Abstract: An audio encoder for encoding an audio signal having a lower frequency band and an upper frequency band includes: a detector for detecting a peak spectral region in the upper frequency band of the audio signal; a shaper for shaping the lower frequency band using shaping information for the lower band and for shaping the upper frequency band using at least a portion of the shaping information for the lower band, wherein the shaper is configured to additionally attenuate spectral values in the detected peak spectral region in the upper frequency band; and a quantizer and coder stage for quantizing a shaped lower frequency band and a shaped upper frequency band and for entropy coding quantized spectral values from the shaped lower frequency band and the shaped upper frequency band.

Abstract: The present invention reduces encoding distortion in frequency domain encoding compared to conventional techniques, and obtains LSP parameters that correspond to quantized LSP parameters for the preceding frame and are to be used in time domain encoding from coefficients equivalent to linear prediction coefficients resulting from frequency domain encoding. When p is an integer equal to or greater than 1, a linear prediction coefficient sequence which is obtained by linear prediction analysis of audio signals in a predetermined time segment is represented as a[1], a[2], . . . , a[p], and ?[1], ?[2], . . . , ?[p] are a frequency domain parameter sequence derived from the linear prediction coefficient sequence a[1], a[2], . . . , a[p], an LSP linear transformation unit (300) determines the value of each converted frequency domain parameter {tilde over (?)}?[i] (i=1, 2, . . . , p) in a converted frequency domain parameter sequence {tilde over (?)}w[1], {tilde over (?)}w[2], . . .

Abstract: An example device in accordance with an aspect of the present disclosure includes a non-linear compensator, an interpolator, a demultiplexer, and a demodulator. The non-linear compensator is to correct a non-linearity of the input bitstream data to obtain linearized bitstream data. The interpolator is to convert a non-constant sample rate of the linearized bitstream data to obtain constant sample rate bitstream data. The demultiplexer is to demultiplex the constant sample rate bitstream data into a first waveform and a second waveform. The demodulator is to demodulate the first waveform and the second waveform.

Abstract: The invention features systems and methods for detecting and mitigating network attacks in a Voice-Over-IP (VoIP) network. A server is configured to receive information related to a mitigation action for a call. The information can include a complexity level for administering an audio challenge-response test to the call and an identification of the call. The server also generates i) a routing label based on the identification of the call, and ii) a script defining a plurality of variables that store identifications of a plurality of altered sound files for the audio challenge-response test. Each altered sound file is randomly selected by the server subject to one or more constraints associated with the complexity level. The server is further configured to transmit the script to a guardian module and the routing label to a gateway.

Abstract: A system for processor wake-up based on sensor data includes an audio buffer, an envelope buffer, and a processor. The audio buffer is configured to store a first data from a sensor. The first data is generated according to a first sampling rate. The envelope buffer is configured to store a second data, which is derived from the first data according to a second sampling rate, which is less than the first sampling rate. The processor is configured to wake up periodically from an idle state and read the second data from the envelope buffer. If the second data indicates an activity, the processor is configured to read the first data from the audio buffer. If the second data does not indicate an activity, the processor is configured to return to the idle state.

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: A location determination system includes a first mobile terminal and a second mobile terminal. The first mobile terminal includes a first processor to acquire a first sound signal, analyze the first sound signal to obtain a first analysis result, and transmit the first analysis result. The second mobile terminal includes a second processor to acquire a second sound signal, analyze the second sound signal to obtain a second analysis result, receive the first analysis result from the first mobile terminal, compare the second analysis result with the first analysis result to obtain a comparison result, and determine whether the first mobile terminal locates in an area in which the second mobile terminal locates, based on the comparison result.

Abstract: Voice activity detection using multiple microphones can be based on a relationship between an energy at each of a speech reference microphone and a noise reference microphone. The energy output from each of the speech reference microphone and the noise reference microphone can be determined. A speech to noise energy ratio can be determined and compared to a predetermined voice activity threshold. In another embodiment, the absolute value of the autocorrelation of the speech and noise reference signals are determined and a ratio based on autocorrelation values is determined. Ratios that exceed the predetermined threshold can indicate the presence of a voice signal. The speech and noise energies or autocorrelations can be determined using a weighted average or over a discrete frame size.

