Abstract:
For a left-channel input audio signal, digital filters  30 LL and  30 LR are provided for reproduction of impulse responses used for locating an acoustic image outside the head of a listener, and for a right-channel input audio signal, digital filters  30 RL and  30 RR are provided for reproduction of impulse responses for locating an acoustic image outside the listener&#39;s head. An addition circuit  7 L adds output signals of the digital filters  30 LL and  30 RL to each other and a second addition circuit  7 R adds output signals of the digital filters  30 LR and  30 RR to each other. Output signals of the addition circuits  7 L and  7 R are supplied to a headphone, whereby the reproduction acoustic images of the left-channel and right-channel input audio signals are located outside the listener&#39;s head. The sampling rate in each of the digital filters  30 LL- 30 RR is changed depending on the section of the response time of the associated impulse response to be reproduced.

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     The present invention relates to a digital signal processing circuit and an audio reproducing device using it such as a headphone device, an earphone device, and a speaker device. 
     2. Description of the Related Art 
     When music or the like is reproduced by supplying an audio signal to a speaker that is located in front of a listener, an acoustic image is located in front of the listener. Even for the same audio signal, its reproduction acoustic image is located inside the head of a listener when reproduction is performed by supplying the audio signal to a headphone; the location of the acoustic image is very unnatural. 
     In view of the above, a headphone device has been developed that locates the acoustic image of an audio signal outside the head. 
     FIG. 1 shows an example of such a headphone device. An analog audio signal SA is supplied to an A/D converter circuit  2  via an input terminal  1  and thereby A/D-converted to a digital audio signal SD. The digital audio signal SD is supplied to digital signal processing circuits  3 L and  3 R, which perform processing for locating an acoustic image outside the head. 
     For example, as shown in FIG. 2, in a case where a sound source SP is located in front of a listener M, a sound that is output from the sound source SP reaches the left ear and the right ear of the listener M along paths represented by transfer functions HL and HR, respectively. 
     The digital signal processing circuits  3 L and  3 R calculate the convolutions of the signal SD and impulse responses obtained by converting the transfer functions HL and HR to time domain signals, respectively. The impulse response can be determined in advance by measurement or calculation. 
     Signals SFL and SFR obtained by the signal processing are supplied to D/A converter circuits  4 L and  4 R and thereby D/A-converted to analog audio signals SL and SR, which are supplied to left and right acoustic units (electro-acoustic conversion elements)  6 L and  6 R of a headphone  6  via headphone amplifiers  5 L and  5 R, respectively. 
     Therefore, when the listener M listens to a reproduction sound of the headphone  6  mounted on his head, the reproduction sound of the headphone  6  is equivalent to a reproduction sound of a signal that has traveled along the paths having the transfer functions HL and HR and hence a state that the acoustic image SP of the reproduction sound is located outside the head of the listener M as shown in FIG. 2 is realized. 
     To realize the transfer functions HL and HR, each of the digital signal processing circuits  3 L and  3 R is given a FIR filter configuration as shown in FIG. 3, for example. As shown in FIG. 3, the signal SD that is output from the A/D converter circuit  2  is supplied to a plurality of cascade-connected delay circuits  3 D via an input terminal  31 . Signals that are output from the input terminal  31  and the delay circuits  3 D are supplied to respective multiplication circuits  3 M, and their multiplication outputs are added together by addition circuits  3 A and a resulting signal is output from an output terminal  37 . 
     Each of the delay circuits  3 D gives a delay of one sampling period (one unit time) τ to the digital audio signal SD. Each of the multiplication circuits  3 M has, as a coefficient, an impulse response at each sampling time that is obtained by converting the transfer function HL or HR to a time domain signal. 
     In reverberation room spaces such as a general room and a concert hall, reflection sounds and reverberation sounds occur in addition to a direct sound from a sound source and hence there is a tendency that a relatively long time is needed for impulse responses of those kinds to attenuate sufficiently. 
     In view of the above, the digital signal processing circuits  3 L and  3 R shown in FIG. 3 are required to have a large number of taps and hence many, for example, 1,024, pairs of delay circuits  3 D and multiplication circuits  3 M are needed. 
     As a result, for example, in a configuration in which the digital signal processing circuits  3 L and  3 R are formed by a DSP, a large-capacity memory is needed for the delay circuits  3 D and hence the IC scale becomes large and the cost increases greatly. Further, since a large number of process steps are needed to implement many multiplication circuits  3 M, a fast calculation process is required. The cost also increases in this respect. 
