Patent Publication Number: US-8989405-B2

Title: Audio signal correction apparatus, audio signal correction method, and audio signal correction program

Description:
CROSS-REFERENCE TO RELATED APPLICATIONS 
     This application is based on and claims the benefit of priority from the prior Japanese Patent Application No. 2011-003403 filed on Jan. 11, 2011, the entire content of which is incorporated herein by reference. 
     BACKGROUND OF THE INVENTION 
     The present invention relates to an audio signal correction apparatus, an audio signal correction method, and an audio signal correction program. 
     An impulsive sound (referred to as an attack sound, hereinafter) produced by hitting a percussion instrument, such as a drum, has a sound level that rises steeply and varies instantaneously. When such an attack sound is recorded once and then reproduced through a speaker, it may happen that a speaker cone does not vibrate instantaneously at the timing at which the attack sound was produced, a reproduced audio signal is deteriorated with slow rise-up of a sound level. This may result in that a reproduced sound is heard with a mild tone and slower rise-up of a sound level than an attack sound. 
     The cause of such a phenomenon may be a smaller number of windings of a coil of a speaker, the deformation of a cone of a speaker, a quantization error in digitalization of audio signals, the cut-off of high-frequency components in digital compression of audio signals, etc. 
     SUMMARY OF THE INVENTION 
     A purpose of the present invention is to provide an audio signal correction apparatus, an audio signal correction method, and an audio signal correction program that achieve the correction of an audio signal that involves an attack sound deteriorated due to digitalization or compression into an audio signal close to an original audio signal. 
     The present invention provides an audio signal correction apparatus comprising: a first differential-value acquisition circuit configured to acquire a first differential value between first current input data and first previous input data in an i number (i being a natural number) of sampling periods before the first current input data, both first input data being of a first digital audio signal that has a sound level of a digital stereo audio signal in a left channel; a second differential-value acquisition circuit configured to acquire a second differential value between second current input data and second previous input data in a j number (j being a natural number) of sampling periods before the second current input data, both second input data being of a second digital audio signal that has a sound level of the digital stereo audio signal in a right channel; a correction coefficient acquisition circuit configured to acquire a first correction coefficient by adding the first and second differential values at a first ratio and acquire a second correction coefficient by adding the first and second differential values at a second ratio; and a correction circuit configured to correct the first digital audio signal by multiplying the first digital audio signal by the first correction coefficient and correct the second digital audio signal by multiplying the second digital audio signal by the second correction coefficient. 
     Moreover, the present invention provides an audio signal correction method comprising: a first differential-value acquisition step of acquiring a first differential value between first current input data and first previous input data in an i number (i being a natural number) of sampling periods before the first current input data, both first input data being of a first digital audio signal that has a sound level of a digital stereo audio signal in a left channel; a second differential-value acquisition step of acquiring a second differential value between second current input data and second previous input data in a j number (j being a natural number) of sampling periods before the second current input data, both second input data being of a second digital audio signal that has a sound level of the digital stereo audio signal in a right channel; a correction coefficient acquisition step of acquiring a first correction coefficient by adding the first and second differential values at a first ratio and acquiring a second correction coefficient by adding the first and second differential values at a second ratio; and a correction step of correcting the first digital audio signal by multiplying the first digital audio signal by the first correction coefficient and correcting the second digital audio signal by multiplying the second digital audio signal by the second correction coefficient. 
     Furthermore, the present invention provides an audio signal correction program stored in a non-transitory computer readable device, the program comprising: a first differential-value acquisition program code of acquiring a first differential value between first current input data and first previous input data in an i number (i being a natural number) of sampling periods before the first current input data, both first input data being of a first digital audio signal that has a sound level of a digital stereo audio signal in a left channel; a second differential-value acquisition program code of acquiring a second differential value between second current input data and second previous input data in a j number (j being a natural number) of sampling periods before the second current input data, both second input data being of a second digital audio signal that has a sound level of the digital stereo audio signal in a right channel; a correction coefficient acquisition program code of acquiring a first correction coefficient by adding the first and second differential values at a first ratio and acquiring a second correction coefficient by adding the first and second differential values at a second ratio; and a correction program code of correcting the first digital audio signal by multiplying the first digital audio signal by the first correction coefficient and correcting the second digital audio signal by multiplying the second digital audio signal by the second correction coefficient. 
