Abstract: A framing processing unit converts a first digital music signal quantized with a first number of quantization bits, into frames. A difference signal calculation unit calculates a first difference signal in which a difference sample is a difference value between two adjacent samples. A flat area detection unit detects a flat area in the first difference signal. A flat area correction unit generate a second difference signal in which the flat area is altered to not be flat. A difference signal averaging unit calculates a difference average value and subtracts the difference average value from each sample value of the second difference signal to generate a third difference signal. A requantization error generation unit generates a requantization error signal. An addition unit adds the requantization error signal to the first digital music signal and outputs a second digital music signal having a second number of quantization bits.

Abstract: An even-numbered harmonic adder adds a correction value to a first adjacent sample that is the next sample after a first local minimum sample, and subtracts a correction value from a second adjacent sample that is one sample before a local maximum sample. The even-numbered harmonic adder adds a correction value to a third adjacent sample that is the next sample after the local maximum sample, and subtracts a correction value from a fourth adjacent sample that is one sample before a second local minimum sample. An odd-numbered harmonic adder subtracts a correction value from the first local minimum sample and the second local minimum sample, and adds a correction value to the local maximum sample.

Abstract: A local extremum calculator detects a local maximum sample and a local minimum sample of a digital audio signal. A number-of-sample detector detects a sample interval between the local maximum sample and the local minimum sample. A difference value calculator calculates difference values between adjacent samples. A correction value calculator calculates a first correction value by multiplying the difference value between the local maximum sample and a first adjacent sample by a coefficient and calculates a second correction value by multiplying the difference value between the local minimum sample and a second adjacent sample by the coefficient. When a periodic signal detector detects that the digital audio signal is a single sine wave, an adder/subtractor does not add the first correction value to the first adjacent sample, and does not subtract the second correction value from the second adjacent sample.

Abstract: An even-numbered harmonic adder adds a correction value to a first adjacent sample that is the next sample after a first local minimum sample, and subtracts a correction value from a second adjacent sample that is one sample before a local maximum sample. The even-numbered harmonic adder adds a correction value to a third adjacent sample that is the next sample after the local maximum sample, and subtracts a correction value from a fourth adjacent sample that is one sample before a second local minimum sample. An odd-numbered harmonic adder subtracts a correction value from the first local minimum sample and the second local minimum sample, and adds a correction value to the local maximum sample.

Abstract: A local extremum calculator detects a local maximum sample and a local minimum sample of a digital audio signal. A number-of-sample detector detects a sample interval between the local maximum sample and the local minimum sample. A difference value calculator calculates difference values between adjacent samples. A correction value calculator calculates a first correction value by multiplying the difference value between the local maximum sample and a first adjacent sample by a coefficient and calculates a second correction value by multiplying the difference value between the local minimum sample and a second adjacent sample by the coefficient. When a periodic signal detector detects that the digital audio signal is a single sine wave, an adder/subtractor does not add the first correction value to the first adjacent sample, and does not subtract the second correction value from the second adjacent sample.

Abstract: A filter generation device according to this embodiment is configured to generate a filter for performing an out-of-head localization process on a high resolution digital audio signal, the filter generation device including: microphones; and a filter generation unit configured to generate a filter corresponding to a transfer characteristic from left and right speakers to the left and right microphones. A predetermined frequency lower than a Nyquist frequency of a sound pickup signal is assumed as a first frequency. The filter generation unit sets an amplitude component in a low frequency band of the filter corresponding to a frequency amplitude characteristic of the sound pickup signal, and the filter generation unit generates an amplitude component in a high frequency band of the filter such that the amplitude component in the high frequency band is allowed to be connected to the amplitude component in the low frequency band.

Abstract: A filter generation device according to this embodiment is configured to generate a filter for performing an out-of-head localization process on a high resolution digital audio signal, the filter generation device including: microphones; and a filter generation unit configured to generate a filter corresponding to a transfer characteristic from left and right speakers to the left and right microphones. A predetermined frequency lower than a Nyquist frequency of a sound pickup signal is assumed as a first frequency. The filter generation unit sets an amplitude component in a low frequency band of the filter corresponding to a frequency amplitude characteristic of the sound pickup signal, and the filter generation unit generates an amplitude component in a high frequency band of the filter such that the amplitude component in the high frequency band is allowed to be connected to the amplitude component in the low frequency band.