     SUMMARY OF THE INVENTION 
     The present invention has been made to solve the above problems in the art, and an object of the invention is therefore to decrease the number of taps of a digital filter used in an audio reproducing device that locates an acoustic image outside the head. 
     To attain the above object, the invention provides a digital signal processing circuit comprising two digital filters for reproducing impulse responses that represent transmission characteristics of a first path from a sound source to a left ear of a listener and a second path from the sound source to a right ear of the listener, respectively, the two digital filters producing output audio signals representing convolutions of an input audio signal and impulse responses obtained by converting transfer functions of the first and second paths to time domain signals, respectively, to thereby allow reproduction of reproduction sounds reflecting the transfer functions of the first and second paths, respectively; and a headphone for receiving the output audio signals produced by the two digital filters, respectively, to thereby locate a reproduction acoustic image of the input audio signal outside a head of the listener, wherein a sampling rate in each of the two digital filters is changed depending on a section of a response time of the associated impulse response. 
     With the above configuration, the number of taps of each digital filter is reduced, resulting in reduction in circuit scale. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a block diagram showing a conventional headphone device; 
     FIG. 2 is a plan view illustrating a relationship between a listener and a sound source located in front of him; 
     FIG. 3 is a block diagram showing an FIR filter as a digital signal processing circuit used in the headphone device of FIG. 1; 
     FIG. 4 is a block diagram showing a headphone device according to an embodiment of the invention; 
     FIG. 5 is a block diagram showing a digital signal processing circuit used in the headphone device of FIG. 4; 
     FIG. 6 is a block diagram showing a headphone device according to another embodiment of the invention; 
     FIG. 7 is a plan view illustrating a relationship between a listener and two sound sources; 
     FIG. 8 is a block diagram showing a speaker device according to a further embodiment of the invention; 
     FIG. 9 is a plan view illustrating how a reproduction acoustic image is located at an arbitrary position; and 
     FIG. 10 is a block diagram showing a digital signal processing circuit according to a modification. 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     In room spaces such as a general room and a concert hall, the reverberation time in a high-frequency range is shorter than in a low-frequency range and high-frequency components generated by a sound source tend to attenuate in a short time. Therefore, where a digital filter is formed by an FIR filter, the number of taps needed to reproduce the frequency characteristic of a transmission system depends on the frequency band; the number of necessary taps decreases as the reproduction frequency band becomes higher, and the former increases as the latter becomes lower. 
     This indicates that more high-frequency components are included in an early portion of an impulse response to be reproduced, and that a late portion of the impulse response can be reproduced with high fidelity even by reproducing only low-frequency components. 
     The present invention greatly decreases the number of taps of a digital signal processing circuit, that is, the number of delay circuits and multiplication circuits, by paying attention to the above point. An embodiment of the invention will be described below. 
     FIG. 4 shows a headphone device according to an embodiment of the invention that makes it possible to locate an acoustic image outside the head. An analog audio signal SA is supplied to an A/D converter circuit  2  via an input terminal  1  and thereby A/D-converted to a digital audio signal SD. The digital audio signal SD is supplied to digital signal processing circuits  30 L and  30 R. The digital signal processing circuits  30 L and  30 R calculate the convolutions of the digital audio signal SD and impulse responses obtained by converting the transfer functions HL and HR to time domain signals, respectively. The impulse response can be determined in advance by measurement or calculation. 
     Signals SFL and SFR produced by the signal processing are supplied to D/A converter circuits  4 L and  4 R and thereby D/A-converted to analog audio signals SL and SR, which are supplied to left and right acoustic units  6 L and  6 R of a headphone  6  via headphone amplifiers  5 L and  5 R, respectively. 
     Therefore, when a listener M (see FIG. 2) listens to a reproduction sound of the headphone  6  mounted on his head, the reproduction sound is equivalent to a reproduction sound of a signal that has traveled along the paths represented by the transfer functions HL and HR and hence the acoustic image of the reproduction sound is located outside the head of the listener M. 
     Further, in this embodiment, each of the digital signal processing circuits  30 L and  30 R is configured as shown in FIG. 5, for example. 
     As shown in FIG. 5, the digital audio signal SD that is output from the A/D converter circuit  2  is supplied to a digital filter  32  via an input terminal  31 . The digital filter  32  reproduces an early portion of an impulse response to be reproduced and outputs a response result as well as a delay result. 