    
    
     
       BRIEF DESCRIPTION OF DRAWINGS 
         FIG. 1  is a block diagram of an audio reproduction apparatus according to an embodiment of the present invention; 
         FIG. 2  is an exemplary block diagram of a DSP of the audio reproduction apparatus shown in  FIG. 1 ; 
         FIG. 3  is a view for explaining an attack-sound emphasizing function of the audio reproduction apparatus shown in  FIG. 1 ; 
         FIG. 4  is an exemplary view of an audio signal output from a decoder of the audio reproduction apparatus shown in  FIG. 1 ; 
         FIG. 5  is an exemplary view of an audio signal output from a DSP of the audio reproduction apparatus shown in  FIG. 1 ; 
         FIG. 6  is a view in which a view of  FIG. 4  is superimposed on that of  FIG. 5 ; 
         FIG. 7  is an exemplary block diagram of a DSP of the audio reproduction apparatus shown in  FIG. 1 ; and 
         FIG. 8  is an exemplary block diagram of circuitry for setting a time constant τ; 
         FIG. 9  is a flow chart explaining an embodiment of a method or a program for attack-sound emphasis according to the present invention. 
     
    
    
     DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS 
     Embodiment of Audio Reproduction Apparatus 
     An embodiment of an audio reproduction apparatus having an audio-signal correction function (for example, an attack-sound emphasizing function) according to the present invention will be explained with reference to  FIG. 1 . 
     It is a precondition in the following description that an audio reproduction apparatus, an embodiment of the present invention, is installed in, for example: a receiving apparatus for digital television broadcasting, to process a signal compressed by AAC (Advanced Audio Coding) so that signal components of 16 KHz or higher are cut off; or a portable terminal, to process a signal compressed by MP3 (MPEG audio layer-3) so that signal components of 8 KHz or higher are cut off. 
     As shown in  FIG. 1 , an audio reproduction apparatus  1 , an embodiment of the present invention, is provided with a sound source  100 , a decoder  110 , a DSP (Digital Signal Processor)  120 , a DAC (Digital Analog Converter)  130 , and a speaker  140 . 
     The sound source  100  is: a receiving apparatus for digital television broadcasting to output a signal encoded by AAC so that signal components of 16 KHz or higher are cut off; or a MP player to output a signal encoded by MP3 so that signal components of 8 KHz or higher are cut off. Accordingly, the sound source  100  outputs lossy-compressed audio data having high-frequency components cut off. Especially, in this embodiment, the sound source  100  outputs lossy-compressed audio data in the left and right channels. 
     The decoder  110  is compatible with a compression technique, such as MC or MP3. The decoder  110  decompresses lossy-compressed audio data in the left and right channels supplied from the sound source  100  with a decompression technique corresponding to AAC or MP3, to convert the audio data into PCM (Pulse Code Modulation) digital audio signals in the left and right channels having high-frequency components cut off. The decompressed digital audio signals in the left and right channels are output to the DSP  120 . 
     The DSP  120  is a processing unit for digital signal processing. In this embodiment, the DSP  120  corrects digital audio signals in the left and right channels decompressed by the decoder  110  into digital audio signal data in the left and right channels having attack sound emphasized. The corrected digital audio signal data in the left and right channels is output to the DAC  130 . 
     The DAC  130  is a converter to convert a digital audio signal into an analog audio signal. In this embodiment, the DAC  130  converts the corrected digital audio signal data in the left and right channels supplied from the DSP  120  into analog audio signals. The analog audio signals are output to the speaker  140  that gives off sounds. 
     The DSP 120  is explained in detail with reference to  FIG. 2 . 
     The DSP 120  processes a digital stereo audio signal having a digital audio signal SL in the left (L) channel and a digital audio signal SR in the right (R) channel. 
     Concerning the digital audio signal SL in the left (L) channel, the DSP 120  is provided with: a buffer  111  that multiplies data (a fragment of a signal) of an input L-channel audio signal SLin by 1; a buffer  112  that multiplies the output signal of a delay element  113  by −1; the delay element  113  that delays the input L-channel audio signal SLin by one sampling period to output a signal sampled in the period that is one sampling period before the current sampling period; an adder  114  that adds the output signals of the buffers  111  and  112 ; an absolute value circuit  115  that takes the absolute value of the output signal of the adder  114 ; multipliers  116  and  117  that amplify the output signal of the absolute value circuit  115  at a specific constant ratio; an adder  118  that adds the output signal of the multiplier  116  and the output signal of a multiplier  127  in the right channel which will be described later; and a multiplier  119  that multiplies the input L-channel audio signal SLin by the output signal of the multiplier  127 , to output an L-channel corrected output signal SLout. 
