Patent Publication Number: US-7592941-B2

Title: Signal processing apparatus and signal processing method

Description:
CROSS REFERENCES TO RELATED APPLICATIONS 
   The present invention contains subject matter related to Japanese Patent Application JP 2007-025920, filed in the Japanese Patent Office on Feb. 5, 2007, the entire contents of which being incorporated herein by reference. 
   BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The present invention relates to a signal processing apparatus for performing signal processing on an audio signal in accordance with a given purpose, and a method therefor. 
   2. Description of the Related Art 
   A so-called noise cancellation system is known that is implemented on a headphone device and used to actively cancel an external noise that comes when a sound of content, such as a tune, is being reproduced via the headphone device. Such noise cancellation systems have been put to practical use. Such noise cancellation systems are broadly classified into a feedback system and a feedforward system. 
   For example, Japanese Patent Laid-open No. Hei 3-214892 describes a structure of a feedback noise cancellation system in which a noise inside a sound tube worn on ears of a user is picked up by a microphone unit provided close to an earphone unit within the sound tube, a phase-inverted audio signal of the noise is generated, and this audio signal is outputted as sound via the earphone unit, so that the external noise is reduced. 
   Meanwhile, Japanese Patent Laid-open No. Hei 3-96199 describes a structure of a feedforward noise cancellation system in which, in essence, a noise is picked up by a microphone attached to the exterior of a headphone device, a characteristic based on a desired transfer function is given to an audio signal of the noise, and a resultant audio signal is outputted from the headphone device. 
   SUMMARY OF THE INVENTION 
   Noise cancellation systems in consumer headphone devices in practical use today are composed of analog circuits, whether they are in accordance with the feedback system or the feedforward system. 
   In order for a noise cancellation effect of the noise cancellation system to be achieved effectively, difference in phase between an external unwanted sound picked up by, for example, a microphone and a sound outputted from a driver for canceling this unwanted sound should be restricted within a certain range. In other words, in the noise cancellation system, a time between input of the external unwanted sound and output of a corresponding cancellation-use sound should be restricted within a certain range. That is, a response speed should be sufficiently fast. 
   When the noise cancellation system is constructed using a digital circuit, however, an A/D converter and a D/A converter need be provided at input and output of the noise cancellation system. A/D converters and D/A converters that are widely used today have too long processing time and cause too long delays to be adopted in the noise cancellation system, and it is difficult to achieve an effective noise cancellation effect therewith. In military and industrial fields, for example, A/D converters and D/A converters that have a significantly high sampling frequency and cause slight delays are used, but these A/D converters and D/A converters are very expensive, and it is not practical to adopt them in consumer devices. This is the reason why the noise cancellation systems today are constructed using an analog circuit instead of a digital circuit. 
   Replacement of the analog circuit by the digital circuit makes it easy to change or switch characteristics or an operation mode, without the need to physically change a constant in a component or replace a component, for example. In addition, in the case of an audio-related system such as the noise cancellation system, the replacement of the analog circuit by the digital circuit has many advantages, such as expected further improvement in sound quality. 
   As such, the present invention aims to allow a noise cancellation system in, for example, a consumer headphone device to be constructed using a digital circuit and achieve a practically sufficient noise cancellation effect. 
   According to one embodiment of the present invention, there is provided a signal processing apparatus including: analog-to-digital conversion means for performing a first delta sigma modulation process of generating a digital signal having a predetermined sampling frequency and a predetermined quantization bit rate of one or more bits based on an input analog signal; signal processing means including a digital filter having a predetermined characteristic for outputting a digital signal having a sampling frequency of n×Fs where n is a natural number and Fs is a predetermined reference sampling frequency and a predetermined quantization bit rate of a bits where a is a natural number greater than one based on the digital signal generated by the analog-to-digital conversion means; and digital-to-analog conversion means including a part for performing a second delta sigma modulation process for outputting a digital signal having a sampling frequency of n×Fs and a predetermined quantization bit rate of b bits where b is a natural number greater than zero and less than a based on a digital signal outputted from the signal processing means. 
   In the above embodiment, first, the output of the (first) delta sigma modulation process is obtained as an output as a result of an analog-to-digital conversion (A/D conversion) process. Then, as signal processing, the digital signal thus obtained is caused to pass through the digital filter in which the filter characteristic is set in accordance with at least a predetermined purpose. The signal outputted as a result of this signal processing is a digital signal with a sampling frequency of n×Fs and a quantization bit rate of a bits where a is a natural number greater than one. A D/A conversion part for accepting input of at least the signal outputted from the digital filter and converting this signal into an analog signal includes the part for performing the (second) delta sigma modulation process, and the digital signal obtained as a result of the above signal processing is inputted to this part for performing the (second) delta sigma modulation process. As a result of the (second) delta sigma modulation process, the digital signal with a sampling frequency of n×Fs and a predetermined quantization bit rate of b bits where b is a natural number greater than zero and less than a is obtained. 
   For example, in a known device for performing an A/D conversion process including a delta sigma modulation process, a signal obtained after the delta sigma modulation process is allowed to pass through a decimation filter, and a digital signal with a reference sampling frequency Fs and a quantization bit rate of two or more bits, for example, is outputted from the device. Meanwhile, in a known device for performing a D/A conversion process including a delta sigma modulation process, first, a signal with the reference sampling frequency Fs (=1 Fs) and a quantization bit rate of two or more bits is subjected to oversampling so as to have a sampling frequency in accord with the delta sigma modulation process. 
   In comparison to a digital signal processing apparatus having the above known devices for performing the A/D conversion process and the D/A conversion process for its input and output, the signal processing apparatus in accordance with the present invention has the following features. That is, in the A/D conversion process, the signal is not caused to pass through the decimation filter, and the signal obtained as a result of the delta sigma modulation process is inputted to the subsequent part (i.e., the signal processing means) for performing the signal processing in accordance with the predetermined purpose. In addition, when the digital signal outputted from the signal processing means is converted into an analog signal, the digital signal is inputted to the part for performing the delta sigma modulation process without being subjected to the oversampling process. In other words, in the signal processing apparatus in accordance with the present invention, the parts for converting the analog signal into the digital signal, subjecting this digital signal to the signal processing in accordance with the predetermined purpose, and converting the resulting digital signal into the analog signal omit decimation in the A/D conversion process and the oversampling process in the D/A conversion process. Due to the omission of at least these processes, a signal propagation time is reduced in the signal processing apparatus in accordance with the present invention, which includes a signal processing system for [A/D conversion−digital signal processing−D/A conversion]. 
