Abstract: A coding unit codes an audio signal by using a vector quantization method to reduce the quantity of data. An audio code having a minimum distance among auditive distances between sub-vectors produced by dividing an input vector and audio codes in a transmission-side code book is selected. A portion corresponding to an element of a sub-vector having a high auditive importance is handled in an audio code selecting unit while neglecting the codes indicating phase information and subjected to comparative retrieval with respect to audio codes in a transmission-side code book. Extracted phase information corresponding to an element portion of the sub-vector is added to the result obtained and output as a code index. Thereby, the calculation amount in the code retrieval of vector quantization and the number of codes in the code book are decreased without lowering the quality of an audio signal.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

Abstract: A portion of an audio signal is separated into multiple frames from which one or more different features are extracted. These different features are used, in combination with a set of rules, to classify the portion of the audio signal into one of multiple different classifications (for example, speech, non-speech, music, environment sound, silence, etc.). In one embodiment, these different features include one or more of line spectrum pairs (LSPs), a noise frame ratio, periodicity of particular bands, spectrum flux features, and energy distribution in one or more of the bands. The line spectrum pairs are also optionally used to segment the audio signal, identifying audio classification changes as well as speaker changes when the audio signal is speech.

Abstract: At least one coded binary audio flux organized into frames is created from digital audio signals which were coded by transforming them from the time domain into the frequency domain. Transform coefficients of the signals in the frequency domain are quantized and coded according to a set of quantizers. The set is determined from a set of values extracted from the signals. The values make up selection parameters of the set of quantizers. The parameters are also present in the frames. A partial decoding state decodes then dequantizes transform coefficients produced by the coding based on a set of quantizers determined from the selection parameters contained in the frames of the coded binary audio flux or of each coded binary audio flux. The partially decoded frames are subjected to processing in the frequency domain. The thus-processed frames are then made available for use in a later utilization step.

Abstract: A method for data compression. An encoder receives data vectors from an original data set. The encoder uses a vector quantization codebook to encode the data vectors into encoded vectors. The codebook is constructed from a compound data set, where the compound data set includes real data vectors and artificial data vectors. The encoded vectors are indexed in the codebook and the indexes are transmitted across communication channels or transmitted to storage.

Abstract: A system and method for applying a linear transformation to classify and input event. In one aspect, a method for classification comprises the steps of capturing an input event; extracting an n-dimensional feature vector from the input event; applying a linear transformation to the feature vector to generate a pool of projections; utilizing different subsets from the pool of projections to classify the feature vector; and outputting a class identity of the classified feature vector. In another aspect, the step of utilizing different subsets from the pool of projections to classify the feature vector comprises the steps of, for each predefined class, selecting a subset from the pool of projections associated with the class; computing a score for the class based on the associated subset; and assigning, to the feature vector, the class having the highest computed score.

Abstract: A line spectral frequency (LSF) coefficient vector quantizer greatly affects wideband speech coding efficiency and performance. An LSF coefficient quantizer of an existing speech codec can be modified into a new structure in which a non-structural vector quantizer and a lattice quantizer are connected in series. Thus, memory capacity and search time required for the LSF coefficient quantizer can be reduced. In addition, a prediction structure and a non-prediction structure can be connected in parallel to stably perform quantization and reduce a quantization transfer error. As a result, an efficient LSF quantizer capable of reducing allocated bits and improving SD can be provided. Moreover, non-structural vector quantization can be performed prior to pyramid vector quantization to convert an input value into a Laplacian model suitable for a pyramid vector quantizer.

Abstract: Linear predictive coding (LPC) filter parameters are determined for use in encoding a voice signal. Samples of a speech signal using a z-transform function are pre-emphasized. The pre-emphasized samples are analyzed to produce LPC reflection coefficients. The LPC reflection coefficients are quantized by a voiced quantizer and by an unvoiced quantizer producing sets of quantized reflection coefficients. Each set is converted into respective spectral coefficients. The set which produces a smaller lag-spectral distance is determined. The determined set is selected to encode the voice signal.

