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: A channel estimation method and system using linear correlation based interference cancellation combined with decision-feedback-equalization (LCIC-DFE) are provided. The channel estimation method includes generating a first correlation sequence by calculating a linear correlation between a baseband sampled complex signal and a locally stored pseudo-noise signal and obtaining a second correlation sequence by iteratively removing inter-path interference from the first correlation sequence and generating a first channel impulse response (CIR) sequence based on the second correlation sequence. And, obtaining a third correlation sequence by removing random-data interference from the second correlation sequence based on the first CIR sequence and a feedback signal and generating a second CIR sequence based on the third correlation sequence.

Abstract: A method and apparatus for estimating the pitch period of a signal. The method includes identifying a first candidate pitch period by performing a search only over a first range of potential pitch periods. The method further includes determining a second candidate pitch period by dividing the first candidate pitch period by an integer, wherein the second candidate pitch period is outside the first range of potential pitch periods. The method further includes selecting as the estimate of the pitch period of the signal the smaller of the candidate pitch periods that is such that portions of the signal separated by that candidate pitch period are well correlated.

Abstract: Apparatus, method, and medium for processing an audio signal using a correlation between bands are provided. The apparatus includes an encoding unit encoding an input audio signal and a decoding unit decoding the encoded input audio signal.

Abstract: A method and apparatus performing blind source separation using frequency-domain normalized multichannel blind deconvolution. Multichannel mixed signals are frames of N samples including r consecutive blocks of M samples. The frames are separated using separating filters in frequency domain in an overlap-save manner by discrete Fourier transform (DFT). The separated signals are then converted back into time domain using inverse DFT applied to a nonlinear function. Cross-power spectra between separated signals and nonlinear-transformed signals are computed and normalized by power spectra of both separated signals and nonlinear-transformed signals to have flat spectra. Time domain constraint is then applied to preserve first L cross-correlations. These alias-free normalized cross-power spectra are further constrained by nonholonomic constraints. Then, natural gradient is computed by convolving alias-free normalized cross-power spectra with separating filters.

Abstract: Possible segmentations for an audio signal are scored based on distortions for feature vectors of the audio signal and the total number of segments in the segmentation. The scores are used to select a segmentation and the selected segmentation is used to identify a starting point and an ending point for a speech signal in the audio signal.

Abstract: A system and method is provided for determining whether a data frame of a coded speech signal corresponds to voice or to noise. In one embodiment, a voice activity detector determines a cross-correlation of data. If the cross-correlation is lower than a predetermined cross-correlation value, then the data frame corresponds to noise. If not, then the voice activity detector determines a periodicity of the cross-correlation and a variance of the periodicity. If the variance is less than a predetermined variance value, then the data frame corresponds to voice. In another embodiment, a method determines energy of the data frame and an average energy of the coded speech signal. If the data frame is one of a predetermined number of initial data frames, then a comparison between the average energy to the energy of the data frame is used to determine whether the data frame is noise or voice.

Abstract: A pitch wave signal creation method as a preliminary process for efficiently coding a speech wave signal having a fluctuated pitch period is provided. A speech signal compressing/expanding apparatus and a speech signal synthesizing apparatus using the method, and a signal processing associated therewith are further provided. The pitch wave creation method of the invention is essentially comprised of a method of detecting the instantaneous pitch period of each pitch wave element of the speech wave signal, and a process of converting a corresponding pitch wave element into a normalized pitch wave element having a predetermined fixed time length by expanding and compressing the pitch wave element on a time axis while retaining its wave pattern based on the each detected instantaneous pitch period. The speech signal having a pitch fluctuation can be compressed in high quality and high efficiency by coding or synthesizing the speech wave signal using the pitch wave signal creation method of the invention.

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: An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.