     To this end, the digital audio signal SD coming from the input terminal  31  is supplied to a predetermined number of cascade-connected delay circuits  321  and a signal S 321  is produced. Signals SD that are output from the input terminal  31  and outputs  321  from the delay circuits  3 D are supplied to respective multiplication circuits  322 , and multiplication outputs of the respective multiplication circuits  322  are added together by addition circuits  323  and a resultant signal S 323  is produced. 
     Each of the delay circuits  321  gives a delay of one sampling period (one unit time) τ to the digital audio signal SD. Each of the multiplication circuits  322  has, as a coefficient, an early portion of an impulse response at each sampling time that is obtained by converting the transfer function HL or HR to a time domain signal. For example, where the sampling frequency for the signal SD is 48 kHz, the number of taps of the digital filter  32  is set at  128 . 
     Therefore, the signal S 321  has both of high-frequency components and low-frequency components of the original analog audio signal SA and corresponds to a late portion of the impulse response to be reproduced. The signal S 323  corresponds to an early portion of the impulse response to be reproduced and includes more high-frequency components of the original analog audio signal SA. 
     The signal S 321  that is output from the last delay circuit  321  is supplied to a down-sampling decimation filter  33 , where it is converted to a digital signal S 33  of a sampling rate of 1/n (n: integer greater than or equal to 2), for example, 1/2. That is, a portion of the signal S 321  corresponding to low-frequency components of the original analog audio signal SA is extracted as the digital signal S 33 . 
     The digital signal S 33  is supplied to a digital filter  34 , which is to reproduce a late portion of the impulse response to be reproduced and output a response result. 
     To this end, the digital signal S 33  that is output from the decimation filter  33  is supplied to a predetermined number of cascade-connected delay circuits  341 . The digital signal S 33  and outputs of the delay circuits  341  are supplied to respective multiplication circuits  342 . Multiplication outputs of the respective multiplication circuits  342  are added together by addition circuits  343  and resultantly a digital signal S 34  is produced. 
     Each of the delay circuits  341  gives a delay of one sampling period 2τ (in this case, n=2) to the digital audio signal S 33 . Each of the multiplication circuits  342  has, as a coefficient, an impulse response at each sampling time that is obtained by converting the transfer function HL or HR to a time domain signal. For example, where the sampling frequency for the digital signal S 33  is 24 kHz, the number of taps of the digital filter  34  is set at  448 . 
     Therefore, the digital signal S 34  corresponds to a late portion of the impulse response to be reproduced and includes more low-frequency components of the original analog audio signal SA. 
     The digital signal S 34  is supplied to an interpolation filter  35  and thereby converted to a digital signal S 35  of the same sampling rate as the original digital audio signal SD. The digital signal S 35  is supplied to an addition circuit  36 . The addition circuit S 36  is also supplied with the signal S 323  that is output from the digital filter  32 . An output signal S 36  of the addition circuit  36  is output from an output terminal  37 . For example, the digital signal processing circuits  30 L and  30 R having the above configuration can be formed by a DSP. 
     With the above configuration, the convolution of the analog audio signal SA (mainly high-frequency components) and an early portion of the impulse response obtained by converting the transfer function HL or HR to a time domain signal is calculated by the digital filter  32 . Convolution of the analog audio signal SA (mainly low-frequency components) and a late portion of the impulse response obtained by converting the transfer function HL or HR to a time domain signal is taken by the digital filters  32  and  34 . 
     Since the signal S 323  mainly as high-frequency components and the signal S 35  mainly as low-frequency components are added to each other by the addition circuit  36 , the signal S 36  as an addition output is equivalent to a signal as would be obtained by calculating the convolution of the analog audio signal SA and the impulse response obtained by converting the transfer function HL or HR to a time domain signal. 
     The signal S 36  is taken out as an output of the digital signal processing circuit  30 L or  30 R and supplied to the D/A converter circuit  4 L or  4 R as described in connection with FIG.  4 . Therefore, when the audio signals SL and SR are reproduced by the headphone  6 , the acoustic image of a reproduction sound can be located outside the head of the listener M. 
     As described above, the digital signal processing circuits  30 L and  30 R can perform the signal processing for locating the acoustic image outside the head during reproduction by the headphone  6 . In this signal processing, while the digital filters  32  and  34  perform convolution processing for locating the acoustic image outside the head for low-frequency components of the analog audio signal SA, the sampling rate of the signal in the digital filter  34  is made 1/2, for example, of that of the signal in the digital filter  32 . Therefore, the number of taps of the digital filter  34  can be decreased. 
     As described in connection with FIG. 3, the number of taps of the digital filter constituting the digital signal processing circuit  3 L or  3 R is 1,024, for example. Since the number of taps of the digital filter  32  shown in FIG. 5 is  128 , according to conventional designing the number of taps of the digital filter  34  would be 896 (=1,024−128). 