   According to another embodiment of the present invention, there is provided a signal processing method including: an analog-to-digital conversion step of performing a first delta sigma modulation process of generating a digital signal having a predetermined sampling frequency and a predetermined quantization bit rate of one or more bits based on an input analog signal; a signal processing step, performed by a digital filter having a predetermined characteristic, of outputting a digital signal having a sampling frequency of n×Fs where n is a natural number and Fs is a predetermined reference sampling frequency and a predetermined quantization bit rate of a bits where a is a natural number greater than one based on the digital signal generated in the analog-to-digital conversion step; and a digital-to-analog conversion step, performed by a part for performing a second delta sigma modulation process, of outputting a digital signal having a sampling frequency of n×Fs and a predetermined quantization bit rate of b bits where b is a natural number greater than zero and less than a based on a digital signal obtained in the signal processing step. 
   With the signal propagation time in the signal processing apparatus in accordance with the present invention, a condition of the response speed for the signal processing system in the noise cancellation system in the headphone device is satisfied. That is, it becomes possible to implement a noise cancellation system using a digital circuit(s) easily. The implementation of the noise cancellation system using the digital circuit(s) makes it possible to implement a feature that is difficult to implement in a noise cancellation system using an analog circuit(s), and also achieve improved sound quality, for example, resulting in increased usefulness for users. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  is a block diagram illustrating an exemplary basic structure of a digital noise cancellation system in a headphone device; 
       FIG. 2  is a block diagram illustrating an exemplary structure of a noise cancellation system in accordance with a first embodiment of the present invention; 
       FIGS. 3A and 3B  illustrate exemplary structures of a noise cancellation-use digital filter in accordance with one embodiment of the present invention; 
       FIG. 4  is a block diagram illustrates an exemplary structure of a noise cancellation system in accordance with a second embodiment of the present invention; 
       FIG. 5  is a block diagram illustrating an exemplary structure of a noise cancellation system in accordance with a third embodiment of the present invention; 
       FIG. 6  is a block diagram illustrating an exemplary structure of a noise cancellation system in accordance with a fourth embodiment of the present invention; 
       FIG. 7  is a block diagram illustrating an exemplary structure of a noise cancellation system in accordance with a fifth embodiment of the present invention; and 
       FIG. 8  is a block diagram illustrating an exemplary structure of a noise cancellation system in accordance with a sixth embodiment of the present invention. 
   

   DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     FIG. 1  illustrates an exemplary structure of a noise cancellation system in a headphone device constructed using digital devices currently known. 
   The structure of the noise cancellation system illustrated in this figure is based on a feedforward system. In the feedforward system, an audio signal is obtained by picking up an external sound, and this audio signal is subjected to an appropriate filtering process to generate a cancellation-use audio signal. Then, this cancellation-use audio signal is combined with an audio signal of a necessary sound, and a resultant audio signal is outputted from a driver as a sound, with the intention to cancel the external sound to achieve noise cancellation. 
   The headphone device (hereinafter simply referred to as a “headphones”)  17  illustrated in this figure is assumed to support dual-channel (L (left) and R (right)) stereo. A system structure as illustrated in this figure corresponds to one of an L channel and an R channel. 
   Note that a reference sampling frequency denoted as Fs (1 Fs) in the following descriptions is assumed to correspond to a sampling frequency of a digital audio source, a sound of which is a sound to be listened to using the headphone device  17 . Specific examples of the digital audio source include a digital audio signal, with Fs=44.1 kHz and a quantization bit rate of 16 bits, recorded on a compact disc (CD). 
   In  FIG. 1 , a microphone  11  is used to pick up an external sound including an external sound (an external noise) that is caused around the headphones  17  and which is to be cancelled. Although not illustrated in this figure, in the case of the feedforward system, this microphone  11  is commonly provided on the exterior of a housing of the headphones  17  corresponding to each of the two (L and R) channels. In this figure, the microphone  11  provided for one of the two (L and R) channels is shown. 
   A signal obtained by the microphone  11  by picking up the external sound is amplified by an amplifier  12 , and is inputted to an A/D converter section  13  in the form of an analog audio signal. 
   The A/D converter section  13  is formed as a single part (device), for example, and converts the input analog audio signal into a digital signal (a PCM signal) by quantizing the input analog audio signal with a sampling frequency of 1 Fs and a quantization bit rate of 16 bits ([1 Fs, 16 bit]) corresponding to those of the digital audio source described below. Then, the A/D converter section  13  outputs the obtained digital signal. 
   For this purpose, as illustrated in  FIG. 1 , the A/D converter section  13  includes a delta sigma (ΔΣ) modulator  13   a , a decimation filter  13   b , and an output buffer  13   c.    
   The analog audio signal inputted to the A/D converter section  13  is first converted into a [64 Fs (=2.8224 MHz), 1 bit] digital signal by the delta sigma modulator  13   a . This [64 Fs, 1 bit] digital signal passes through the decimation filter  13   b , e.g., a finite impulse response (FIR) filter, to be converted into a [1 Fs, 16 bit] digital signal, and then is amplified by the output buffer  13   c . A resultant signal outputted from the output buffer  13   c  is outputted from the A/D converter section  13 . 
   The [1 Fs, 16 bit] digital signal outputted from the A/D converter section  13  is inputted to a digital signal processor (DSP)  14 . 
   The DSP  14  is formed as a single chip part, for example, and performs a necessary digital signal process for generating at least an audio signal of a sound to be outputted from a driver  17   a  of the headphone device  17 . As will be understood from the following descriptions, the audio signal to be outputted from the driver  17   a  of the headphone device  17  is composed of a combination of an audio signal of the digital audio source and the audio signal (i.e., the cancellation-use audio signal) for allowing the external sound picked up by the microphone  11  to be cancelled. 
   In  FIG. 1 , as a signal processing functional block contained in the DSP  14 , a noise cancellation-use digital filter  14   a  is shown. 
   The digital signal outputted from the A/D converter section  13 , i.e., the digital audio signal of the external sound picked up by the microphone  11 , is inputted to the noise cancellation-use digital filter  14   a . Then, this signal inputted is used to generate, as an audio signal of a sound to be outputted from the driver  17   a , an audio signal (i.e., the cancellation-use audio signal) of a sound that will contribute to canceling the external sound that will arrive at an ear, corresponding to the driver  17   a , of a user wearing the headphones. The cancellation-use audio signal in the simplest form is, for example, an audio signal that is in inverse relation to the audio signal inputted to the noise cancellation-use digital filter  14   a , i.e., the audio signal obtained by picking up the external sound, in terms of characteristics and phase. In practice, additional characteristics are given to the cancellation-use audio signal, taking account of transfer characteristics of circuits, spaces, and so on in the noise cancellation system. 