Abstract: A non-spatial speech detection system includes a plurality of microphones whose output is supplied to a fixed beamformer. An adaptive beamformer is used for receiving the output of the plurality of microphones and one or more processors are used for processing an output from the fixed beamformer and identifying speech from noise though the use of an algorithm utilizing a covariance matrix.

Abstract: A method for decoding an audio signal in a decoder having a CELP-based decoder element including a fixed codebook component, at least one pitch period value, and a first decoder output, wherein a bandwidth of the audio signal extends beyond a bandwidth of the CELP-based decoder element. The method includes obtaining an up-sampled fixed codebook signal by up-sampling the fixed codebook component to a higher sample rate, obtaining an up-sampled excitation signal based on the up-sampled fixed codebook signal and an up-sampled pitch period value, and obtaining a composite output signal based on the up-sampled excitation signal and an output signal of the CELP-based decoder element, wherein the composite output signal includes a bandwidth portion that extends beyond a bandwidth of the CELP-based decoder element.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: A method for overlap-adding signals useful for performing frame loss concealment (FLC) in an audio decoder as well as in other applications. The method uses a dynamic mix of windows to overlap two signals whose normalized cross-correlation may vary from zero to one. If the overlapping signals are decomposed into a correlated component and an uncorrelated component, they are overlap-added separately using the appropriate window, and then added together. If the overlapping signals are not decomposed, a weighted mix of windows is used. The mix is determined by a measure estimating the amount of cross-correlation between overlapping signals, or the relative amount of correlated to uncorrelated signals.

Abstract: Provided are an encoding apparatus and a decoding apparatus of a multi-channel signal. The encoding apparatus of the multi-channel signal may process a phase parameter associated with phase information between a plurality of channels constituting the multi-channel signal, based on a characteristic of the multi-channel signal. The encoding apparatus may generate an encoded bitstream with respect to the multi-channel signal using the processed phase parameter and a mono signal extracted from the multi-channel signal.

Abstract: Speech recognition device uses visual information to narrow down the range of likely adaptation parameters even before a speaker makes an utterance. Images of the speaker and/or the environment are collected using an image capturing device, and then processed to extract biometric features and environmental features. The extracted features and environmental features are then used to estimate adaptation parameters. A voice sample may also be collected to refine the adaptation parameters for more accurate speech recognition.

Abstract: A first-pitch metric function based on a first audio sample and a second pitch-metric function based on a second audio sample may be determined. The first and second pitch-metric functions may have either local minima or local maxima that correspond to candidate pitch values of the first and the second audio samples, respectively. The first and the second pitch-metric functions may be transformed to generate a first and a second transformed pitch-metric function, respectively. A correlation function based on a correlation between the first and the second transformed pitch-metric function may also be determined. A lower-dimensionality representation of the correlation function may further be determined. The lower-dimensionality representation may convey information indicative of pitch dynamics between the first and second audio sample. A computing device having a processor and a memory may perform an action based on the information indicative of the pitch dynamics.

Abstract: An automatic marking method for Karaoke vocal accompaniment is provided. In the method, pitch, beat position and volume of a singer are compared with the original pitch, beat position and volume of the theme of a song to generate a score of pitch, a score of beat and a score of emotion respectively, so as to obtain a weighted total score in a weighted marking method. By using the method, the pitch, beat position and volume error of each section of the song sung by the singer can be exactly worked out, and a pitch curve and a volume curve can be displayed, so that the singer can learn which part is sung incorrectly and which part needs to be enhanced. The present invention also has the advantages of dual effects of teaching and entertainment, high practicability and technical advancement.

