JP2007088568A - Audio unit - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an "audio unit" in which output audio sound can be prevented from becoming unnatural. <P>SOLUTION: The audio unit 100 comprises an SL signal generating section 20 for extracting a component in an L signal having high correlation with an R signal by updating the filter coefficients of an adaptive filter using an adaptive algorithm and generating a surround L signal by subtracting that component from the L signal, an SR signal generating section 30 for extracting a component in the R signal having high correlation with the L signal and generating a surround R signal by subtracting that component from the R signal, an addition section 70 for adding an L signal of a predetermined level to the surround L signal, and section 74 for adding an R signal of a predetermined level to the surround R signal. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to an audio device that generates a surround signal from a two-channel stereo signal.

2. Description of the Related Art Conventionally, an audio apparatus that generates a surround signal from a two-channel stereo signal is known (for example, see Patent Document 1). In this audio apparatus, the surround signals SL and SR are generated by passing the input stereo signals INL and INR through an adaptive decorrelator. For example, this adaptive decorrelator is realized by adaptive signal processing using an FIR filter.
Japanese Unexamined Patent Publication No. 2003-333698 (page 3-6, FIG. 1-13)

  By the way, in the audio device disclosed in Patent Document 1, the surround signals SL and SR are generated based on the two-channel stereo signals. In order to update the filter coefficients of the FIR filter used for generating these surround signals. When the value of the step size parameter μ to be used is small, if the level of the input stereo signal becomes small, the filter coefficient is not updated, and the audio sound reproduced corresponding to the surround signal becomes unnatural. In particular, such a phenomenon is likely to occur when a DSP (digital signal processing device) that performs operations such as FIR filters at a fixed point is used. In order to avoid such an inconvenience, it is necessary to increase the value of the step size parameter μ to some extent. However, if this is done, the update rate of the filter coefficient of the FIR filter becomes too fast, and the input stereo signals INL and INR There is a strong tendency to extract uncorrelated components included, and the degree of separation with respect to these stereo signals INL and INR may become too strong, which in turn causes a problem that the audio sound becomes unnatural. It was.

  The present invention has been created in view of the above points, and an object of the present invention is to provide an audio device capable of preventing an output audio sound from becoming unnatural.

  In order to solve the above-described problem, the audio apparatus of the present invention receives an L signal and an R signal, which are two-channel stereo signals, and extracts a component having a high correlation with the L signal in the R signal. A first surround signal generating means for generating a first surround signal by subtracting from the signal, and extracting a component having a high correlation with the R signal in the L signal and subtracting the second surround signal from the R signal. Second surround signal generation means for generating, and the first surround signal generation means correlates with the L signal in the R signal by updating the filter coefficient of the adaptive filter using an adaptive algorithm. The second surround signal generating means updates the filter coefficient of the adaptive filter using an adaptive algorithm, and thereby the R signal in the L signal is extracted. First correction means for extracting a component having a high correlation with the signal and adding an L signal of a predetermined level to the first surround signal output from the first surround signal generation means; and a second surround signal Second correction means for adding an R signal of a predetermined level to the second surround signal output from the generation means. As a result, the tendency that the first and second surround signal generating means extract a component uncorrelated with the L signal or the R signal becomes strong, and the audio sound becomes unnatural. This can be prevented by adding the R signal.

  Further, the ratio of adding the L signal or the R signal by the first and second correction means described above is as follows: the weighting coefficient corresponding to the L signal and the R signal is a, and the weighting coefficient corresponding to the first and second surround signals It is desirable to satisfy the relationship of a <b, where b is b. By adding the L signal or the R signal so as to satisfy such a relationship, it is possible to prevent the degree of separation from becoming too strong for the L signal or the R signal while ensuring the effect of the surround signal. .

  Further, the first level adjusting means for adjusting the level of the output signal of the first correcting means and the levels of the first surround signal and the L signal, and the level of the output signal of the second correcting means, It is desirable to further include second level adjusting means for adjusting each level of the second surround signal and the R signal. Thereby, it is possible to prevent the signal level from fluctuating by adding the L signal or the R signal.