   The noise cancellation-use digital filter  14   a  is formed, for example, as an FIR filter, and is configured to accept input of a signal with a quantization bit rate of 16 bits and multiply the signal by a 16-bit coefficient. Thus, a signal outputted from the noise cancellation-use digital filter  14   a  is in [1 Fs, 16 bit] form as is the signal inputted. 
   The signal of the digital audio source is also inputted to the DSP  14 . This signal of the digital audio source is a digital audio signal in the [1 Fs, 16 bit] form, and is, in a combiner  14   b  within the DSP  14 , combined with (added to) the cancellation-use audio signal, which is also in the [1 Fs, 16 bit] form, outputted from the noise cancellation-use digital filter  14   a.    
   In such a manner, the digital audio signal composed of the combination of the signal of the digital audio source and the cancellation-use audio signal is obtained by the combiner  14   b . This digital audio signal is outputted from the DSP  14  and inputted to a D/A converter section  15  in the subsequent stage. 
   The D/A converter section  15  is also formed as a single chip part, for example, and is used to convert a digital signal in the form resulting from conversion by the A/D converter section  13  described above into an analog signal. For example, as illustrated in  FIG. 1 , the D/A converter section  15  includes an oversampling filter  15   a , a delta sigma modulator  15   b , and an analog low-pass filter (LPF)  15   c.    
   The [1 Fs, 16 bit] digital signal inputted to the D/A converter section  15  is subjected to an oversampling process by the oversampling filter  15   a  to be converted into a digital signal in [64 Fs, 16 bit] form. The resultant signal is outputted to the delta sigma modulator  15   b.    
   The delta sigma modulator  15   b  converts the input digital signal into a 1-bit signal. In other words, the delta sigma modulator  15   b  converts the input digital signal into a digital signal in [64 Fs, 1 bit] form, and outputs the resulting signal. Then, the [64 Fs, 1 bit] digital signal outputted from the delta sigma modulator  15   b  passes through the analog LPF  15   c , so that an analog audio signal is obtained as an output of the analog LPF  15   c . That is, the [1 Fs, 16 bit] digital audio signal inputted to the D/A converter section  15  is converted into the analog audio signal, and this analog audio signal is outputted from the D/A converter section  15 . 
   The analog audio signal outputted from the D/A converter section  15  is inputted to a power amplifier  16 . The power amplifier  16  amplifies the input audio signal and outputs the amplified audio signal to drive the driver  17   a , corresponding to one ear, of the headphones  17 . 
   A sound outputted from the driver  17   a  driven in such a manner is composed of a combination of a sound component corresponding to the digital audio source and a sound component corresponding to the noise cancellation-use audio signal. In this sound, the sound component corresponding to the noise cancellation-use audio signal serves to cancel the external sound that comes from an outside to the ear corresponding to the driver  17   a . As a result, in a sound heard by the ear, corresponding to the driver  17   a , of the user wearing the headphones, the external sound is cancelled, ideally, so that the sound of the digital audio source is relatively emphasized. 
   In the structure as illustrated in  FIG. 1 , an A/D converter, a DSP, a D/A converter, and so on which are readily available for general (e.g., consumer) use are used. Therefore, this structure is a natural choice today when actually constructing a digital noise cancellation system for an audio source such as a CD, for example. 
   However, it is known that it is practically difficult to obtain a sufficient noise cancellation effect with the above structure. This is because actual devices that serve as the A/D converter section  13  and the D/A converter section  15  have a significantly long signal processing time, i.e., a significantly long input-output delay. Originally, these devices are devised to simply process an audio signal of an audio source, such as of a tune, and therefore the delay caused by signal processing has not produced a problem. However, when such devices are adopted in the noise cancellation system, the delay is too large to be neglected. 
   That is, with regard to the noise cancellation system as a whole constructed using such devices, a time (i.e., a response speed) between picking up of the external sound by the microphone  11  and the output of the sound from the driver involves a significant delay. Because of this delay, it is difficult to cancel the external sound with the sound component for noise cancellation outputted from the driver, for example. If the sampling frequency is 44.1 KHz and the delay corresponds to a time of 40 samples, even the A/D converter section  13  alone causes a phase delay of greater than 180° concerning a signal at a frequency greater than approximately 550 Hz, for example. When the delay is so large, the noise cancellation effect is hard to obtain, and also a phenomenon of the external sound being emphasized may arise. 
   As described above, in accordance with the structure of the digital noise cancellation system as illustrated in  FIG. 1 , a sufficient noise cancellation effect is obtained within a limited frequency range of approximately 550 Hz or lower. In the case where a standard range of 20 Hz to 20 kHz is set as an audible range, for example, the noise cancellation effect is obtained within a very narrow frequency range. That is, a practically sufficient noise cancellation effect is not obtained. This is why noise cancellation systems in headphone devices in practical use today are in analog form. 
   As noted previously, however, when there is a desire to provide various features such as the change or switch of the characteristics or the operation mode of the noise cancellation system or a desire for improved sound quality, the digital form is preferable to the analog form because the above desires are more easily fulfilled by the digital form. Thus, the digital noise cancellation system has great merit. 
   Hereinafter, a structure of a digital noise cancellation system in a headphone device in accordance with one embodiment of the present invention will be described. Despite its digital form, this digital noise cancellation system does not have the above-described delay problem and can be put to practical use. 
     FIG. 2  illustrates an exemplary structure of a noise cancellation system in a headphone device in accordance with a first embodiment of the present invention. Note that, in  FIG. 2 , components having their counterparts in  FIG. 1  are assigned the same reference numerals as those of their counterparts in  FIG. 1 , and descriptions thereof will be omitted or simple descriptions thereof will be provided. 
   In the structure as illustrated in  FIG. 2 , instead of the A/D converter section  13  as shown in  FIG. 1 , an A/D converter section  20  is provided as a part for converting the analog audio signal of the external sound (i.e., the external noise) obtained by the microphone  11  and the amplifier  12  into a digital signal. 