     The elements that constitute the DSP 120  in the left channel will be described in detail. 
     It is defined in the following description that data SL(t) is a fragment of the input L-channel audio signal SLin sampled in a sampling period t and data SL(t−1) is a fragment of the input L-channel audio signal SLin sampled in the period that is one sampling period before the sampling period t for the data SL(t). 
     In accordance with the definition, when the L-channel audio signal SLin is input, the buffer  111  outputs the data SL(t). The buffer  112  multiplies output data SL(t−1) of the delay element  113  by −1 to output data −SL(t−1). The delay element  113  delays the input L-channel audio signal SL by one sampling period to output the data SL(t−1) sampled in the period that is one sampling period before the sampling period t for the data SL(t). 
     The adder  114  adds the output data SL(t) of the buffer  111  and the output data −SL(t−1) of the buffer  112 , to output data (a differential value) SL(t)−SL(t−1). The absolute value circuit  115  takes the absolute value of the output data SL(t)−SL(t−1) of the adder  114  to output data |(SL(t)−SL(t−1)|. 
     The multiplier  116  multiplies the output data |SL(t)−SL(t−1)| of the absolute value circuit  115  by a specific multiplier A to output data A·|SL(t)−SL(t−1)|. The multiplier  117  multiplies the output data |SL(t)−SL(t−1)| of the absolute value circuit  115  by a specific multiplier B to output data B·|SL(t)−SL(t−1)|. It is preferable that the multiplier A is larger than the multiplier B. 
     The adder  118  adds, by weighted addition, the output data A·|SL(t)−SL(t−1)| of the multiplier  116  and output data B·|SR(t)−SR(t−1)| of the multiplier  127  in the right channel which will be described later, to output data (a correction coefficient) A·|SL(t)−SL(t−1)|+B·|SR(t)−SR(t−1)|. 
     The multiplier  119  multiplies the data SL(t) and the output data A·|SL(t)−SL(t−1)|+B·|SR(t)−SR(t−1)| of the adder  118  to correct the data SL(t) to output corrected data SL(t) ·{A·|SL(t)−SL(t−1)|+B·|SR(t)−SR(t−1)|} that is the output data of the DSP  120  in the left channel. 
     Next, concerning the digital audio signal SR in the right (R) channel, the DSP 120  is provided with: a buffer  121  that multiplies data (a fragment of a signal) of an input R-channel audio signal SRin by 1; a buffer  122  that multiplies the output signal of a delay element  123  by −1; the delay element  123  that delays the input R-channel audio signal SRin by one sampling period to output a signal sampled in the period that is one sampling period before the current sampling period; an adder  124  that adds the output signals of the buffers  121  and  122 ; an absolute value circuit  125  that takes the absolute value of the output signal of the adder  124 ; multipliers  126  and  127  that amplify the output signal of the absolute value circuit  125  at a specific constant ratio; an adder  128  that adds the output signal of the multiplier  126  and the output signal of the multiplier  117  in the left channel; and a multiplier  129  that multiplies the input R-channel audio signal SRin by the output signal of the adder  128 , to output a R-channel corrected output signal SRout. 
     The elements that constitute the DSP 120  in the right channel will be described in detail. 
     It is defined in the following description that data SR(t) is a fragment of the input R-channel audio signal SRin sampled in a sampling period t and data SR(t−1) is a fragment of the input R-channel audio signal SRin sampled in the period that is one sampling period before the sampling period t for the data SR(t). 
     In accordance with the definition, when the R-channel audio signal SRin is input, the buffer  121  outputs the data SR(t). The buffer  122  multiplies output data SR(t−1) of the delay element  123  by −1 to output data −SR(t−1). The delay element  123  delays the input R-channel audio signal SR by one sampling period to output the data SR(t−1) sampled in the period that is one sampling period before the sampling period t for the data SR(t). 
     The adder  124  adds the output data SR(t) of the buffer  121  and the output data −SR(t−1) of the buffer  122 , to output data (a differential value) SR(t)−SR(t−1). The absolute value circuit  125  takes the absolute value of the output data SR(t)−SR(t−1) of the adder  124  to output data |SR(t)−SR(t−1)|. 