Abstract: A speech encoder that analyzes and classifies each frame of speech as being periodic-like speech or non-periodic like speech where the speech encoder performs a different gain quantization process depending if the speech is periodic or not. If the speech is periodic, the improved speech encoder obtains the pitch gains from the unquantized weighted speech signal and performs a pre-vector quantization of the adaptive codebook gain GP for each subframe of the frame before subframe processing begins and a closed-loop delayed decision vector quantization of the fixed codebook gain GC. If the frame of speech is non-periodic, the speech encoder may use any known method of gain quantization.

Abstract: A decoder for packetized speech with differential quantization of line spectral frequencies and fixed-codebook gain conceals erased frames with interpolation of future and past frames by reconstruct future frame predicted parameters from presumed interpolations of erased frame parameters.

Abstract: A system and method reduces the effects of the bit-error induced distortion of decoded voice transmission by assigning vectors that are close or similar in Euclidean distance to respective indices that are close in Hamming distance. The system calculates a first distortion sum of the distance error induced by single, double or N bit error possibilities, switches vector assignments and calculates a second distortion sum. If the second sum is less than the first sum the vector swap is maintained.

Abstract: A system controller (106) is for transferring a low bit rate digital voice message. The system controller generates from an analog voice signal representing the voice message a set of speech model parameters, and generates a first derived set of speech model parameters from a first subset of the set of speech model parameters, the first derived set encoding the voice signal at a second voice quality and second vocoder rate that are less, respectively, than a first voice quality and vocoder rate. The system controller transmits (3610) the low bit rate-digital voice message comprising the first derived set of speech model parameters to a communication receiver (114). The communication receiver requests (3640) an incremental message when the quality of the voice message is unsatisfactory. The system controller generates and transmits (3555, 3650) an incremental message-and the communication receiver uses (3660) the incremental message to generate a higher quality voice message.

Abstract: A receiver is used in decoding a received encoded signal. The received encoded speech signal is encoded using excitation linear prediction. The receiver receives the encoded speech signal. The encoded speech signal comprises a code, a pitch lag and a line spectral pair index. An innovation sequence is produced by selecting a code from each of a plurality of codebooks based on the code index. A line spectral pair quantization of a speech signal is determined using the line spectral pair index. A pitch lag is determined using the pitch lag index. A speech signal is reconstructed using the produced innovation sequence, the determined line spectral pair quantization and pitch lag.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A speech coder for high quality coding speech signals at low bit rates is disclosed. An excitation quantization unit 12 expresses an excitation signal in terms of a combination of a plurality of pulses. A codebook (i.e., an amplitude codebook 13) collectively quantizes either amplitude or position of pulses, and executes excitation signal quantization other parameter by making retrieval of the codebook.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: According to one aspect of the invention, a method is provided in which audio samples representing an input audio signal are received. The input audio samples are transformed into a vector of spectral values in a frequency domain. A value of a quantizing parameter is determined that satisfies one or more criteria based, at least in part, on a modified Newtonian search process, the determined value of the quantizing parameter being used to quantize the respective vector of spectral values to generate a vector of quantized values.

Abstract: A linear predictive analysis-by-synthesis encoder includes a search algorithm block (50) and a vector quantizer (58) for vector quantizing optimal gains from a plurality of subframes in a frame. The internal encoder states are updated (50, 52, 54, 56) using the vector quantized gains.

Abstract: A communication system (20) employs fixed rate channel-optimized, trellis-coded quantization (COTCQ) at a plurality of diverse encoding bit rates. COTCQ is performed through a COTCQ encoder (40) and COTCQ decoder (54). The COTCQ encoder and decoder (40,54) each include a codebook table (62) having at least one codebook (64) for each encoding bit rate. Each codebook (64) is configured in response to the bit error probability of the channel (26) through which the communication system (20) communicates. The bit error probability influences codebooks through the calculation of channel transition probabilities for all combinations of codewords (90) receivable from the channel (26) given all combinations of codewords (90) transmittable through the channel (26). Channel transition probabilities are responsive to base channel transition probabilities and the hamming distances between indices for codewords within subsets of the transmittable and receivable codewords.