   The A/D converter section  20  is formed as a single chip part, for example, and, as illustrated in  FIG. 2 , includes a delta sigma modulator  21 . The input analog signal is converted by the delta sigma modulator  21  into a digital signal in [64 Fs (=2.8224 MHz), 1 bit] form. Then, the digital signal outputted from the delta sigma modulator  21  is outputted from the A/D converter section  20  and inputted to a noise cancellation-use digital filter  30  in the subsequent stage. 
   As with the noise cancellation-use digital filter  14   a  as shown in  FIG. 1 , the noise cancellation-use digital filter  30  has a function of generating a noise cancellation-use audio signal. That is, using the digital audio signal of the external sound supplied from the A/D converter section  20 , the noise cancellation-use digital filter  30  generates an audio signal corresponding to a sound that has a characteristic for canceling the external sound that will arrive at the ear, corresponding to the driver  17   a , of the user wearing the headphones. 
   Note that the digital audio signal inputted to the noise cancellation-use digital filter  14   a  as shown in  FIG. 1  and the digital audio signal outputted from the noise cancellation-use digital filter  14   a  are both in the [1 Fs, 16 bit] form. On the other hand, the signal inputted to the noise cancellation-use digital filter  30  as shown in  FIG. 2  is in the [64 Fs, 1 bit] form, while the signal outputted from the noise cancellation-use digital filter  30  is in the [64 Fs, 16 bit] form. The noise cancellation-use digital filter  30  can be formed by an FIR digital filter, for example, and accordingly, the signal outputted therefrom is in multi-bit form. In this embodiment, the quantization bit rate is set at 16 bits. The form of the signal outputted from the noise cancellation-use digital filter  30  is determined to be [64 Fs, 16 bit] in order that, as will be understood from the following description, the form of this signal may coincide with the form of the signal of the digital audio source, [64 Fs, 16 bit], with which this signal will be combined. 
   In this embodiment, instead of being contained in the DSP or the like, the noise cancellation-use digital filter  30  is an independent portion and formed as a single part, for example. The cancellation-use audio signal outputted from the noise cancellation-use digital filter  30  is inputted to a D/A converter section  40 . 
   The D/A converter section  40  as shown in  FIG. 2  is also formed as a single part, for example. Similar to the D/A converter section  15  as shown in  FIG. 1 , the D/A converter section  40  includes an oversampling filter  41 , a delta sigma modulator  43 , and an analog LPF  44 . However, unlike the D/A converter section  15  as shown in  FIG. 1 , a combiner  42  is additionally provided between the oversampling filter  41  and the delta sigma modulator  43 . 
   In this embodiment, as illustrated in  FIG. 2 , the signal of the digital audio source is inputted to the oversampling filter  41 . Accordingly, the oversampling filter  41  converts an audio signal component corresponding to the digital audio source from the [1 Fs, 16 bit] form to the [64 Fs, 16 bit] form. 
   Then, the combiner  42  combines the audio signal of the digital audio source with the noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30 , which are both in the [64 Fs, 16 bit] form, and outputs a resultant [64 Fs, 16 bit] digital signal to the delta sigma modulator  43 . 
   The delta sigma modulator  43  accepts input of the [64 Fs, 16 bit] digital signal outputted from the combiner  42 , converts this signal into a [64 Fs, 1 bit] digital signal, and outputs the resultant signal. 
   The digital signal outputted from the delta sigma modulator  43  passes through the analog LPF  44  to be converted into an analog audio signal, and this resultant analog audio signal is outputted from the D/A converter section  40 . 
   The analog audio signal thus obtained is amplified by the power amplifier  16 , and the driver  17   a  is driven by a resultant signal. 
   In accordance with this structure, the signal outputted from the combiner  42  is composed of a combination of the audio signal of the digital audio source and the noise cancellation-use audio signal, and therefore, a sound eventually outputted from the driver  17   a  is composed of a combination of a sound component for canceling the external sound and a reproduced sound of the digital audio source as in the case of  FIG. 1 . That is, the noise cancellation system in accordance with the feedforward system is properly constructed. 
   Concerning the structure of  FIG. 2 , focus will now be placed on a noise processing system in which the external sound is picked up by the microphone  11  and the sound component for canceling the noise is outputted from the driver. Then, it can be said that the output from the delta sigma modulator  21  that forms an A/D conversion part (i.e., the A/D converter section  20 ) is inputted to the noise cancellation-use digital filter  30 , while the output from the noise cancellation-use digital filter  30  is inputted to the delta sigma modulator  43  that forms a D/A conversion part (i.e., the D/A converter section  40 ). 
   Thus, compared to the structure of  FIG. 1 , the noise processing system in the structure of  FIG. 2  does not include the decimation filter on the A/D conversion side or the oversampling filter on the D/A conversion side. 
   As noted previously, in the structure of  FIG. 1 , the delays in the A/D converter section  13  and the D/A converter section  15  are large. Regarding causes of these delays, a delay cased by the decimation filter  13   b  is dominant in the A/D converter section  13  and a delay caused by the oversampling filter  15   a  is dominant in the D/A converter section  15 . 
   The present embodiment has been designed in view of this fact. That is, in order to exclude the influence of the delay caused by the decimation filter on the A/D conversion side and the delay caused by the oversampling filter on the D/A conversion side in the noise processing system, the input of the noise cancellation-use digital filter  30  is directly connected to the delta sigma modulator  21  (i.e., the A/D converter section  20 ) and the output of the noise cancellation-use digital filter  30  is directly connected to the delta sigma modulator  43  (within the D/A converter section  40 ). 
   In this manner, in the noise processing system, the dominant causes of the delays on both the D/A conversion side and the A/D conversion side are eliminated, so that the delay in the noise processing system is significantly reduced. Accordingly, the sound frequency range for which noise cancellation works effectively is significantly enlarged, and as a result, the practically sufficient noise cancellation effect is obtained. That is, the digital noise cancellation system in the headphone device that can be put to practical use is achieved. 
   Moreover, in the present embodiment, the noise cancellation-use digital filter  30  is so constructed as to reduce the delay to achieve a more excellent noise cancellation effect. 
   Exemplary structures of the noise cancellation-use digital filter  30  that causes the reduced delay will now be described below. 
   First, in the case where an FIR digital filter (i.e., an FIR filter) is normally adopted as the noise cancellation-use digital filter  30 , a structure as illustrated in  FIG. 3A  is adopted. 