Abstract: A system and method may be configured to analyze audio information derived from an audio signal. The system and method may track sound pitch across the audio signal. The tracking of pitch across the audio signal may take into account change in pitch by determining at individual time sample windows in the signal duration an estimated pitch and a representation of harmonic envelope at the estimated pitch. The estimated pitch and the representation of harmonic envelope may then be implemented to determine an estimated pitch for another time sample window in the signal duration with an enhanced accuracy and/or precision.

Abstract: A method for determining an overlap and add length estimate comprises determining a plurality of correlation values of a plurality of ordered frequency domain samples obtained from a data frame; comparing the correlation values of a first subset of the samples to a first predetermined threshold to determine a first edge sample; comparing the correlation values of a second subset of the samples to a second predetermined threshold to determine a second edge sample; using the first and second edge samples to determine an overlap and add length estimate; and providing the overlap and add length estimate to an overlap and add circuit.

Abstract: An encoding apparatus comprises a frame processor (105) which receives a multi channel audio signal comprising at least a first audio signal from a first microphone (101) and a second audio signal from a second microphone (103). An ITD processor 107 then determines an inter time difference between the first audio signal and the second audio signal and a set of delays (109, 111) generates a compensated multi channel audio signal from the multi channel audio signal by delaying at least one of the first and second audio signals in response to the inter time difference signal. A combiner (113) then generates a mono signal by combining channels of the compensated multi channel audio signal and a mono signal encoder (115) encodes the mono signal. The inter time difference may specifically be determined by an algorithm based on determining cross correlations between the first and second audio signals.

Abstract: A speech signal processing system which outputs a speech feature, divides an input speech signal into frames so that each pair of consecutive frames have a frame shift length equal to at least one period of the speech signal and have an overlap equal to at least a predetermined length, applies discrete Fourier transform to each of the frames, calculates a CSP coefficient for the pair, searches a predetermined search range in which a speech wave lags a period equal to at least one period to obtain the maximum value of the CSP coefficient for the pair, and generates time-series data of the maximum CSP coefficient values arranged in the order in which the frames appear. A method and a computer readable article of manufacture for the implementing the same are also provided.

Abstract: The present disclosure relates to coding and decoding technologies, and discloses a preprocessing method, a preprocessing apparatus, and a coding device. The preprocessing method includes: obtaining characteristic information of a current frame signal; identifying whether the current frame signal requires no coding operation of removing LTC according to the characteristic information of the current frame signal and preset information; and if identifying that the current frame signal requires no coding operation of removing LTC, performing the coding operation of removing STC for the current frame signal; and if identifying that the current frame signal requires the coding operation of removing LTC, performing the coding operations of removing both LTC and STC for the current frame signal. Through the technical solution provided herein, the coding operation of removing LTC is performed for only part of the input frame signals.

Abstract: A system and method may be configured to process an audio signal. The system and method may track pitch, chirp rate, and/or harmonic envelope across the audio signal, may reconstruct sound represented in the audio signal, and/or may segment or classify the audio signal. A transform may be performed on the audio signal to place the audio signal in a frequency chirp domain that enhances the sound parameter tracking, reconstruction, and/or classification.

Abstract: A method of providing a quality measure for an output voice signal generated to reproduce an input voice signal, the method comprising: partitioning the input and output signals into frames; for each frame of the input signal, determining a disturbance relative to each of a plurality of frames of the output signal; determining a subset of the determined disturbances comprising one disturbance for each input frame such that a sum of the disturbances in the subset set is a minimum; and using the set of disturbances to provide the measure of quality.

Abstract: A sound analysis device comprises: a sound parameter calculation unit operable to acquire an audio signal and calculate a sound parameter for each of partial audio signals, the partial audio signals each being the acquired audio signal in a unit of time; a category determination unit operable to determine, from among a plurality of environmental sound categories, which environmental sound category each of the partial audio signals belongs to, based on a corresponding one of the calculated sound parameters; a section setting unit operable to sequentially set judgement target sections on a time axis as time elapses, each of the judgment target sections including two or more of the units of time, the two or more of the units of time being consecutive; and an environment judgment unit operable to judge, based on a number of partial audio signals in each environmental sound category determined in at least a most recent judgment target section, an environment that surrounds the sound analysis device in at least the