     The multiplier  126  multiplies the output data |SR(t)−SR(t−1)| of the absolute value circuit  125  by the multiplier A to output data A·SR(t)−SR(t−1)|. The multiplier  127  multiplies the output data |SR(t)−SR(t−1)| of the absolute value circuit  125  by the multiplier B to output data B·|SR(t)−SR(t−1)|. 
     The adder  128  adds, by weighted addition, the output data A·|SR(t)−SR(t−1)| of the multiplier  126  and output data B·|SL(t)−SL(t−1)| of the multiplier  117  in the left channel, to output data (a correction coefficient) A·|SR(t)−SR(t−1)|+B·SL(t)−SL(t−1)|. 
     The multiplier  129  multiplies the data SR(t) and the output data A·|SR(t)−SR(t−1)|+B·|SL(t)−SL(t−1)| of the adder  128  to correct the data SR(t) to output corrected data SR(t) ·{A·|SR(t)−SR(t−1)|+B·|SL(t)−SL(t−1)|} that is the output data of the DSP  120  in the right channel. 
     In  FIG. 2 , the buffers  111  and  112 , the delay element  113 , and the adder  114  constitute a first differential-value acquisition circuit that acquires a first differential value SL(t)−SL(t−1) between first current input data SL(t) and first previous input data SL(t−1) in an i number (i being a natural number, that is t in the embodiment) of sampling periods before the first current input data SL(t), both first input data SL(t) and SL(t−1) being of a first digital audio signal SLin that has a sound level of a digital stereo audio signal in the left channel. 
     Also, in  FIG. 2 , the buffers  121  and  122 , the delay element  123 , and the adder  124  constitute a second differential-value acquisition circuit that acquires a second differential value SR(t)−SR(t−1) between second current input data SR(t) and second previous input data SR(t−1) in a j number (j being a natural number, that is t in the embodiment) of sampling periods before the second current input data, both second input data SR(t) and SR(t−1) being of a second digital audio signal SRin that has a sound level of the digital stereo audio signal in the right channel. 
     Moreover, in  FIG. 2 , the absolute value circuits  115  and  125 , the multipliers  116 ,  117 ,  126  and  127 , and the adders  118  and  128  constitute a correction coefficient acquisition circuit that acquires a first correction coefficient A·|SL(t)−SL(t−1)|+B·|SR(t)−SR(t−1)| by adding the first and second differential values SL(t)−SL(t−1) and SR(t)−SR(t−1) at a first ratio (the multiplier A:B, A&gt;B) and acquires a second correction coefficient A·|SR(t)−SR(t−1)|+B·|SL(t)−SL(t−1)| by adding the first and second differential values SL(t)−SL(t−1) and SR(t)−SR(t−1) at a second ratio (B:A). 
     Furthermore, in  FIG. 2 , multipliers  119  and  129  constitute a correction circuit that corrects the first digital audio signal SLin by multiplying the first digital audio signal SLin by the first correction coefficient A·|SL(t)−SL(t−1)|+B·|SR(t)−SR(t−1)| and corrects the second digital audio signal SRin by multiplying the second digital audio signal SRin by the second correction coefficient A·|SR(t)−SR(t−1)|+B·|SL(t)−SL(t−1)|. 
     Described next is an operation of the audio reproduction apparatus  1  shown in  FIG. 1 . 
     The sound source  100  outputs to the decoder  110  L- and R-channel lossy-compressed audio data having high-frequency components cut off. The decoder  110  decodes the L- and R-channel lossy-compressed audio data into decompressed L- and R-channel digital audio signals having high-frequency components cut off. The L- and R-channel digital audio signals are then input to the DSP 120 . 
     The DSP 120  corrects the L- and R-channel digital audio signals with attack-sound emphasis to output attack-sound-emphasized L- and R-channel digital audio signals. 
     The correction of digital audio signals at the DSP  120  in the left channel is described in detail with respect to  FIG. 2 . 
     At the buffer  111 , the data SL(t) of the input L-channel audio signal SLin multiplied by 1 in the sampling period t. At the buffer  112 , the data SL(t−1) of the audio signal SLin sampled in the period that is one sampling period before the sampling period t for the data SL(t) is multiplied by −1. The output data of the buffers  111  and  112  are added to each other by the adder  114  to be the data SL(t)−SL(t−1). Accordingly, obtained through these operations is a differential value xL(t) between the current data and data at one sampling before the current data for the input L-channel audio signal SLin. 