Abstract: A pitch detection method and apparatus, the pitch detection apparatus includes: a data rearrangement unit which rearranges voice data on the basis of a center peak of the voice data included in a single frame; a decomposition unit which decomposes rearranged voice data into even symmetrical components on the basis of a center peak; a pitch determination unit which obtains a segment correlation value between a reference point and at least one or more local peaks in relation to even symmetrical components, and determines the location of a local peak corresponding to a maximum segment correlation value among the obtained segment correlation values, as a pitch period.

Abstract: A digital audio signal can be processed to have a constant pitch for a range of playback rates. A playback rate associated with a portion of the digital audio signal can be received, which differs from the rate at which the digital audio signal was recorded. A channel of the digital audio signal can be divided into a plurality of windows and one or more cross-correlation evaluations can be performed for at least a first window and a second window of the channel to determine an appropriate overlap. Further, the first cross-correlation evaluation can be performed using an ideal overlap. One or more sample values associated with a first waveform included in the first window can be averaged with one or more corresponding sample values associated with a second waveform included in the second window for the determined appropriate overlap to generate a rate-modified audio signal, which can be played.

Abstract: In binaural stereo coding, only one monaural channel is encoded. An additional layer holds the parameters to retrieve the left and right signal. An encoder is disclosed which links transient information extracted from the mono encoded signal to parametric multi-channel layers to provide increased performance. Transient positions can either be directly derived from the bit-stream or be estimated from other encoded parameters (e.g. window-switching flag in mp3).

Abstract: A low complexity packet loss concealment method for use in voice-over-IP speech transmission calculates a cross-correlation of previous speech data to estimate the pitch period of the previous speech when speech frames have been lost. A tap interval used to calculate the cross-correlation is dynamically adapted, thereby reducing the computational complexity of the process. In addition, the pitch period estimation is bypassed completely when it is determined not to be necessary, as a result of the speech being unvoiced or silence. A waveform “bending” operation is performed into the current frame without inserting any algorithmic delay into each frame.

Abstract: A multiple step adaptive method for time scaling. Synthesizing S3[n] signal from signal S1[n]signal and S2[n]signal. Comprising following steps: (a) calculating a first magnitude of a cross-correlation function of S1[n]signal and S2[n]signal according to a first index; (b) comparing the first magnitude with a threshold value; (c) if first magnitude is smaller than threshold value, calculating a first reference magnitude of cross-correlation function of S1[n]signal and S2[n]signal according to a first reference index behind the first index by a first determined number, or calculating a second reference magnitude of the cross-correlation function of the S1[n] signal and the S2[n] signal according to a second reference index behind the first index by a second number; (d) synthesizing the S3[n] signal by adding S1[n]signal to the S2[n] signal in accordance with a maximum index corresponding to a largest magnitude among all the magnitudes calculated in (c).

Abstract: A method, system, and computer program product are provided for detecting at least one signal of interest within an input signal. At least one set of samples from the input signal is multiplied by at least one set of samples representing a complex conjugate of the input signal to obtain a series of correlation samples. Sets of one or more consecutive correlation samples are summed to obtain a series of pulse sums. A plurality of non-consecutive pulse sums are integrated to obtain a search value.

Abstract: A random code vector reading section and a random codebook of a conventional CELP type speech coder/decoder are respectively replaced with an oscillator for outputting different vector streams in accordance with values of input seeds, and a seed storage section for storing a plurality of seeds. This makes it unnecessary to store fixed vectors as they are in a fixed codebook (ROM), thereby considerably reducing the memory capacity.

Abstract: A system, computer-readable medium and generated signal are disclosed for extending the bandwidth of a first signal (i.e., a narrowband signal) such as a speech signal. The system produces a second signal from a first signal by computing first area coefficients from a first signal, generating second area coefficients from the first area coefficients and generating a second signal using the second area coefficients. The first signal may be a narrowband signal and second signal may be a wideband signal. The first area coefficients may be narrowband coefficients and the second area coefficients may be wideband area coefficients.

