SRES, SRES1), and on the output side to set the gain (KU) - with an amplifier (CSS, CSS1). 10. The device according to claim 9, characterized in that it contains a filter for improving signal quality (UKF, UKF1), designed to amplify the frequencies of the modified audio signal (MAC). 11. The device according to claim 9 or 10, characterized in that the said device is integrated or contained in an electronic device for issuing or reproducing audio signals.

Images