Abstract: In an active muffler having improved response characteristics, a speaker section includes a diaphragm adapted to generate sound, a voice coil for driving the diaphragm, and a distance sensor to detect the movement of the diaphragm. A light generated by the LED is reflected by the diaphragm, the reflected light is detected by a phototransistor to thereby measure the distance to the diaphragm, so that the movement of the diaphragm is detected. Noise is detected by a microphone, and a signal having opposite phase to that of the noise is generated by an opposite-phase generating section. The difference between the opposite-phase signal and the signal of the distance to the speaker from the distance sensor is calculated and inputted to a PID control section. Such a difference indicates the delay of the speaker movement. Feedback control is performed in a direction in which the difference is canceled out.

Abstract: A high frequency signal interpolation apparatus provides, with a simple structure, a high-quality digital audio signal through interpolation of high frequency signals missing due to compression. The high frequency signal interpolation apparatus includes a peak value detection and holding circuit configured to detect a peak value of a digital audio signal provided to an input terminal by sampling the digital audio signal and generate a square wave signal by holding the detected peak value; a high-pass filter configured to extract a higher harmonic component from the generated square wave signal; and an adder configured to add the extracted higher harmonic component to the digital audio signal provided to the input terminal.

Abstract: A signal corresponding to a short-period change and a signal corresponding to a long-period change of a sound signal are detected, and optimal quantization is performed based on the combination of the two signals. In an ADPCM encoding apparatus (100), a differential value dn between a 16-bit input signal Xn and a decoded signal Yn-1 of one sample ago is calculated by a subtractor (102). Thereafter, the 16-bit differential value dn is adaptively quantized by an adaptive quantizing section (103), so as to be converted to a (1 to 8)-bit length-variable ADPCM value Dn. Thereafter, the ADPCM value Dn is compression-encoded by a compression-encoding section (108) to generate a signal D?n, and the signal D?n is framed by a framing section (130) and outputted. Further, in an ADPCM decoding apparatus, a framed input signal is subjected to a reverse of the aforesaid process so as to be decoded.

Abstract: A favorable high-frequency signal is generated and practical high-frequency signal interpolation is implemented through simple processing. A digital audio signal reproduced by an instrument, which also carries out compression, is supplied as an original signal to an input terminal 1, and this original signal is then supplied to a digital sample and hold circuit 3 via a band-pass filter 2. The signal from the digital sample and hold circuit 3 is supplied to a ±1 multiplier 6, which then alternately inverts sign bits. The harmonic components of this signal in which the sign bits are inverted alternately are extracted by a high-pass filter (HPF) 7. Meanwhile, the original signal from the input terminal 1 is supplied to a delay circuit 8 equivalent to the processing time consumed by the aforementioned digital sample and hold circuit 3 and related circuits, forming an adjusted, delayed signal.

Abstract: An acoustic characteristic control apparatus supplies music signal, for example, to input terminal connected to a band-pass filter and a peaking filter. In a zero-cross detection circuit, a pulse signal corresponding to a period while a signal is positive is formed. A pulse-width measuring circuit output a signal corresponding to a pulse width. Next, the output of the pulse-width measuring circuit is inputted to one comparator and another comparator. The one comparator discriminates a time when the pulse width is equal to or larger than a first setting value, and the another comparator discriminates a time when the pulse width is equal to or smaller than a second setting value. The comparator is connected to the up terminal and the down terminal of an up/down counter. The output of the up/down counter is connected to the peaking filter through the subtractor, and acoustic characteristics of the peaking filter is controlled according to the count value of the up/down counter.

Abstract: A signal corresponding to a short-period change and a signal corresponding to a long-period change of a sound signal are detected, and optimal quantization is performed based on the combination of the two signals. In an ADPCM encoding apparatus (100), a differential value dn between a 16-bit input signal Xn and a decoded signal Yn-1 of one sample ago is calculated by a subtractor (102). Thereafter, the 16-bit differential value dn is adaptively quantized by an adaptive quantizing section (103), so as to be converted to a (1 to 8)-bit length-variable ADPCM value Dn. Thereafter, the ADPCM value Dn is compression-encoded by a compression-encoding section (108) to generate a signal D?n, and the signal D?n is framed by a framing section (130) and outputted. Further, in an ADPCM decoding apparatus, a framed input signal is subjected to a reverse of the aforesaid process so as to be decoded.

Abstract: An acoustic characteristic control apparatus supplies music signal, for example, to input terminal connected to a band-pass filter and a peaking filter. In a zero-cross detection circuit, a pulse signal corresponding to a period while a signal is positive is formed. A pulse-width measuring circuit output a signal corresponding to a pulse width. Next, the output of the pulse-width measuring circuit is inputted to one comparator and another comparator. The one comparator discriminates a time when the pulse width is equal to or larger than a first setting value, and the another comparator discriminates a time when the pulse width is equal to or smaller than a second setting value. The comparator is connected to the up terminal and the down terminal of an up/down counter. The output of the up/down counter is connected to the peaking filter through the subtractor, and acoustic characteristics of the peaking filter is controlled according to the count value of the up/down counter.

Abstract: In an active muffler having improved response characteristics, a speaker section includes a diaphragm adapted to generate sound, a voice coil for driving the diaphragm, and a distance sensor to detect the movement of the diaphragm. A light generated by the LED is reflected by the diaphragm, the reflected light is detected by a phototransistor to thereby measure the distance to the diaphragm, so that the movement of the diaphragm is detected. Noise is detected by a microphone, and a signal having opposite phase to that of the noise is generated by an opposite-phase generating section. The difference between the opposite-phase signal and the signal of the distance to the speaker from the distance sensor is calculated and inputted to a PID control section. Such a difference indicates the delay of the speaker movement. Feedback control is performed in a direction in which the difference is canceled out.

Abstract: A favorable high-frequency signal is generated and practical high-frequency signal interpolation is implemented through simple processing. A digital audio signal reproduced by an instrument, which also carries out compression, is supplied as an original signal to an input terminal 1, and this original signal is then supplied to a digital sample and hold circuit 3 via a band-pass filter 2. The signal from the digital sample and hold circuit 3 is supplied to a ±1 multiplier 6, which then alternately inverts sign bits. The harmonic components of this signal in which the sign bits are inverted alternately are extracted by a high-pass filter (HPF) 7. Meanwhile, the original signal from the input terminal 1 is supplied to a delay circuit 8 equivalent to the processing time consumed by the aforementioned digital sample and hold circuit 3 and related circuits, forming an adjusted, delayed signal.

Abstract: A quality high frequency signal is generated through simple processing, and practical high frequency signal interpolation is carried out. A digital audio signal reproduced by an apparatus, which carries out compression, is provided to an input terminal 1 as an original signal. This original signal is sent to a peak value detection and holding circuit 2, which detects and holds a peak value and then generates a square wave signal; wherein a higher harmonic component is included in the square wave signal. This higher harmonic component is extracted by a high pass filter (HPF) 3. On the other hand, the original signal from the input terminal 1 is sent to a delay circuit 4, which delays it for a time equivalent to the processing time of the peak value detection and holding circuit 2 described above. The resulting delayed, aligned signal is sent to a low pass filter (LPF) 5, which then generates a high frequency component-removed signal.