  Further, the first and second surround signal generating means described above are preferably performed by fixed point arithmetic by a DSP. When fixed-point arithmetic is performed, the update of the filter coefficient of the adaptive filter in the first and second surround signal generation means tends to stop and the audio sound using the surround signal may become unnatural. By adding the L signal or the R signal to the signal, it is possible to improve the unnatural audio sound.

  A third surround signal generating means for generating, as a third surround signal, a signal obtained by averaging the first and second surround signals output from the first and second surround signal generating means; It is desirable to further include third correction means for adding an L signal and an R signal of a predetermined level to the third surround signal output from the three surround signal generation means. As a result, the audio sound becomes unnatural even when a third surround signal (for example, a surround signal corresponding to the back center) is generated using the first and second surround signals to output the audio sound. Can be prevented.

  In addition, it is preferable that a plurality of sets of the first and second correction units described above are provided, and the ratio of adding the L signal or the R signal of a predetermined level in the first and second correction units is different for each group. Accordingly, even when two or more sets of surround signals (for example, surround signals corresponding to front L, R and back L, R) are generated using the first and second surround signals, audio sound is output. It is possible to prevent the audio sound from becoming unnatural.

  An audio apparatus according to an embodiment to which the present invention is applied will be described below with reference to the drawings.

[First Embodiment]
FIG. 1 is a diagram illustrating the configuration of the audio apparatus according to the first embodiment. An audio apparatus 100 shown in FIG. 1 is mounted on a vehicle and includes an audio signal generation unit 200, amplifiers 71, 72, 75, 76, 81, 82, and 83, and addition units 70, 74, and 80. Seven (6.1 ch) speakers 110, 112, 114, 120, 122, 130, 132 are connected to the audio device 100. The operation of each component included in the audio apparatus 100 is performed by arithmetic processing using a DSP (Digital Signal Processor).

  The audio signal generator 200 includes adders 10, 13, 17, amplifiers 11, 12, 14, 15, 18, 19, LPF (low-pass filter) 16, SL signal generator 20, SR signal generator 30, and delay circuit 26. , 36.

  The adder 10 adds input stereo signals (L signal, R signal). The amplifiers 11 and 12 provided in the preceding stage of the adding unit 10 are for gain adjustment, and are input to the adding unit 10 after the gain adjustment of each of the L signal and the R signal is performed by 50% distribution. The added signal is output from the speaker 110 as a center speaker installed in front of the listener. In front of the speaker 110, a digital-analog converter that converts digital audio data into an analog signal, an amplifier that drives the speaker 110 based on the analog signal output from the digital-analog converter, and the like are actually provided. However, these configurations are omitted in FIG. The same applies to the other speakers.

  Similar to the adding unit 10, the adding unit 13 adds the stereo signals after gain adjustment by the amplifiers 14 and 15. The added signal is output from the speaker 112 as a subwoofer installed in the lower part of the listener's seat after extracting the low frequency component through the LPF 16. In this embodiment, the stereo signals are simply added and output from the speaker 110. However, the method for generating the signal output from the speaker 110 is not limited to this, and other methods may be used.

  The SL signal generation unit 20 generates a surround L signal (SL signal) based on the input L signal and R signal, and outputs the surround L signal (SL signal) from the speaker 130 installed on the left side of the listener. The SR signal generation unit 30 generates a surround R signal (SR signal) based on the input L signal and R signal and outputs the surround R signal (SR signal) from the speaker 132 installed on the right side of the listener. The generation of the surround L signal and the surround R signal described above is performed by updating the filter coefficient value of the adaptive filter using the LMS algorithm.

  Note that the L signal of the input stereo signal passes through the delay circuit 26 and is then directly output from the speaker 120 installed in the left front of the listener. Further, the R signal in the input stereo signal passes through the delay circuit 36 and is directly output from the speaker 122 installed on the right front side of the listener.

