Abstract: When there are missing voice-transmission-signals, a repetition-section calculating unit sets a plurality of repetition sections of different lengths that are determined to be similar to the voice-transmission-signals preceding the missing voice-transmission-signal, the repetition sections being determined with respect to stationary voice-transmission-signals stored in a normal signal storage unit, the stationary voice-transmission-signals being selected from the previously input voice-transmission-signals. A controller generates a concealment signal using the repetition sections.

Abstract: In an echo processing method and device which can detect an accurate echo section without effects of a far end signal, an echo delay, and a reduction of an echo cancellation amount, a signal of a specified frequency band is generated in conformity with a near end signal, and the signal of the specified frequency band is added to the near end signal to form a transmitting signal. Receiving signals are separated into the signal of the specified frequency band and a signal of a band other than the specified frequency band. An echo section is detected based on the signal of the specified frequency band separated. An echo component in the signal of the band other than the specified frequency band is removed and a level of the echo component is detected based on the near end signal in the echo section.

Abstract: In a pitch conversion method and device which can reduce data throughput while suppressing a degradation of sound quality due to a pitch conversion as much as possible, an input signal pitch pattern per predetermined processing unit and a target pitch pattern are inputted, and a degradation degree indicating how a waveform of the input signal degrades upon pitch conversion from the input signal pitch pattern to the target pitch pattern is calculated. Alternatively, a degradation degree corresponding to a voice state and a phonemic type of the input signal is extracted from a database in which all of combinations of voice states and phonemic types estimated are associated with the degradation degrees to be recorded. Then, a pitch converter which performs a pitch conversion with small data throughput and a pitch converter which performs a pitch conversion with large data throughput are switched over depending on the degradation degree.

Abstract: A data decryption apparatus that decrypts encrypted data, includes a first data-receiving unit that receives a first data set, in which information on an encryption specification is embedded, through a first communication path; a time-information obtaining unit that obtains time information on a reception of the first data set by the first data receiving unit; a time-information storage unit that stores the time information with the information on the encryption specification associated therewith; a second data-receiving unit that receives a second data set through a second communication path, the second data set being encrypted based on the encryption-specification and appended by time information on performing data encryption; and an encryption-specification selecting unit that selects an encryption specification for use in decryption of the second data set based on the time information stored in the time-information storage unit and the time information appended to the second data set.

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: In a signal processing method and device which enhance a following speed of an estimated noise in a steep rise section of a noise level and generate little estimation error of a noise spectrum due to an influence of voice in a voice section, a time domain signal that is sampled data of an input signal is extracted, the time domain signal is converted into a frequency domain signal per frame, and an input spectrum is calculated. Furthermore, a minimum value of the input spectrum is acquired, so that a noise spectrum that is a frequency domain signal of a noise component included in the input voice signal is estimated. Moreover, the input spectrum is compared with the noise spectrum, so that whether a section is in a noise section or a mixed section where voice and noise are mixed is determined.

Abstract: A noise suppressor includes a frequency division part dividing an input signal into bands and outputting band signals; an amplitude calculation part determining amplitude components of the band signals; a noise estimation part estimating an amplitude component of noise contained in the input signal and determining an estimated noise amplitude component for each band; a weighting factor generation part generating a different weighting factor for each band; an amplitude smoothing part determining smoothed amplitude components that are the amplitude components of the band signals temporally smoothed using the weighting factors; a suppression calculation part determining a suppression coefficient from the smoothed amplitude component and the estimated noise amplitude component for each band; a noise suppression part suppressing the band signals based on the suppression coefficients; and a frequency synthesis part synthesizing and outputting the band signals of the bands after the noise suppression output from the

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: The invention relates to speech speed conversion, and provides a speech speed converting device and a speech speed converting method for changing a speed of voice without degrading the voice quality, without changing characteristics, regarding a signal containing voice. The speech speed converting device includes: a voice classifying unit that is input with voice waveform data and a voice code based on a linear prediction, and that classifies the input signal based on the characteristic of the input signal; and a speed adjusting unit that selects either one of or both a speed conversion processing using the voice waveform and a speed conversion processing using the voice code, based on the classification, and that changes a speech speed of the input signal using the selected speed converting method.

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: 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: 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: An audio coder that improves audio quality by reducing a quantization error. When a code corresponding to a sampled value of an audio signal is determined, a candidate code storage section stores all combinations of candidate codes in a neighborhood interval of the sampled value. A local decoder generates reproduced signals by decoding the codes stored in the candidate code storage section. An error evaluation section calculates, for each candidate code, a sum of squares of differentials between input sampled values and reproduced signals, detects a combination of candidate codes by which a smallest sum is obtained, that is to say, which minimizes a quantization error, and outputs a code included in the detected combination of candidate codes.

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: 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: An audio coder that improves audio quality by reducing a quantization error. When a code corresponding to a sampled value of an audio signal is determined, a candidate code storage section stores all combinations of candidate codes in a neighborhood interval of the sampled value. A local decoder generates reproduced signals by decoding the codes stored in the candidate code storage section. An error evaluation section calculates, for each candidate code, a sum of squares of differentials between input sampled values and reproduced signals, detects a combination of candidate codes by which a smallest sum is obtained, that is to say, which minimizes a quantization error, and outputs a code included in the detected combination of candidate codes.

Abstract: In a packet receiving method and device which convert a voice packet received into a voice, a receiving packet buffer temporarily stores a voice packet received; a plurality of parameter information monitors respectively determine different buffer adjustment values for determining a buffering amount of the receiving packet buffer based on one or more pieces of parameter information obtained from the voice packet temporarily stored; a buffer adjustment value determiner determines a receiving buffer adjustment value from the plural buffer adjustment values; and a buffer controller controls the buffering amount based on the receiving buffer adjustment value.

Abstract: A code separation/decoding unit restores a vocal tract characteristic sp1 and a vocal source signal r1. A vocal tract characteristic modification unit modifies the vocal tract characteristic sp1 and outputs the modified vocal tract characteristic sp2. In this method, an emphasized vocal tract characteristic sp2 is generated to output by applying formant emphasis directly to the vocal tract characteristic sp1 for instance. A signal synthesis unit synthesizes the modified vocal tract characteristic sp2 and the vocal source signal r1 to generate and output an output voice, s.

Abstract: Communications from a transmission side to a reception side neither changing the format of voice code data nor requiring another transmission path or increasing the transmission quantity of control information are controlled utilizing information obtained on the reception side. A system includes a first communication equipment provided with a control information embedding unit for embedding control information that is used for a control of communications from a communication partner to the own communication equipment and that is obtained on the own communication equipment side in the communication data to be transmitted to the communication partner side and a second communication equipment provided with a communication control unit for controlling communications to the first communication equipment side using control information transmitted from the first communication equipment.

Abstract: In the voice synthesizer, which obtains a voice by emphasizing a specific part of a sentence, an emphasis degree deciding unit extracts a word or a collocation to be emphasized from among respective words or respective collocations on the basis of an extracting reference with respect to the each word or the each collocation included in a sentence and deciding an emphasis degree of the extracted word or the extracted collocation. An acoustic processing unit synthesizes a voice having an emphasis degree that is decided by the emphasis degree deciding unit provided to the word to be emphasized or the collocation to be emphasized. Whereby the emphasized part of the word or the collocation can be obtained automatically on the basis of the extracting reference such as a frequency of appearance and a level of importance of the word or the collocation, further, improves an operation-ability.