Abstract: An apparatus and method for selecting a set of channels from a plurality channels in a signal processor, the method comprising sampling each one of a plurality of channels and obtaining a binary representation of each one of the samples, arranging each one of the binary representations of samples into a series of bit planes from a most significant bit plane containing the most significant bit of each binary representation, to a least significant bit plane containing the least significant bit of each binary representation, determining those bit planes having binary representations that conform to a predetermined value criteria, and selecting a set of channels by summing bits from each one of those determined bit planes that conform to the predetermined value criteria.

Abstract: A speaker intent analysis system and method for validating the truthfulness and intent of a plurality of participants' responses to questions. A computer stores, retrieves, and transmits a series of questions to be answered audibly by participants. The participants' answers are received by a data processor. The data processor analyzes and records the participants' speech parameters for determining the likelihood of dishonesty. In addition to analyzing participants' speech parameters for distinguishing stress or other abnormality, the processor may be equipped with voice recognition software to screen responses that while not dishonest, are indicative of possible malfeasance on the part of the participants. Once the responses are analyzed, the processor produces an output that is indicative of the participant's credibility. The output may be sent to proper parties and/or devices such as a web page, computer, e-mail, PDA, pager, database, report, etc. for appropriate action.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: The present invention discloses a method for performing a frame erasure concealment to a higher-band signal, including: calculating a periodic intensity of a higher-band signal with respect to a lower-band signal; judging whether the periodic intensity of the higher-band signal is higher than or equal to a preconfigured threshold; if the periodic intensity of the higher-band signal is higher than or equal to the preconfigured threshold, using a pitch period repetition method to perform the frame erasure concealment to the higher-band signal of a current lost frame; and if the periodic intensity of the higher-band signal is lower than the preconfigured threshold, using a previous frame data repetition method to perform the frame erasure concealment to the higher-band signal of the current lost frame. The present invention further discloses a device for performing a frame erasure concealment to a higher-band signal and a speech decoder. The problem that the quality of the voice signal is lowered is avoided.

Abstract: One embodiment of the present invention provides a post-processing method of a modulation envelope resulting from an interference of two harmonics in a filter band. According to one embodiment, the method comprising filtering the modulation envelope with a band-pass filter bank, wherein a combination of demodulation and application of the band-pass filter on the modulation envelope enables use of identical techniques for resolved and unresolved harmonics. One embodiment of the present invention provides a method of determining whether a frequency band of an input signal includes unresolved harmonics. According to a further embodiment, in response to a determination that the frequency band includes unresolved harmonics, the method comprises obtaining a modulation envelope of the frequency band by demodulating the frequency band, obtaining one or more frequency bands from the modulation envelope, and determining an evidence value that one of the frequency bands originates from one of fundamental frequencies.

Abstract: A mixed time-domain/frequency-domain coding device and method for coding an input sound signal, wherein a time-domain excitation contribution is calculated in response to the input sound signal. A cut-off frequency for the time-domain excitation contribution is also calculated in response to the input sound signal, and a frequency extent of the time-domain excitation contribution is adjusted in relation to this cut-off frequency. Following calculation of a frequency-domain excitation contribution in response to the input sound signal, the adjusted time-domain excitation contribution and the frequency-domain excitation contribution are added to form a mixed time-domain/frequency-domain excitation constituting a coded version of the input sound signal. In the calculation of the time-domain excitation contribution, the input sound signal may be processed in successive frames of the input sound signal and a number of sub-frames to be used in a current frame may be calculated.

Abstract: A scalable decoder capable of avoiding deterioration in subjective quality of a listener. The scalable decoder for decoding core layer encoding data and extension layer encoding data including an extension layer gain coefficient, wherein a voice analysis section detects variation in power of a core layer decoding voice signal being obtained from the core layer encoding data, a gain attenuation rate calculating section (140) sets the attenuation intensity variable depending on variation in power, and a gain attenuation section (143) attenuates the extension layer gain coefficient in a second period preceding a first period according to a set attenuation intensity when extension layer encoding data in the first period is missing, thus interpolating the extension layer gain coefficient in the first period.