Abstract: An apparatus and a method for symbol decoding of baseband data in a wireless communications network is disclosed, and specifically CCK subsymbol prediction and symbol demodulation that occurs at 5.5 Mbps or 11 Mbps. The apparatus is configured to demodulate or predict the data differently, depending on the modulation rate. If the data was modulated at 11 Mbps, the ?3 rotator is rotated through each of its possible phase values and symbol correlation takes four clock cycles to complete. If the data was modulated at 5.5 Mbps, ?3 is not rotated with a set value of 0 within the correlator architecture, thereby saving power and reducing symbol correlation and subsymbol prediction to a single cycle while in such transmission mode.

Abstract: A nonlinear overlap method for time scaling to synthesize an S1[n] and an S2[n] into an S3[n] is disclosed. The S1[n] and the S2[n] having N1 and N2 signals respectively. The nonlinear overlap method includes the following steps: (a) delaying the S2[n] by a predetermined number and forming an S5[n], (b) establishing a correlogram of a cross-correlation function of the S1[n] and S5[n], and (c) setting S3[n] as a number of S1[n] when 0<=n<; as a number formed by overlap-adding the S1[n] and an S4[n] in a weighting manner when (the predetermined number+the maximum index+the first threshold)<=n<(N1?a second threshold); and as a number of S4 wherein the first and second thresholds are not equal to zero at the same time, and the S4[n] is formed by delaying the S5[n] by the maximum index.

Abstract: A method and apparatus for enhancing the receiving and information identification functions of multiple access communications systems by employing one or more optical processors configured as a bank of 1-D correlators. The present invention is particularly useful in a DS/SS CDMA communications system, resulting in a multiuser CDMA system that approaches carrier to noise performance (C/N) as opposed to being limited by multiple access interference (MAI). The correlators are arranged in parallel to detect and/or demodulate the received signal, in conjunction with one or more complex algorithms to perform near-optimum multiuser detection, perform multipath combining and/or perform carrier Doppler compensation.

Abstract: The present invention provides a method and system for computerized synchronization of an audio stream having a first synchronized textual representation usable for subtitling of the audio stream in the original language, with a second synchronized textual representation which can be used as an alternative subtitle including a transcription of the original language into another language. Time synchronous links can be built between the audio stream and, for instance, the textual representations of the words spoken in the audio stream. More particularly, the second representation can inherit the synchronization between the audio stream and the first representation using structure association information determined between the first and the second representation.

Abstract: A system for the detection of narrowband signals in wideband noise that combines information across two frequency channels that straddle the frequency of the target signal. Two band pass filters having center frequencies that straddle the frequency of the target signal and that have phase transfer functions that differ by 180 degrees relative to each other at the frequency of the target signal. The presence of the target signal is detected by performing a running cross-correlation of the outputs of saturating, non-linearities that follow from the filters, and determining when the output of the running cross-correlator drops below a predetermined threshold due to the phase shift between the two filter responses caused by the presence of the target signal.

Abstract: A computationally efficient and robust pitch detection and tracking system and related methods are presented. According to certain exemplary implementations a method is presented comprising identifying an initial set of pitch period candidates using a first estimation algorithm, filtering the initial set of candidates and passing the filtered candidates through a second, more accurate pitch estimation algorithm to generate a final set of pitch period candidates from which the most likely pitch value is selected.

Abstract: A method for preparing a speech signal for encoding comprises determining whether the spectral content of an input speech signal is representative of a defined spectral characteristic (e.g., a defined characteristic slope). A frequency specific filter component of a weighting filter is controlled based on the determination of the spectral content of the speech signal or/and its location in the encoder. A core weighting filter component of the weighting filter may be maintained regardless of the spectral content of the speech signal.