Abstract: According to the invention, quantization encoding is conducted using the probability density function of the source, enabling fixed, variable and adaptive rate encoding. To achieve adaptive encoding, an update is conducted with a new observation of the data source, preferably with each new observation of the data source, preferably with each new observation of the data source. The current probability density function of the source is then estimated to produce codepoints to vector quantize the observation of the data source.

Abstract: A codebook excited linear prediction coding system providing improved digital speech coding for high quality speech at low bit rates with side-by-side codebooks for segments of the modeled input signal to reduce the complexity of the codebook search. A linear predictive filter responsive to an input signal desired to be modeled is used for identifying a basis vector from a first codebook over predetermined intervals as a subset of the input signal. A long term predictor and a vector quantizer provide synthetic excitation of modeled waveform signal components corresponding to the input signal desired to be modeled from side-by-side codebooks by providing codevectors with concatenated signals identified from the basis vector over the predetermined intervals with respect to the side-by-side codebooks. Once a codevector is identified, the codebook at the next segment is searched and a concatenation of codevectors is provided by selecting up to but not including the current segment.

Abstract: A speech and musical signal codec employing a band splitting technique encodes sound source signals of each of a plurality of bands using a small number of bits. The codec includes a second pulse position generating circuit, to which an index output by a minimizing circuit and a first pulse position vector P{overscore ( )}=(P1, P2, . . . , PM) are input, for revising the first pulse position vector using a pulse position revision quantity d{overscore ( )}i=(di1, di2, . . . , diM) specified by the index and outputting the revised vector to a second sound source generating circuit as a second pulse position vector P{overscore ( )}t=(P1+di1, P2+di2, . . . , PM+diM).

Abstract: Wideband speech is synthesized from a bandlimited speech signal, for example from speech which has been transmitted via the public switched telephone network. Due to the nature of the vocal tract, there is a correlation between a bandlimited signal and those parts of an original wideband speech signal which are missing from that signal. Narrowband speech is characterized in terms of estimated formant frequencies provided by a peak picker. The frequency of formants in speech give a good indication, for voiced sounds, as to the shape of the vocal tract. The set of frequencies provided by the peak picker is used to access a codebook which provides synthesis parameters for use by a synthesizer.

Abstract: A method for encoding speech at a low bit rate. The method assembles parameters on N consecutive frames to form a super-frame. A vector quantization of transition frequencies of a voicing during each super-frame is made. Only the most frequent configurations are transmitted without deterioration and the least frequent configurations are replaced by the configuration that is the nearest in terms of absolute error among most frequent configurations. The pitch is encoded in carrying out a scalar quantization of only one value of the pitch for each super-frame. The energy is encoded in selecting only a reduced number of values in assembling these values in sub-packets quantized by vector quantization. The spectral envelope parameters are encoded by vector quantization in selecting only a determined number of filters. The untransmitted energy values are recovered in the synthesis part by interpolation or extrapolation from transmitted values. Such a method may find particular application in vocoders.

Abstract: A method of searching an MP-MLQ fixed codebook through bit predetermination includes the steps of generating a target vector with amplitude, reducing time to search an optimal pulse array through the bit predetermination and searching all of pulses if two errors have an identical value.

Abstract: A method for objective speech quality assessment that accounts for phonetic contents, speaking styles or individual speaker differences by distorting speech signals under speech quality assessment. By using a distorted version of a speech signal, it is possible to compensate for different phonetic contents, different individual speakers and different speaking styles when assessing speech quality. The amount of degradation in the objective speech quality assessment by distorting the speech signal is maintained similarly for different speech signals, especially when the amount of distortion of the distorted version of speech signal is severe. Objective speech quality assessment for the distorted speech signal and the original undistorted speech signal are compared to obtain a speech quality assessment compensated for utterance dependent articulation.