Abstract: A disclosed received voice playback apparatus includes a characteristic acquiring unit configured to acquire first frequency characteristic values obtained by resolving digital vocal signals that are based on received vocal signals into predetermined frequency bands, wherein each first frequency characteristic value corresponds to one of the predetermined frequency bands; a setting unit configured to obtain second frequency characteristic values, wherein each second frequency characteristic value is set for one of the predetermined frequency bands; a computing unit configured to compute a gain for each of the predetermined frequency bands based on a difference between the first frequency characteristic value and the second frequency characteristic value; and a characteristic changing unit configured to change the first frequency characteristic values of the digital vocal signals by multiplying the digital vocal signals by each of the gains corresponding to one of the predetermined frequency bands of the digit

Abstract: Characteristic features are extracted from an audio sample based on its acoustic content. The features can be coded as fingerprints, which can be used to identify the audio from a fingerprints database. The features can also be used as parameters to separate the audio into different categories.

Abstract: A wideband speech encoder according to one embodiment includes a lowband encoder and a highband encoder. The lowband encoder is configured to encode a lowband portion of a wideband speech signal as a set of filter parameters and an encoded excitation signal. The highband encoder is configured to calculate values for coding parameters that specify a spectral envelope and a temporal envelope of a highband portion of the wideband speech signal. The temporal envelope is based on a highband excitation signal that is derived from the encoded excitation signal. In one such example, the temporal envelope is based on a difference in levels between the highband portion and a synthesized highband signal, wherein the synthesized highband signal is generated according to the highband excitation signal and a set of highband filter parameters.

Abstract: Methods and apparatuses for encoding a signal and decoding a signal and a system for encoding and decoding are provided. The method for encoding a signal includes performing a classification decision process on high frequency signals of input signals, adaptively encoding the high frequency signals according to the result of the classification decision process, and outputting a bitstream including codes of low frequency signals of the input signals, adaptive codes of the high frequency signals, and the result of the classification decision process. The classification decision process is performed on the high frequency signals, and adaptive encoding or adaptive decoding is performed according to the result of the classification decision process, so the quality of voice and audio output signals is improved.

Abstract: A system distinguishes a primary audio source and background noise to improve the quality of an audio signal. A speech signal from a microphone may be improved by identifying and dampening background noise to enhance speech. Stochastic models may be used to model speech and to model background noise. The models may determine which portions of the signal are speech and which portions are noise. The distinction may be used to improve the signal's quality, and for speaker identification or verification.

Abstract: A system and method for automatically adjusting floor controls for a conversation is provided. Audio streams are received, which each originate from an audio source. Floor controls for a current configuration including at least a portion of the audio streams are maintained. Conversational characteristics shared by two or more of the audio sources are determined. Possible configurations for the audio streams are identified based on the conversational characteristics. An analysis of the current configuration and the possible configurations is performed. A change threshold is applied to the analysis. When the analysis satisfies the change threshold, the floor controls are automatically adjusted. The audio streams are mixed into one or more outputs based on the adjusted floor controls.

Abstract: An encoding device and an encoding method are provided for encoding by reducing the number of samples to be processed when encoding higher-band spectrum data according to lower-band spectrum data in a wide-band signal. The device and the method can obtain a high-quality decoded signal even if a large quantization distortion is caused in the lower-band spectrum data. When encoding higher-band spectrum data in a signal to be encoded, according to lower-band spectrum data in the signal, only for a part (a head portion) of the higher-band spectrum data, the lower-band spectrum data after being quantized is subjected to approximate partial search and higher-band spectrum data is generated according to the search result.

Abstract: A voice processing apparatus has a storage device that stores registration information containing a characteristic parameter of a given voice. The voice processing apparatus is further provided with a judgment unit, a management unit and a notification unit. The judgment unit judges whether an input voice is appropriate or not for creating or updating the registration information based on a degree of a difference between an inter-band correlation matrix of an input voice acquired this time and an inter-band correlation matrix of another input voice that is judged as being appropriate last time. The management unit creates or updates the registration information based on a characteristic parameter of the input voice when the judgment unit judges that the input voice is appropriate. The notification unit notifies a speaker of the input voice when the judgment unit judges that the input voice is inappropriate.

Abstract: A sound encoder having an improved quantization performance while suppressing an increase of the bit rate to a lowest level. In a second layer encoder, a standard deviation calculator calculates a standard deviation ?c of a first layer decoding spectrum after decoding a scale factor ratio multiplication and outputs the standard deviation ?c to a selector. The selector selects a linear transform function as a function for a nonlinear transform of a residual spectrum according to the standard deviation ?c A nonlinear transform function selects one of prepared nonlinear transform functions #1 to #N according to a result of the selection by the selector, and outputs the selected one to an inverse transformer. The inverse transformer subjects an inverse transform (expansion) to a residual spectrum candidate that is stored in a residual spectrum code book using the nonlinear transform function outputted from the nonlinear transform function and outputs the result to an adder.