Abstract: A system, computer readable medium, and method for sampling a speech signal; dividing the sampled speech signal into overlapped frames; extracting first pitch information from a frame using frequency domain analysis; providing at least one pitch candidate, each being associated with a spectral score, from the first pitch information, each of the at least one pitch candidate representing a possible pitch estimate for the frame; extracting second pitch information from the frame using a time domain analysis; providing a correlation score for the at least one pitch candidate from the second pitch information; and selecting one of the at least one pitch candidate to represent the pitch estimate of the frame. The system, computer readable medium, and method are suitable for speech coding and for distributed speech recognition.

Abstract: One embodiment of the present invention includes a circular shift register, K storage elements, and a code register. The circular shift register having N data samples circularly shifts a first data sample of the N data samples into a data position at a first clock frequency. The N data samples correspond to signal received from one of K satellites in a global positioning system (GPS). The N data samples are loaded into the circular shift register at a second clock frequency. The K storage elements store K code sequences, respectively. Each of the K code sequences has N code samples and includes a first code sample being written at a code position corresponding to the data position at a third clock frequency. The K storage elements correspond to the K satellites. The code register stores the N code samples loaded from one of the K storage elements at a fourth clock frequency. The fourth clock frequency is K times faster than the first clock frequency.

Abstract: An application server generates and maintains a server-side data record, also referred to as a “brownie”, that includes application state information and user attribute information for multiple users within a single session controlled by a web-based browser. The brownie includes a session identifier that uniquely identifies the session, and a subsession identifier that uniquely identifies each corresponding user of the application session. As each new user is added to the session, for example by initiating a call to the new user, the application server stores the subsession identifier and corresponding application state information for the new user in the same brownie. In response to receiving a second web page request from the browser that includes the session identifier, the application server initiates a new web application instance, and recovers the brownie from the memory based on the session identifier included in the second page request.

Abstract: In order to efficiently retrieve AV data by using signal characteristics as retrieval conditions, in a first step, a comparison and determination section computes a correlation coefficient (degree of similarity) of a spectrum coefficient of coded audio data and a spectrum coefficient of a sample waveform, and extracts correlation coefficients such that the value of the computed spectrum coefficient is larger than a threshold value which is set in the first step, and assumes them to be retrieval results. In a second step, the comparison and determination section determines whether or not the retrieval result is satisfactory. When it is determined that the number of pieces of audio data retrieved in the first step is equal to or greater than the predetermined threshold value and the retrieval result is not satisfactory, the process proceeds to a third step.

Abstract: In a method for tracking pitch in a speech signal, first and second window vectors are created from samples taken across first and second windows of the speech signal. The first window is separated from the second window by a test pitch period. The energy of the speech signal in the first window is combined with the correlation between the first window vector and the second window vector to produce a predictable energy factor. The predictable energy factor is then used to determine a pitch score for the test pitch period. Based in part on the pitch score, a portion of the pitch track is identified.

Abstract: A sample speech is analyzed by a speech analyzing unit to obtain sample characteristic parameters, and a coding distortion is calculated from the sample characteristic parameters in each of a plurality of coding modules. The sample characteristic parameters and the coding distortions are statistically processed by a statistical processing unit to obtain a coding module selecting rule. Thereafter, when a speech is analyzed by the speech analyzing unit to obtain characteristic parameters, an appropriate coding module is selected by a coding module selecting unit from the coding modules according to the coding module selecting rule on condition that a coding distortion for the characteristic parameters is minimized in the appropriate coding module. Thereafter, the characteristic parameters of the speech are coded in the appropriate coding module, and a coded speech is obtained. When the coded speech is decoded, a reproduced speech is obtained.

Abstract: For use in a synthesizer having a wave source that produces a periodic wave, frequency shifting circuitry for frequency-shifting the periodic wave and waveshaping circuitry for transforming the periodic wave into a waveform containing a formant, the frequency-shifting causing displacement of the formant, a circuit for, and method of, compensating for the displacement and a synthesizer employing the circuit or the method. In one embodiment, the circuit includes bias circuitry, coupled to the wave source and the frequency shifting circuitry, that introduces a bias into the periodic wave based on a degree to which the frequency shifting circuitry frequency shifts the periodic wave, the bias reducing a degree to which the formant is correspondingly frequency-shifted.

