Abstract: A hierarchical lossless encoding and decoding technology for digital signals such as of music data, audio data, or the like. A lossless reproduced signal is made identical to an input signal even when the processing accuracy in an encoding apparatus and the processing accuracy in a decoding apparatus are different from each other. the encoding apparatus transmits low-bit-rate encoded data produced by encoding an input signal, lossless encoded data produced by effecting lossless encoding on a differential signal between a low-bit-rate decoded signal decoded from the low-bit-rate encoded data and the input signal, and corrective information extracted from the low-bit-rate decoded signal to respective input terminals of the decoding apparatus. A low-bit-rate decoder decodes the low-bit-rate encoded data. A lossless decoder decodes the differential signal. A corrector corrects the low-bit-rate decoded signal based on the corrective information.

Abstract: A mapping transform unit (11) subjects input audio signals to a mapping transform and generates frequency region signals that take frequency as a variable; a code amount designation unit (12) supplies a preset coding bit rate as a code amount output; a frequency region signal compression encoder (13), based on the code amount, subjects input frequency region signals to a compression encoding process and generates a bitstream; and a bandwidth-limiting unit (10) executes a bandwidth-limiting processing in which a part of the frequency zone covered by frequency region signals is allotted to an attenuation frequency zone, and in which the value of the frequency region signal is multiplied by an attenuation coefficient having a value less than 1 in the attenuation frequency zone to attenuate the frequency region signal in the attenuation frequency zone, and supplies the frequency region signals that have undergone the bandwidth-limiting processing to the frequency region signal compression encoder (13).

Abstract: A random number generating apparatus is equipped with a random number memory section in which a plurality of sets of random numbers, whose total energy R, has a fixed value are stored. N random numbers are generated on the basis of more than one set of random numbers read from the random number memory section, and further multiplied by a square root of (T×M÷R÷N) to obtain a random number sequence having a predetermined energy. A normalization coefficient S can be obtained with a simple process without any usage of the calculation of division and square root multiplication, thereby enabling a small-scale apparatus to be realized. Accordingly, the random number generating apparatus has a decreased scale in the circuit arrangement capable of outputting the random numbers having a predetermined energy.

Abstract: A bit-rate converting apparatus and a method thereof, in which bit-rate conversion is executed by low computational complexity, are provided. The bit-rate conversion is executed in a frequency domain, and psycho-acoustic analysis is not needed by using information included in an inputted bit-stream before the bit-rate conversion is applied. With this, the computational complexity is lowered. And in order that many equal values are not contained in a frequency domain signal, which is inputted to a quantizing means, a quantized value before inverse quantizating is applied is modified, or an inverse quantized value after the inverse quantizing was applied is modified. With this, fine control for the bit-rate is made to be easy.

Abstract: In a coded speech decoding system, an n-channel time domain speech signal is converted to a frequency domain speech signal. A predetermined weighting adding process is executed on the frequency domain speech signal for each of a plurality of different transfer functions. The frequency domain speech signal obtained through the weighting adding process is converted to an m-channel (m<n) time domain speech signal. A predetermined windowing processing is executed on the time domain speech signal.

Abstract: To provide an audio signal quantization device in which an amount of arithmetical operation can be reduced. A 0.75-power computing apparatus includes: an inverse number computing unit; a first −0.5-power computing unit; a multiplication unit; a second −0.5-power computing unit; and a program storage medium that is connected to an outside, whereby a program stored in the program storage medium is executed to compute the 0.75-power in a quantization computing expression. The inverse number computing unit, the first −0.5-power computing unit, the multiplication unit, and the second −0.5-power computing unit operate in such a manner that a multiplication between an infinity and zero which leads its operation result to an indefinite value is inhibited.

Abstract: In a data compression device having a plurality of Huffman coders given identification (ID) values, respectively, to produce Huffman codes together with the corresponding ID codes obtained from the ID values, a re-coding portion is included to change a sequence of the ID values so that a reduction is accomplished about a code amount of the Huffman codes and the ID codes obtained from the ID value sequence and is operable to re-code an input data sequence again by Huffman coders indicated by the changed ID values. Such re-coded Huffman codes and ID codes based on the changed ID values are produced as output data signals.

Abstract: An audio coding device for coding input signals of a plurality of channels over a plurality of time sections is disclosed. The device includes a parallel transform unit, which can store input signals of two or more channels and also input signals of two or more time sections in one SIMD variable and collectively perform transform on the input signals of the two or more channels and the input signals of the two or more time sections by using SIMD processor. An audio decoding device includes a parallel in verse transform unit, which can store transformed signals of two or more channels in one SIMD variable and collectively perform inverse transform with respect to the two or more channels by using SIMD processor.

Abstract: A sub-band dividing unit divides an input signal into a plurality of frequency bands, and outputs a plurality of sub-band signals. A scaling unit calculates a scaling factor related to a reference value for each of the sub-band signals, and uniformly adjusts the dynamic range thereof. An auditory-sense-analysis bit allocating unit performs weighting conforming to an equal-loudness curve for each of the sub-band signals, and then calculates the amount of bit allocation to equalize a weighted quantization error in the individual sub-band signals. A quantization unit performs quantization calculations. A bitstream generating unit generates a bitstream together with header information and auxiliary information.

Abstract: A coded voice signal format converting apparatus is provided which is capable of converting a signal format of a coded voice signal by computations in reduced amounts. In the coded voice signal format converting apparatus, in a second coding device is employed a quantizing accuracy information converting section to which a first quantizing accuracy information output from a quantizing accuracy information decoding section in a first decoding device is input. Second mapping signal is quantized by a mapped signal coding section to produce a coded voice signal and the first quantizing accuracy information is converted so that it can be used by mapped signal coding section to determine a second quantizing accuracy information.

Abstract: An encoding system is disclosed, which includes an input terminal for receiving an input signal, a signal converting portion for converting the input signal into frequency-domain signals and grouping several frequency-domain signals into blocks, an analyzing portion for analyzing the input signal and the frequency-domain signals and obtaining an allowable error, a selecting portion for selecting one of a plurality of quantizing portions that quantizes frequency-domain signals of each block corresponding to the allowable error, a plurality of quantizing portions for quantizing frequency-domain signals corresponding to selection information and calculating amplitude information and codes, at least one pulse quantizing portion for encoding only part of the frequency domain signals, a multiplexing portion for multiplexing the amplitude information, codes, and selection information, and an output terminal for outputting the multiplexed signal.

Abstract: In an adaptive transform coding system and/or and adaptive transform decoding system, coding efficiency in the case where a small number of quantized values having large absolute value are present, is improved. The adaptive transform coding system codes the small number of quantized values having large absolute values and other quantized values are coded separately. More particularly, the adaptive transform coding system includes a selector (6) discriminating the small number of quantized value having large absolute value from other quantized value, a pulse coding means for coding the small number of quantized values having large absolute values (8) and the pulse decoding means (16) for decoding the same, a coding means (7) for coding the quantized value other than those having large absolute values and a decoding means (15) decoding the same, and a synthesis means (18) for synthesizing the small number of quantized values having large absolute value and other quantized values.