Abstract: An information providing system includes a terminal device and an identification device which are linked with each other via a network. When a music signal is transmitted from the terminal device to the identification device via the network, an identification code extracting unit extracts an identification code corresponding to a category which is inserted as a digital watermark in the music signal. A feature extracting unit extracts features of the music signal. A database search unit compares the features of information which belongs to the category corresponding to the identification code extracted by the identification code extracting unit with the features extracted by the feature extracting unit. If a match is found, the music title stored in the database search unit is read according to the features, and is then output to the terminal device.

Abstract: An apparatus and method for performing transform encoding, in which time domain samples can overlap one another by any desired percentage and can be added so that signals may be reproduced completely. In the apparatus and method, the linear/nonlinear prediction analysis section 3 receives an audio signal from the input terminal 2 and effectuates linear or nonlinear prediction on the audio signal, generating a prediction residual. The constancy inferring section 7 infers the constancy of the audio signal. The block-length determining section 8 determines the length of an MDCT block from the constancy of the input signal, which the section 7 has inferred. The MDCT section 5 receives M time domain samples supplied from the buffer 4 and having the prediction residual. The MDCT section 5 applies the block length determined by the section 8, performing MDCT transform on the time domain samples, thus generating MDCT coefficients. The quantization section 6 quantizes the MDCT coefficients.

Abstract: First, at step Si, i=0 is set. At step S2, data comprising M samples is fetched. At step S3, an M-point DFT is applied to the data fetched at step S2 above. At step S4, an obtained y(k) is multiplied by a twist coefficient w(i, k). The result is placed in y(k). At step S5, the value in y(k) is overwritten to an array x which contains original data. The above processing is repeated N/M times through steps S6 and S7 until all the input data is processed. At step S9, an FFT with N/M (=2n) points is performed within the range of 0≦k<M. Finally, at step S12, a sort is performed.

Abstract: An apparatus and a method for encoding an input signal on the time base through orthogonal transform involves removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal is sent to a normalization circuit section and a LPC analysis circuit. The normalization circuit section removes the correlation of the signal waveform and takes out the residue by an LPC inverse filter and pitch inverse filter and sends the residue to an orthogonal transform circuit section. The LPC parameters from the LPC analysis circuit and the pitch parameters from the pitch analysis circuit are sent to a bit allocation calculation circuit.

Abstract: A tuner 1 receives a TV-broadcast signal from an antenna 6. The TV-broadcast signal is supplied to a demodulator 2. The demodulator 2 demodulates the audio signal and video signal, both contained in the TV-broadcast signal. A CM-detecting section 4 is provided. In the section 4, characteristic patterns are generated at intervals of 15 seconds, 30 seconds and 60 seconds, from the audio and video signals. Then, CM candidates are detected in accordance with the characteristic pattems. Characteristic data representing the characteristics of a CM is extracted from each CM candidate or the audio and video signals preceding and following the CM candidate. The section 4 detects a CM from the characteristic data thus extracted.

Abstract: A signal processing device and a signal processing method can accurately detect CMs out of broadcast signals and also be used for storing, accessing, retrieving and viewing/listening to CMs. CM detecting section (202) detects CM sections out of a broadcast signal. CM extracting section (201) extracts the signal part for the CM section from the broadcast signal on the basis of the CM section detection signal (202a) of the CM detecting section (202) and CM characteristics extracting section (203) extracts the characteristic value of the CM. Then, CM recording section (205) records the signal of the CM section and the characteristic value. CM index generating section (206) generates CM index information by using the signal of the CM section and the characteristic value and characteristics comparing section (204) determines agreement/disagreement of CMs.

Abstract: An apparatus and a method for encoding an input signal on the time base through orthogonal transform, comprising a step of removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal 10 is sent to normalization circuit section 11 and (LPC) analysis circuit 39. The normalization circuit section 11 removes the correlation of the signal waveform and takes out the residue by means of LPC inverse filter 12 and pitch inverse filter 13 and sends the residue to orthogonal transform circuit section 25. The LPC parameters from the lop analysis circuit 39 and the pitch parameters from the pitch analysis circuit 15 are sent to bit allocation calculation circuit 41.

Abstract: An apparatus and a method for encoding an input signal on the time base through orthogonal transform, comprising a step of removing the correlation of signal waveform on the basis of the parameters obtained by means of linear predictive coding (LPC) analysis and pitch analysis of the input signal on the time base prior to the orthogonal transform. The time base input signal from input terminal 10 is sent to normalization circuit section 11 and (LPC) analysis circuit 39. The normalization circuit section 11 removes the correlation of the signal waveform and takes out the residue by means of LPC inverse filter 12 and pitch inverse filter 13 and sends the residue to orthogonal transform circuit section 25. The LPC parameters from the lop analysis circuit 39 and the pitch parameters from the pitch analysis circuit 15 are sent to bit allocation calculation circuit 41.

Abstract: An audio signal processing apparatus and method using pitch information to change a length of predictive residual signals while maintaining continuity and thereby enabling conversion of a reproduction speed without changing a pitch and enabling a conversion of speed by a small amount of calculation, comprising shortening or extending residual signals on a time axis while maintaining pitch information, cutting out signals and connecting of different pitch sections in the respective frames based on resemblance of signals at the time of shortening, and extending predictive residual signals in respective frames by extrapolation at the time of extension. An audio signal compressed or expanded on the time axis can be reproduced without changing the pitch by synthesizing an audio signal by an LPC synthesis filter based on the generated new predictive residual signals.

