Abstract: In order to improve the accuracy of an excitation source for a band-spreading apparatus and to generate a wide-band signal having no gaps, an &agr; band-widening section generates a prediction coefficient &agr;W of a wide-band speech signal from a prediction coefficient &agr;N of a narrow-band speech signal. An oversampling apparatus oversamples a narrow-band speech signal sndN. An interpolation section generates an adaptive signal excPW of a wide-band speech signal from an adaptive signal excPN of the narrow-band speech signal. A zero-filling section generates a noise signal of a wide-band speech signal from a noise signal excNN of the narrow-band speech signal. A noise addition section adds a noise signal which is a gap of the wide-band speech signal and generates a noise signal excNW. An adder generates an excitation source excPW for the wide-band speech signal from the adaptive signal excPW and the noise signal excNW of the wide-band speech signal.

Abstract: An echo canceller is provided in which a down-sampling circuit converts a 16-kHz sampling frequency of a wide-band voice signal output to an 8-kHz sampling frequency of a narrow-band voice signal supplied at an input terminal, an adaptive filter estimates, from the wide-band voice signal whose sampling frequency has been down-sampled to 8 kHz in the down-sampling circuit, an echo signal coming from a speaker to a microphone and having an echo path characteristic imparted to the echo signal by an echo path filter, and a subtraction circuit subtracts from the microphone input signal the echo signal having been estimated by the adaptive filter.

Abstract: An object of the invention is to process image data the quality by simply uploading the image data prepared by the user himself himself to a server of a provider without using a particular hardware equipped with an image quality correction function, without the need of purchasing and using a software for correcting the image quality, and without caring about anything. In a system including personal computers 2a, 2b, 2c, - - - of the clients and a server 1 of a provider, when image data are uploaded to the server 1 of the provider from the personal computers 2a, 2b, 2c, - - - of the clients, the quality image of the image data is corrected by the server 1.

Abstract: A bandwidth expanding method and apparatus in which frequency characteristics of high-frequency components of broad band signals can be adjusted to the liking of the user, overflow due to addition is prevented from occurring without power variations being perceived by a user, the number of broad band formants is reduced, and emphasis is attached to the rough structure of the spectrum, so that the produced broad band speech signals can be improved in quality. To this end, in a speech bandwidth expansion device, frequency characteristics of the frequency components not less than 3400 Hz are adjusted by preset alterable parameter values and summed to the original narrow band speech components. If overflow has occurred in a sample, the high-range gain of the sample is lowered to a level below the overflow level before proceeding to addition.

Abstract: A signal processing method enabling reproduction of a broadband signal free of aliasing which transforms a plurality of discrete signals obtained by sampling an identical continuous signal using sampling phases different in a one-dimensional direction and containing a basic spectral component contained in the continuous signal and imaging components other than the basic spectral component to the frequency domain in a Fourier transforming circuit and shifts the phase in a spatial shift circuit, then solves predetermined simultaneous equations in a basic spectrum calculation circuit, finds complex numbers to be multiplied with the phase shifted plurality of signals, multiplying the calculated corresponding complex numbers with the plurality of signals, and adding the results of the multiplication to generate a signal free of the aliasing.

Abstract: A method and apparatus for sound synthesizing and sound band expanding of a narrow band input signal uses wide-band voiced and unvoiced sound code books and also uses narrow-band voiced and unvoiced sound code books. Coded input sound parameters are decoded and quantized using the narrow-band voiced and unvoiced sound code books and are then de-quantized using the wide-band voiced and unvoiced sound code books. The sound is synthesized based on the de-quantized data and a so-called innovation-related parameter formed by a zero-filling circuit filing zeros between samples of the framed input signal, so that the result is an upsampled aliased wide-band signal used with the de-quantized data to synthesize the sound.

Abstract: In a radio communication such as a radio telephone system or the like, a transmission channel which can accommodate a change in a transmission capacity is set. While a terminal apparatus and a base station are in communication for transmitting a predetermined information using a predetermined transmission channel, a signal requesting to set another transmission channel is transmitted using a part of the predetermined transmission channel to initiate communication between the terminal apparatus and the base station through the other transmission channel.

Abstract: A method and apparatus for voiced/unvoiced decision for judging whether an input speech signal is voiced or unvoiced. The input parameters for performing the voiced/unvoiced (V/UV) decision are comprehensively judged in order to enable high-precision V/UV decision by a simplified algorithm. Parameters for the voiced/unvoiced (V/UV) decision include the frame-averaged energy of the input speech signal lev, the normalized autocorrelation peak value r0r, the spectral similarity degree pos, the number of zero crossings nZero, and the pitch lag pch. If these parameters are denoted by x, these parameters are converted by function calculation circuits using a sigmoid function g(x) represented byg(x)=A/(1+exp (-(x-b)/a))where A, a, and b are constants differing with each input parameter. Using the parameters converted by this sigmoid function g(x), the voiced/unvoiced decision is made a V/UV decision circuit.

