Abstract: A sampling rate converting method and apparatus for converting the sampling frequency of a 1-bit digital data obtained by a .SIGMA..DELTA. modulator to 32 kHz, 48 kHz, 96 kHz or 192 kHz without producing jitter. A decimation filter 3 decimates the sampling frequency of 1-bit digital data of 2.8224 MHz supplied from an input terminal 2 by 1/21-tuple decimation. An interpolation filter 4 oversamples the frequency of the output of the decimation filter 3 by quintuple oversampling with an integer ratio of 1:5. A decimation filter 5 decimates the frequency of the output of the interpolation filter 4 by 1/21-tuple decimation with an integer ratio of 21:1.

Abstract: An audio signal recording and reproduction apparatus enabling a low frequency band signal component to be reproduced by existing reproduction apparatuses and a high frequency band component to be processed as well. A digital audio signal is divided into a low frequency band signal component and high frequency band signal component by a frequency band division filter. The low band signal component is subjected to straight PCM processing to maintain compatibility with existing reproduction methods. The high frequency band signal component is subjected to high frequency coding through a requantization circuit and entropy coding circuit. A comparison circuit compares the straight PCM signals and the high efficiency coded signals to determine if they will fit into a predetermined bit length and, when not fitting, causes the requantization circuit to perform requantization to shorten the bit length.

Abstract: A high efficiency coding method and apparatus includes quantization which takes into account correlation of input signals of plural channels in coding the input signals of the respective channels simultaneously. In the high efficiency coding method and apparatus, spectrum data from MDCT circuits or coefficient data are transmitted to another channel via an other-channel transmission circuit, and data from another channel are transmitted to correlation adaption circuits. The data from correlation adaption on circuits are transmitted to adaptive bit allocation circuits, respectively, wherein bit allocation in coding of the input signals of one channel takes into account the input signals of another channel.

Abstract: An apparatus and method for processing an input digital signal in which an input digital audio signal at an input terminal 11 is supplied via a one-block delay unit 17 to a difference unit 20, an output of which is processed by noise-shaping circuit 21 and restored by a near-instantaneous compressor 26, a near-instantaneous expander 27 and an accumulator 28 so as to be fed back to the difference unit 20 to find a difference between a signal of a directly preceding sample and a signal of the current sample. The noise shaping circuit 21 modifies the spectrum of the quantization noise by taking aural characteristics, such as equal-loudness characteristics or masking characteristics, into account for diminishing the level of the quantization noise as perceived by the ear. The noise level as perceived may be reduced in a manner desirable for the sound quality without changing the existing format.

Abstract: A signal encoding apparatus for cutting out blocks of an input signal waveform at a predetermined time interval and converting the in-block signal by mutually independent conversion axes for encoding, wherein the waveform cutting block length along the time axis is changed according to the input signal.

Abstract: A decoder for reproducing information from an encoded signal that has been frequency-domain transformed by a transform conversion circuit in response to audio signal samples grouped into time-domain blocks based on selected block time lengths, includes a reverse conversion transformation circuit that groups the frequency-domain transfer coefficients into transform blocks based on the block time length information used in the encoding operation and applies the inverse discrete transform function to generate time-domain signal sample blocks therefrom.

Abstract: A system for encoding audio signals, transmitting or recording the encoded audio signals, and then decoding the transmitted audio signal includes a circuit for determining the block length of the input audio signals based upon detected values of predetermined portions of the input signal and then performing an orthogonal conversion of the determined block length signal, in which the signal is orthogonally transformed from a time-domain signal to a frequency-domain signal. Following quantization, the converted and encoded signal is transmitted and upon performing reverse quantization, a reverse orthogonal converting circuit converts the signal back to the time-domain in keeping with the signal providing information concerning the various block lengths that were determined during encoding. The converted time-domain signals are then connected into a smooth waveform to form output audio signals.

Abstract: There is provided a quantizing error reducer for an audio signal, which is constructed to feed a quantizing error back to the input side of a quantizer through a noise filter, wherein the coefficient of the noise filter is set on the basis of information relating to the equiloudness curve, thereby making it possible to reduce the noise in the hearing sense.

Abstract: A signal encoding apparatus for cutting out blocks of an input signal waveform at a predetermined time interval and converting the in-block signal by mutually independent conversion axes for encoding, wherein the waveform cutting block length along the time axis is changed according to the input signal.

Abstract: This invention relates to a digital signal encoding apparatus in which the width of the range in selected to be wider for the higher frequency range of the digital signals divided into a plurality of regions and in which the encoded signals are synthesized for the respective ranges, wherein encoding is controlled as a function of the output detecting the characteristics of the components of the divided frequency ranges and the detection time interval is selected to be longer for the lower frequency to enable more efficient encoding to be performed as a function of the properties of input digital signals.