Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

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: 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: 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: A signal compression method and apparatus are provided. The signal compression method includes: multiplying an input signal by a window function; calculating original autocorrelation coefficients of a windowed input signal; calculating a white-noise correction factor or a lag-window according to the original autocorrelation coefficients, and calculating modified autocorrelation coefficients according to the original autocorrelation coefficients, the white-noise correction factor and the lag-window; calculating linear prediction coefficients according to the modified autocorrelation coefficients; and outputting a coded bit stream according to the linear prediction coefficients.

Abstract: A method of and system for transient noise rejection for improved speech recognition. The method comprises the steps of (a) receiving audio including user speech and at least some transient noise associated with the speech, (b) converting the received audio into digital data, (c) segmenting the digital data into acoustic frames, and (d) extracting acoustic feature vectors from the acoustic frames. The method also comprises the steps of (e) evaluating the acoustic frames for transient noise on a frame-by-frame basis, (f) rejecting those acoustic frames having transient noise, (g) accepting as speech frames those acoustic frames having no transient noise and, thereafter, (h) recognizing the user speech using the speech frames.

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: 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: It is provided a voice activity decision apparatus capable of accurately performing the decision on the state being associated with a sound interval or a silence interval also in terms of the input signal having many aperiodic components and/or plural mixed different periodic components. The apparatus 1 comprises: an autocorrelation calculating unit 11 for calculating autocorrelation values of an input signal; a delay calculating unit 12 for calculating plural delays at which autocorrelation values calculated by the autocorrelation calculating unit 11 become maximums; a noise deciding unit 13 for deciding whether the input signal is a noise or not based on the plurality of delays calculated by the delay calculating unit 12; and an activity decision unit 14 for performing the activity decision in terms of the input signal based on results of decision by the noise deciding unit 13 and the input signal.

Abstract: A method (100) includes receiving (101) an input digital audio signal comprising a narrow-band signal. The input digital audio signal is processed (102) to generate a processed digital audio signal. A high-band energy level corresponding to the input digital audio signal is estimated (103) based on a transition-band of the processed digital audio signal within a predetermined upper frequency range of a narrow-band bandwidth. A high-band digital audio signal is generated (104) based on the high-band energy level and an estimated high-band spectrum corresponding to the high-band energy level.

Abstract: A signal compression method includes: multiplying an input signal by a window function, calculating original autocorrelation coefficients of a windowed input signal. The method also includes calculating a white-noise correction factor or a lag-window according to the original autocorrelation coefficients, and calculating modified autocorrelation coefficients according to the original autocorrelation coefficients, the white-noise correction factor and the lag-window. The method further includes calculating linear prediction coefficients according to the modified autocorrelation coefficients, and outputting a coded bit stream according to the linear prediction coefficients.

Abstract: A method, database, and article of manufacture comprising a plurality of audio fingerprints. Each audio fingerprint contains characteristic information about a corresponding audio frame and is produced by filtering the corresponding audio frame into frequency bands, resampling the filtered audio signals at a nonlinear timescale, transforming the resampled audio signals for each frequency band to produce a feature vector for the frequency band, and computing the audio fingerprint based on the set of feature vectors, and one or more index values for one or more of the audio fingerprints, where the audio fingerprints are organized according to their index values.

Abstract: A method and an apparatus for detecting a pitch in input voice signals by using a spectral auto-correlation. The pitch detection method includes: performing a Fourier transform on the input voice signals after performing a pre-processing on the input voice signals, performing an interpolation on the transformed voice signals, calculating a spectral difference from a difference between spectrums of the interpolated voice signals, calculating a spectral auto-correlation by using the calculated spectral difference, determining a voicing region based on the calculated spectral auto-correlation, and extracting a pitch by using the spectral auto-correlation corresponding to the voicing region.

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 first signal of two signals to be compared for similarity is divided into small areas and one small area is selected for calculating the correlation with a second signal using a correlative method. Then, the quantity of translation, expansion rate and similarity in an area where the similarity, which is the square of the correlation value, reaches its maximum, are found. Values based on the similarity are integrated at a position represented by the quantity of translation and expansion rate. Similar processing is performed with respect to all the small areas, and at a peak where the maximum integral value of the similarity is obtained, its magnitude is compared with a threshold value to evaluate the similarity. The small area voted for that peak can be extracted.