     However, since the sampling frequency of the signal in the digital filter  34  is made 1/2 of that of the signal in the digital filter  32 , the number of taps of the digital filter  34  can be halved to  448  for signal processing of the same response time. Therefore, the total number of taps of the digital filters  32  and  34  can be decreased to 576 (128+448). 
     The decrease in the number of taps of the digital filter  34  results in reduction in the scale of the digital signal processing circuits  30 L and  30 R. For example, when they are formed by a DSP, the capacity of a memory that constitutes the delay circuits  321  and  341  can be decreased and hence the IC scale can be reduced. This leads to reduction in both cost and power consumption. 
     Further, since locating the acoustic image of a reproduction sound during reproduction by the headphone  6  is realized by using such digital filters, the cost of the headphone device can also be reduced. 
     FIG. 6 shows a case where the invention is applied to a headphone device in which 2-channel stereo audio signals are reproduced by a headphone and the acoustic image of a reproduction sound is located outside the head of a listener who is mounted with the headphone. 
     Left-channel and right-channel analog audio signals SAL and SAR are supplied to A/D converter circuits  2 L and  2 R via input terminals  1 L and  1 R and thereby converted to digital audio signals SDL and SDR, respectively. The digital audio signal SDL is supplied to digital signal processing circuits  30 LL and  30 LR and the digital audio signal SDR is supplied to digital signal processing circuits  30 RL and  30 RR. 
     The digital signal processing circuits  30 LL- 30 RR are configured in the same manner as the digital signal processing circuits  30 L and  30 R shown in FIG.  5 . The digital signal processing circuits  30 LL- 30 RR perform signal processing for locating the acoustic image outside the head on the audio signals SDL and SDR so as to produce, when the audio signals SL and SR are reproduced by a headphone  6 , a reproduction sound field close to that as would be obtained by reproducing the audio signals SAL and SAR by speakers. 
     For example, as shown in FIG. 7, when sound sources SPL and SPR are located at a front-left position and a front-right position with respect to a listener M, respectively, a sound that is output from the sound source SPL reaches the left ear and the right ear of the listener M along paths represented by transfer functions HLL and HLR, respectively, and a sound that is output from the sound source SPR reaches the left ear and the right ear of the listener M along paths represented by transfer functions HRL and HRR, respectively, where 
     HLL: transfer function of the path from the sound source SPL to the left ear of the listener M; 
     HLR: transfer function of the path from the sound source SPL to the right ear of the listener M; 
     HRL: transfer function of the path from the sound source SPR to the left ear of the listener M; and 
     HRR: transfer function of the path from the sound source SPR to the right ear of the listener M. 
     In view of the above, the digital signal processing circuit  30 LL calculates the convolution of the audio signal SDL and an impulse response obtained by converting the transfer function HLL to a time domain signal. The digital signal processing circuit  30 LR calculates the convolution of the audio signal SDL and an impulse response obtained by converting the transfer function HLR to a time domain signal. The digital signal processing circuit  30 RL calculates the convolution of the audio signal SDR and an impulse response obtained by converting the transfer function HRL to a time domain signal. The digital signal processing circuit  30 RR calculates the convolution of the audio signal SDR and an impulse response obtained by converting the transfer function HRR to a time domain signal. 
     Output signals of the digital signal processing circuits  30 LL and  30 RL are supplied to an addition circuit  7 L and thereby added to each other. Output signals of the digital signal processing circuits  30 LR and  30 RR are supplied to an addition circuit  7 R and thereby added to each other. Output signals of the addition circuits  7 L and  7 R are supplied to respective D/A converter circuits  4 L and  4 R and thereby D/A-converted to analog audio signals SL and SR, which are supplied to left and right acoustic units  6 L and  6 L of the headphone  6  via headphone amplifiers  5 L and  5 R, respectively. 
     Therefore, a reproduction sound field approximately equal to that as would be obtained when the audio signals SAL and SAR were supplied to speakers located at the front-left position and the front-right position with respect to the listener M is reproduced by supplying the audio signals SL and SR to the headphone  6 . The acoustic image of a reproduction sound is located outside the head of the listener M. 
     Also in this case, since the digital signal processing circuits  30 LL- 30 RR can be configured in the same manner as shown in FIG. 5, for example, the circuit scale can be reduced, resulting in reduction in both cost and power consumption. 
