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: A gain adjusting method and a gain adjusting device for adjusting gain of a processed voice signal that is obtained by signal processing an input voice signal are disclosed. According to the gain adjusting method, a masking property of the processed voice signal is computed, and gain is adjusted for every frequency if the frequency is masked according to the masking property, while canceling a difference between the processed voice signal and the input voice signal where the frequency is not masked.

Abstract: An encoding apparatus compresses a stereo signal using a sum signal and a difference signal of a left component signal and a right component signal of the stereo signal. The encoding apparatus includes a calculating unit that calculates complexity of the sum signal and complexity of the difference signal; a setting unit that sets, based on the complexity, an allocation rate of bits to be allocated in quantizing the sum signal and the difference signal; and a quantizing unit that quantizes the sum signal and the difference signal based on the allocation rate.

Abstract: An audio encoding apparatus comprising: a power calculation unit that calculates a power fluctuation ratio based on the input signal; a calculation unit that calculates a prediction gain fluctuation ratio based on the input signal; and a block length judging unit that selects one of encoding using a long block mode segmenting an input signal into frames each consisting of a predetermined number of samples and encoding each of the frames, and encoding using a short block mode segmenting each of the frames into short blocks and encoding each of the short blocks, based on the power fluctuation ratio and the prediction gain fluctuation ratio.

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: 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: 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: 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: 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: 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 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.