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: A compressor device for a compander system has a level detecting/control device and a pre-emphasis device for carrying out an adaptive pre-emphasis filtering. The invention is also directed to an expander device for a compander system with a level detecting/control device and a de-emphasis device for carrying out an adaptive de-emphasis filtering.

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 VF detecting apparatus capable of highly accurate vocal fry (VF) detection includes: a very-short-term peak detection processing unit framing a speech signal with a first frame of a first frame length and first frame shift amount and detecting each power peak; a short-term periodicity detecting unit framing the speech signal with a second frame of a second frame length longer than the first frame length and a second frame shift amount larger than the first frame length and determining presence/absence of periodicity in each of the resulting frame; a periodicity checking unit for detecting power peaks in those frames determined to have no periodicity, from among the detected power peaks; and a similarity checking unit for detecting, for each of the selected power peaks, neighboring power peaks having high cross-correlation and detecting the section therebetween as the VF section.

Abstract: A method for supporting an encoding of an audio signal is shown, wherein at least a first and a second coder mode are available for encoding a section of the audio signal. The first coder mode enables a coding based on two different coding models. A selection of a coding model is enabled by a selection rule which is based on signal characteristics which have been determined for a certain analysis window. In order to avoid a misclassification of a section after a switch to the first coder mode, it is proposed that the selection rule is activated only when sufficient sections for the analysis window have been received. The invention relates equally to a module in which this method is implemented, to a device and a system comprising such a module and to a software program product including a software code for realizing the proposed method.

Abstract: Methods, digital systems, and computer readable media are provided for determining a predominant fundamental frequency of a frame of an audio signal by finding a maximum absolute signal value in history data for the frame, determining a number of bits for downshifting based on the maximum absolute signal value, computing autocorrelations for the frame using signal values downshifted by the number of bits, and determining the predominant fundamental frequency using the computed autocorrelations.

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

Abstract: The fundamental frequency of a harmonic signal is estimated by forming a fundamental frequency hypothesis (f0?). A comb filter is provided based on the fundamental frequency hypothesis. The harmonic signal is filtered using the comb filter. The fundamental frequency hypothesis is tested for each tooth in the comb filter. A signal indicating an estimated fundamental frequency of the provided harmonic signal may be outputted based on the testing.

Abstract: There is provided a scalable encoding device capable of realizing a bandwidth scalable LSP encoding with high performance by improving the conversion performance from narrow band LSPs to wide band LSPs.

Abstract: An enhancement system extracts pitch from a processed speech signal. The system estimates the pitch of voiced speech by deriving filter coefficients of an adaptive filter and using the obtained filter coefficients to derive pitch. The pitch estimation may be enhanced by using various techniques to condition the input speech signal, such as spectral modification of the background noise and the speech signal, and/or reduction of the tonal noise from the speech signal.

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: An encoding method includes extracting background noise characteristic parameters within a hangover period; for a first superframe after the hangover period, performing background noise encoding based on the extracted background noise characteristic parameters; for superframes after the first superframe, performing background noise characteristic parameter extraction and DTX decision for each frame in the superframes after the first superframe; and for the superframes after the first superframe, performing background noise encoding based on extracted background noise characteristic parameters of the current superframe, background noise characteristic parameters of a plurality of superframes previous to the current superframe, and a final DTX decision. Also, a decoding method and apparatus and an encoding apparatus are disclosed.

Abstract: A signal processing device for processing an input signal includes gain calculating means and feature quantity calculating means. The gain calculating means is configured to obtain information indicating magnitude of noise to be added to the input signal on a basis of periodicity information indicating periodicity of the input signal and power of the input signal. The feature quantity calculating means is configured to obtain periodicity information of a noise-added signal obtained by adding noise having magnitude corresponding to the gain information to the input signal as a feature quantity of the input signal.

Abstract: In one configuration, erasure of a significant frame of a sustained voiced segment is detected. An adaptive codebook gain value for the erased frame is calculated based on the preceding frame. If the calculated value is less than (alternatively, not greater than) a threshold value, a higher adaptive codebook gain value is used for the erased frame. The higher value may be derived from the calculated value or selected from among one or more predefined values.