Abstract: The invention describes a method of deriving a set of features (S) of an audio input signal (M), which method comprises identifying a number of first-order features (f1, f2, . . . , ff) of the audio input signal (M), generating a number of correlation values (?1, ?2, . . . , ?I) from at least part of the first-order features (f1, f2, . . . , ff), and compiling the set of features (S) for the audio input signal (M) using the correlation values (?1, ?2, . . . , ?I). The invention further describes a method of classifying an audio input signal (M) into a group, and a method of comparing audio input signals (M, M?) to determine a degree of similarity between the audio input signals (M, M?).

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: An encoder improves inter-channel prediction (ICP) performance in scalable stereo sound encoding using an ICP. In the encoder, ICP analysis units use, as reference signal candidates, a frequency coefficient in the low-band portion of a side residual signal, a frequency coefficient in each sub-band portion of a monaural residual signal, and a frequency coefficient in the low-band portion of the monaural residual signal, respectively, and perform an ICP analysis between the these respective candidates and a frequency coefficient in each sub-band portion of the side residual signal to generate first, second, and third ICP coefficients.

Abstract: A packet loss concealment method and system is described that attempts to reduce or eliminate destructive interference that can occur when an extrapolated waveform representing a lost segment of a speech or audio signal is merged with a good segment after a packet loss. This is achieved by guiding a waveform extrapolation that is performed to replace the bad segment using a waveform available in the first good segment or segments after the packet loss. In another aspect of the invention, a selection is made between a packet loss concealment method that performs the aforementioned guided waveform extrapolation and one that does not. The selection may be made responsive to determining whether the first good segment or segments after the packet loss are available and also to whether a segment preceding the lost segment and the first good segment following the lost segment are deemed voiced.

Abstract: A method of providing a quality measure for an output voice signal generated to reproduce an input voice signal, the method comprising: partitioning the input and output signals into frames; for each frame of the input signal, determining a disturbance relative to each of a plurality of frames of the output signal; determining a subset of the determined disturbances comprising one disturbance for each input frame such that a sum of the disturbances in the subset set is a minimum; and using the set of disturbances to provide the measure of quality.

Abstract: An apparatus calculates a number of spectral envelopes to be derived by a spectral band replication (SBR) encoder, wherein the SBR encoder is adapted to encode an audio signal using a plurality of sample values within a predetermined number of subsequent time portions in an SBR frame extending from an initial time to a final time, the predetermined number of subsequent time portions being arranged in a time sequence given by the audio signal. The apparatus has a decision value calculator for determining a decision value, the decision value measuring a deviation in spectral energy distributions of a pair of neighboring time portions. The apparatus further has a detector for detecting a violation of a threshold by the decision value and a processor for determining a first envelope border between the pair of neighboring time portions when the violation of the threshold is detected.

Abstract: An efficient encoded representation of a first and a second input audio signal can be derived using correlation information indicating a correlation between the first and the second input audio signals, when a signal characterization information, indicating at least a first or a second, different characteristic of the input audio signal is additionally considered. Phase information indicating a phase relation between the first and the second input audio signals is derived, when the input audio signals have the first characteristic. The phase information and a correlation measure are included into the encoded representation when the input audio signals have the first characteristic, and only the correlation information is included into the encoded representation when the input audio signals have the second characteristic.

Abstract: A method of and apparatus for image analysis for picture loss detection in fields or frames in video or film content makes use of different correlation characteristics of picture images and non-picture images to detect picture loss. A measure of self correlation of a plurality of image data samples, and a measure of the correlation of the plurality of image data samples with a mean value are determined, and a positive detection of picture loss is based on a comparison between the two correlation measures.

Abstract: A synchronization sequence (preamble) that is known to the receiver forms as an integral part of packet-based digital communication systems. The first operation in such digital communication systems is the detection of the beginning of a valid signal (packet). A system, apparatus, and method for a scheme to robustly detect the preamble are provided having a hierarchical cross-correlator in combination with a second stage delayed auto-correlator using the output of the cross-correlator as an input to the second stage correlator.