     Further, since locating the acoustic image of a reproduction sound during reproduction by the headphone  6  is realized by using such digital filters, the cost of the headphone device can also be reduced. 
     FIG. 8 shows a case of locating a reproduction acoustic image at an arbitrary position by using two speakers. 
     As shown in FIG. 8, an analog audio signal SA is supplied, via an input terminal  1 , to an A/D converter circuit  2  and thereby A/D-converted to a digital audio signal SD, which is supplied to digital signal processing circuits  30 L and  30 R. The digital signal processing circuits  30 L and  30 R calculate the convolutions of the digital audio signal SD and impulse responses obtained by converting transfer functions described below to time domain signals, respectively. 
     Signals SFL and SFR produced by the above signal processing are supplied to D/A converter circuits  4 L and  4 R and thereby D/A-converted to analog audio signals SL and SR. The analog audio signals SL and SR are supplied to left-channel and right-channel speakers  9 L and  9 R that are located at a front-left position and a front-right position with respect to a listener via speaker amplifiers  8 L and  8 R, respectively. 
     The digital signal processing circuits  30 L and  30 R perform the following processing. Now, as shown in FIG. 9, assume a case of equivalently reproducing a sound source SPX at an arbitrary position by using sound sources SPL and SPR that are located at a front-left position and a front-right position with respect to a listener M. 
     The sound sources SPL and SPR can be given by 
       SPL ={( HXL×HRR−HXR×HRL )/( HLL×HRR−HLR×HRL )}× SPX   (1) 
     
       
           SPR ={( HXR×HLL−HXL×HLR )/( HLL×HRR−HLR×HRL )}× SPX   (2) 
       
     
     where 
     HLL: transfer function of the path from the sound source SPL to the left ear of the listener M; 
     HLR: transfer function of the path from the sound source SPL to the right ear of the listener M; 
     HRL: transfer function of the path from the sound source SPR to the left ear of the listener M; 
     HRR: transfer function of the path from the sound source SPR to the right ear of the listener M; 
     HXL: transfer function of the path from the sound source SPX to the left ear of the listener M; and 
     HXR: transfer function of the path from the sound source SPX to the right ear of the listener M. 
     Therefore, the reproduction acoustic image of the audio signal SA can be located at the position of the sound source SPX by supplying an input audio signal SXA corresponding to the sound source SPX to a speaker located at the position of the sound source SPL via a filter as an implementation of the transfer function portion of Equation (1) and supplying the input audio signal SXA to a speaker located at the position of the sound source SPR via a filter as an implementation of the transfer function portion of Equation (2). 
     The digital signal processing circuits  30 L and  30 R calculate the convolutions of the received digital audio signal SD and impulse responses obtained by converting transfer functions similar to the transfer function portions of Equations (1) and (2) to a time domain signal, respectively. For example, the digital signal processing circuits  30 L and  30 R are configured as shown in FIG.  5 . 
     The reproduction acoustic image of the analog audio signal SA can thus be located at the position of the sound source SPX. 
     Also in this case, since the digital signal processing circuits  30 L and  30 R can be configured in the same manner as shown in FIG. 5, for example, the circuit scale can be reduced, resulting in reduction in both cost and power consumption. 
     In the embodiment of FIGS. 4 and 5, for example, the delay circuits  321  and  341  and the decimation filter  33  of the digital signal processing circuit  30 L handle the same signal as those of the digital signal processing circuit  30 R. Therefore, as shown in FIG. 10, for example, the delay circuits  321  and  341  and the decimation filter  33  of the digital signal processing circuit  30 L can be commonized with those of the digital signal processing circuit  30 R. 
     For the same reason, in the embodiment of FIGS. 6 and 5, the delay circuits  321  and  341  and the decimation filter  33  of the digital signal processing circuit  30 LL can be commonized with those of the digital signal processing circuit  30 LR and the delay circuits  321  and  341  and the decimation filter  33  of the digital signal processing circuit  30 RL can be commonized with those of the digital signal processing circuit  30 RR. 
     It is noted that the invention can similarly be applied to a case of handling audio signals that are multi-channel stereo signals of four or more channels. 
     As described above, in the invention, while an early portion of an impulse response to be reproduced is processed at a dominant sampling rate, a late portion of the impulse response is processed at a sampling rate that is 1/n of the dominant one in the same response time as the early portion. Therefore, the number of taps of digital filters necessary for the process can be reduced. 
     As a result, the circuit scale of the digital filters can be reduced, which leads to reduction in both cost and power consumption. Further, the cost of a headphone device or a speaker device using such digital filters can also be reduced.