Nippon Telegraph and Telephone Corporation

Naoki Iwakami

Takehiro Moriya

Takeshi Mori

Akio Jin

Satoshi Miki

Kazuaki Chikira

Shigeaki Sasaki

Kazunori Mano

Kazunaga Ikeda

Hitoshi Ohmura

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.

Methods, apparatuses and recorded medium for reversible encoding and decoding

In multichannel acoustic signal coding and decoding, left- and right-channel signals are alternately interleaved for each sample to generate a one-dimensional signal sample sequence. The one-dimensional signal sample sequence is subjected to coding based on correlation. In coding, the left- and right-channel signals may preferably be interleaved after reducing an imbalance in power between input channels. In such an instance, a power imbalance is introduced between the decoded left- and right-channel signal sample sequences

Apparatus and method for efficiently coding plural channels of an acoustic signal at low bit rates

In a CELP coding scheme, p-order LPC coefficients of an input signal are transformed into n-order LPC cepctrum coefficients c.sub.j (S.sub.2), which are modified into n-order modified LPC cepstrum coefficients c.sub.j ' (S.sub.3). Log power spectral envelopes of the input signal and a masking function suited thereto are calculated (FIGS. 3B, C), then they are subjected to inverse Fourier transform to obtain n-order LPC cepstrum coefficients, respectively, (FIGS. 3D, E), then the relationship between corresponding orders of the LPC cepstrum coefficients is calculated, and the modification in step S.sub.3 is carried out on the basis of the relationship. The modified coefficients c.sub.j are inversely transformed by the method of least squares into m-order LPC coefficients for use as filter coefficients of a perceptual weighting filter. This concept is applicable to a postfilter as well.

Method for the modification of LPC coefficients of acoustic signals

An input acoustic signal is subjected to modified discrete cosine transform processing to obtain its spectrum characteristics. Linear prediction coefficients are derived from the input acoustic signal in a linear prediction coding analysis part, and the prediction coefficients are subjected to Fourier transform in a spectrum envelope calculation part to obtain the envelope of the spectrum characteristics of the input acoustic signal. In a normalization part the spectrum characteristics are normalized by the envelope thereof to obtain residual coefficients. Another normalization part normalizes the residual coefficients by a residual-coefficients envelope predicted in a residual-coefficients envelope calculation part, thereby obtaining fine structure coefficients, which are vector-quantized in a quantization part. A de-normalization part de-normalizes the quantized fine structure coefficients. The residual-coefficients envelope calculation part uses the reproduced residual coefficients to predict the envelope of residual coefficients of the subsequent frame.

Acoustic signal transform coding method and decoding method having a

Power normalization parts calculate the average powers of input signals of a plurality of channels for each frame and divide the signals by the calculated average powers to generate normalized signals and, at the same time, generate weights corresponding to the normalization gains. The normalized signals of the plurality of channels are combined in a combining part into predetermined sequences and outputted therefrom as one or more interleaved signal vectors. The combining part combines the weights from the power normalization part into the same sequences of the normalized signal and outputs one or more interleaved weight vectors. In a vector quantization part the signal vectors are vector quantized by the interleaved weight vectors corresponding thereto, respectively, and quantization indexes and normalization indexes are outputted as results of coding.

Growing community of inventors

Yokohama, Japan

Naoki Iwakami