Abstract: A sound event detecting module for detecting whether a sound event with characteristic of repeating is generated. A sound end recognizing unit recognizes ends of sounds according to a sound signal to generate sound sections and multiple sets of feature vectors of the sound sections correspondingly. A storage unit stores at least M sets of feature vectors. A similarity comparing unit compares the at least M sets of feature vectors with each other, and correspondingly generates a similarity score matrix, which stores similarity scores of any two of the sound sections of the at least M of the sound sections. A correlation arbitrating unit determines the number of sound sections with high correlations to each other according to the similarity score matrix. When the number is greater than one threshold value, the correlation arbitrating unit indicates that the sound event with the characteristic of repeating is generated.

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 audio encoding device which can improve encoding performance while performing division search on an algebraic codebook in an audio encoding. In a distortion minimizing unit (112) of a CELP encoding device: a maximum correlation value calculation unit (221) calculates a correlation value by using each pulse and a target signal in each candidate position for four pulses constituting the fixed codebook so as to acquire a maximum value of the correlation value for each pulse and calculates a maximum correlation value by using the maximum value of the correlation value; a sorting unit (222) divides the four pulses into two subsets each having two pulses; and a search unit (224) performs a division search on the fixed codebook and acquires a code indicating the positions and polarities of the four pulses where the encoding distortion is minimum.

Abstract: Provided is an audio coding apparatus and method that can selectively apply a operation mode of a coding module for stereo or multi-channel representation according to input signal characteristics of each channel, when voice or music signals are transmitted using an audio codec in portable terminals capable of stereo or multi-channel input and output. The audio coding apparatus includes a down-mixer for down-mixing multi-channel audio signals into mono signals; a coder for coding the mono signals; an input channel correlation analyzer for deciding whether to give them stereo effect based on their signal distribution characteristics, and outputting a control signal indicating whether to perform stereo representation process; and a stereo representation unit for performing stereo representation process onto the multi-channel audio signals when the control signal indicating to perform stereo representation process.

Abstract: Frame mapping-based cross-lingual voice transformation may transform a target speech corpus in a particular language into a transformed target speech corpus that remains recognizable, and has the voice characteristics of a target speaker that provided the target speech corpus. A formant-based frequency warping is performed on the fundamental frequencies and the linear predictive coding (LPC) spectrums of source speech waveforms in a first language to produce transformed fundamental frequencies and transformed LPC spectrums. The transformed fundamental frequencies and the transformed LPC spectrums are then used to generate warped parameter trajectories. The warped parameter trajectories are further used to transform the target speech waveforms in the second language to produce transformed target speech waveform with voice characteristics of the first language that nevertheless retain at least some voice characteristics of the target speaker.

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: In one embodiment, a method of receiving a decoded audio signal that has a transmitted pitch lag is disclosed. The method includes estimating pitch correlations of possible short pitch lags that are smaller than a minimum pitch limitation and have an approximated multiple relationship with the transmitted pitch lag, checking if one of the pitch correlations of the possible short pitch lags is large enough compared to a pitch correlation estimated with the transmitted pitch lag, and selecting a short pitch lag as a corrected pitch lag if a corresponding pitch correlation is large enough. The postprocessing is performed using the corrected pitch lag. In another embodiment, when the existence of irregular harmonics or wrong pitch lag is detected, a coded-excited linear prediction (CELP) postfilter is made more aggressive.

Abstract: A scalable decoder device (50) for signals representing audio comprises a primary decoder (21) connected to an input (40). The primary decoder (21) is arranged to provide a primary decoded signal (23) based on received parameters (4). A primary postfilter (31) is connected to the primary decoder (23) to provide a primary postfiltered signal (32). A secondary enhancement decoder (45) is connected to the input (40) and arranged to provide a secondary decoded enhancement signal (44). The device further comprises a combiner arrangement (55), arranged for combining the primary postfiltered signal (32) and a signal (53) based on the secondary decoded enhancement signal (44) into an output signal (6) to be provided at an output (6). The combining is made with an adaptable strength relation between contributions from the two signals. A method for decoding coded signals representing audio operates in analogy with the scalable decoder device (50).

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 method and device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses distributed over the pulse positions. In the algebraic codebook searching method and device, a reference signal for use in searching the algebraic codebook is calculated. In a first stage, a position of a first pulse is determined in relation with the reference signal and among the number of pulse positions. In each of a number of stages subsequent to the first stage, (a) an algebraic codebook gain is recomputed, (b) the reference signal is updated using the recomputed algebraic codebook gain and (c) a position of another pulse is determined in relation with the updated reference signal and among the number of pulse positions.

Abstract: An apparatus for analyzing a sound signal is based on an ear model for deriving, for a number of inner hair cells, an estimate for a time-varying concentration of transmitter substance inside a cleft between an inner hair cell and an associated auditory nerve from the sound signal so that an estimated inner hair cell cleft contents map over time is obtained. This map is analyzed by means of a pitch analyzer to obtain a pitch line over time, the pitch line indicating a pitch of the sound signal for respective time instants. A rhythm analyzer is operative for analyzing envelopes of estimates for selected inner hair cells, the inner hair cells being selected in accordance with the pitch line, so that segmentation instants are obtained, wherein a segmentation instant indicates an end of the preceding note or a start of a succeeding note. Thus, a human-related and reliable sound signal analysis can be obtained.