  FIG. 2 is a diagram illustrating a detailed configuration of the SL signal generation unit 20 and the SR signal generation unit 30. As shown in FIG. 2, the SL signal generation unit 20 includes an FIR filter 21, an adaptive filter (ADF) 22, an addition unit 23, and an LMS algorithm processing unit 24. The FIR filter 21 is used as a delay circuit (delay means), and delays an input L signal by a time corresponding to the number of taps (for example, in the case of 32 taps, a time corresponding to half of 16 taps). Output. The adaptive filter 22 has the same configuration as the FIR filter, and multiplies the input R signal by a predetermined tap coefficient W and outputs the result. The adding unit 23 is an adding unit, and subtracts the signal output from the adaptive filter 22 from the L signal output from the FIR filter 21 to output an error signal e. The LMS algorithm processing unit 24 is LMS algorithm processing means, and varies the filter coefficient of the adaptive filter 22 using the LMS algorithm so that the power of the error signal e output from the adding unit 23 is minimized. Further, the error signal e output from the adder 23 is extracted as it is as a surround L signal (SL signal).

  FIG. 3 is a diagram showing a detailed configuration of the adaptive filter 22. As shown in FIG. 3, the adaptive filter 22 includes a plurality of delay elements 221, a multiplying unit 222 that multiplies a signal held in each delay element 221 by a variable filter coefficient, and each of the multiplying units 222. And an adder 223 for adding outputs. The values of the filter coefficients (multipliers) of the plurality of multipliers 222 are updated by the LMS algorithm processor 24.

  The LMS algorithm processing unit 24 updates the value of the filter coefficient of the adaptive filter 22 so that the power of the error signal e output from the adding unit 23 is minimized, and the adaptive filter 22 receives the input R signal component. The value of the filter coefficient is updated so as to extract a component having a high correlation with the L signal. That is, the LMS algorithm processing unit 24 receives the R signal and the error signal e output from the adding unit 23, and the RMS algorithm is processed by the RMS algorithm by processing the R signal and the error signal e. A filter coefficient update command is output from the processing unit 24 to each multiplication unit 222 in the adaptive filter 22, and the value of the filter coefficient superimposed on the signal held in each delay element 221 is changed.

  In this way, a component having a high correlation with the L signal in the R signal is extracted by the adaptive filter 22, and this component is subtracted from the L signal by the adding unit 23. Therefore, the error signal e output from the adder 23 includes only a component that is not highly correlated with the R signal in the L signal, and this is used as the surround L signal.

  By the way, the LMS algorithm is an algorithm using the instantaneous square error as an evaluation amount, and the LMS algorithm processing unit 24 updates the value of the filter coefficient W according to the following expression.

W (n + 1) = W (n) +2 μ · e (n) · R (n) (1)
Here, μ is a step size parameter. When this value is set larger, the convergence of the filter coefficient W becomes faster. Conversely, when this value is set smaller, the convergence of the filter coefficient W becomes slower.

  As is clear from the equation (1), when the value of the step size parameter μ used to update the filter coefficient W of the adaptive filter 22 is small, the level of the input L signal and R signal becomes small. The filter coefficient W is not updated, and the audio sound reproduced in response to the surround signal becomes unnatural. In particular, when the operation of the SL signal generation unit 20 is performed with a fixed point, such a phenomenon is likely to occur. In order to avoid such inconvenience, it is necessary to increase the value of the step size parameter μ to some extent. However, if this is done, the update speed of the filter coefficient W of the adaptive filter 22 becomes too fast, and the input L signal and R There is a strong tendency to extract uncorrelated components included in the signal, and the degree of separation with respect to the L signal may be too strong. For this reason, in this embodiment, by adding the FL signal having the highest correlation with the input L signal at a predetermined ratio, it is possible to improve the identification of the SL signal that is too separated from the L signal (FL signal). I am trying.

  Specifically, the addition unit 70 adds the SL signal and the FL signal after gain adjustment by the amplifiers 71 and 72. For example, the SL signal is added at a rate of 80% and the FL signal is added at a rate of 20%. Then, the added signal is output from the speaker 130.