Abstract: Processing an audio signal associated with a sound recording made available to be rendered to an end user is disclosed. The audio signal is received. A high-order perceptual attribute of the audio signal as rendered is changed by modifying the audio signal. The modification may be based on real-time analysis of the audio signal.

Abstract: A signal processing apparatus includes a decoding unit, an analyzing unit, a synthesizing unit, and a selecting unit. The decoding unit decodes an input encoded audio signal and outputs a playback audio signal. When loss of the encoded audio signal occurs, the analyzing unit analyzes the playback audio signal output before the loss occurs and generates a linear predictive residual signal. The synthesizing unit synthesizes a synthesized audio signal on the basis of the linear predictive residual signal. The selecting unit selects one of the synthesized audio signal and the playback audio signal and outputs the selected audio signal as a continuous output audio signal.

Abstract: A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

Abstract: A signal classifying method and apparatus are disclosed. The signal classifying method includes: obtaining a spectrum fluctuation parameter of a current signal frame determined as a foreground frame, and buffering the spectrum fluctuation parameter; obtaining a spectrum fluctuation variance of the current signal frame according to spectrum fluctuation parameters of all buffered signal frames, and buffering the spectrum fluctuation variance; and calculating a ratio of signal frames whose spectrum fluctuation variance is above or equal to a first threshold to all the buffered signal frames, and determining the current signal frame as a speech frame if the ratio is above or equal to a second threshold or determining the current signal frame as a music frame if the ratio is below the second threshold.

Abstract: Methods of using individually distinctive patterns of voice characteristics to identify a speaker include computing the reassigned spectrogram of each of at least two voice samples, pruning each reassigned spectrogram to remove noise and other computational artifacts, and comparing (either visually or with the aid of a processor) the strongest points to determine whether the voice samples belong to the same speaker.

Abstract: The present invention provides a method, a computer-software-product and an apparatus for enabling a determination of speech related audio data within a record of digital audio data. The method comprises steps for extracting audio features from the record of digital audio data, for classifying one or more subsections of the record of digital audio data, and for marking at least a part of the record of digital audio data classified as speech. The classification of the digital audio data record is performed on the basis of the extracted audio features and with respect to at least one predetermined audio class.

Abstract: To increase channel capacity, mobile phone carriers have deployed speech coders, such as Advanced MultiBand Excitation coding (AMBE), in networks to reduce the bit rate of each call. One undesired consequence of employing such speech coders is that the voice quality can be much worse as compared to higher bit-rate speech coders. A method or corresponding apparatus in an example embodiment of the present invention performs voice quality enhancement transparently within a network by detecting use of a coder applying rate reduction to a speech signal and known to have an adverse effect on a coded speech signal. Upon detection of the use of such coder, the coded speech signal is corrected based on components introduced into the coded speech signal due to the rate reduction. As a result of applying the voice quality enhancement, adverse effects of speech coders can be reduced, while maintaining high quality voice signals.

Abstract: A method and apparatus include a voice activity detection module configured to detect silent frames, and a codec mode selection module configured to determine a codec mode. The voice activity detection module includes a receiver configured to receive a frame, a first determiner configured to determine a first set of parameters from the frame, and a providing unit configured to provide the first set of parameters to the codec mode selection module. The codec mode selection module includes a second determiner configured to determine a second set of parameters in dependence on the first set of parameters, and a selector configured to select a codec mode in dependence on the second set of parameters.

Abstract: Audio coding and decoding apparatuses and methods which support fine granularity scalability (FGS) using harmonic information of a high-band audio signal or wideband error audio signal when performing wideband audio coding and decoding, and recording mediums on which the methods are stored. The audio coding method includes detecting harmonics of a high-band audio signal or wideband error audio signal of an input audio signal; determining an order of the detected harmonics; and coding the detected harmonics based on the determined order.