   Specifically, referring to  FIG. 3A , in the case where the noise cancellation-use digital filter  30  is formed as an 8-tap FIR filter, seven delay devices D 1  to D 7  are connected in series to form a shift register. In addition, coefficient multipliers h 0  to h 7  and an adder P are provided. The coefficient multipliers h 0  to h 7  receive outputs from the shift register, i.e., data inputted to the delay device D 1  and data outputted from the delay devices D 1  to D 7 , respectively, and multiply the received data by a predetermined coefficient. The adder P adds outputs from these coefficient multipliers h 0  to h 7  together. Since the digital signal inputted to the noise cancellation-use digital filter  30  is in the [64 Fs, 1 bit] form, the delay devices D 1  to D 7  and the coefficient multipliers h 0  to h 7  are configured to accept input of 1-bit signals. Since the digital signal outputted from the noise cancellation-use digital filter  30  should be in the [64 Fs, 16 bit] form, 16-bit coefficients are set in the coefficient multipliers h 0  to h 7  so that the outputs from the coefficient multipliers h 0  to h 7  will be 16-bit data, and these outputs are added together by the adder P. 
   It can be said that, in accordance with the structure as illustrated in  FIG. 3A , 8-bit data, i.e., an arrangement of the data inputted to the delay device D 1  and the data outputted from the delay devices D 1  to D 7 , is converted into a 16-bit bit pattern that is linearly associated with a bit pattern of the 8-bit data, and the 16-bit bit pattern is outputted. Based on this fact, the noise cancellation-use digital filter  30  can also be constructed of the delay devices D 1  to D 7  and a ROM  60  as illustrated in  FIG. 3B . 
   In  FIG. 3B , 8-bit data is constructed of 1-bit data inputted to the delay device D 1  and seven pieces of 1-bit data outputted from the delay devices D 1  to D 7 , respectively, which can be considered as being outputted from the shift register at the same time, and this 8-bit data is used to specify an address in the ROM  60 . Since there are 256 bit patterns that can be expressed by 8 bits, addresses 0 to 255 are set in the ROM  60 . In the ROM  60 , appropriate 16-bit bit patterns are stored so as to be associated with the addresses 0 to 255. 
   In accordance with the above structure, for each sample, one of the addresses 0 to 255 is specified for the ROM  60  and data of a 16-bit bit pattern corresponding to the specified address is read from the ROM  60 . The 16-bit data thus read is outputted from the noise cancellation-use digital filter  30  in the present embodiment. 
   In accordance with the above structure, the coefficient multipliers h 0  to h 7  and the adder P as illustrated in  FIG. 3A  are omitted, and processes performed by the coefficient multipliers h 0  to h 7  and the adder P are realized by reading the data of the 16-bit bit pattern from the specified address in the ROM  60 . Thus, circuitry is simplified. 
   The noise cancellation-use digital filter  30  that causes the reduced delay can also be realized by being formed as a minimum phase shift filter, for example. This can be realized by, with the structure as illustrated in  FIG. 3A , for example, setting a pattern of the coefficients set in the coefficient multipliers h 0  to h 7  so as to form a minimum phase shift filter. Alternatively, the noise cancellation-use digital filter  30  may be formed by an infinite impulse response (IIR) digital filter. One characteristic of the IIR filter is that a delay amount is small as a result. 
   In the present embodiment, the sampling frequency of the signal outputted from the noise cancellation-use digital filter  30  is set as follows. 
   First, the D/A converter section  40  is configured to convert a digital audio signal as a PCM signal in the [1 Fs, 16 bit] form into an analog signal, and the oversampling filter converts the signal into the [64 Fs, 16 bit] form. That is, the sampling frequency of the signal obtained after oversampling is set at 64 Fs. Accordingly, the delta sigma modulator  43 , which follows the oversampling filter, is configured to convert the signal in the [64 Fs, 16 bit] form into a 1-bit signal. Thus, the output from the delta sigma modulator  43  is in the [64 Fs, 1 bit] form. 
   Moreover, in the present embodiment, the noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30  is directly inputted to the delta sigma modulator  43  in the D/A converter section  40 , without passing through the oversampling filter. This is why the noise cancellation-use audio signal should be in a [sampling frequency, quantization bit rate] form corresponding to that of the signal inputted to the delta sigma modulator  43  (and outputted from the oversampling filter). Thus, the cancellation-use audio signal outputted from the noise cancellation-use digital filter  30  as illustrated in  FIG. 2  is in the [64 Fs, 16 bit] form. Regarding the sampling frequency, the noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30  should have the same sampling frequency as the signal outputted from the delta sigma modulator  43 . 
   In the present embodiment, the sampling frequency after oversampling, i.e., the sampling frequency of the signal (i.e., the noise cancellation-use audio signal) outputted from the noise cancellation-use digital filter  30 , is assumed to be 64 Fs. However, the present invention is not limited to this. This sampling frequency should be greater than the sampling frequency, 1 Fs, of the PCM signal of the digital audio source, but as long as it is, any frequency value that allows the reproduced sound to have a sufficient quality, for example, may be set as the above sampling frequency. More specifically, on the assumption that the sampling frequency of the PCM signal of the digital audio source is 1 Fs, the sampling frequency of the noise cancellation-use audio signal (i.e., the sampling frequency after oversampling) is set at 2 n  Fs where n is a natural number greater than 0, for example. In practice, it is desirable that this sampling frequency be set at 4 Fs or higher. 
   Next, an exemplary structure of a noise cancellation system in accordance with a second embodiment of the present invention will now be described below with reference to  FIG. 4 . Note that, in  FIG. 4 , components having their counterparts in  FIG. 2  are assigned the same reference numerals as those of their counterparts in  FIG. 2 , and descriptions thereof will be omitted. 
   First, a basic structure of the second embodiment will now be described below. 
   In  FIG. 4 , broadly speaking, a D/A converter section  40 A includes the oversampling filter  41 , the combiner  42 , the delta sigma modulator  43 , a pulse width modulation (PWM) modulator  45 , and the analog LPF  44 . That is, compared to the D/A converter section  40  as illustrated in  FIG. 2 , the PWM modulator  45  is additionally inserted between the delta sigma modulator  43  and the analog LPF  44 . 
   Moreover, the oversampling filter  41  in the D/A converter section  40 A is configured to accept input of a [1 Fs, 16 bit] signal of the digital audio source and convert this signal into [16 Fs, 16 bit] form. 
   Thus, the combiner  42  in the D/A converter section  40 A combines a [16 Fs, 16 bit] digital signal with another [16 Fs, 16 bit] digital signal. That is, a noise cancellation-use audio signal outputted from a noise cancellation-use digital filter  30  should be not in the [64 Fs, 16 bit] form as in  FIG. 2  but in the [16 Fs, 16 bit] form. 