Abstract: A waveform correction processor corrects the waveform of a first digital audio signal (a CD signal, for example) having a first sampling frequency. A bit depth and sampling frequency converter converts the first digital audio signal with the waveform corrected by the first waveform correction processor to a second digital audio signal (a high-resolution digital audio signal, for example) having a second sampling frequency, which is higher than the first sampling frequency. The waveform correction processor corrects the waveform of the second digital audio signal.

Abstract: A waveform correction processor corrects the waveform of a first digital audio signal (a CD signal, for example) having a first sampling frequency. A bit depth and sampling frequency converter converts the first digital audio signal with the waveform corrected by the first waveform correction processor to a second digital audio signal (a high-resolution digital audio signal, for example) having a second sampling frequency, which is higher than the first sampling frequency. The waveform correction processor corrects the waveform of the second digital audio signal.

Abstract: A video-audio recording and reproducing apparatus (101) has a built-in stereo microphone (21a, 21b) and an external microphone connection terminal (32). The external microphone connection terminal (32) is connected to a binaural microphone (3) to be attached to the ears of a photographer (300). When the binaural microphone (3) is used to collect ambient sounds, an audio signal to be recorded on a recording medium is switched from an audio signal from the built-in stereo microphone (21a, 21b) to a binaural audio signal from the binaural microphone (3). The photographer (300) puts the binaural microphone (3 (31a, 31b)) on his or her ears and collects ambient sounds around the photographer (300) including a sound emanating from an object. The object is photographed with a camera unit (11). The recording medium records the binaural audio signal, a photographed video signal, and a binaural flag signal.

Abstract: A video-audio recording and reproducing apparatus (101) has a built-in stereo microphone (21a, 21b) and an external microphone connection terminal (32). The external microphone connection terminal (32) is connected to a binaural microphone (3) to be attached to the ears of a photographer (300). When the binaural microphone (3) is used to collect ambient sounds, an audio signal to be recorded on a recording medium is switched from an audio signal from the built-in stereo microphone (21a, 21b) to a binaural audio signal from the binaural microphone (3). The photographer (300) puts the binaural microphone (3 (31a, 31b)) on his or her ears and collects ambient sounds around the photographer (300) including a sound emanating from an object. The object is photographed with a camera unit (11). The recording medium records the binaural audio signal, a photographed video signal, and a binaural flag signal.

Abstract: A coding unit for coding an audio signal by using an adaptive orthogonal transformation and a coding method comprises calculating the sum total of power of an input audio signal in a time duration of an orthogonal transform length or over the length, adaptively controlling the minimum audible level as a threshold level corresponding to the power level on a frequency axis after the orthogonal transformation or on a time axis before the orthogonal transformation, excepting samples having a power level under the threshold level from orthogonally transformed samples and quantizing remaining samples. The threshold level of the minimum audible level is adaptively controlled corresponding to an observed level of an input audio signal. Even if the input audio signal has various power levels, it is possible to usually set the adaptively audible value (the threshold level), thereby effectively reducing a coding amount.

Abstract: A system for coding and decoding an audio signal by using an orthogonal and inverse orthogonal transformation of a block unit, includes a coding unit having a circuit for obtaining a power level of the audio signal of a segment unit having a predetermined time interval shorter than the block unit, a circuit for generating a gain control signal from the power level, a circuit for performing a predetermined adaptive gain control responsive to the gain control signal to generate and output the adaptive gain control signal to a decoding unit, thereby performing a pre-treatment, and a coding portion for coding the adaptive gain control signal by using the orthogonal transformation to generate and output a coded signal; and the decoding unit having a decoding portion for decoding the coded signal, dequantizing and inversely and orthogonally transforming a decoded audio signal, and a circuit for performing an inverse gain control for the decoded audio signal responsive to the adaptive gain control signal from the ad