  The same applies to the SR signal generation unit 30. That is, the SR signal generation unit 30 includes an FIR filter 31, an adaptive filter (ADF) 32, an addition unit 33, and an LMS algorithm processing unit 34. The FIR filter 31 is used as a delay circuit, and delays and outputs the input R signal by a time corresponding to the number of taps (for example, in the case of 32 taps, a half of the time corresponding to 16 taps). The adaptive filter 32 has the same configuration as the FIR filter, and multiplies the input L signal by a predetermined tap coefficient W and outputs the result. The adder 33 subtracts the signal output from the adaptive filter 32 from the R signal output from the FIR filter 31, and outputs an error signal e. The LMS algorithm processing unit 34 varies the filter coefficient of the adaptive filter 32 using the LMS algorithm so that the power of the error signal e output from the adding unit 33 is minimized. Further, the error signal e output from the adding unit 33 is extracted as it is as a surround R signal (SR signal) and output from the speaker 132.

  The LMS algorithm processing unit 34 updates the value of the filter coefficient of the adaptive filter 32 so that the power of the error signal e output from the adding unit 33 is minimized. In the adaptive filter 32, the component of the input L signal The value of the filter coefficient is updated so as to extract a component having a high correlation with the R signal. In other words, the LMS algorithm processing unit 34 is supplied with the L signal and the error signal e output from the adding unit 33, and the LMS algorithm is processed by the LMS algorithm. A filter coefficient update command is output from the processing unit 34 to each multiplication unit in the adaptive filter 32, and the value of the filter coefficient superimposed on the signal held in each delay element is changed.

  Thus, the adaptive filter 32 extracts a component having a high correlation with the R signal in the L signal, and this component is subtracted from the L signal by the adding unit 33. Therefore, the error signal e output from the adder 33 includes only a component that is not highly correlated with the L signal in the R signal, and this is used as the surround R signal.

  By the way, the LMS algorithm is an algorithm using the instantaneous square error as an evaluation amount, and the LMS algorithm processing unit 34 updates the value of the filter coefficient W according to the following equation.

W (n + 1) = W (n) +2 μ · e (n) · L (n) (2)
Here, μ is a step size parameter. When this value is set larger, the convergence of the filter coefficient W becomes faster. Conversely, when this value is set smaller, the convergence of the filter coefficient W becomes slower.

  As is clear from the equation (2), when the value of the step size parameter μ used to update the filter coefficient W of the adaptive filter 32 is small, the level of the input L signal and R signal is small. The filter coefficient W is not updated, and the audio sound reproduced in response to the surround signal becomes unnatural. In particular, such a phenomenon is likely to occur when the operation of the SR signal generator 30 is performed with a fixed point. In order to avoid such an inconvenience, it is necessary to increase the value of the step size parameter μ to some extent. However, if this is done, the update speed of the filter coefficient W of the adaptive filter 32 becomes too fast, and the input L signal and R There is a strong tendency to extract uncorrelated components included in the signal, and the degree of separation with respect to the R signal may be too strong. For this reason, in this embodiment, by adding the FL signal having the highest correlation with the input L signal at a predetermined ratio, the specific improvement of the SR signal that is too strongly separated from the R signal (FR signal) is improved. I am trying.

  Specifically, the adder 74 adds the SR signal and the FR signal after gain adjustment by the amplifiers 75 and 76. For example, the SR signal is added at a rate of 80% and the FR signal is added at a rate of 20%. Then, the signal after addition is output from the speaker 132.

  The adding unit 17 shown in FIG. 1 adds the SL signal and the SR signal after gain adjustment by the amplifiers 18 and 19. The added signal (BC signal) is output from a speaker 114 serving as a back center speaker installed behind the listener. By the way, the BC signal generated by using the SL signal and the SR signal in this way is the same in the tendency that the separation described for the SL signal and the SR signal is too strong. For this reason, in this embodiment, the BC signal, the FL signal, and the FR signal after gain adjustment by the amplifiers 81, 82, and 83 are added by the adding unit 80. For example, the BC signal is added at a rate of 90%, and the FL signal and the FR signal are added at a rate of 5%. Then, the signal after addition is output from the speaker 114.