     The differential value xL(t) is supplied to the absolute value circuit  115  that takes an absolute value |xL(t)|. The absolute value |xL(t)| of the differential value xL(t) is amplified by the multiplier A (for example, 0.8) at the multiplier  116  to be data A·|xL(t)|. The data A·|xL(t)| is supplied to the adder  118 . Also supplied to the adder  118  is data B·|xR(t)| in the right channel, which is obtained by amplifying an absolute value |xR(t)| of a differential value xR(t) between the current data and data at one sampling before the current data for the input R-channel audio signal SRin by the multiplier B (for example, 0.2) at the multiplier  127 . The data A·|xL(t)| and B·|xR(t)| are added to each other by the adder  118  to be data (a correction efficient) A·|xL(t)|+B·|xR(t)|. 
     The data SL(t) of the input L-channel audio signal SLin is then multiplied by the output data A·|xL(t)|+B·|xR(t)| of the adder  118  at the multiplier  119  so that the level of the data SL(t) is corrected, thus level-corrected data SL(t)·A·|xL(t)|+B·|xR(t)| is output. 
     These operations are performed for sequential input L-channel digital audio data SL(t), SL(t+1), SL(t+2), . . . , for level corrections or adjustments. 
     The correction of digital audio signals at the DSP  120  in the right channel is also performed at the elements  123  to  129  ( FIG. 2 ), in the same way as the digital audio signals in the left channel, the level of the data SR(t) of the input R-channel audio signal SRin is corrected based on: the data obtained by multiplying the absolute value |xR(t)| of the differential value xR(t) between the current data and data at one sampling before the current data by the multiplier A (for example, 0.8); and the data obtained by amplifying the absolute value |xL(t)| of the differential value xL(t) for the input L-channel audio signal SRin by the multiplier B (for example, 0.2). 
     The multipliers A and B (weighting coefficients) may be equal to each other or they may be different from each other, that is, the multiplier A may be larger than the multiplier B, and vise versa. Nevertheless, it is preferable that the multiplier A is larger than the multiplier B. Specific constants (ratios) different between the left and right channels may also be used. The same multiplier A is used for both of the left and right channels. Likewise, the same multiplier B is used for both of the left and right channels. 
     Through the operations described above, the level-corrected L- and R-channel audio signals SLout and SRout are supplied to the speaker  140 , via the DAC  130 , that gives off sounds based on the audio signals SLout and SRout. 
     Discussed next is the absolute value |xL(t)| of the differential value xL(t) and the absolute value |xR(t)| of the differential value xR(t) obtained at the absolute value circuits  115  and  125 , respectively. 
     The absolute value |xL(t)| expresses the change in data amount of the current audio data SL(t) to the audio data SL(t−1) in one sampling period before the current audio data SL(t), in the left channel. Likewise, the absolute value |xR(t)| expresses the change in data amount of the current audio data SR(t) to the audio data SR(t−1) in one sampling period before the current audio data SR(t), in the right channel. 
     When the change discussed above is positive and large (that is, the sound level rises steeply) for the L-channel audio data SL(t), through the operations described above, the L-channel audio data SL(t) is multiplied by the value obtained by weighted addition to the absolute value |xL(t)| of the differential value xL(t) and the absolute value |xR(t)| of the differential value xR(t). Therefore, the L-channel output sound level increases. 
     Moreover, when the change discussed above is positive and large (that is, the sound level rises steeply) for the R-channel audio data SR(t), through the operations described above, the R-channel audio data SR(t) is multiplied by the value obtained by weighted addition to the absolute value |xR(t)| of the differential value xR(t) and the absolute value |xL(t)| of the differential value xL(t). Therefore, the R-channel output sound level increases. 
     When the change discussed above is positive but small (that is, the sound level rises not so steeply), the same operations as described are performed. However, since the absolute values |xL(t)| and |xR(t)| are both small, the output sound level does not increase, or changes little. 
     The same operation as for the positive and large change described above is also performed when the change discussed above is negative and large, that is, the sound level rises steeply. 
     Explained next in detail is how an attack sound is emphasized by the attack-sound emphasizing function of the audio reproduction apparatus  1  described above. 
     It is supposed that an original signal having an original waveform indicated by a solid line in  FIG. 3  is input to the audio reproduction apparatus  1  in the left channel. It is further supposed that the original signal is a PCM (Pulse Code Modulation) audio signal decoded by an MP-3 decoder from lossy-compressed audio data compressed by MP3, having high-frequency components cut and dynamics lost. 
