Abstract: There is disclosed an audio coding apparatus which has a wideband encoder and noise canceller. The encoder includes a high-frequency audio coder and low-frequency audio coder. The low-frequency audio coder includes a low-frequency noise canceller. When the high-frequency audio coder is disabled, the noise canceller is disabled, and allows a digital audio signal to pass through it and outputs that signal to the encoder. When the high-frequency audio coder is enabled, the low-frequency noise canceller is disabled, and allows a digital audio signal to pass through it.

Abstract: A prediction parameter analysis apparatus comprises a windowing part which generates a short time input signal by subjecting an input signal or a signal derived from the input signal to windowing, a component removal part which removes an unnecessary component from the short time input signal to generate a modified short time input signal, an autocorrelation coefficient computation part which computes autocorrelation coefficients based on the modified short time input signal, and a prediction parameter computation part which computes prediction parameters based on the autocorrelation coefficients.

Abstract: A speech encoding method of generating a synthesized speech signal by using an excitation signal generated by using an adaptive codebook storing a past excitation signal includes the steps of modifying an excitation signal used to generate a synthesized speech signal by filter processing, and storing the modified excitation signal in the adaptive codebook.

Abstract: When the speaker is registered or verified, the supplying part outputs an interference sound signal to cause the loudspeaker to emit interference sound. In the sound inputted to the microphone, the canceling part cancels the component corresponding to the interference sound fed back to the microphone. Using the sound in which the component corresponding to the interference sound has been canceled, the collating part registers or verifies the speaker.

Abstract: In a background noise/speech classification method, whether a digital input signal input through an input terminal is background noise or speech is decided by a background noise/speech decision section on the basis of calculated frame power and a calculated LSP coefficient which are obtained by supplying the input signal to a feature amount calculation section and estimated frame power and an estimated LSP coefficient obtained by an estimated feature amount update section. Thereafter, the estimated feature amount update section updates the estimated frame power and the estimated LSP coefficient by using the frame power and the LSP coefficient obtained by the feature amount calculation section to prepare for the next frame.

Abstract: A speech encoding method, apparatus and program wherein an input speech signal is divided into a plurality of frames each having a predetermined length, each of the frames is subdivided into a plurality of subframes, a predictive pitch period of a subframe in a to-be-encoded current frame is obtained by using pitch periods of at least two frames of the current frame and past and future frames with respect to the current frame; a pitch period of a subframe in the current frame is obtained by using the predictive pitch period, a relative pitch pattern codebook storing a plurality of relative pitch patterns representing fluctuations in pitch periods of a plurality of subframes is prepared, and a change in pitch period of plural subframes is expressed with one relative pitch pattern selected from the relative pitch pattern codebook.

Abstract: A prediction parameter analysis apparatus comprises a windowing part which generates a short time input signal by subjecting an input signal or a signal derived from the input signal to windowing, a component removal part which removes an unnecessary component from the short time input signal to generate a modified short time input signal, an autocorrelation coefficient computation part which computes autocorrelation coefficients based on the modified short time input signal, and a prediction parameter computation part which computes prediction parameters based on the autocorrelation coefficients.

Abstract: Processing for producing encoded output representing information about a pitch period of an input speech signal is performed. The pitch period of a previously entered speech signal is stored in a buffer. A search range-determining portion determines a range in which a current pitch period is analyzed, according to the pitch period of the previously entered speech signal. A presently entered speech signal is applied from a speech input terminal. A pitch analysis portion makes a pitch analysis of candidates for the pitch period contained in the determined search range. Information about the pitch period is delivered from an output terminal and stored in the buffer for subsequent processing. The pitch period of the speech signal can be calculated with a small amount of calculation and represented with a small amount of information.

Abstract: A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.

Abstract: A speech encoding method in which information representing characteristics of a synthesis filter is generated based on an input speech signal in units of one frame. A pitch vector is generated from an adaptive codebook containing past excitation signals, and a first number of reduced pulse position candidates are generated by selecting a first number of pulse positions from a number of possible pulse positions in each of sub-frames obtained by dividing the frame, where a density of the reduced pulse position candidates is high where the pitch vector has a large power and decreases in accordance with a decrease in the power.

Abstract: A speech encoding method of generating a synthesized speech signal by using an excitation signal generated by using an adaptive codebook storing a past excitation signal includes the steps of modifying an excitation signal used to generate a synthesized speech signal by filter processing, and storing the modified excitation signal in the adaptive codebook.