Abstract: A scalable encoder enabling improvement of the encoding efficiency in the second layer and improvement of the quality of the original signal decoded using the encoding signal in the second layer. A predictive coefficient encoder of the scalable encoder has a predictive coefficient codebook where candidates of the predictive coefficient are recorded. After searching the predictive coefficient codebook, the scale factor of the first layer decoded signal inputted from a scale factor calculator is multiplied, and a predictive coefficient which most approximates the multiplication result to the scale factor of the original signal inputted from the scale factor calculator is determined and encoded, and the coded code is inputted to a multiplexer.

Abstract: A method and system for refining an estimated pitch period estimate based on a coarse pitch useful for performing frame loss concealment in an audio decoder as well as for other applications. A normalized correlation at the coarse pitch lag is computed and used as the current best candidate. The normalized correlation is then evaluated at the midpoint of the refinement pitch range on either side of the current best candidate. If the normalized correlation at either midpoint is greater than the current best lag, the midpoint with the maximum correlation is selected as the current best lag. After each iteration, the refinement range is decreased by a factor of two and centered on the current best lag. This bisectional search continues until the pitch has been refined to an acceptable tolerance or until the refinement range has been exhausted. During each step of the bisectional pitch refinement, the signal is decimated to reduce the complexity of computing the normalized correlation.

Abstract: Methods and systems for sample rate conversion convert a sampled signal to a higher data rate signal. Conversion pulses are received, having a conversion rate that is higher than the sample rate of the sampled signal. Sample points are then reconstructed from the sampled signal, in real time, on either side of a conversion pulse. An interpolation is performed between the reconstructed sample points, at the time of the conversion pulse. The interpolation results are outputted in real time. The process is repeated for additional conversion pulses. The outputted interpolated amplitudes form the higher data rate signal having a data rate equal to the conversion rate. Sample rate conversion is thus performed in real time according to the higher data rate clock, rather than with fixed ratios. As a result, when the higher data rate clock is affected by, for example, jitter or other frequency variations, the higher data rate samples immediately track the lower data rate samples.

Abstract: In a formants extracting method capable of precisely obtaining formants as resonance frequencies of voice with less computational complexity, the method includes searching a maximum value by a spectral peak-picking method, judging whether the number of formants corresponding to a zero at the obtained maximum point are two, and analyzing a pertinent root by roots polishing when the number of the formants are judged as two. The number of the formants are judged by applying Cauchy's integral formula, wherein Cauchy's integral formula is not applied repeatedly but only once at a surrounding portion of the maximum value in a z-domain.

Abstract: A similarity degree estimation method is performed by two processes. In a first process, an inter-band correlation matrix is created from spectral data of an input voice such that the spectral data are divided into a plurality of discrete bands which are separated from each other with spaces therebetween along a frequency axis, a plurality of envelope components of the spectral data are obtained from the plurality of the discrete bands, and elements of the inter-band correlation matrix are correlation values between the respective envelope components of the input voice. In a second process, a degree of similarity is calculated between a pair of input voices to be compared with each other by using respective inter-band correlation matrices obtained for the pair of the input voices through the inter-band correlation matrix creation process.

Abstract: The present invention is based on the finding that parameters including a first set of parameters of a representation of a first portion of an original signal and including a second set of parameters of a representation of a second portion of the original signal can be efficiently encoded, when the parameters are arranged in a first sequence of tuples and in a second sequence of tuples, wherein the first sequence of tuples comprises tuples of parameters having two parameters from a single portion of the original signal and wherein the second sequence of tuples comprises tuples of parameters having one parameter from the first portion and one parameter from the second portion of the original signal. An efficient encoding can be achieved using a bit estimator to estimate the number of necessary bits to encode the first and the second sequence of tuples, wherein only the sequence of tuples is encoded, that results in the lower number of bits.

Abstract: Methods and systems of identifying speech sound features within a speech sound are provided. The sound features may be identified using a multi-dimensional analysis that analyzes the time, frequency, and intensity at which a feature occurs within a speech sound, and the contribution of the feature to the sound. Information about sound features may be used to enhance spoken speech sounds to improve recognizability of the speech sounds by a listener.

Abstract: A Voice User Interface (VUI) or Interactive Voice Response (IVR) system utilizes three levels of navigation (e.g. Main Menu, Services, and Helper Commands) in presenting information units arranged in sets. The units are “spoken” by a system in a group to a human user and the group of information at each level is preceded by a tone that is unique to the level. When navigating the levels, the tones of the levels are in a musical progression, e.g. the three-note blues progression I, IV, V, for preceding the groups of information, respectively. The musical progression returns to the tonic of the musical key when the navigation returns to the level one of the first group of information.