Abstract: A method and apparatus to encode and decode an audio/speech signal is provided. An inputted audio signal or speech signal may be transformed into at least one of a high frequency resolution signal and a high temporal resolution signal. The signal may be encoded by determining an appropriate resolution, the encoded signal may be decoded, and thus the audio signal, the speech signal, and a mixed signal of the audio signal and the speech signal may be processed.

Abstract: A speech recognition system includes: a speed level classifier for measuring a moving speed of a moving object by using a noise signal at an initial time of speech recognition to determine a speed level of the moving object; a first speech enhancement unit for enhancing sound quality of an input speech signal of the speech recognition by using a Wiener filter, if the speed level of the moving object is equal to or lower than a specific level; and a second speech enhancement unit enhancing the sound quality of the input speech signal by using a Gaussian mixture model, if the speed level of the moving object is higher than the specific level. The system further includes an end point detection unit for detecting start and end points, an elimination unit for eliminating sudden noise components based on a sudden noise Gaussian mixture model.

Abstract: A method of transmitting speech data to a remote device in a distributed speech recognition system, includes the steps of: dividing an input speech signal into frames; calculating, for each frame, a voice activity value representative of the presence of speech activity in the frame; grouping the frames into multiframes, each multiframe including a predetermined number of frames; calculating, for each multiframe, a voice activity marker representative of the number of frames in the multiframe representing speech activity; and selectively transmitting, on the basis of the voice activity marker associated with each multiframe, the multiframes to the remote device.

Abstract: Methods and systems for monitoring mobile communication terminals. A correlation system selects candidate communication terminals to be monitored, and then attempts to identify whether the candidate terminals are indeed operated by target users. Following successful correlation of a candidate terminal with a target user, various surveillance actions can be performed with respect to the terminal. Correlation of candidate communication terminals with target users is based on identification of speech key-phrases. When evaluating a given candidate terminal, the system analyzes speech that is communicated via the candidate terminal and attempts to detect one or more of the speech key-phrases in the analyzed speech.

Abstract: A fixed codebook searching apparatus, includes a convolution operator, implemented by at least one processor, that convolves an impulse response of a perceptually weighted synthesis filter with an impulse response vector that has values at negative times, to generate a second impulse response vector that has values at negative times. A matrix generator, implemented by at least one processor, generates a Toeplitz-type convolution matrix using the second impulse response vector generated by the convolution operator. A searcher, implemented by at least one processor, performs a codebook search by maximizing a term using the Toeplitz-type convolution matrix.

Abstract: A game apparatus includes a CPU core for creating an input envelope and a registered envelope. The input envelope has a plurality of envelope values detected from a voice waveform input in real time through a microphone. The registered envelope has a plurality of envelope values detected from a voice waveform previously input. Both of the input envelope and the registered envelope are stored in a RAM. The CPU core evaluates difference of the envelope values between the input envelope and the registered envelope. When an evaluated value satisfies a condition, the CPU core executes a process according to a command assigned to the registered envelope.

Abstract: In one aspect, a method of processing a voice signal to extract information to facilitate training a speech synthesis model is provided. The method comprises acts of detecting a plurality of candidate features in the voice signal, performing at least one comparison between one or more combinations of the plurality of candidate features and the voice signal, and selecting a set of features from the plurality of candidate features based, at least in part, on the at least one comparison. In another aspect, the method is performed by executing a program encoded on a computer readable medium. In another aspect, a speech synthesis model is provided by, at least in part, performing the method.

Abstract: There is provided an audio encoding device capable of effectively encoding stereo audio in audio encoding having monaural-stereo scalable configuration. In this device, a correlation degree comparison unit (304) calculates correlation in a first channel (correlation degree between the past signal and the current signal in the first channel) from the first channel audio signal and calculates correlation in a second channel (correlation degree between the past signal and the current signal in the second channel) from the second channel audio signal. The correlation in the first channel is compared to the correlation in the second channel. A channel having the greater correlation is selected.

Abstract: Methods for estimating speech model parameters are disclosed. For pulsed parameter estimation, a speech signal is divided into multiple frequency bands or channels using bandpass filters. Channel processing reduces sensitivity to pole magnitudes and frequencies and reduces impulse response time duration to improve pulse location and strength estimation performance. These methods are useful for high quality speech coding and reproduction at various bit rates for applications such as satellite and cellular voice communication.

Abstract: There is provided an audio encoding device capable of effectively encoding a stereo audio even when a correlation between channels of the stereo audio is small. In the device, a monaural signal generation unit (110) generates a monaural signal by using a first channel signal and a second channel signal contained in the stereo signal. An encoding channel selection unit (120) selects one of the first channel signal and the second channel signal. An encoding unit including a monaural signal encoding unit (112), a first channel encoding unit (122), a second channel encoding unit (124), and a switching unit (126) encodes the generated monaural signal to obtain core-layer encoded data and encodes the selected channel signal to obtain extended layer encoded data corresponding to the core-layer encoded data.