Abstract: A speech encoding method includes generating information representing characteristics of a synthesis filter, and generating an excitation signal for exciting the synthesis filter, the excitation signal including a pulse train generated by setting one or more pulses at a predetermined number of pulse positions selected from a plurality of pulse position candidates adaptively changed in accordance with the characteristics of the speech signal. A speech decoding method includes inputting the excitation signal to a synthesis filter for reconstructing a speech signal.

Abstract: In a signal processing apparatus, a speech coder includes, as three sections for coding speech data by different algorithm, an Algorithm-A coding section, an Algorithm-B coding section and an Algorithm-C coding section. A noise suppressor includes, as three sections for suppressing background noise by different algorithm, an Algorithm-X noise suppress section, an Algorithm-Y noise suppress section and an Algorithm-Z noise suppress section. A suppress algorithm switching control section controls switching on the basis of information from a coding algorithm switching control section such that an optimal one of the noise suppress sections may function in association with the coding section activated in the speech coder.

Abstract: In a background noise/speech classification method, whether a digital input signal input through an input terminal is background noise or speech is decided by a background noise/speech decision section on the basis of calculated frame power and a calculated LSP coefficient which are obtained by supplying the input signal to a feature amount calculation section and. estimated frame power and an estimated LSP coefficient obtained by an estimated feature amount update section. Thereafter, the estimated feature amount update section updates the estimated frame power and the estimated LSP coefficient by using the frame power and the LSP coefficient obtained by the feature amount calculation section to prepare for the next frame.

Abstract: In a background noise/speech classification method, whether a digital input signal input through an input terminal is background noise or speech is decided by a background noise/speech decision section on the basis of calculated frame power and a calculated LSP coefficient which are obtained by supplying the input signal to a feature amount calculation section and estimated frame power and an estimated LSP coefficient obtained by an estimated feature amount update section. Thereafter, the estimated feature amount update section updates the estimated frame power and the estimated LSP coefficient by using the frame power and the LSP coefficient obtained by the feature amount calculation section to prepare for the next frame.

Abstract: A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.

Abstract: On the basis of an autocorrelation coefficient calculated by an autocorrelation coefficient computation section from an input speech signal, an LSF computation section computes LSF parameters F(k) (k=1, 2, . . . , N). A modified logarithmic transformation section performs on the LSF parameters a logarithmic transformation with offset defined by f(k)=logC (1+A.times.F(k)) to obtain modified logarithmic LSF parameters f(k). The resulting modified logarithmic LSF parameters are quantized by a quantization section to provide quantized LSF parameters fq(k). Codes representing the quantized LSF parameters fq(k) are outputted. An inverse transformation defined by Fq(k)=(C.sup.fq(k) -1)/A is performed on the LSF parameters fq(k) to output LSF parameters Fq(k) on the general frequency scale.

Abstract: In a formant emphasis method of emphasizing the formant as the spectral peak of an input speech signal and attenuating the spectral valley of the input speech signal, a spectrum emphasis filter performs processing for emphasizing the formant of the input speech signal and attenuating the valley of the input speech signal. A first-order variable characteristic filter whose characteristic adaptively changes in accordance with the characteristic of the input speech signal and a first-order fixed characteristic filter compensate a spectral tilt included in an output signal from the spectrum emphasis filter.

Abstract: This invention provides a novel speech coding system which recursively executes a filter-applied "Toeplitz characteristic" by causing a drive signal (i.e., an excitation signal) to be converted into a "Toeplitz matrix" when detecting a pitch period in which distortion of the input vector and the vector subsequent to the application of filter-applied computation to the drive signal vector in the pitch forecast called either "closed loop" or "compatible code book" is minimized. The vector quantization method substantially making up the speech coding system of the invention is characteristically used by the system.

Abstract: A speech signal is input to an excitation signal generating section, a prediction filter and a prediction parameter calculator. The prediction parameter calculator calculates a predetermined number of prediction parameters (LPC parameter or reflection coefficient) by an autocorrelation method or covariance method, and supplies the acquired prediction parameters to a prediction parameter coder. The codes of the prediction parameters are sent to a decoder and a multiplexer. The decoder sends decoded values of the codes of the prediction parameters to the prediction filter and the excitation signal generating section. The prediction filter calculates a prediction residual signal, which is the difference between the input speech signal and the decoded prediction parameter, and sends it to the excitation signal generating section.