     With the attack-sound emphasizing function of the DSP 120 , as described above, a differential value SL(t)−SL(t−1) is obtained for a signal level SL(t) in the current sampling period t and a signal level SL(t−1) in a sampling time t−1 just before the current sampling period t. Then, the sampled value in the current sampling period t is corrected to be a corrected sampled value SL(t)·{A·|SL(t)−SL(t−1)|+B·|SR(t)−SR(t−1)|}, as described above. With the processing, the sampled value in the current sampling period t is increased as shown in  FIG. 3 . Then, audio data having the corrected sampled value is output to the DAC 130  from the DSP  120 . Accordingly, the original waveform indicated by the solid line in  FIG. 3  is changed to an analog waveform obtained by the attack-sound emphasizing function and indicated by a broken line, having an attack sound emphasized. The analog waveform having the attack sound emphasized is output the speaker  140  that gives off a sharp and dynamic attack sound. 
     Explained next is how much an attack sound is emphasized by the attack sound emphasizing function of the audio reproduction apparatus  1  described above. 
       FIG. 4  shows an example of audio signals continuously output from the decoder  110 , with the time (sec) and level on the abscissa and ordinate, respectively.  FIG. 5  shows audio signals continuously output from the DSP 120  in response to the audio signals of  FIG. 4 , with the time (sec) and level on the abscissa and ordinate, respectively. 
       FIG. 6  is a view in which a view of  FIG. 4  is superimposed on that of  FIG. 5 , with a curve CA (indicated by a broken line) indicating the audio signals output from the decoder  110  and a curve CB (indicated by a solid line) indicating the audio signals output from the DSP 120 . It is understood from  FIG. 6  that specific data having a level increased very much with respect to data one sampling period before the specific data is corrected to have a level increased further. 
     As described above, according to the audio reproduction apparatus  1 , the embodiment of the present invention, an attack sound having a sound level rising up steeply and a volume varying instantaneously is reproduced as a sharper and clearer attack sound having a sound level rising up steeply. 
     Moreover, the audio reproduction apparatus  1 , the embodiment of the present invention, has the following advantages: The DSP 120  is not equipped with filters which would otherwise cause phase delay or error, thus achieving real-time correction of audio signals with very light load processing. The DSP 120  performs the correction to raise the level higher for a sound with a steeper rising level, thus outputting a corrected sound that does not give an adverse effect to the characteristics of the speaker  140 , such as conversion loss. The DSP 120  is not equipped with feedback circuits which would otherwise cause oscillation, thus outputting sounds of stable levels. The DSP 120  corrects audio signals not based on the level difference in either the left or right channel but based on the level difference in both of the left and right channels. Therefore, the levels of the audio signals rise instantaneously with almost no movement of sound image between the left and right channels, thus the reproduction of a real attack sound is achieved. 
     As described above in detail, according to the audio reproduction apparatus  1 , the embodiment of the present invention, an attack sound portion of an audio signal is corrected to have a waveform closer to an original sound (an original audio signal). Therefore, a shaper, clearer and more realistic attack sound that is closer to the original sound can be reproduced. 
     (Variation to Audio Reproduction Apparatus) 
     Described next is a variation to the audio reproduction apparatus  1 , the embodiment of the present invention. 
     An audio reproduction apparatus  2 , a variation of the present invention, is provided with a sound source  100 , a decoder  110 , a DSP  120   a , a DAC  130 , and a speaker  140 , connected to one another in the same manner as the audio reproduction apparatus  1  shown in  FIG. 1 , with the same reference numerals given to the same or analogous elements as those of  FIG. 1 . 
     Different from the DSP  120  of the audio reproduction apparatus  1  shown in  FIG. 2 , the DSP  120   a  of the audio reproduction apparatus  2  is equipped with time constant circuits  11 A and  12 A as shown in  FIG. 7 , with the same reference numerals given to the same or analogous elements as those of  FIG. 2 . 
     In detail, as shown in  FIG. 7 , the time constant circuit  11 A is provided between the adder  118  and the multiplier  119  in the left channel and the time constant circuit  12 A is provided between the adder  128  and the multiplier  129 . The time constant circuit  11 A receives the output signal of the adder  118 , varies the response speed of the output signal, and outputs a signal with a varied response speed to the multiplier  119 . The time constant circuit  12 A receives the output signal of the adder  128 , varies the response speed of the output signal, and outputs a signal with a varied response speed to the multiplier  129 . 