Abstract: In a speech coding apparatus, an input device inputs a mixed speech signal of a plurality of speakers. A separating device analyzes period characteristics of the input mixed speech signal, and separates the same signal into a plurality of single speech signals each associated with a corresponding one of the speakers, based on a result of the analysis. A first extracting device extracts source speech characteristic parameters included in each of the single speech signals. A second extracting device extracts a generic vocal-tract characteristic parameter from the input mixed speech signal. In a speech decoding apparatus, a first input device inputs the source speech characteristic parameters for each of the speakers. A second input device inputs the vocal-tract characteristic parameter.

Abstract: Methods are presented for reducing the processing required for CELP speech encoders which have multiple fixed stochastic codebook subframes corresponding to a single adaptive codebook subframe. The search for the optimum excitation vector in the fixed stochastic codebook requires calculating terms involving correlation of the target speech sample and the fixed stochastic codebook excitation vector as well as energy terms involving only the fixed stochastic codebook excitation vector, and for this class of CELP encoders it is possible to simplify the calculations to reduce their complexity and to make advantageous use of an adaptive energy lookup table. In addition, linear interpolation may be employed to estimate values for the adaptive energy lookup table and further reduce the computational burden.

Abstract: The invention concerns a voice activity detection device in which an input speech signal (x(n)) is divided in subsignals (S(s)) representing specific frequency bands and noise (N(s)) is estimated in the subsignals. On basis of the estimated noise in the subsignals, subdecision signals (SNR(s)) are generated and a voice activity decision (V.sub.ind) for the input speech signal is formed on basis of the subdecision signals. Spectrum components of the input speech signal and a noise estimate are calculated and compared. More specifically a signal-to-noise ratio is calculated for each subsignal and each signal-to-noise ratio represents a subdecision signal (SNR(s)). From the signal-to-noise ratios a value proportional to their sum is calculated and compared with a threshold value and a voice activity decision signal (V.sub.ind) for the input speech signal is formed on basis of the comparison.

Abstract: A system that synchronously segments a speech waveform using pitch period and a center of the pitch waveform. The pitch waveform center is determined by finding a local minimum of a centroid histogram waveform of the low-pass filtered speech waveform for one pitch period. The speech waveform can then be represented by one or more of such pitch waveforms or segments during speech compression, reconstruction or synthesis. The pitch waveform can be modified by frequency enhancement/filtering, waveform stretching/shrinking in speech synthesis or speech disguise. The utterance rate can also be controlled to speed up or slow down the speech.

Abstract: A low-complexity method and apparatus for performing signal decomposition in a low bit-rate WI speech encoder. A time-ordered sequence of sets of time-domain parameters is generated based on samples of a speech signal to be coded, each set of time-domain parameters corresponding to a waveform characterizing the speech signal. A cross correlation is then performed between two or more of said sets of time-domain parameters to produce a set of signals which represents relatively high rates of evolution of characterizing waveform shape across the time-ordered sequence of sets. Finally, the speech signal is coded based on the produced set of signals. A set of signals which represents relatively low rates of evolution of characterizing waveform shape across the time-ordered sequence of sets may also be produced.

Abstract: A pitch period extracting apparatus includes a microcomputer which determines a sampling frequency for an A/D converter, and a range of delay times for calculating autocorrelative values on the basis of the sampling frequency. For example, the delay times are set within a range of 20 samples.ltoreq.k.ltoreq.100 samples in a case of 8 kHz, and a range of 15 samples.ltoreq.k.ltoreq.75 samples in a case of 6 kHz. The microcomputer calculates the autocorrelative values of speech signal data stored in a buffer memory, and outputs a delay time at which a maximum autocorrelative value is obtainable as a pitch period of an inputted speech signal.

