Abstract: An encoding device (100) includes: a transforming unit (120) operable to extract a part of an inputted audio signal at predetermined time intervals and to transform each extracted part to produce a plurality of windows composed of short blocks; a judging unit (137) operable to compare the windows with one another to judge whether there is a similarity of a predetermined degree and to replace a high frequency part of a first window, which is one of the produced windows, with values “0” when there is the similarity, wherein the first window and a second window share a high frequency part of the second window, which is also one of the produced windows; a first quantizing unit (131) operable to quantize the produced windows after replacing operation; and a first encoding unit (132) operable to encode the quantized windows to produce encoded data; and a stream output unit (140) operable to output the produced encoded data.

Abstract: An audio data input unit (310) of an encoding device 300 splits an audio data string into contiguous 4,096 samples of audio data, and a transforming unit (320) transforms the split audio data into spectral data in a frequency domain. A data dividing unit (330) divides the spectral data into that in a lower frequency band and that in a higher frequency band at 11.025 kHz (f1) as a boundary. The spectral data in the lower frequency band is quantized and encoded by a first quantizing unit (340) and an encoding unit (350) as usual. A second quantizing unit (345) generates sub information indicating a characteristic of the spectral data in the higher frequency band, and a second encoding unit (355) encodes the sub information. A stream output unit (390) integrates the codes obtained by the first and second encoding units (350), (355), and outputs the integrated one. Here, the f1 is a half or less of a sampling frequency f2 at which the audio data string is created.

Abstract: An encoder of the present invention includes: G storage sections for storing G groups of data; a selection section for selecting one of H Huffman codebooks having codebook numbers for each of the groups of data; G encoding sections Huffman-encoding the G groups of data by using the selected Huffman codebook; and an encoding section for encoding the codebook number of each Huffman codebook selected. The selection section includes a calculation section for calculating a code length and a control section for selecting one of the Huffman codebooks. When the Huffman codebook selected is an unsigned codebook, a number of bits required for sign information has previously been added to the calculated code length.

Abstract: An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing t

Abstract: An encoding apparatus includes a band gain encoding section for calculating an average amplitude of a frequency spectrum stream corresponding to each of a plurality of frequency bands so as to generate a first code representing the average amplitude of the frequency spectrum stream; an encoding band determination section for determining at least one frequency band, for which the corresponding frequency spectrum stream is to be quantized and encoded from among the plurality of frequency bands; a spectrum encoding section for quantizing and encoding the frequency spectrum stream of each of the at least one frequency band determined by the encoding band determination section so as to generate a second code; and an encoded stream generation section for generating an encoded stream based on the first code and the second code.

Abstract: An encoding apparatus includes a quantized spectral sequence generation section for generating a quantized spectral sequence by quantizing an audio signal with a predetermined quantization precision, and a circulating code vector quantization section for outputting a spectral sequence code containing circulating position identification information indicating how much a reference spectral sequence is circulated to obtain a circulant quantized spectral sequence which is most similar to the quantized spectral sequence.

Abstract: An exponent part extraction section extracts a bit series from the exponent part of an inputted floating point data. A mantissa part extraction section extracts the uppermost K bits from the mantissa part of the floating point data. A first conversion section inputs the output e from the exponent part extraction section and outputs the value of a function X(e) thereof. A second conversion section inputs the output f from the mantissa part extraction section and outputs the value of a function Y(f) thereof. A multiplier section multiplies together these values. By setting suitable tables in advance in the first and the second conversion sections, the calculation of the vˆ p for an item v of floating point data can be performed.

Abstract: An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization means having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing th

Abstract: An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing t

Abstract: The present invention is a signal processing device which performs parallel processes A and B efficiently. There is a deviation in the throughputs of the process A and B in processing an audio signal. First to Nth sub signal processing sections have capabilities to complete the process A within a period (N×T). A main signal processing sections has a capability to complete the process B within a period T. Efficient signal processing can be achieved by processing an input digital signal by means of distinct sub signal processing devices one after another and then processing the signal by the main signal processing section. (N×T).

Abstract: Apparatus for expanding the bandwidth of speech signals such that a narrowband speech signal is input and digitized, the spectral envelope information and residual information are extracted from the digitized signal by linear predictive coding analysis, the spectral envelope information is expanded into wideband information by a spectral envelope converter, the residual information is expanded into wideband information by a residual converter, the converted spectral envelope information and residual information are combined to produce a wideband speech signal, frequency information not contained in the input signal is extracted from the obtained wideband speech signal by a filter, and the resulting signal is added to the original digitized input signal, and the obtained signal is converted into an analog signal as the output signal of the apparatus.

Abstract: A sound reproduction system for automatically conducting a tone control for an audio signal according to the present invention includes: an input terminal for inputting an audio signal; a judgment circuit for determining a ratio between a stereophonic signal component and a monophonic signal component of the input audio signal and for outputting a judgment signal; a signal processing section for receiving the input audio signal and processing the input audio signal so as to generate an output signal having uniform transmission characteristics irrespective of the listening position; an adder for receiving the input audio signal and the output signal and for adding the input audio signal with the output signal at a certain addition ratio based on the judgment signal so as to generate an added signal; and a loudspeaker for receiving the added signal and for reproducing the added signal in a plurality of positions.

Abstract: The speech detection apparatus comprises: a reference model maker for extracting a plurality of parameters for a speech detection from training data, and for making a reference model based on the parameters; a parameter extractor for extracting the plurality of parameters from each frame of an input audio signal; and a decision device for deciding whether or not the audio signal is speech, by comparing the parameters extracted from the input audio signal with the reference model. The reference model maker makes the reference model for each phoneme.

Abstract: A digital signal recording apparatus includes: a level encoder for encoding a digital signal into the first to N-th level data, N being an integer greater than 1; a solid state memory having a data storage area and an auxiliary information storage area, the data storage area being writable for the first to N-th level data; and a write controller for, when the writable area in the data storage area is exhausted, freeing at least a part of an area in the data storage area in which level data other than the first level data is stored, for storing level data including at least the first level data into the freed area in the data storage area, and for storing auxiliary information indicating attributes of the stored level data into the auxiliary information storage area.

Abstract: A sound field controller for generating apparent sound sources by adjusting the amplitude and delay time of a sound signal so that the sound will be perceived by plural listeners as sound coming from a location separated from the specific location of the front speakers, and for additionally controlling the effect of the apparent sound sources by evaluating the attributes of the source sound signal.