Abstract: ITU Recommendation G.729 Annex E teaches in the implementation of a fixed codebook search to determine the selected sample combination providing the minimal difference between the original input speech and the reconstructed speech after implementation of the codec. A large number of sample sets are processed and the difference between the original input signal and the reconstructed signal for each set is determined and stored in a register. Under certain conditions, the register can overflow resulting in invalid difference values. When such a condition occurs, the fixed codebook search cannot determine the sample combination providing the minimal mean square error between the weighted input speech and the weighted reconstructed speech. An initialization vector for the codvec vector is used to provide valid data which conforms to the G.729 Annex E specifications and minimizes changes to the G.729 source code while providing robust quality signal processing in the event of register overflow condition.

Abstract: To transmit data over a mobile telephone speech channel a source encoder is replaced by a transcoder, a conversion table and/or a concatenation circuit in order to choose from the words produced by the source encoder the ones that are the most robust and which can without difficulty withstand speech synthesis followed by an inverse analysis to reconstitute streams of the bits of the data to be transmitted.

Abstract: A vector quantizing apparatus, a decoding apparatus, a vector quantization method, and a decoding method are provided. Upon encoding of a speech signal by the vector quantization apparatus and method, the advantages of vector quantization are maximized by quantizing the speech signal using KLT-based classified codebooks and the eigenvalues and eigenvectors of the speech signal. The vector quantization apparatus includes a codebook group, a Karhunen-Loéve Transform (KLT) unit, first and second selection units and a transmission unit. The codebook group has a plurality of codebooks that store the code vectors for a speech signal, and the codebooks are classified using KLT domain statistics for the speech signal. The KLT unit transforms an input speech signal to a KLT domain. The first selection unit selects an optimal codebook from the codebooks in the codebook group on the basis of the eigenvalue set of the covariance matrix of the input speech signal obtained by KLT.

Abstract: A fast search method for LSP (Linear Spectrum Pair) quantization is provided. The fast search method in accordance with an embodiment of the present invention includes the following steps. A first step is obtaining a target vector and a code vector. The target vector and the code vector are converted for ordering property. A second step is generating a code book having the ordering property for sub-matrices by utilizing the target vector and the code vector. A third step is selecting a particular line for determining a search scope in the code books and sorting the code book in descending order with respect to component values of the particular line. A fourth step is determining the search scope by utilizing the ordering property of the target vector and the sorted code vectors. The fifth step is obtaining an error standard by utilizing the target vector and the code vector, and obtaining an optimal code vector by utilizing the error standard within the determined search scope.

Abstract: The processing volume for codebook search for vector quantization is diminished by sending data representing an envelope of spectral components of the harmonics from a spectrum evaluation unit 148 of a sinusoidal analytic encoder 114 to a vector quantizer 116 for vector quantization, so that the degree of similarity between an input vector and all code vectors stored in the codebook is found by approximation for pre-selecting a smaller number of code vectors. From these pre-selected code vectors, such a code vector minimizing an error with respect to the input vector is ultimately selected. In this manner, a smaller number of candidate code vectors are pre-selected by pre-selection involving simplified processing and subsequently subjected to ultimate selection with high precision.

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: In a CELP coder a comparison between a target signal and a plurality of synthetic signals is made. The synthetic signal is derived by filtering a plurality of excitation sequences by a synthesis filter having parameters derived from the target signal. The excitation signal which results in a minimum error between the target signal and the synthetic signal is selected. The search for the best excitation signal requires a substantial computational complexity. To reduce the complexity a preselection of a small number of excitation sequences is made by selecting a small number of excitation sequences resembling the most a backward filtered target signal. With this small number of excitation sequences a full complexity search is made. Due to the reduced number of excitation sequences involved in the final selection the required computational complexity is reduced.

Abstract: A variable dimension spectral magnitude quantization apparatus and method using a predictive and mel scale binary vector is provided.