   Thus, in this embodiment, the noise cancellation-use digital filter  30  is configured to give the input signal a characteristic as a noise cancellation-use audio signal, and perform a decimation process so that the input signal with a sampling frequency of 64 Fs will be outputted with a sampling frequency of 16 Fs. In other words, the noise cancellation-use digital filter  30  is configured to have a function as a noise cancellation-use filter and an additional function as a decimation filter  30   a . While some structures are conceivable for fulfilling both the functions, one of the most efficient structures is to cause the noise cancellation-use digital filter to function as the decimation filter as well, taking advantage of the fact that the noise cancellation-use digital filter has a characteristic of an LPF. The decimation filter also has the characteristic of the LPF. 
   The combiner  42  combines the signal of the digital audio source, which has been subjected to oversampling by the oversampling filter  41  to have the [16 Fs, 16 bit] form, with the [16 Fs, 16 bit] noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30 , and a resultant signal is inputted to the delta sigma modulator  43 . 
   In this embodiment, the delta sigma modulator  43  converts the input signal into a [16 Fs, 5 bit] signal with a quantization bit rate of 5 bits instead of 1 bit. This [16 Fs, 5 bit] signal is inputted to the PWM modulator  45  and subjected to PWM modulation therein, and is allowed to pass through the analog LPF  44  to be converted into an analog audio signal, which is outputted from the D/A converter section  40 A. That is, a D/A conversion part of the second embodiment has a structure in accordance with a structure of a class-D amplifier. 
   Variants of the second embodiment are conceivable as follows. 
   For example, referring to  FIG. 4 , the oversampling filter  41  may be configured to include multiple upsampling circuits  46   a  to  46   d  connected in series. In this example, each of the upsampling circuits  46   a  to  46   d  is configured to double the sampling frequency, and since four such upsampling circuits are connected in series, the input signal in the [1 Fs, 16 bit] form is converted into the [16 (=2×2×2×2) Fs, 16 bit] form and thus outputted from the oversampling filter  41 . 
   Moreover, the noise cancellation-use digital filter  30  is configured to convert the input signal with a sampling frequency of 64 Fs into a 16-bit signal with a sampling frequency lower than 16 Fs, such as 8 Fs, 4 Fs, or 2 Fs, by means of the decimation filter  30   a , and the resultant signal is outputted from the noise cancellation-use digital filter  30 . This signal is inputted to an appropriate one of the upsampling circuits in the oversampling filter  41  in accordance with the sampling frequency of this signal. 
   In the case where the signal outputted from the noise cancellation-use digital filter  30  is in [8 Fs, 16 bit] form, for example, a combiner  47   c  is provided in front of the upsampling circuit  46   d  in the oversampling filter  41 , and the combiner  47   c  combines the signal outputted from the noise cancellation-use digital filter  30  with the signal outputted from the upsampling circuit  46   c  to output a resultant signal to the upsampling circuit  46   d . In accordance with this structure, the combiner  47   c  combines the signal of the digital audio source as upsampled to [8 Fs, 16 bit] with the signal, which is also in the [8 Fs, 16 bit] form, outputted from the noise cancellation-use digital filter  30 . Then, the resultant signal passes through the upsampling circuit  46   d  to finally become an [16 Fs, 16 bit] audio signal, which is inputted to the delta sigma modulator  43  (in this case, the combiner  42  can be omitted). 
   Similarly, in the case where the signal outputted from the noise cancellation-use digital filter  30  is in [4 Fs, 16 bit] form, a combiner  47   b  is provided in front of the upsampling circuit  46   c  in the oversampling filter  41 , and the combiner  47   b  combines the signal outputted from the noise cancellation-use digital filter  30  with the signal outputted from the upsampling circuit  46   b  to output a resultant signal to the upsampling circuit  46   c.    
   In the case where the signal outputted from the noise cancellation-use digital filter  30  is in [2 Fs, 16 bit] form, a combiner  47   a  is provided in front of the upsampling circuit  46   b  in the oversampling filter  41 , and the combiner  47   a  combines the signal outputted from the noise cancellation-use digital filter  30  with the signal outputted from the upsampling circuit  46   a  to output a resultant signal to the upsampling circuit  46   b.    
   In the above variants, the number of operation steps per sampling period is increased, for example, and therefore, in the case where a necessary amount of operation in one sampling period in the noise cancellation-use digital filter  30  has been increased, a desired filter characteristic can be achieved without the need to increase a clock frequency of the system. 
   Note that it has been noted concerning the first embodiment that the sampling frequency of the noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30  should be the same as the sampling frequency of the signal handled by the delta sigma modulator  43  in the D/A converter section  40 . In the above-described variants, however, the sampling frequency of the noise cancellation-use audio signal is lower than the sampling frequency of the signal handled by the delta sigma modulator  43 . If the upsampling circuit(s) within the oversampling filter  41  through which the noise cancellation-use audio signal passes is regarded as a component of the noise cancellation-use digital filter, however, it can be said that the sampling frequency of the noise cancellation-use audio signal is the same as the sampling frequency of the signal handled by the delta sigma modulator  43  in the D/A converter section  40 A. 
   In the structures of the above variants, the noise cancellation-use audio signal passes through a part of the oversampling filter  41 , resulting in an additional delay compared to when the noise cancellation-use audio signal does not pass through the oversampling filter  41  at all, for example. However, compared to the structure of  FIG. 1 , in which the noise cancellation-use audio signal passes throughout the oversampling filter  15   a , the effect of reduced delay in the D/A converter section is achieved. 
   Next, an exemplary structure of a third embodiment of the present invention will now be described below with reference to  FIG. 5 . Note that, in  FIG. 5 , components that have their counterparts in  FIGS. 2 and 4  are assigned the same reference numerals as those of their counterparts, and descriptions thereof will be omitted. 
   Compared to the structure of the first embodiment as illustrated in  FIG. 2 , a noise cancellation system as illustrated in  FIG. 5  additionally includes a level adjuster  51 , a noise analyzer  52 , and a level detector  53 . With this structure, a noise cancellation operation is performed in accordance with contents of the external sound and the signal of the digital audio source, and so on as described below. 
   The level adjuster  51  is inserted between the output of the noise cancellation-use digital filter  30  and the input of the combiner  42 . That is, the level adjuster  51  accepts input of the audio signal outputted from the noise cancellation-use digital filter  30 , adjusts a level of the audio signal, and outputs the level-adjusted audio signal to the combiner  42 . 