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: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

Abstract: There is provided a speech encoder for performing an algorithm that comprises obtaining (205) a plurality of open-loop pitch candidates from a current frame of a speech signal, the plurality of open-loop pitch candidates including a first open-loop pitch candidate and a second open-loop pitch candidate; obtaining (205) a voicing information from one or more previous frames; and selecting (280) one of the plurality of open-loop pitch candidates as a final pitch of the current frame using the voicing information from the one or more previous frames. In one aspect, the voicing information from the one or more previous frames includes a previous pitch of the one or more previous frames. In a further aspect, selecting the final pitch of the current frame includes selecting (210) an initial open-loop pitch from that has the maximum long-term correlation value.

Abstract: A stereo sound decoding apparatus wherein lost-frame compensation performance has been improved to enhance the quality of decoded sounds.

Abstract: A system improves the speech intelligibility and the speech quality of a speech segment. The system includes a dynamic controller that detects a background noise from an input by modeling a signal. A variable gain amplifier adjusts the variable gain of the amplifier in response to an output of dynamic controller. A shaping filter adjusts a speech signal by tilting portions of the speech signal of the dynamic controller.

Abstract: An apparatus for decoding a signal and method thereof are disclosed, by which the audio signal can be controlled in a manner of changing/giving spatial characteristics (e.g., listener's virtual position, virtual position of a specific source) of the audio signal. The present invention includes receiving an object parameter; extracting object information by parsing the received object parameter; generating a control parameter using the extracted object information and control information including at least one of user control information, default control information, device control information, and device information; and, generating a rendering parameter determining a position and level of an object in an output signal using the object parameter and the control parameter.

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: A method for identifying a frame type is disclosed. The present invention includes receiving current frame type information, obtaining previously received previous frame type information, generating frame identification information of a current frame using the current frame type information and the previous frame type information, and identifying the current frame using the frame identification information. And, a method for identifying a frame type is disclosed. The present invention includes receiving a backward type bit corresponding to current frame type information, obtaining a forward type bit corresponding to previous frame type information, generating frame identification information of a current frame by placing the backward type bit at a first position and placing the forward type bit at a second position.

Abstract: Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.

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 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 system and method are disclosed for modifying an audio signal. A pitch associated with the audio signal is detected. A portion of the audio signal that is associated with the detected pitch is modified. Controlling the modification of a primary audio signal is disclosed. The level of a secondary audio signal is monitored. Modification of the primary audio signal is enabled if the level of the secondary audio signal rises above a first prescribed threshold at a time when the primary audio signal is not being modified. Modification of the primary audio signal is disabled if the level of the secondary audio signal drops below a second prescribed threshold at a time when the primary audio signal is being modified.

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: 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: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

Abstract: A triangular microphone assembly (101) for use in a vehicle accessory includes a mirror housing (106) adapted for attachment to the interior of the vehicle. A mirror is disposed in an opening of the mirror housing (106) and a plurality of virtual digital microphones (108a, 108b, 108c) are arranged in a substantially triangular configuration in the mirror housing (106). A digital signal processor (DSP) (537) is used for receiving signals from the plurality of digital microphones (108a, 108b, 108c) such that the digital microphones exhibit directional characteristics for reducing undesirable noise in at least one direction by normalizing the phase of the received signals as a function of signal frequency.

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 method and apparatus for processing audio for playback to provide a smooth transition between a beginning region of an audio track and an end region of a previous audio track is disclosed. A quantity representative of a chromagram is calculated for each of the audio tracks and the mixing points for the beginning and end regions of each audio track are identified. A quantity representative of a chromagram at the mixing point of the beginning region of the audio track and a quantity representative of a chromagram at the mixing point of the end region of the previous audio track are correlated to determine an order of audio tracks for playback and/or to determine the duration of the mix transition.

Abstract: A multi-channel audio decoder provides a reduced complexity processing to reconstruct multi-channel audio from an encoded bitstream in which the multi-channel audio is represented as a coded subset of the channels along with a complex channel correlation matrix parameterization. The decoder translates the complex channel correlation matrix parameterization to a real transform that satisfies the magnitude of the complex channel correlation matrix. The multi-channel audio is derived from the coded subset of channels via channel extension processing using a real value effect signal and real number scaling.

Abstract: Methods for estimating speech model parameters are disclosed. For pulsed parameter estimation, a speech signal is divided into multiple frequency bands or channels using bandpass filters. Channel processing reduces sensitivity to pole magnitudes and frequencies and reduces impulse response time duration to improve pulse location and strength estimation performance. These methods are useful for high quality speech coding and reproduction at various bit rates for applications such as satellite and cellular voice communication.

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: A method of processing a decoded speech (DS) signal including successive DS frames, each DS frame including DS samples. The method comprises: adaptively filtering the DS signal to produce a filtered signal; gain-scaling the filtered signal with an adaptive gain updated once a DS frame, thereby producing a gain-scaled signal; and performing a smoothing operation to smooth possible waveform discontinuities in the gain-scaled signal.