Mitsubishi Denki Kabushiki Kaisha

Mitsubishi Electric Corporation

Hirohisa Tasaki

Tadashi Yamaura

Satoru Furuta

Shinya Takahashi

Bunkei Matsuoka

Kunio Nakajima

Takashi Sudo

Masaru Kimura

Ikuo Kajiyama

Katsushi Seza

Filters---N divide an input signalinto N band-limited signals---N, and multipliers---N carry out dynamic range control of the N band-limited signals---N, respectively. After that, filters---N eliminate odd harmonics caused by the dynamic range control, and a signal synthesis unitcombines the signals passing through the filters---N into a single output signal

Dynamic range control apparatus

A band separating unitcarries out a band division of a plurality of power spectra into which an input signal is converted by a time-to-frequency converting unitto combine power spectra into each subband, and a band representative component generating unitdefines a power spectrum having a maximum among the plurality of power spectra within each subband as a representative power spectrum. A noise suppression amount generating unitcalculates an amount of noise suppression for each subband by using the representative power spectrum and a noise spectrum, and a noise suppressing unitsuppresses the amplitudes of the power spectra according to the amount of noise suppression.

Noise suppression device

A noise suppression device includes: a power spectrum calculator converting an input signal of time domain into power spectra of frequency domain; a voice/noise determination unit determining whether the power spectra indicate voice or noise; a noise spectrum estimation unit estimating noise spectra of the power spectra; a period component estimation unit analyzing a harmonic structure constituting the power spectra and estimating periodical information about the power spectra; a weighting coefficient calculator calculating a weighting coefficient for weighting the power spectra; a suppression coefficient calculator calculating a suppression coefficient for suppressing noise included in the power spectra; a spectrum suppression unit suppressing amplitude of the power spectra in accordance with the suppression coefficient; and an inverse Fourier transformer converting the power spectra output by the spectrum suppression unit into a signal of time domain to generate a noise-suppressed signal.

A noise suppressor selects, for individual frequency components, maximums by comparing a plurality of noise suppressed spectraanda plurality of noise suppressing unitsandoutput, thereby obtaining an output spectrumhaving the frequency components selected as its components. A first noise suppressing unitgenerates a noise suppressed spectrumby multiplying an input spectrumby amplitude suppression gains, and makes the amplitude suppression gains greater than most of the amplitude suppression gains in a noise signal intervals of a second noise suppressing unit

Noise suppressor

A correction spectrum calculation unitobtains a correction spectrum by smoothing an estimated noise spectrum in accordance with the degree of its variations, and a suppression quantity limiting coefficient calculation unitdecides a suppression quantity limiting coefficient from the correction spectrum. A suppression quantity calculation unitobtains a suppression coefficient based on the suppression quantity limiting coefficient, and the spectrum suppression unitcarries out amplitude suppression of spectral components of an input signal.

A synthesis filtersynthesizes a plurality of wide-band speech signals by combining wide-band phoneme signals and sound source signals from a speech signal code book, and a distortion evaluation unitselects one of the wide-band speech signals with a minimum waveform distortion with respect to an up-sampled narrow-band speech signal output from a sampling conversion unit. A first bandpass filterextracts a frequency component outside a narrow-band of the wide-band speech signal and a band synthesis unitcombines it with the up-sampled narrow-band speech signal.

Speech signal restoration device and speech signal restoration method

A speech coding apparatus comprises a repetition period pre-selecting unit for generating a plurality of candidates for the repetition period of a driving excitation source by multiplying the repetition period of an adaptive excitation source by a plurality of constant numbers, respectively, and for pre-selecting a predetermined number of candidates from all the candidates generated. A driving excitation source coding unit provides both excitation source location information and excitation source polarity information that minimize a coding distortion, for each of the predetermined number of candidates, and provides an evaluation value associated with the minimum coding distortion for each of the predetermined number of candidates. A repetition period coding unit compares the evaluation values provided for the predetermined number of candidates with one another, selects one candidate from the predetermined number of candidates according to the comparison result, and furnishes selection information indicating the selection result, excitation source location code, and polarity code.

Speech coding apparatus and speech decoding apparatus

A sound encoder multiplexes a plurality of codes into a sound code in an order determined by a multiplexing order determination unit (), and a sound decoder demultiplexes the sound code into a plurality of codes one by one in an order determined by a demultiplexing order determination unit ().

Sound decoder and sound decoding method with demultiplexing order determination

Sound encoder and sound encoding method with multiplexing order determination

A speech encoding apparatus calculates encoding distortion of a noise-like fixed code vector and multiplies the encoding distortion by a fixed weight corresponding to the noise-like degree of the noise-like fixed code vector, calculates encoding distortion of a non-noise-like fixed code vector and multiplies the encoding distortion by a fixed weight corresponding to the non-noise-like fixed code vector, and selects the fixed excitation code associated with multiplication result with a smaller value.

Speech encoding system, and speech encoding method

A variable length code multiplexer multiplexes a plurality of input codes () including a variable length code into a multiplexed code () and outputs the multiplexed code (). The multiplexer includes a forward direction multiplexing unit () for multiplexing the plurality of input codes () one by one so that they are running in a forward direction from a head of the multiplexed code () to a tail of the multiplexed code (), a backward direction multiplexing unit () for multiplexing the plurality of input codes () one by one so that they are running in a backward direction from the tail of the multiplexed code () to the head of the multiplexed code (), and a selection unit () for selecting either the forward direction multiplexing unit () or the backward direction multiplexing unit () for each of the plurality of input codes ().

Variable length code multiplexer and variable length code demultiplexer

An sound encoder accepts a sound signal and then produces a plurality of codes which represent the sound signal on a frame-by-frame basis. The sound encoder determines the order in which the plurality of codes is to be multiplexed into a multiplexed code based on one of the plurality of codes on a frame-by-frame basis, multiplexes the plurality of codes one by one into a multiplexed code in the determined order, and acquires an error correction code for the multiplexed code. The sound encoder then outputs the multiplexed code including the acquired error correction code added to the end thereof as a sound code.

Sound encoder and sound decoder

In order to achieve a speech encoding method and device of high quality, which are small in local occurrence of abnormal noise in decoded speech, the speech encoding method and device include: fixed excitation generating meansfor generating a plurality of fixed excitations; a first distortion calculating portionfor calculating a distortion related to a waveform defined between a signal to be encoded which is obtained from the input speech and a synthetic vector which is obtained from the fixed excitation as a first distortion for each of the fixed excitations; a second distortion calculating portionfor calculating a second distortion different from the first distortion which is defined between the signal to be encoded and the synthetic vector determined from the fixed excitation for each of the fixed excitations; an evaluation value calculating portionfor calculating a given evaluation value for search by using the first distortion and the second distortion for each of the vectors; and searching meansfor selecting the fixed excitation that minimizes the evaluation value for search and outputting a code which is associated with the selected fixed excitation in advance.

Voice encoding method and apparatus

A sound encoder and sound decoder that encode and decode, respectively, variable length codes on a frame by frame basis, the coding including main codes and auxiliary codes in which auxiliary codes are multiplexed or demultiplexed in a same fixed order to determine the order of multiplexing and demultiplexing the main codes which are used to determine where the codes are to be placed in the sound code.

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Tokyo, Japan

Hirohisa Tasaki