Abstract: There is provided a pitch lag predictor for use by a speech decoder to generate a predicted pitch lag parameter. The pitch lag predictor comprises a summation calculator configured to generate a first summation based on a plurality of previous pitch lag parameters, and a second summation based on a plurality of previous pitch lag parameters and a position of each of the plurality of previous pitch lag parameters with respect to the predicted pitch lag parameter; a coefficient calculator configured to generate a first coefficient using a first equation based on the first summation and the second summation, and a second coefficient using a second equation based on the first summation and the second summation, wherein the first equation is different than the second equation; and a predictor configured to generate the predicted pitch lag parameter based on the first coefficient and the second coefficient.

Abstract: A method of estimating a pitch period of a first portion of a signal wherein the first portion overlaps a previous portion. The method comprises computing a first autocorrelation value for part of the first portion not overlapping the previous portion. The method further comprises retrieving a stored second autocorrelation value for part of the first portion overlapping the previous portion, the second autocorrelation value having been computed during estimation of a pitch period of the previous portion. The method further comprises forming a combined autocorrelation value using the first and second autocorrelation values, and selecting the estimated pitch period in dependence on the combined autocorrelation value.

Abstract: According to one embodiment, sound quality control processing for speech or music is performed by calculating various kinds of characteristic parameters to determine a speech signal and a music signal from an input audio signal and determining the input audio signal closer to the speech signal or music signal based on a score difference between a sum of scores provided to characteristic parameters indicating the speech signal and that of scores provided to characteristic parameters indicating the music signal.

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: Autocorrelation values are determined as a basis for an estimation of a pitch lag in a segment of an audio signal. A first considered delay range for the autocorrelation computations is divided into a first set of sections, and first autocorrelation values are determined for delays in a plurality of sections of this first set of sections. A second considered delay range for the autocorrelation computations is divided into a second set of sections such that sections of the first set and sections of the second set are overlapping. Second autocorrelation values are determined for delays in a plurality of sections of this second set of sections.

Abstract: A method for evaluating the processing delay of a speech signal contained in data packets received in a receiver terminal having a telephony module during a voice call to a terminal sending the data packets over a packet-switched network. The method includes the step of obtaining from the received data packets a stream of audio packets containing the speech signal. Within a predetermined decoding time, the stream of obtained audio packets is decoded and a first reconstituted speech signal is created. At least a portion of the speech reconstituted by the telephony module is duplicated to create a second reconstituted speech signal. The time difference between the first and the second reconstituted speech signals is determined. The processing delay of the speech signal in the receiver terminal is calculated from at least the determined time difference between the reconstituted first and second speech signals and the predetermined decoding time.

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: A pitch estimating method and apparatus in which mixture Gaussian distributions based on candidate pitches having high period estimating values are generated, a mixture Gaussian distribution having a high likelihood is selected and dynamic programming is executed so that the pitch of the speech signal can be accurately estimated.

Abstract: A method for transcoding a bit stream encoded according to a mixed-excitation linear prediction (MELP) standard to a bit stream encoded according to a linear predictive coding (LPC) standard, including: decoding a bit stream into a first set of vocoder parameters compatible with the MELP standard; transforming the first set of vocoder parameters into a second set of vocoder parameters compatible with the LPC standard without converting the first set of vocoder parameters to an analog or digital waveform representation; and encoding the second set of vocoder parameters into a bit stream compatible with the LPC vocoder standard.

Abstract: A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Abstract: In accordance with the present invention, computer implemented methods and systems are provided for representing and modeling the temporal structure of audio signals. In response to receiving a signal, a time-to-frequency domain transformation on at least a portion of the received signal to generate a frequency domain representation is performed. The time-to-frequency domain transformation converts the signal from a time domain representation to the frequency domain representation. A frequency domain linear prediction (FDLP) is performed on the frequency domain representation to estimate a temporal envelope of the frequency domain representation. Based on the temporal envelope, one or more speech features are generated.

Abstract: The present invention provides an innovative technique for rapidly and accurately determining whether two audio samples match, as well as being immune to various kinds of transformations, such as playback speed variation. The relationship between the two audio samples is characterized by first matching certain fingerprint objects derived from the respective samples. A set (230) of fingerprint objects (231,232), each occurring at a particular location (242), is generated for each audio sample (210). Each location (242) is determined in dependence upon the content of the respective audio sample (210) and each fingerprint object (232) characterizes one or more local features (222) at or near the respective particular location (242). A relative value is next determined for each pair of matched fingerprint objects. A histogram of the relative values is then generated. If a statistically significant peak is found, the two audio samples can be characterized as substantially matching.

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.