Abstract: A voice intensifier capable of reducing abrupt changes in the amplification factor between frames and realizing excellent sound quality with less noise feeling by dividing input voices into the sound source characteristic and the vocal tract characteristic, so as to individually intensify the sound source characteristic and the vocal tract characteristic and then synthesize them before being output.

Abstract: In a voice packet communication system, a voice packet loss concealment device compensates for the deterioration of voice quality due to voice packet loss. In the device, a detecting section detects a loss of a voice packet and outputting information; an estimating section estimates the voice characteristics of the lost segment using a pre-loss voice packet received before the lost segment or a post-loss voice packet received after the lost segment; a pitch signal generating section generates a pitch signal having the voice characteristics; and a lost packet generating section outputs the pitch signal generated by the pitch signal generating section, with the voice characteristics estimated by the estimating section, which allows abnormal noise and feeling of mute, subjective deterioration of naturalness and continuity to be improved, and the voice packet loss concealment to be further improved.

Abstract: A noise reduction apparatus includes an analysis unit for converting input into a signal of a frequency area, a suppression unit for suppressing the signal, and a synthesis unit for synthesizing a signal of a time area. The apparatus further includes an estimation unit for estimating, using the output of the analysis unit, information corresponding to at least pure voice element excluding noise element in an input voice signal as voice information which is the basic voice information for calculation of a suppression gain of a signal, and a unit for calculating a suppression gain corresponding to the output of the estimation unit and the analysis unit and providing it for the suppression unit.

Abstract: A voice activity detector that detects talkspurts in a given signal at a high accuracy, so as to improve the quality of voice communication. A frequency spectrum calculator calculates frequency spectrum of a given input signal. A flatness evaluator evaluates the flatness of this power spectrum by, for example, calculating the average of power spectral components and then adding up the differences between those components and the average. The resultant sum of differences, in this case, is used as a flatness factor of the spectrum. A voice/noise discriminator determines whether the input signal contains a talkspurt or not, by comparing the flatness factor of the frequency spectrum with a predetermined threshold.

Abstract: A data embedding device for embedding data in a speech code obtained by encoding a speech in accordance with a speech encoding method based on a voice generation process of a human being, includes an embedding judgment unit, every speech code, judging whether or not data should be embedded in the speech code, and an embedding unit embedding data in two or more parameter codes of a plurality of parameter codes constituting the speech code for which it is judged by the embedding judgment unit that the data should be embedded.

Abstract: A speech transcoder includes a codebook in which a plurality of algebraic codes conforming to a second encoding method to serve as conversion candidates of the algebraic code of a first speech code, and a limiting unit for limiting the plurality of algebraic codes stored in the algebraic codebook to at least one algebraic code having a value equal to that of embedded data embedded in a second speech code to limit the conversion candidates, a determination unit for determining an element code corresponding to a converted speech code from the limited conversion candidates.

Abstract: When a voice encoding apparatus embeds any data in voice code, the apparatus determines whether data embedding condition is satisfied using a first element code from among element codes constituting the voice code, and a threshold value. If the data embedding condition is satisfied, the apparatus embeds optional data in the voice code by replacing a second element code with the optional data. When a voice decoding apparatus extracts data that has been embedded in voice code, the apparatus determines whether data embedding condition is satisfied using a first element code from among element codes constituting the voice code, and a threshold value. If the data embedding condition is satisfied, the apparatus determines that optional data has been embedded in the second element code portion of the voice code and extracts this embedded data.

Abstract: When a voice encoding apparatus embeds any data in encoded voice code, the apparatus determines whether data embedding condition is satisfied using a first element code from among element codes constituting the encoded voice code, and a threshold value. If the data embedding condition is satisfied, the apparatus embeds optional data in the encoded voice code by replacing a second element code with the optional data. When a voice decoding apparatus extracts data that has been embedded in encoded voice code, the apparatus determines whether data embedding condition is satisfied using a first element code from among element codes constituting the encoded voice code, and a threshold value. If the data embedding condition is satisfied, the apparatus determines that optional data has been embedded in the second element code portion of the encoded voice code and extracts this embedded data.