Abstract: In a CELP coder a comparison between a target signal and a plurality of synthetic signals is made. The synthetic signal is derived by filtering a plurality of excitation sequences from a one dimensional codebook by a synthesis filter having parameters derived from the target signal. The excitation signal that results in a minimum error between the target signal and the synthetic signal is selected. In order to reduce the complexity of the search for the best excitation signal, the selection is done in two stages. First a preselection of a small number of excitation sequences is made by selecting only every L.sup.th codebook entry for preselecting a plurality of excitation sequences. Thereafter, with this small number of excitation sequences, a fill complexity search is made in which all excitation sequences surrounding the preselected ones are involved in the selection.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: In a CELP coder, a comparison between a target signal and a plurality of synthetic signals is made. The synthetic signal is derived by filtering a plurality of excitation sequences by a synthesis filter having parameters derived from the target signal. The excitation signal which results in a minimum error between the target signal and synthetic signal is selected. The search for the best excitation signal requires a substantial computational complexity. To reduce the complexity, a preselection of a small number of excitation sequences is made using a reduced complexity synthesis filter. With this small number of excitation sequences, a full complexity search is made. Due to the reduced number of excitation sequences involved in the final selection, the required computational complexity is reduced.

Abstract: A lattice-structured multiple description vector quantization (LSMDVQ) encoder generates M descriptions of a signal to be encoded, each of the descriptions being transmittable over a corresponding one of M channels. The encoder is configured based at least in part on a distortion measure which is a function of a central distortion and at least one side distortion. For example, if M=2, the distortion measure may be an average mean-squared error (AMSE) function of the form ƒ(D0, D1, D2), where D0 is a central distortion resulting from reconstruction based on receipt of both a first and a second description, and D1 and D2 are side distortions resulting from reconstruction using only a first description and a second description, respectively. Further performance improvements may be obtained through perturbation of the lattice points.

Abstract: Input waveform data are processed by Fast Fourier Transform or otherwise to separate a harmonic wave component from the waveform data while a non-harmonic wave component is separated by subtracting the harmonic wave component from the waveform data. Vector quantization is performed on the harmonic wave component by selecting and using one of prestored harmonic vectors as a representative vector for the harmonic wave component, and vector quantization is performed on the non-harmonic wave component, independently of the harmonic wave component, by selecting and using one of prestored non-harmonic vectors as a representative vector for the non-harmonic wave component. Then, using harmonic and non-harmonic vectors indicated by vector information of a waveform to be reproduced, waveforms of harmonic and non-harmonic wave components are generated separately and then combined to thereby reproduce/generate the waveform.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: In this speech encoding system, the limiter circuit is input with the delay of adaptive codebook obtained for the previous subframe, and the pitch cycle search range is limited so that the delay of adaptive codebook obtained for the previous subframe is not discontinuous to the delay of adaptive codebook to be obtained for the current subframe, and the pitch cycle search range limited is output to the pitch calculation circuit. The pitch calculation circuit is input with output signal Xw(n) of the perceptual weighting circuit and the pitch cycle search range output from the limiter, calculating the pitch cycle Top, then outputting at least one pitch cycle Top to the adaptive codebook circuit.

Abstract: A speech compression system capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech is disclosed. The speech compression system optimizes the bandwidth consumed by the bitstream by balancing the desired average bit rate with the perceptual quality of the reconstructed speech. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. The codecs are selectively activated based on a rate selection. In addition, the full and half-rate codecs are selectively activated based on a type classification. Each codec is selectively activated to encode and decode the speech signals at different bit rates emphasizing different aspects of the speech signal to enhance overall quality of the synthesized speech.

Abstract: A multi-rate speech codec supports a plurality of encoding bit rate modes by adaptively selecting encoding bit rate modes to match communication channel restrictions. In higher bit rate encoding modes, an accurate representation of speech through CELP (code excited linear prediction) and other associated modeling parameters are generated for higher quality decoding and reproduction. To achieve high quality in lower bit rate encoding modes, the speech encoder departs from the strict waveform matching criteria of regular CELP coders and strives to identify significant perceptual features of the input signal. The encoder generates pluralities of codevectors from a single, normalized codevector by shifting or other rearrangement. As a result, searching speeds are enhanced, and the physical size of a codebook built from such codevectors is greatly reduced.