   The digital audio signal of the external sound outputted from the A/D converter section  20  is inputted to both the noise cancellation-use digital filter  30  and the noise analyzer  52 . The noise analyzer  52  performs an analysis process on the digital audio signal concerning a tone color, a tone quality, a level, and so on of the external sound as noise. Based on a result of this analysis, the noise analyzer  52  determines an optimum coefficient in the noise cancellation-use digital filter  30  and an optimum level of the noise cancellation-use audio signal, and, based on a result of this determination, outputs a coefficient control signal Sc 1  to the noise cancellation-use digital filter  30  to instruct the noise cancellation-use digital filter  30  to set the determined coefficient, and outputs a signal level control signal Sc 2  for specifying the determined level of the noise cancellation-use audio signal to the level adjuster  51 . 
   Meanwhile, the signal of the digital audio source is inputted to both the D/A converter section  40  and the level detector  53 , and the level detector  53  detects a level of the input signal. As to a technique for detecting the level, the level detector  53  may detect absolute values of the audio signal and determine an envelope obtained by the absolute values of the detected levels to be a detected level, for example. Then, based on a result of this detection, the level detector  53  determines an optimum level of the noise cancellation-use audio signal so that sound of the signal of the digital audio source will be heard excellently, and outputs a signal level control signal Sc 3  for specifying the determined level to the level adjuster  51 . Note that the level of the noise cancellation-use audio signal thus determined has a value that will not cause data overflow when the noise cancellation-use audio signal is combined with the signal of the digital audio source. 
   In accordance with the control signals thus outputted, the noise cancellation-use digital filter  30  changes the coefficient, and the level adjuster  51  adjusts the level of the noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30 . As a result, in accordance with variation in the condition of the external sound and variation in the level of the signal of the digital audio source, the optimum coefficient of the noise cancellation-use digital filter  30  and the optimum level of the noise cancellation-use audio signal are set, so that a nearly optimum noise cancellation effect is obtained at all times. 
   Note that the level detector  53  may alternatively accept input of the signal outputted from the oversampling filter  41  and detect a level of this signal. 
     FIG. 6  illustrates an exemplary structure of a fourth embodiment of the present invention. Note that, in  FIG. 6 , components that have their counterparts in  FIGS. 2 ,  4 , and  5  are assigned the same reference numerals as those of their counterparts, and descriptions thereof will be omitted. 
   In the fourth embodiment, a digital microphone  70  is adopted as a part for picking up the external sound and converting the external sound into a digital audio signal. 
   The digital microphone  70  is formed as a single part, for example. As illustrated in  FIG. 6 , the digital microphone  70  includes a microphone  71 , an amplifier  72 , and a delta sigma modulator  73 . In functional terms, the microphone  71  and the amplifier  72  are equivalent to the microphone  11  and the amplifier  12  as illustrated in  FIG. 2 , for example, and are used to obtain an analog audio signal of the external sound. The analog audio signal thus obtained is inputted to the delta sigma modulator  73  and converted therein into a [64 Fs, 1 bit] digital signal, and this signal is outputted from the digital microphone  70 . This signal outputted from the digital microphone  70  is inputted to the noise cancellation-use digital filter  30 . Physically, the digital microphone  70  as described above is provided on the housing of the headphone device  17  such that the microphone  71  is capable of picking up the external sound. 
     FIG. 7  illustrates an exemplary structure of a fifth embodiment of the present invention. Note that, in  FIG. 7 , components that have their counterparts in  FIGS. 2 ,  4 ,  5 , and  6  are assigned the same reference numerals as those of their counterparts, and descriptions thereof will be omitted. 
   It has been assumed in the above-described embodiments that the digital audio source is the CD or the like which provides the PCM digital audio signal in the [1 Fs, 16 bit] form. The [1 Fs, 16 bit] form of the digital audio signal is one predominant form today, but besides, signals in a so-called direct stream digital (DSD) format, such as digital audio signals in the [64 Fs, 1 bit] form recorded on a Super Audio CD (SACD) or the like, which correspond to a signal after delta-sigma modulation, have come to be handled as substance of audio content. The structure in accordance with the fifth embodiment corresponds to the case where the digital audio source provides such a signal in the DSD format. 
   The digital audio source as illustrated in  FIG. 7  provides a signal in a [64 Fs, 1 bit] DSD format. A bit extender  81  is provided so that this signal can be combined by a combiner  82  with a [64 Fs, 16 bit] noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30 . The bit extender  81  accepts input of the [64 Fs, 1 bit] signal from the digital audio source, performs a bit extension process for extending the quantization bit rate to 16 bits, thereby converting the signal into a [64 Fs, 16 bit] signal, and outputs the resultant signal to the combiner  82 . 
   Note that the above bit extension process performed by the bit extender  81  refers to a process of converting a 1-bit signal in the DSD format, for example, i.e., a signal that can take two values, 1 and 0, into a 16-bit signal, 0x0400 (0.5) or 0xC000 (−0.5). Therefore, the bit extender  81  can be formed by a digital filter having the characteristic of the LPF, and further, the bit extender  81  may have the structure as illustrated in  FIG. 3B , which uses the ROM. 
   A signal resulting from the combining of the above two signals by the combiner  82  is inputted to a D/A converter section  40 B. Compared to the D/A converter section  40  as illustrated in  FIG. 2 , for example, the D/A converter section  40 B does not include the oversampling filter. The combiner  82  as illustrated in  FIG. 7  corresponds to the combiner  42  within the D/A converter section  40  in  FIG. 2 , but in this embodiment, the combiner  82  is an independent part and is not contained in the D/A converter section  40 B. 
   The audio signal which is composed of the combination of the signal of the digital audio source and the cancellation-use audio signal and outputted from the combiner  82  passes through the delta sigma modulator  43  and the LPF within the D/A converter section  40 B to be converted into an analog signal, and this analog signal is outputted to the power amplifier  16 . 
     FIG. 8  illustrates an exemplary structure of a sixth embodiment of the present invention. Note that, in  FIG. 8 , components having their counterparts in  FIGS. 2 ,  4 ,  5 ,  6 , and  7  are assigned the same reference numerals as those of their counterparts, and descriptions thereof will be omitted. 
   Noise cancellation systems in headphone devices are broadly classified into the feedforward system and the feedback system. The first to fifth embodiments described above are based on the feedforward system. However, the present invention is also applicable to the feedback system, instead of the feedforward system. As such, the exemplary structure in accordance with the sixth embodiment is based on the feedback system. 