Abstract: It is so arranged that a voice code can be converted even between voice encoding schemes having different subframe lengths. A voice code conversion apparatus demultiplexes a plurality of code components (Lsp1, Lag1, Gain1, Cb1), which are necessary to reconstruct a voice signal, from voice code in a first voice encoding scheme, dequantizes the codes of each of the components and converts the dequantized values of code components other than an algebraic code component to code components (Lsp2, Lag2, Gp2) of a voice code in a second voice encoding scheme. Further, the voice code conversion apparatus reproduces voice from the dequantized values, dequantizes codes that have been converted to codes in the second voice encoding scheme, generates a target signal using the dequantized values and reproduced voice, inputs the target signal to an algebraic code converter and obtains an algebraic code (Cb2) in the second voice encoding scheme.

Abstract: Disclosed is a voice encoding method having a synthesis filter implemented using linear prediction coefficients obtained by dividing an input signal into frames each of a fixed length, and subjecting the input signal to linear prediction analysis in the frame units, generating a reconstructed signal by driving said synthesis filter by a periodicity signal output from an adaptive codebook and a pulsed signal output from an algebraic codebook, and performing encoding in such a manner that an error between the input signal and said reproduced signal is minimized, wherein there are provided an encoding mode 1 that uses pitch lag obtained from an input signal of a present frame and an encoding mode 2 that uses pitch lag obtained from an input signal of a past frame. Encoding is performed in encoding mode 1 and encoding mode 2, the mode in which the input signal can be encoded more precisely is decided frame by frame and encoding is carried out on the basis of the mode decided.

Abstract: A gain unit scales a code vector Ci output from a configuration variable code book by a gain g after the positions of non-zero samples are controlled according to an index and transmission parameter p. A linear prediction synthesis filter input the multiplication result, and outputs a regenerated signal gACi. A subtracter outputs an error signal E by subtracting the regenerated signal gACi from an input signal X. A error power evaluation unit computes an error power according to an error signal E. The above described processes are performed on all code vectors Ci and gains g. The index i of the code vector Ci and the gain g with which the error power is the smallest are computed and transmitted to the decoder.

Abstract: A first CN code (silence code) obtained by encoding a silence signal, which is contained in an input signal, by a silence compression function of a first speech encoding scheme is transcoded to a second CN code of a second speech encoding scheme without decoding the first CN code to a CN signal. For example, the first CN code is demultiplexed into a plurality of first element codes by a code demultiplexer, the first element codes are each transcoded to a plurality of second element codes that constitute the second CN code, and the second element codes obtained by this transcoding are multiplexed to output the second CN code.

Abstract: The invention relates to a voice activity detecting device and a voice activity detecting method. An object of the invention is to adapt to various characteristics of noise which may possibly be superimposed on an aural signal to thereby reliably discriminate between an active voice segment and a non-active voice segment. For this purpose, the voice activity detecting device comprises: a speech-segment inferring section 11 for determining the probability that each of active voice frames given in order of time sequence belongs to the active voice segment, based on the statistical characteristic of the aural signal; a quality monitoring section 12 for monitoring the quality of the aural signal for each active voice frame, and a speech-segment determining section 13 for weighting the determined probability with the above quality to obtain for each active voice frame the accuracy that the active voice frame belongs to the active voice segment.

Abstract: The invention enables a source speech signal to be coded in an optimal way. An adaptive codebook (Ba) stores a series of signal vectors of a past speech signal. A vector extraction means extracts a signal vector and neighboring vectors adjacent thereto. A high-order long-term prediction synthesis filter produces a long-term predicted speech signal (Sna−1) from the signal vector and its neighboring vectors. A filter coefficients calculation means calculates filter coefficients of the long-term prediction synthesis filter. A perceptual weighting synthesis filter obtains a reproduced coded speech signal (Sna) from the long-term predicted speech signal (Sna−1). An error calculation means calculates the error (En) of the reproduced coded speech signal (Sna) with reference to the speech signal (Sn). A minimum error detection means finds a minimum error point that yields the smallest error among the calculated errors.