Abstract: A speech processing apparatus includes a plurality of microphones which receive speech produced by a first sound source to obtain first speech signals for a plurality of channels having one-to-one correspondence with the plurality of microphones, a calculation unit configured to calculate a first characteristic amount indicative of an inter-channel correlation of the first speech signals, a storage unit configured to store in advance a second characteristic amount indicative of an inter-channel correlation of second speech signals for the plurality of channels obtained by receiving speech produced by a second sound source by the plurality of microphones, and a collation unit configured to collate the first characteristic amount with the second characteristic amount to determine whether the first sound source matches with the second sound source.

Abstract: A method of refining a pitch period estimation of a signal, the method comprising: for each of a plurality of portions of the signal, scanning over a predefined range of time offsets to find an estimate of the pitch period of the portion within the predefined range of time offsets; identifying the average pitch period of the estimated pitch periods of the portions; determining a refined range of time offsets in dependence on the average pitch period, the refined range of time offsets being narrower than the predefined range of time offsets; and for a subsequent portion of the signal, scanning over the refined range of time offsets to find an estimate of the pitch period of the subsequent portion.

Abstract: An audio data transmitting/receiving apparatus for realizing a high-quality frame compensation in audio communications. In an audio data transmitting apparatus (10), a delay part (104) subjects multi-channel audio data to a delay process that delays the L-ch encoded data relative to the R-ch encoded data by a predetermined delay amount. A multiplexing part (106) multiplexes the audio data as subjected to the delay process. A transmitting part (108) transmits the audio data as multiplexed. In an audio data receiving apparatus (20), a separating part (114) separates, for each channel, the audio data received from the audio data transmitting apparatus (10). A decoding part (118) decodes, for each channel, the audio data as separated. If there has occurred a loss or error in the audio data as separated, then a frame compensating part (120) uses one of the L-ch and R-ch encoded data to compensate for the loss or error in the other encoded data.

Abstract: A method, apparatus, and medium for classifying a speech signal and a method, apparatus, and medium for encoding the speech signal using the same are provided. The method for classifying a speech signal includes calculating classification parameters from an input signal having block units, calculating a plurality of classification criteria from the classification parameters, and classifying the level of the input signal using the plurality of classification criteria. The classification parameters include at least one of an energy parameter of the input signal, a cross-correlation parameter between a specific block of a present frame and the input signal, and an integrated cross-correlation parameter obtained by accumulating the cross-correlation parameter.

Abstract: A speech analysis apparatus analyzing prosodic characteristics of speech information and outputting a prosodic discrimination result includes an input unit inputting speech information, an acoustic analysis unit calculating relative pitch variation and a discrimination unit performing speech discrimination processing, in which the acoustic analysis unit calculates a current template relative pitch difference, determining whether a difference absolute value between the current template relative pitch difference and a previous template relative pitch difference is equal to or less than a predetermined threshold or not, when the value is not less than the threshold, calculating an adjacent relative pitch difference, and when the adjacent relative pitch difference is equal to or less than a previously set margin value, executing correction processing of adding or subtracting an octave of the current template relative pitch difference to calculate the relative pitch variation by applying the relative pitch differe

Abstract: This invention involves time-scale modification of audio signals. The invention describes overlap and add time scale modification with variable input and output buffer sizes. Seamless speed change is achieved by keeping track of previously processed data to avoid discontinuities during playback speed transitions.

Abstract: An apparatus, method, and computer program are capable of receiving and cross-correlating a first audio signal and a second audio signal. This produces a cross-correlated signal, which is used to identify a plurality of parameters associated with at least one of the first and second audio signals. The parameters are used to generate an indicator identifying an extent to which the first and second audio signals match.