     In the case of adjusting the response speed to be slower, the time constant circuits  11 A and  11 B may delay or integrate the input signal, or suppress high-frequency components of the input signal. 
     Although the operation of the audio reproduction apparatus  2  is basically the same as the audio reproduction apparatus  1 , the audio reproduction apparatus  2  can vary the speed of rise-up (the response speed) of a signal, that is, the dynamic characteristics of a signal. In other words, when a level difference between differential values xL(t) and xR(t) is large, the audio reproduction apparatus  2  starts the correction of audio signals at the time of detecting the large level difference and gradually decreases the degree of the correction over a specific period. 
     The time constants of the time constant circuits  11 A and  11 B are adjusted to vary the response speed of a signal, which has the following advantages and disadvantages: The smaller the time constant to increase the response speed, the steeper the rise of a signal, which is advantageous in adequately outputting a sound with rapid change, such as a attack sound, whereas disadvantageous in lower sound reproducibility. On the other hand, the larger the time constant to decrease the response speed, the slower the rise of a signal, which is disadvantageous in inadequately outputting a sound with rapid change, such as a attack sound, whereas advantageous in higher sound reproducibility. 
     The sound reproducibility discussed above is defined as follows: The sound reproducibility is low when a sound is processed only at the point at which the sound level rises, with the continuity between the processed sound and the next sound after the process being not smooth and hence not natural when given off by the speaker  140 . On the other hand, the sound reproducibility is high when a sound at the point at which the sound level rises and the next sound are processed, with the continuity between the processed sounds being smooth and hence natural when given off by the speaker  140 . 
     The audio reproduction apparatus  2  may be equipped with a setting circuit  12  for adjusting a time constant τ of the time constant circuits  11 A and  11 B, as shown in  FIG. 8 . The time constant τ may be set by user input or may be set to a value corresponding to a user ID input by a user. Or the time constant τ may be set to a value corresponding to genre information carried by a reproduced signal supplied from the sound source  100 . 
     As described above, the variation to the audio reproduction apparatus  2  allows a user to set the response speed to any value in accordance with how much high-frequency components have been cut off or with a user&#39;s favorite genre of music. 
     (Embodiment of Audio Reproduction Method and Program) 
     Described above are the embodiment of audio reproduction apparatus and its variations equipped with the DSP  120  ( 120   a ) having the attack-sound emphasizing function. Not only by the DSP  120 , the attack sound emphasizing function can be achieved with an ordinary processor (CPU) that executes a program for a process which will be described blow. The program is preferably stored in a storage medium, such as a RAM or ROM implemented with the CPU in an audio reproduction apparatus. 
     An audio reproduction apparatus in this case has the circuit configuration the same as that of  FIG. 1 , except for the CPU in place of the DSP 120 . 
     An attack-sound emphasizing process executed by the CPU is explained with reference to  FIG. 9 . 
     Firstly, a variable t that indicates a sampling period is substituted with zero, in step S 101 . Next, audio signals SL(t) and SR(t) in the left and right channels, respectively, are input and stored associated with the variable t, in step S 102 . It is then determined whether the variable t is zero, in step S 103 . 
     If it is determined that the variable t is zero (Yes in step S 103 ), there is only one piece of audio data for each of the left and right channels, and hence the differential values xL(t) and xR(t) cannot be obtained. Therefore, the variable t is incremented by +1 in step S 104  and then the process retunes to step S 102  to repeat the steps described above. 
     On the other hand, if it is determined that the variable t is not zero (No in step S 103 ), xL(t)=|SL(t)−SL(t−1)| and xR(t)=|SR(t)−SR(t−1)| are calculated in the left and right channels, in step S 105 , that are the absolute vales of a differential value between current audio data SL(t) and audio data SL(t−1) obtained in one sampling period before the data SL(t) and a differential value between current audio data SR(t) and audio data SR(t−1) obtained in one sampling period before the data SR(t), respectively. 
     The absolute vales in the left and right channels are combined to obtain multipliers ML(t)=A·xL(t)+B·xR and MR(t)=A·xR(t)+B·xL which are then stored, in step S 106 . Next, in step S 107 , multipliers are selected from among the obtained multipliers according to the time constant τ. For example, if the time constant τ corresponds to n sampling periods, selected are multipliers ML(t−n) and MR(t−n). 