Abstract: The invention relates to a voice activity detecting device and a voice activity detecting method. An object of the invention is to adapt to various characteristics of noise which may possibly be superimposed on an aural signal to thereby reliably discriminate between an active voice segment and a non-active voice segment. For this purpose, the voice activity detecting device comprises: a speech-segment inferring section 11 for determining the probability that each of active voice frames given in order of time sequence belongs to the active voice segment, based on the statistical characteristic of the aural signal; a quality monitoring section 12 for monitoring the quality of the aural signal for each active voice frame, and a speech-segment determining section 13 for weighting the determined probability with the above quality to obtain for each active voice frame the accuracy that the active voice frame belongs to the active voice segment.

Abstract: Disclosed is a voice encoding method having a synthesis filter implemented using linear prediction coefficients obtained by dividing an input signal into frames each of a fixed length, and subjecting the input signal to linear prediction analysis in the frame units, generating a reconstructed signal by driving said synthesis filter by a periodicity signal output from an adaptive codebook and a pulsed signal output from an algebraic codebook, and performing encoding in such a manner that an error between the input signal and said reproduced signal is minimized, wherein there are provided an encoding mode 1 that uses pitch lag obtained from an input signal of a present frame and an encoding mode 2 that uses pitch lag obtained from an input signal of a past frame. Encoding is performed in encoding mode 1 and encoding mode 2, the mode in which the input signal can be encoded more precisely is decided frame by frame and encoding is carried out on the basis of the mode decided.

Abstract: Disclosed is a voice code conversation apparatus to which voice code obtained by a first voice encoding method is input for converting this voice code to voice code of a second voice encoding method. The apparatus includes a code separating unit for separating, from the voice code based upon the first voice encoding method, codes of a plurality of components necessary to reconstruct a voice signal, code converters for dequantizing the codes of each of the components and then quantizing the dequantized values by the second voice encoding method to thereby generate codes, and a code multiplexer for multiplexing the codes output from respective ones of the code converters and transmitting voice code based upon the second voice encoding method.

Abstract: Speech encoding using searching of a code book for a code that matches an input speech signal, and speech decoding using the code book are disclosed. A random series of code samples is stored in a buffer memory such as a ring buffer memory, and a basic vector generation unit generates basic vectors by applying an arbitrary shift to each of code series retrieved from the random series. Generation of the basic vectors may be performed according to, for example, an overlapping vector generation process. A code book generation unit extends the basic vectors contained in a basic vector unit according to a structuring process so as to produce a tree-structured delta code book. The basic vector generation unit may extend the basic vectors based on pitch parameters or a center clipping threshold.

Abstract: A speech coding apparatus includes multipliers and prediction filters which successively process a plurality of signal vectors obtained from an index 2.sup.M and dimension N code book to obtain a reproduced speech signal. Error detectors are provided which find the error between the input speech signal and reproduced speech signal. Evaluators are also provided which calculate the optimum signal vectors giving the smallest errors. The multipliers are connected to a reduced code book, which is constituted of n number of code book blocks of index 2.sup.M/n and dimension N/n (where n is an integer of two or more). There are n number of multipliers, n number of prediction filters, n number of error detectors, and n number of evaluators corresponding to the code book blocks.

Abstract: A speech coding apparatus which selects an optimum code from a code book, the optimum code giving the minimum magnitude of error signal between the input signal and the reproduced signal obtained by a filter calculation using a linear prediction parameter from a linear predictive analysis unit with respect to the codes of the code book, wherein the code book is formed by thinning to 1/M (M being an integer of two or more) the plurality of sampling values constituting the codes. To compensate for the deterioration of the quality of the reproduced signal caused by thinning the sampling values in this way, an additional linear predictive analysis unit is further introduced and use made of an amended linear prediction parameter instead of the linear prediction parameter from the originally provided linear predictive analysis unit.