  The delay circuit 26 is for adjusting a delay time generated by passing through the SL signal generation unit 20, and for example, when a delay of 16 taps is generated by the SL signal generation unit 20, the same delay time is set. Is done. Similarly, the delay circuit 36 is for adjusting a delay time generated by passing through the SR signal generation unit 30. For example, when a delay of 16 taps is generated by the SR signal generation unit 30, the same delay is used. Time is set. Instead of these delay circuits 26 and 36, delay circuits are individually inserted before or after the amplifiers 71, 72, 75, 76, 81 to 83, and the SL signal generation unit 20 or the SR signal generation unit. The delay time generated by 30 may be adjusted.

  The SL signal generator 20 described above is the first surround signal generator, the SR signal generator 30 is the second surround signal generator, the adder 70 is the first corrector, and the adder 74 is the second surround. As the correction means, the amplifiers 71 and 72 are the first level adjustment means, the amplifiers 75 and 76 are the second level adjustment means, the adder 17 and the amplifiers 18 and 19 are the third surround signal generation means, and the adder Reference numeral 80 corresponds to the third correction means.

  In this way, the SL signal generation unit 20 and the SR signal generation unit 30 tend to extract components that are not correlated with the L signal or the R signal, and the audio sound becomes unnatural. This can be prevented by adding signals or FR signals. In particular, when the SL signal generation unit 20 and the SR signal generation unit 30 are performed by fixed-point arithmetic using a DSP, the update of the filter coefficient W of the adaptive filters 22 and 32 is easily stopped, and the SL signal and the SR signal are used. Although the audio sound may become unnatural, it can be improved by adding the FL signal or the FR signal to the SL signal or the SR signal.

  In addition, by setting the ratio of adding the FL signal or FR signal to a predetermined value or less (for example, 20%), the degree of separation becomes too strong with respect to the FL signal or FR signal while ensuring the effect of the surround signal. Can be prevented. Note that the ratio of adding the FL signal or FR signal to the SL signal or SR signal is as follows: a is a weighting coefficient corresponding to the FL signal and FR signal, and b is a weighting coefficient corresponding to the SL signal and SR signal. By satisfying the relationship of a <b, it is possible to prevent the degree of separation from becoming too strong with respect to the FL signal or the FR signal while ensuring the effect of the surround signal.

  In addition, by providing the amplifier 71 and the like in the previous stage of the addition unit 70 and the like to perform signal level adjustment, it is possible to prevent the signal level from fluctuating by adding the FL signal or the FR signal. The level adjustment may be performed in the subsequent stage of the addition unit 70 and the like.

  Also, when a third surround signal (for example, a surround signal corresponding to the back center) is generated using the SL signal and the SR signal to output an audio sound, the FL signal and the FR signal at a predetermined level are added. Therefore, it is possible to prevent the audio sound from becoming unnatural.

[Second Embodiment]
FIG. 4 is a diagram illustrating the configuration of the audio apparatus according to the second embodiment. The audio device 100A shown in FIG. 4 has a configuration that generates a BC signal (amplifiers 18, 19, 81, 82, and 83, and addition units 17 and 80) in the audio device 100 shown in FIG. The difference is that an adder 90 and amplifiers 91 and 92 for generating an L signal (BL signal) and an adder 94 and amplifiers 95 and 96 for generating a back R signal (BR signal) are added.

  When the SL signal and the BL signal are compared, the degree of correlation with the L signal (FL signal) is much weaker in the BL signal than in the SL signal. Specifically, in order to realize such a relationship, the adder 90 adds the SL signal and the FL signal after gain adjustment by the amplifiers 91 and 92. For example, the SL signal is added at a rate of 90% and the FL signal is added at a rate of 10%. Then, the signal after addition is output from the speaker 140.

  Similarly, when the SR signal and the BR signal are compared, the degree of correlation with the R signal (FR signal) is even weaker in the BR signal than in the SR signal. Specifically, in order to realize such a relationship, the adder 94 adds the SR signal and the FR signal after gain adjustment by the amplifiers 95 and 96. For example, the SR signal is added at a rate of 90% and the FR signal is added at a rate of 10%. Then, the signal after addition is output from the speaker 142.