Abstract: A method and apparatus (100) for pitch-epoch-synchronous source-filter speech encoding by means of error component modeling methods (310) which capture fundamental orthogonal (uncorrelated) basis elements of an excitation source waveform. A periodic waveform model (318) along with four orthogonal error waveforms, desirably including phase error (319), ensemble error (321), standard deviation error (323), and mean error (324) waveforms, are incorporated together to form a complete description of the excitation. These error waveforms (319,321, 323, 324) represent those portions of the excitation that are not represented by the purely periodic model. By thus orthogonalizing the error components, the perceptual effect of each element is isolated from the composite set, and can thus be encoded separately.

Abstract: Harmonics coefficients are estimated in primary coefficients of an orthogonal transform of a speech or a music input signal by using a pitch frequency extracted from the input signal and are quantized into a harmonics code vector. Residue coefficients are calculated by removing the harmonics coefficients from the primary coefficients and quantized into residue code vectors and gain code vectors. It is possible to search harmonics excitation pulses at the harmonics locations for harmonics quantization into the harmonics code vector. On the other hand, it is possible to estimate the harmonics coefficients or excitation pulses by using quantized LSP parameters and to calculate secondary coefficients for use in weighting the harmonics quantization and residue quantization and, if applicable, in excitation pulse search.

Abstract: A method used by a controller (112) and a communication receiver (122) for performing a time-scale modification of a signal comprised of an input stream of signal representations to form an output stream of signal representations. An output block of S.sub.o signal representations is selected from the output stream for use in overlapping signal representations from the output block with S.sub.o contiguous signal representations of the input stream, the S.sub.o contiguous signal representations beginning with a predetermined signal representation. The overlapping process includes determining (406) a maximum similarity measure between selected portions of the input and the output streams, applying (410) a weighting function to each, and summing the results of each. The overlapping process further includes replacing (412) a selected portion of the output stream with the latter summation, and selecting (414) a new set of samples from the input stream to append to the output stream.

Abstract: A synthesis method in which that part of each interval of the original signal which contains the fundamental information is left unchanged, and only the remaining part of the interval is altered. In this way, not only is processing time reduced, but the natural sound of the synthetic signal is also improved. The main part of the interval is an exact reproduction of the original signal. At least the waveforms associated to voiced sounds are subdivided into a plurality of intervals, corresponding to the responses of the vocal duct to a series of excitation impulses of the vocal cords, synchronous with the fundamental frequency of the signal. Each interval is subjected to a weighting. The signals resulting from the weighting are replaced with a replica thereof shifted in time by an amount that depends on a prosodic information. The synthesis is then carried out by overlapping and adding the shifted signals.

Abstract: A vector quantizer for a speech coder for coding speech signals at low bit rates. The vector quantizer includes an auto-correlation calculation circuit for calculating an impulse response of a weighting function for each sub-interval of an input signal vector. The vector quantizer also includes a weighted cross-correlation calculation circuit for calculating a weighted cross-correlation of the weighted input signal vector and the weighted codevector having a code length equal to that of the input signal vector. The vector quantizer further includes a weighted auto-correlation calculation circuit for calculating an auto-correlation of the weighted codevectors, by using respective auto-correlations of the impulse responses, the codevectors and the cross-correlations.

Abstract: A pitch period extracting apparatus includes a microcomputer which determines a sampling frequency for an A/D converter, and a range of delay times for calculating autocorrelative values on the basis of the sampling frequency. For example, the delay times are set within a range of 20 samples?k?100 samples in a case of 8 kHz, and a range of 15 samples?k?75 samples in a case of 6 kHz. The microcomputer calculates the autocorrelative values of speech signal data stored in a buffer memory, and outputs a delay time at which a maximum autocorrelative value is obtainable as a pitch period of an inputted speech signal.