Abstract: A pitch extraction method and apparatus whereby the pitch of a speech signal having various characteristics can be extracted accurately. The frame-based input speech signal, band-limited by an HPF 12 and an LPF 16, is sent to autocorrelation computing units 13, 17 where autocorrelation data is found. The pitch lag is computed and normalized in the pitch intensity/pitch lag computing units 14, 18. The pitch reliability of the input speech signals, limited by the HPF 12 and the LPF 16, is computed in elevation parameter calculation units. A selection unit 20 selects one of the parameters obtained from the input speech signal, limited by the HPF 12 and the LPF 16, using the pitch lag and the evaluation parameter.

Abstract: A signal decoding method and apparatus in which the speech signal reproducing speed is controlled without changing the phoneme or the pitch, in which the apparatus has a data number convertor for converting the number of orthogonal transform coefficients entering a transmission signal input terminal from N to M, an inverse orthogonal transform unit for inverse orthogonal-transforming the M number of the orthogonal transform coefficients obtained by the data number convertor, and a linear predictive coding synthesis filter for performing predictive synthesis based on the short-term prediction residuals obtained by the inverse orthogonal transform unit. For an input signal, short-term prediction residuals are found and are orthogonally transformed to form the orthogonal transform coefficients at a rate of N coefficients per transform unit. The frequency positions of the N transform coefficients may be rearranged to M values by M/N or by oversampling to change N to M.

Abstract: A method and apparatus for reproducing speech signals at a controlled speed and for synthesizing speech includes a dividing unit that divides the input speech into time segments and an encoding unit that discriminates whether each of the speech segments is voiced or unvoiced. Based on the results of the discrimination, the encoding unit performs sinusoidal synthesis and encoding for voiced segments and vector quantization by closed-loop search for an optimum vector using an analysis-by-synthesis method for unvoiced segments in order to find encoded parameters. A period modification unit modifies the length of time associated with each signal segment and calculates a set of modified encoded parameters.

Abstract: A speech encoding method and apparatus for encoding an input speech signal on a block-by-block or frame-by-frame basis wherein short-term prediction residuals are found and then sinusoidal analytic encoding parameters are produced based on those short-term prediction residuals. Perceptually weighted vector quantization is performed for voiced blocks or frames by encoding their sinusoidal frequency or analytic harmonic magnitudes and, in the case of unvoiced blocks or frames, the time waveforms of the unvoiced blocks are encoded.

Abstract: An encoding apparatus in which an input speech signal is divided into blocks and encoded in units of blocks. The encoding apparatus includes an encoding unit for performing CELP encoding having a noise codebook memory containing having codebook vectors generated by clipping Gaussian noise and codebook vectors obtained by learning using the code vectors generated by clipping the Gaussian noise as initial values. The encoding apparatus enables optimum encoding for a variety of speech configurations.

Abstract: A method and apparatus for encoding an input signal, such as a broad-range speech signal, in which a number of decoding operations with different bit rates are enabled for assuring a high encoding bit rate and for minimizing deterioration of the reproduced sound even with a low bit rate. The signal encoding method includes a band-splitting step for splitting an input signal into a number of bands and a step of encoding signals of the bands in a different manner depending on signal characteristics of the bands. Specifically, a low-range side signal is taken out by a low-pass filter from an input signal entering a terminal, and analyzed for Linear Predictive coding by an Linear Predictive coding analysis quantization unit. After finding the Linear Predictive coding residuals, as short-term prediction residuals by an Linear Predictive coding inverted filter, the pitch is found by a pitch analysis circuit. Then, pitch residuals are found by long-term prediction by a pitch inverted filter.

Abstract: A speech decoding method and apparatus for decoding encoded speech signals and subsequently post-filtering the decoded signals, wherein the filter coefficient of a spectral shaping filter in a post-filter fed with an encoded and subsequently decoded speech signal is updated with a sub-frame period, while the gain of a gain adjustment circuit for correcting gain changes caused by the spectral shaping is updated with a frame period that is eight times as long as the sub-frame period. This achieves switching of the filter coefficient so as to be changed smoothly with a higher follow-up speed, while suppressing level changes otherwise caused by frequent gain switching. The result is improved characteristics of a post-filter used for spectral shaping of a decoded signal supplied from the signal decoder and more effective post-filter processing.

Abstract: An instruction for operation mode control of a VTR 40 and information on the video recording reservation is voice inputted. The voice input is recognized by a voice recognition circuit 13 and is fed to a control circuit 15. The control circuit 15 controls the VTR 40 in response to the instruction or information of the voice input and causes an animation character generating circuit 16 to generate a video image of an animation character AC for displaying it on the screen of a CRT display 30. A message from the animation character AC is voice synthesized in a voice synthesizing circuit 19 and the synthesized voice is outputted from a speaker 20.The electronic equipment can be operated as if the user were talking with the animation character, so that a natural man-machine interface can be realized.