     The input audio data SL(t) and SR(t) are then multiplied by the selected multipliers ML(t) and MR(t), respectively, to obtain output signals OL(t) and OR(t), in step S 108 . 
     It is then determined whether there is audio data in the next sampling period, in step S 109 . 
     If it is determined that there is audio data in the next sampling period (Yes in step S 109 ), the process returns to step S 102  to repeat the steps described above. On the other hand, if it is determined that there is no audio data in the next sampling period (No in step S 109 ), the attack-sound emphasizing process ends. 
     With the attack-sound emphasizing process described above, the correction of sounds having attack sounds emphasized that have been deteriorated due to lossy-compressed can be performed. 
     In the description above, a differential value between two pieces of audio data appearing one after another is obtained for acquiring the change in audio signals SL and SR in the left and right channels, respectively. However, not only the differential value between two pieces of audio data appearing one after another, any value can be obtained in this invention as far as substantial differential values that represent the change in audio signals SL and SR in the left and right channels, respectively, can be obtained. 
     For example, an audio signal may be corrected with the acquisition of differential values between current audio data and audio data one sampling period before, the current audio data and audio data two sampling periods before, . . . , and the current audio data and audio data n sampling periods before, through a plurality (n) of stages of delay elements, in each of the left and right channels. 
     The correction with the acquisition of differential values through n pieces of audio data can be achieved, in  FIG. 2 , with an n number of delay elements  113  sequentially provided in the left channel. In this case, the adder  114  outputs xL(t)=W1·{SL(t)−SL(t−1)}+W2·{SL(t−1)−SL(t−2)}+ . . . +Wn·{SL(t−n+1)−SL(t−n)}. Or the adder  114  may output xL(t)=W1·{SL(t)−SL(t−1)}+W2·{SL(t1)−SL(t−2)}+ . . . +Wn·{SL(t)−SL(t−n)}. W1 to Wn are weights which can be set freely. Moreover, the adder  114  may obtain Σij·{(SL(t−i)−SL(t−j)} (i=0 to n+1, j=1 to n, i&lt;j). The same is applied to the right channel. 
     Moreover, the average or maximum value of differential values between current audio data and audio data one sampling period before, the current audio data and audio data two sampling periods before, . . . , and the current audio data and audio data n sampling periods before may be used as the differential value x for the correction of audio signals. 
     In  FIGS. 2 and 7 , the absolute value circuits  115  and  125  may be omitted. 
     In the above description, input audio signals are multiplied by multipliers that are correction coefficients obtained by the adders  118  and  128 . The multipliers may be a value obtained by applying some factors to the correction coefficients. For example, the multipliers may be obtained by adding a specific bias value to the correction coefficients. 
     Moreover, a switching circuit may be provided to: determine whether audio data supplied from the sound source  100  ( FIG. 1 ) is lossy-compressed audio data and turn on the attack-sound emphasizing function explained with reference to  FIG. 2  or  7  (or supplies the audio data to the attack-sound emphasizing circuit of  FIG. 2  or  7 ) when determined that the audio data is lossy-compressed data; whereas, if not, turn off the attack-sound emphasizing function (or not supply the audio data to the attack-sound emphasizing circuit). 
     Furthermore, a program running on a computer to achieve the attack-sound emphasizing function described with respect to  FIG. 2  or  7  (or the process described with respect to  FIG. 9 ) may be retrieved from a storage medium (a flexible disc, a CD-ROM, a DVD-ROM, etc.). Or the program may be transferred from a storage medium of a server on a communication network, such as the Internet, and installed in a computer. 
     Moreover, the attack-sound emphasizing function or process may be achieved with OS (Operating System) and an application program that is stored in a storage medium or apparatus. 
     Furthermore, the program running on a computer to achieve the attack-sound emphasizing function or process may be carried by a carrier wave and delivered over a communication network. In this case, the program may be posted on BBS (Bulletin Board System) on a communication network. The program is then delivered or downloaded over the network to a computer that executes the program like other application programs under control by the OS to perform the attack-sound emphasizing function or process. 
     As described above in detail, the present invention achieves the correction of an audio signal that involves an attack sound deteriorated due to digitalization or compression into an audio signal close to an original audio signal. 
     It is further understood by those skilled in the art that the foregoing description is a preferred embodiment of the disclosed device or method and that various changes and modifications may be made in the invention without departing from the spirit and scope thereof.