Nippon Telegraph and Telephone Corporation

Akio Jin

Takehiro Moriya

Takeshi Mori

Kazunaga Ikeda

Naoki Iwakami

Kazuaki Chikira

Noboru Harada

Dai Yang

A down samplerdown samples a digital signal in the sampling frequency thereof from 96 kHz to 48 kHz on a frame-by-frame basis. The converted signal is compression encoded and output as a main code Im. An up samplerconverts a partial signal corresponding to the main code Im to a signal having the original sampling frequency 96 kHz, for example. An error signal between the up sampled signal and an input digital signal is generated. An array converting and encoding unitarray converts bits of sample chains of the error signal, thereby outputting an error code Pe. On a decoding side, a high fidelity reproduced signal is obtained based on the main code Im and the error code Pe, or a reproduced signal is obtained based on the main code Im only.

Method, apparatus, and program for encoding digital signal, and method, apparatus, and program for decoding digital signal

Digital signal encoding method, decoding method, encoding device, decoding device, digital signal encoding program, and decoding program

At the coder side, bits of samples of each frame of an input digital signal are concatenated every digit common to the samples across each frame to generate equi-order bit sequences, which are output as packets. At the decoding side, the input equi-order sequences are arranged inversely to their arrangement at the coder side to reconstruct sample sequences. When a packet dropout occurs, a missing information compensating partcorrect the reconstructed sample sequences in a manner to reduce an error between the spectral envelope of the reconstructed sample sequence concerned and a known spectral envelope.

Digital signal coding and decoding methods and apparatuses and programs therefor

A sample sequence &#x394;S similar to a first or last sample sequence of the current frame is extracted from its samples Sand concatenated, as an alternative sample sequence AS, to each of the front and back of the current frame, and the current frame with the alternative sample sequence concatenated thereto is subjected to filtering or prediction coding to obtain processing result Sof the current frame. In the case of prediction coding, auxiliary information, which indicates which part of the current frame was used as the alternative sample sequence, is also output. By this, filtering, autoregressive prediction coding and decoding, which require processing extending over preceding and succeeding frames as in an interpolation filter, can be concluded in the current frame with substantially no degradation of the continuity and coding efficient of the reconstructed signal.

Digital signal processing method, processor thereof, program thereof, and recording medium containing the program

Digital signal samples X in a floating-point format, each of which is composed of 1 bit of sign, 8 bits of exponent E and 23 bits of mantissa M, are converted through rounding by an integer formatting partinto digital signal samples Y in an integer format, the sequence of the digital signal samples Y is losslessly compression-coded by a compressing partinto a code sequence Ca, and the code sequence Ca is output. The digital signal samples Y are converted by a floating point formatting partinto digital signal samples X' in the floating-point format, a difference signal &#x394;X indicating the difference between the digital signal sample X&#x2032; and the digital signal sample X is determined by a subtraction part, the difference signal &#x394;X is losslessly coded, and the resulting code sequence Cb is output.

Floating point type digital signal reversible encoding method, decoding method, devices for them, and programs for them

An input signal is time-frequency transformed, then the frequency-domain coefficients are divided into coefficient segments of about 100 Hz width to generate a sequence of coefficient segments, and the sequence of coefficient segments is split into subbands each consisting of plural coefficient segments. A threshold value is determined based on the intensity of each coefficient segment in each subband. The intensity of each coefficient segment is compared with the threshold value, and the coefficient segments are classified into low- and high-intensity groups. The coefficient segments are quantized for each group, or they are flattened respectively and then quantized through recombination.

Audio signal coding and decoding methods and apparatus and recording media with programs therefor

An object of the invention is to provide a method for compressing digital input signals at high compression efficiency and reproducing the input data perfectly. The method includes the steps of: converting a digital input signal in each frame to bitstreams according to a sign-magnitude format; deblocking the bitstreams into individual bits; joining each bit in a time sequence while retaining an identical chronological order of bits in all the frames; and reversibly encoding each bitstream obtained by joining the bits. And, the reversible decoding method includes the steps of: reversibly decoding a reversible code sequence in each frame; deblocking the bitstreams obtained by reversible decoding into individual bits; joining each bit in a time sequence while retaining an identical chronological order of bits in all the frames; and joining successive frames obtained by joining the bits.

Growing community of inventors

Kokubunji, Japan

Akio Jin