   As schematically shown in  FIG. 8 , in the case of the feedback system, the microphone  11  is so provided as to pick up a sound outputted from the driver  17   a  near the ear of the user wearing the headphones. A sound picked up in this case contains the sound outputted from the driver and an external sound component that has intruded into the housing of the headphone device and is arriving at the ear of the user wearing the headphone device, for example. A signal of the sound thus picked up is amplified by the amplifier  12  to become an analog audio signal, and this analog audio signal is converted into a [64 Fs, 1 bit] digital audio signal by the delta sigma modulator  21  within the A/D converter section  20 , and the resultant digital audio signal is inputted to the noise cancellation-use digital filter  30 . 
   The noise cancellation-use digital filter  30  gives a necessary characteristic to the input signal, for example, to generate, as the noise cancellation-use audio signal, an audio signal of a sound having a characteristic for canceling the external sound that will arrive at the ear, corresponding to the driver  17   a , of the user wearing the headphones. This process is generally a process of giving a transfer function −β for noise cancellation to the signal of the sound picked up. Then, the noise cancellation-use audio signal generated is inputted to the combiner  42  provided at the back of the oversampling filter  41  within a D/A converter section  40 C. 
   Compared to the D/A converter section  40  as illustrated in  FIG. 2 , the D/A converter section  40 C additionally includes an equalizer  48 , which is provided in front of the oversampling filter  41 . The equalizer  48  gives a characteristic based on a transfer function 1+β to the signal of the digital audio source. In the case of the feedback system, the noise cancellation-use audio signal outputted from the noise cancellation-use digital filter  30  contains a component corresponding to the external sound and also a component corresponding to the sound of the digital audio source outputted from the driver  17   a  and picked up by the microphone  11 . That is, a characteristic corresponding to a transfer function 1/1+β is given to the component of the sound of the digital audio source. Accordingly, the equalizer  48  is configured to give the characteristic based on the transfer function 1+β, which is the inverse of 1/1+β, to the signal of the digital audio source in advance. Thus, when the signal outputted from the oversampling filter  41  is combined with the noise cancellation-use audio signal by the combiner  42 , the above transfer characteristic 1/1+β is cancelled. Thus, a signal outputted from the combiner  42  is composed of a combination of a signal component having a characteristic for canceling the external sound and a signal component corresponding to the original signal of the digital audio source. Then, the signal outputted from the combiner  42  passes through the delta sigma modulator  43  and the analog LPF  44  to be converted into an analog audio signal, and this analog audio signal is amplified by the power amplifier  16 , and the driver  17   a  is driven by a resulting signal to output a corresponding sound. 
   As described above, in the case of the feedback system, the sound outputted from the driver and the external sound component that has intruded into the housing of the headphones are picked up near the ear of the user wearing the headphones to generate the signal used for noise cancellation. Then, this signal used for noise cancellation is outputted from the driver so as to involve negative feedback. As a result, a sound that contributes to canceling the external sound to relatively emphasize the sound of the digital audio source will reach the ear, corresponding to the driver  17   a , of the user wearing the headphone device. 
   Note that, in the above-described embodiments, the A/D converter section, the noise cancellation-use digital filter, the D/A converter section, and so on are independent parts, and the combination of these parts forms the noise cancellation system. However, all or some of these parts may be integrated into a single part, for example. 
   It is assumed in the above-described embodiments that the sound that is originally to be heard is the sound of the digital audio source, i.e., a digitized audio signal in a certain form. Specifically, as noted previously, it is assumed in the above-described embodiments that the sound that is originally to be heard is the sound of the digital audio signal recorded on the CD, the SACD, or the like, for example. It is needless to say, however, that the sound that is originally to be heard may initially be in the form of an analog audio signal. In this case, this analog audio signal is converted into a digital signal via A/D conversion, and this digital signal is inputted to the D/A converter section  40  ( 40 B,  40 C) as the signal of the digital audio source in the above-described embodiments, for example. 
   Note that the sampling frequency and the quantization bit rate handled by each of the digital signal processing blocks in the noise cancellation system are not necessarily identical between the above-described embodiments. As will be understood from this fact, as long as the noise cancellation system can be formed properly, the sampling frequency and the quantization bit rate handled by each of the digital signal processing blocks in the noise cancellation system may be changed as necessary. 
   Also note that, in the above-described embodiments, the noise cancellation system is based on either the feedforward system or the feedback system. However, the structures in accordance with the above-described embodiments can also be applied to a noise cancellation system in accordance with a combination of the feedforward system and the feedback system. Such a noise cancellation system can be achieved, for example, by adding, to the structure as illustrated in  FIG. 8 , a noise cancellation-use signal processing system in accordance with the feedforward system as composed of the microphone  11 , the amplifier  12 , the A/D converter section  20 , and the noise cancellation-use digital filter  30  as illustrated in  FIG. 2 , for example. In this case, the signal outputted from the noise cancellation-use digital filter  30  corresponding to the feedforward system is additionally combined with the signal of the digital audio source by the combiner  42  as illustrated in  FIG. 8 . 
   No specific mention has been made about how the parts for signal processing that form the noise cancellation systems in accordance with the above-described embodiments are implemented. The manner of implementation may be determined arbitrarily depending on the structure, use, or the like of an apparatus or system to which a noise cancellation system in accordance with the present invention is applied. 
   For example, in the case where a headphone device that fulfills a noise cancellation function by itself is constructed, most of the parts that form the noise cancellation system may be contained in the housing of the headphone device. In the case where a noise cancellation system is formed by a combination of a headphone device and an external device such as an adapter, at least one part other than the microphone and the driver may be provided in the external device such as the adapter. 
   Further, in the case where a noise cancellation system is implemented on a mobile phone device, a network audio communication device, an audio player, or the like that is configured to reproduce audio content and output the reproduced content to a headphones terminal, for example, at least one part other than the microphone and the driver may be provided in such a device. 
   Also note that it is assumed in the above-described embodiments that an audio signal having a signal characteristic for canceling the noise is generated in the noise cancellation-use digital filter. However, an inverting amplifier may be adopted as the amplifier  12 , and the noise cancellation-use digital filter may be formed as a digital filter having a desired frequency characteristic, such as an LPF, for example. In this case also, an equivalent noise cancellation-use signal can be obtained. 
   It should be understood by those skilled in the art that various modifications, combinations, sub-combinations and alterations may occur depending on designs and other factors insofar as they are within the scope of the appended claims or the equivalents thereof.