Abstract: System, method and computer-readable medium are disclosed for using filters signal processing. The system includes a module that receives information regarding a first filter, a module that receives information regarding a second filter, and a module that receives date to indicate switching between the first filter and the second filter across the spectrum of the received audio signal, and a module that processes the received audio signal according to the received data and switching between the first filter and the second filter, wherein at least one of the first filter and the second filter represent a merger of two initial filters.

Abstract: Generic and specific C-to-E binaural cue coding (BCC) schemes are described, including those in which one or more of the input channels are transmitted as unmodified channels that are not downmixed at the BCC encoder and not upmixed at the BCC decoder. The specific BCC schemes described include 5-to-2, 6-to-5, 7-to-5, 6.1-to-5.1, 7.1-to-5.1, and 6.2-to-5.1, where “0.1” indicates a single low-frequency effects (LFE) channel and “0.2” indicates two LFE channels.

Abstract: An apparatus and method for improving a sound quality of a portable terminal are provided. Particularly, an apparatus and method for tuning to a voice quality that a user desires using the portable terminal, in case where the user determines an abnormal sound quality, are provided. The apparatus includes a sound quality improving unit for selecting a Digital Signal Processing (DSP) filter value corresponding to a voice quality that a user desires among a plurality of DSP filters for controlling the voice quality according to a user's voice quality.

Abstract: A method for determining an upperband speech signal from a narrowband speech signal is disclosed. A list of narrowband line spectral frequencies (LSFs) is determined from the narrowband speech signal. A first pair of adjacent narrowband LSFs that have a lower difference between them than every other pair of adjacent narrowband LSFs in the list is determined. A first feature that is a mean of the first pair of adjacent narrowband LSFs is determined. Upperband LSFs are determined based on at least the first feature using codebook mapping.

Abstract: A vocoder system for improving the performance expression of an output sound while lightening the computational load. The system includes formant detection means and division means in which the center frequencies have been fixed. The modulation level with which the levels of each of the frequency bands that have been divided in the division means are set by a setting means based on the levels of each of the frequency bands that correspond to those that have been detected in the formant detection means and formant information with which the formants are changed. Therefore, it is possible to improve the performance expression of the output sound with a light computational load and without the need to calculate and change the filter figure of each filter for each sample in order to change the center frequency and bandwidth of each of the filters comprising the division means.

Abstract: A degree of voicing is extracted using the characteristic of harmonic peaks existing in a constant period by converting an input speech or audio signal to a speech signal of the frequency domain, selecting the greatest peak in a first pitch period of the converted speech signal as a harmonic peak, thereafter selecting a peak having the greatest spectral value among peaks existing in each peak search range of the speech signal as a harmonic peak, extracting harmonic spectral envelope information by performing interpolation of the selected harmonic peaks, extracting non-harmonic spectral envelope information by performing interpolation of the non-harmonic peaks, and comparing the two pieces of envelope information to each other.

Abstract: Various technologies and techniques are disclosed for providing a numeric tower that represents a structure supporting statically defined numeric data types. The numeric data types each are operable to implement a different but accurate representation of a particular value. Numeric operations are supported for the numeric tower that can be performed with any of the statically defined numeric data types. The numeric tower is extensible, and allows for additional statically defined numeric data types to be added, as well as operations. The numeric tower is also operable to detect overflow situations. For example, suppose a result of an operation will result in an overflow situation because the operation does not fit within a range supported by the particular numeric type. The system converts the numeric type to a different one of the numeric data types when the result does not fit within a range supported by the first one.

Abstract: Prevalence detection is advantageously applied to the result of specific spectral discrimination to adaptively determine prevalent frequencies existing within an audio signal containing speech. Prevalent frequencies in this audio signal so isolated are attenuated in a highly selective manner, thus reducing the masking potential of pervasive resonances and obfuscative energy within the speech itself over low energy language-imparting speech elements.

Abstract: A scalable encoding apparatus, a scalable decoding apparatus and the like are disclosed which can achieve a band scalable LSP encoding that exhibits both a high quantization efficiency and a high performance. In these apparatuses, a narrow band-to-wide band converter receives and converts a quantized narrow band LSP to a wide band, and then outputs the quantized narrow band LSP as converted (i.e., a converted wide band LSP parameter) to an LSP-to-LPC converter. The LSP-to-LPC converter converts the quantized narrow band LSP as converted to a linear prediction coefficient and then outputs it to a pre-emphasizer. The pre-emphasizer calculates and outputs the pre-emphasized linear prediction coefficient to an LPC-to-LSP converter. The LPC-to-LSP converter converts the pre-emphasized linear prediction coefficient to a pre-emphasized quantized narrow band LSP as wide band converted, and then outputs it to a prediction quantizer.