Abstract: A digital filter design apparatus for noise suppression by spectral subtraction includes a first spectrum estimator for determining a high frequency resolution noisy speech power spectral density estimate from a noisy speech signal block. A second spectrum estimator determines a high frequency resolution background noise power spectral density estimate from a background noise signal block. Averaging units form a piece-wise constant noisy speech power spectral density estimate and a piece-wise constant background noise power spectral density estimate. These averaging units are controlled by devices for adapting the length of individual segments to the shape of the high frequency-resolution noisy speech power spectral density estimate and for using the same segmentation in both piecewise constant estimates.

Abstract: A method of performing an excitation Vector Quantization (VQ) in a Noise Feedback Coding environment involves reorganizing a calculation of an energy of an error vector for each of a plurality of candidate excitation vectors of a codebook. The energy of the error vector is a cost function that is minimized during a search of the codebook for a best candidate excitation VQ vector. The reorganization includes expanding a Mean Squared Error (MSE) term of the error vector, excluding an energy term that is invariant to the candidate excitation vector, and pre-computing energy terms of ZERO-STATE responses of the candidate excitation vectors that are invariant to sub-vectors of a subframe. Another method searches a signed codebook. Both methods use correlation techniques.

Abstract: A method for searching a codebook which predicts a residual element of an input voice signal includes combining each track of the input signal, forming track units including at least two tracks, and determining a pulse code for each track. The method further includes calculating energy for each track using an energy formula including a vector dot product, arranging or selecting codewords in a small track energy order, and searching or selecting an optimal pulse for a single- or double-pulse track of the selected codeword.

Abstract: An encoder which encodes a voice in accordance with LD-CELP (Low-Delay Code Excited Linear Prediction) of the ITU-T Recommendation G.728. When a vibration wave is encoded by vector quantization, the code is secretly combined with other data. The encoder stores dividing key data kidx by which 128 types of representative vector data (waveform codes) yj; j=0, 1, . . . , 127 are labeled with 0 or 1 in order from the uppermost bit. If the bit is “0”, the vectors are quantized by using only the waveform codes yj corresponding to the bit “0” of the dividing key data kidx as the selection objects. If the bit is “1”, the vectors are quantized by using only the waveform codes yj corresponding to the bit “1” of the dividing key data kidx as the selection objects. Thus, the outputted voice code is combined with another datum bit.

Abstract: A scheme to judge emphasized speech portions, wherein the judgment is executed by a statistical processing in terms of a set of speech parameters including a fundamental frequency, power and a temporal variation of a dynamic measure and/or their derivatives. The emphasized speech portions are used for clues to summarize an audio content or a video content with a speech.

Abstract: Data is transmitted using multiple description vector quantization by first quantizing the source vector at a lattice vector quantizer. After quantization, a labeling function is applied to the quantized source vector, creating a plurality of data streams. Each data stream is encoded and transmitted over a separate channel. Furthermore, the encoded data is decoded by first retrieving a representation of a sublattice point from a set of sublattice points from the data stream and then, determining a single data code word being associated with the retrieved representation of the sublattice point.

Abstract: Representative vectors Z1i and Z2j are selected from code-books codebooks CB1 and CB1 CB2, respectively, and multiplied by weighting coefficient vectors w1 and w2 of the same number of dimensions as those of the representative vectors, whereby weighted representative vectors Z1iw1 and Z2jw2 are generated. These weighted representative vectors are vector combined into a combined vector yij, and a combination of the representative vectors is selected by a control part in such a manner as to minimize the distance between the combined vector yij and an input vector X. The weighting coefficient vectors w1 and w2 each have a maximum component in a different dimension and are selected so that the sum of diagonal matrixes W1 and W2 using components of the weighting coefficient vectors as their diagonal elements becomes a constant multiple of the unit matrix.