Abstract: Disclosed is a stereo audio encoding device capable of improving a spatial image of a decoded audio in stereo audio encoding. In this device, an original cross correlation calculation unit (101) calculates a mutual relationship coefficient (C1) between the original L channel signal and the original R channel signal. A stereo audio reconfiguration unit (104) subjects the inputted L channel signal and the R channel signal to encoding and decoding so as to generate an L channel reconfigured signal (L?) and an R channel reconfigured signal (R?). A reconfiguration cross correlation calculation unit (105) calculates a cross correlation coefficient (C2) between the L channel reconfigured signal (L?) and the R channel reconfigured signal (R?). A cross correlation comparison unit (106) calculates and outputs a comparison result &agr; between the cross correlation coefficient (C1) and the cross correlation coefficient (C2).

Abstract: Systems, methods, and apparatus described include waveform alignment operations in which a single set of evaluated cosines and sines is used to calculate cross-correlations of two periodic waveforms at two different phase shifts.

Abstract: A two microphone noise reduction system is described. In an embodiment, input signals from each of the microphones are divided into subbands and each subband is then filtered independently to separate noise and desired signals and to suppress non-stationary and stationary noise. Filtering methods used include adaptive decorrelation filtering. A post-processing module using adaptive noise cancellation like filtering algorithms may be used to further suppress stationary and non-stationary noise in the output signals from the adaptive decorrelation filtering and a single microphone noise reduction algorithm may be used to further provide optimal stationary noise reduction performance of the system.

Abstract: A system capable of separating sound source signals with high precision while improving a convergence rate and convergence precision. A process of updating a current separation matrix Wk to a next separation matrix Wk+1 such that a next value J(Wk+1) of a cost function is closer to a minimum value J(W0) than a current value J(Wk) is iteratively performed. An update amount ?Wk of the separation matrix is increased as the current value J(Wk) of the cost function is increased and is decreased as a current gradient ?J(Wk)/?W of the cost function is rapid. On the basis of input signals x from a plurality of microphones Mi and an optimal separation matrix W0, it is possible to separate sound source signals y(=W0·x) with high precision while improving a convergence rate and convergence precision.

Abstract: A modified synchronized overlap add (SOLA) algorithm for performing high-quality, low-complexity audio time scale modification (TSM) is described. The algorithm produces good output audio quality with a very low complexity and without producing additional audible distortion during dynamic change of the audio playback speed. The algorithm may achieve complexity reduction by performing the maximization of normalized cross-correlation using decimated signals. By updating the input buffer and the output buffer in a precise sequence with careful checking of the appropriate array bounds, the algorithm may also achieve seamless audio playback during dynamic speed change with a minimal requirement on memory usage.

Abstract: A feature extraction apparatus includes a spectrum calculating unit that calculates, based on an input speech signal, a frequency spectrum having frequency components obtained at regular intervals on a logarithmic frequency scale for each of frames that are defined by regular time intervals, and thereby generates a time series of the frequency spectrum; a cross-correlation coefficients calculating unit that calculates, for each target frame of the frames, a cross-correlation coefficients between frequency spectra calculated for two different frames that are in vicinity of the target frame and a predetermined frame width apart from each other; and a shift amount predicting unit that predicts a shift amount of the frequency spectra on the logarithmic frequency scale with respect to the predetermined frame width by use of the cross-correlation coefficients.

Abstract: A system and method are disclosed for extending the bandwidth of a narrowband signal such as a speech signal. The method applies a parametric approach to bandwidth extension but does not require training. The parametric representation relates to a discrete acoustic tube model (DATM). The method comprises computing narrowband linear predictive coefficients (LPCs) from a received narrowband speech signal, computing narrowband partial correlation coefficients (parcors) using recursion, computing Mnb area coefficients from the partial correlation coefficient, and extracting Mwb area coefficients using interpolation. Wideband parcors are computed from the Mwb area coefficients and wideband LPCs are computed from the wideband parcors.

Abstract: Parameters for distributions of a hidden trajectory model including means and variances are estimated using an acoustic likelihood function for observation vectors as an objection function for optimization. The estimation includes only acoustic data and not any intermediate estimate on hidden dynamic variables. Gradient ascent methods can be developed for optimizing the acoustic likelihood function.