  As described above, even when two or more sets of surround signals are generated using the SL signal and the SR signal to output the audio sound, it is possible to prevent the audio sound from becoming unnatural. Further, by changing the ratio of the FL signal and the FR signal of a predetermined level to be added in each group, it becomes easy to expand to a surround system having a larger number of speakers.

  In addition, this invention is not limited to the said embodiment, A various deformation | transformation implementation is possible within the range of the summary of this invention. In the first embodiment described above, the case where a 6.1 ch surround system is configured by applying the present invention has been described. However, when a 5.1 ch surround system is configured, a back center configuration (addition) is used. The units 17, 80, the amplifiers 18, 19, 81 to 83, and the speaker 114) may be deleted.

  In the second embodiment described above, two sets of surround sounds (a set of SL signal, SR signal and a set of BL signal and BR signal) are generated. However, three or more sets of surround sounds are generated. Also good. In this case, in order to generate the surround sound of the set to be added, a necessary number of configurations combining the adder 90 and the amplifiers 91 and 92 are added, and the gain adjustment value for each amplifier is different from the other sets. You can make it.

It is a figure which shows the structure of the audio apparatus of 1st Embodiment. It is a figure which shows the detailed structure of SL signal generation part and SR signal generation part. It is a figure which shows the detailed structure of an adaptive filter. It is a figure which shows the structure of the audio apparatus of 2nd Embodiment.

Explanation of symbols

10, 13, 17, 70, 74, 80 Adder 11, 12, 14, 15, 18, 19, 71, 72, 75, 76, 81, 82, 83 Amplifier 16 LPF (low pass filter)
20 SL signal generator 26, 36 Delay circuit 30 SR signal generator 100 Audio device 110, 112, 114, 120, 122, 130, 132 Speaker

Claims (6)

The L signal and the R signal, which are two-channel stereo signals, are input, and a first surround signal is generated by extracting a component highly correlated with the L signal from the R signal and subtracting it from the L signal. First surround signal generation means; and second surround signal generation means for generating a second surround signal by extracting a component highly correlated with the R signal in the L signal and subtracting the component from the R signal. In an audio device having
The first surround signal generation means extracts a component highly correlated with the L signal in the R signal by updating a filter coefficient of an adaptive filter using an adaptive algorithm,
The second surround signal generation means extracts a component highly correlated with the R signal in the L signal by updating a filter coefficient of an adaptive filter using an adaptive algorithm,
First correction means for adding the L signal at a predetermined level to the first surround signal output from the first surround signal generation means;
Second correction means for adding the R signal of a predetermined level to the second surround signal output from the second surround signal generation means;
An audio device comprising: In claim 1,
The ratio of adding the L signal or the R signal by the first and second correction means corresponds to the weighting coefficient corresponding to the L signal and the R signal, and corresponds to the first and second surround signals. An audio apparatus characterized by satisfying a <b when a weighting coefficient is b. In claim 1 or 2,
First level adjusting means for adjusting the level of the output signal of the first correcting means and each level of the first surround signal and the L signal;
Second level adjusting means for adjusting the level of the output signal of the second correcting means and each level of the second surround signal and the R signal;
An audio apparatus further comprising: In any one of Claims 1-3,
The audio apparatus according to claim 1, wherein the first and second surround signal generation means are performed by a fixed-point operation by a DSP. In any one of Claims 1-4,
Third surround signal generation means for generating, as a third surround signal, a signal obtained by averaging the first and second surround signals output from the first and second surround signal generation means;
Third correction means for adding the L signal and the R signal at a predetermined level to the third surround signal output from the third surround signal generation means;
An audio apparatus further comprising: In any one of Claims 1-5,
A plurality of sets of the first and second correction means are provided, and a ratio of adding the L signal or the R signal at a predetermined level in the first and second correction means is made different for each set. Audio device.

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