Abstract: A small-sized audio signal reproducing apparatus for hearing reproduced audio signals with the aid of a headphone is disclosed. Digitized and compression encoded audio signals, stored in a semiconductor memory, are read out so as to undergo a decoding operation, which is an inverse operation to compression encoding, to reproduce the audio signals, and the reproduced signals are heard by the headphone. The apparatus may be significantly reduced in size and weight as compared to the apparatus in which a tape or a disk is used as the recording medium.

Abstract: A phase quantization method and apparatus in which the phase information of the input signal such as at the time of the sinusoidal synthesis encoding can be quantized efficiently. The phase of the input signal derived from speech signals from an input terminal 11 is found by a phase detection unit 12 and scalar-quantized by a scalar quantizer 13. The spectral amplitude weighting k of each harmonics is calculated by a weighting calculation unit 18 based on the LPC coefficients from a terminal 17. Using the weighting k, a bit allocation calculation unit 19 calculates an optimum number of quantization bits of respective harmonics to send the calculated optimum number to the scalar quantizer 13.

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: An apparatus and procedure for performing phase detection in which one-pitch cycle of an input signal waveform is cut out on a time axis. The cut-out one pitch cycle is filled with zeroes to form 2N samples (where N is an integer, 2N is equal to or greater than the number of samples of the one-pitch cycle), and the samples are subjected to an orthogonal conversion such as fast Fourier transform, whereby a real and imaginary part are used to calculate tan−1 to obtain a basic phase information. This basic phase is subjected to linear interpolation to obtain phases of respective higher harmonics of the input signal waveform.

Abstract: A pitch detection method and apparatus capable of realizing high-precision pitch detection even for speech signals in which half-pitch or double-pitch exhibits stronger autocorrelation than the pitch for detection. An input speech signal is judged as to voicedness or unvoicedness and a voiced portion and an unvoiced portion of the input speech signal are encoded by a sinusoidal analytic encoding unit 114 and by a code excitation encoding unit 120, respectively, for producing respective encoded outputs. The sinusoidal analytic encoding unit 114 performs pitch search on the encoded outputs for finding the pitch information from the input speech signal and sets the high-reliability pitch information based on the detected pitch information. The results of pitch detection are determined based on the high-reliability pitch information.

Abstract: Phase detection apparatus and method, wherein a one-pitch cycle is cutout of an input signal waveform and then that cutout one-pitch cycle is dimension converted into 2.sup.k -sample data. The data conversion is performed on respective higher harmonic components of the input signal according to a real part and an imaginary part of the orthogonal data that has been converted.

Abstract: A speech analysis method and a speech encoding method and apparatus in which, even if the harmonics of the speech spectrum are offset from integer multiples of the fundamental wave, the amplitudes of the harmonics can be evaluated correctly for producing a playback output of high clarity. To this end, the frequency spectrum of the input speech is split on the frequency axis into plural bands in each of which pitch search and evaluation of amplitudes of the harmonics are carried out simultaneously using an optimum pitch derived from the spectral shape. Using the structure of an harmonics as the spectral shape, and based on the rough pitch previously detected by an open-loop rough pitch search, a high-precision pitch search comprised of a first pitch search for the frequency spectrum in its entirety and a second pitch search of higher precision than the first pitch search is carried out.

Abstract: A speech encoding method, a speech decoding method and corresponding apparatus capable of outputting non-buzzing spontaneous playback speech in a voiced portion includes a sinusoidal analysis encoding unit on the decoder side that detects the pitch of the voiced portion of the input speech signal. The pitch intensity information, which is a parameter containing the information representing not only the pitch intensity of the input speech signal but also the information representing proximity to the voiced speech or the unvoiced speech of the speech signal, is generated by a voiced/unvoiced (V/UV) discrimination unit and pitch intensity information generating circuit. The pitch intensity data is sent along with the encoded speech signal to the encoding side which then adds the noise component controlled on the basis of the pitch intensity information to the voiced portion of the encoded speech signal in a voiced speech synthesis portion and decodes and outputs the resulting signal.

Abstract: A signal coding apparatus including normalization circuit 101 for carrying out a linear prediction or the like to extract features of an input signal and for carrying out whitening. The signal whitened is transmitted to a T/F (time axis/frequency axis) conversion circuit 102 where the signal is subjected to a conversion such as the Modified Discrete Cosine Transform so as to obtain a coefficient y on the frequency axis. This coefficient y is supplied to a quantization (scalar quantization (SQ) and vector quantization (VQ)) circuit 103. A bit allocation circuit 104 uses the coefficient y and parameters such as a Linear Predictive Coding coefficient and a pitch from the normalization circuit 101 for carrying out quantization bit allocation for each coefficient. The quantization (SQ and VQ) circuit 103, according to this allocation bit, controls whether to carry out the SQ or VQ for each coefficient.

Abstract: A speech synthesizing method and apparatus arranged to use a sinusoidal waveform synthesis technique provide for preventing degradation of acoustic quality caused by the shift of the phase when synthesizing a sinusoidal waveform. A decoding unit decodes the data from an encoding side. The decoded data is transformed into the voiced/unvoiced data through a bad frame mask unit. Then, an unvoiced frame detecting circuit detects an unvoiced frame from the data. If there exist two or more continuous unvoiced frames, a voiced sound synthesizing unit initializes the phases of a fundamental wave and its harmonic into a given value such as 0 or .pi./2. This makes it possible to initialize the phase shift between the unvoiced and the voiced frames at a start point of the voiced frame, thereby preventing degradation of acoustic quality such as distortion of a synthesized sound caused by dephasing.

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.