Abstract: In order to detect a degree of voicing of a speech signal, an input speech signal is converted to a speech signal in the frequency domain, a pitch value is calculated from the speech signal, a plurality of harmonic peaks existing in the speech signal are detected, and a difference obtained by comparing the pitch value to an interval between adjacent harmonic peaks among the detected harmonic peaks is detected as the degree of voicing included in the speech signal.

Abstract: An apparatus and method of low bit rate encoding and reproducing. The method includes transforming input audio signals in a time domain into spectral signals in a frequency domain, extracting important-spectrum components from the spectral signals in the frequency domain, and quantizing the important-spectrum components, extracting residual-spectrum components other than the important-spectrum components from the spectral signals in the frequency domain, and calculating and quantizing a noise level of the residual-spectrum components, and encoding the quantized important-spectrum components and the quantized noise level losslessly, and outputting encoded bitstreams.

Abstract: The invention provides a method for estimating a fundamental frequency of a speech signal comprising the steps of receiving a signal spectrum of the speech signal, filtering the signal spectrum to obtain a refined signal spectrum, determining a cross-power spectral density using the refined signal spectrum and the signal spectrum, transforming the cross-power spectral density into the time domain to obtain a cross-correlation function, and estimating the fundamental frequency of the speech signal based on the cross-correlation function.

Abstract: Techniques and tools related to delayed or lost coded audio information are described. For example, a concealment technique for one or more missing frames is selected based on one or more factors that include a classification of each of one or more available frames near the one or more missing frames. As another example, information from a concealment signal is used to produce substitute information that is relied on in decoding a subsequent frame. As yet another example, a data structure having nodes corresponding to received packet delays is used to determine a desired decoder packet delay value.

Abstract: A speaker intent analysis system and method for validating the truthfulness and intent of a plurality of participants' responses to questions. A computer stores, retrieves, and transmits a series of questions to be answered audibly by participants. The participants' answers are received by a data processor. The data processor analyzes and records the participants' speech parameters for determining the likelihood of dishonesty. In addition to analyzing participants' speech parameters for distinguishing stress or other abnormality, the processor may be equipped with voice recognition software to screen responses that while not dishonest, are indicative of possible malfeasance on the part of the participants. Once the responses are analyzed, the processor produces an output that is indicative of the participant's credibility. The output may be sent to proper parties and/or devices such as a web page, computer, e-mail, PDA, pager, database, report, etc. for appropriate action.

Abstract: Embodiments of a system and method for encoding audio data have been described. In one embodiment, the method includes transforming frequency domain data in a plurality of signal windows of an audio dataset from a cosine/sine format to a magnitude/cosine/sine format. The magnitude/cosine/sine format disproportionately represents a magnitude of the frequency domain data over a phase of the frequency domain data. The above transformation may be a pre-processing stage of vector quantization usable to produce a codebook.

Abstract: A speech analysis device which accurately analyzes an aperiodic component included in speech in a practical environment where there is background noise includes: a frequency band division unit which divides, into bandpass signals each associated with a corresponding one of frequency bands, an input signal representing a mixed sound of background noise and speech; a noise interval identification unit which identifies a noise interval and a speech interval of the input signal; an SNR calculation unit which calculates an SN ratio; a correlation function calculation unit which calculates an autocorrelation function of each bandpass signal; a correction amount determination unit which determines a correction amount for an aperiodic component ratio, based on the calculated SN ratio; and an aperiodic component ratio calculation unit which calculates, for each frequency band, an aperiodic component ratio of the aperiodic component, based on the determined correction amount and the calculated autocorrelation function.

Abstract: An apparatus for processing a signal and method thereof are disclosed. Data coding and entropy coding are performed with interconnection, and grouping is used to enhance coding efficiency. The present invention includes the steps of obtaining a difference value and index information and entropy-decoding the index information and identifying an entropy table corresponding to the entropy-decoded index information and entropy-decoding the difference value using the identified entropy table and obtaining data using a reference value corresponding to a plurality of data and the decoded difference value.