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704 - Data processing: speech signal processing, linguistics, language translation, and audio compression/decompression

704200000 - SPEECH SIGNAL PROCESSING

704201000 - For storage or transmission

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Class / Patent application numberDescriptionNumber of patent applications / Date published
704206000 Specialized information 174
Entries
DocumentTitleDate
20130030797EFFICIENT TEMPORAL ENVELOPE CODING APPROACH BY PREDICTION BETWEEN LOW BAND SIGNAL AND HIGH BAND SIGNAL - This invention provides a more efficient way to quantize temporal envelope shaping of high band signal by benefiting from energy relationship between low band signal and high band signal; if low band signal is well coded or it is coded with time domain codec such as CELP, temporal envelope shaping information of low band signal can be used to predict temporal envelope shaping of high band signal; the temporal envelope shaping prediction can bring significant saving of bits to precisely quantize temporal envelope shaping of high band signal. This prediction approach can be combined with other specific approach to further increase the efficiency and save mores bits.01-31-2013
20130030796AUDIO ENCODING APPARATUS AND AUDIO ENCODING METHOD - An audio encoding apparatus that allows a decoded signal exhibiting an excellent sound quality to be obtained on a decoding side. In the audio encoding apparatus (01-31-2013
20100161322ENCODING AND DECODING APPARATUSES FOR IMPROVING SOUND QUALITY OF G.711 CODEC - An encoding apparatus and a decoding apparatus for reducing the quantization error of a G.711 codec and improving sound quality are provided. The encoding apparatus includes a G.711 encoder which generates a G.711 bitstream by encoding an input audio signal; an enhancement-layer encoder which chooses one of a static bit allocation method and a dynamic bit allocation method that can produce less quantization error based on the input audio signal and the G.711 bitstream, and outputs an enhancement-layer bitstream including encoded additional mantissa information obtained by using the chosen bit allocation method; and a multiplexer which multiplexes the G.711 bitstream and the enhancement-layer bitstream. Therefore, it is possible to reduce the quantization error of a G.711 codec and improve sound quality.06-24-2010
20120173230SPEECH DECODING APPARATUS FOR PRODUCING AN EXCITATION SIGNAL AND A SYNTHESIS FILTER - A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.07-05-2012
20120245928GATEWAY APPARATUS, RELAY METHOD, PROGRAM, FEMTO SYSTEM - In order to avoid deterioration in sound quality caused by band shortage between an HNB-GW and an HNB, in a femto system 09-27-2012
20130085751VOICE COMMUNICATION SYSTEM ENCODING AND DECODING VOICE AND NON-VOICE INFORMATION - In a voice coding apparatus of a voice communication system, feature parameters of background noise in background noise sections of an input signal stream are extracted and background noise is encoded into a comfortable-noise code, and embedding positions where additional information is to be embedded are determined according to the values of the extracted feature parameters. Additional information is embedded into the embedding positions thus determined of the voice or comfortable-noise code, which will be transmitted to a voice decoding apparatus in the system. In the decoding apparatus, the transmitted code is separated into voice and background noise sections to be decoded. From the background noise sections, the values of the feature parameters are found out and used to reference a correspondence relationship table to determine the embedding positions where the additional information is embedded. The additional information is extracted at the embedding positions thus determined to be restored.04-04-2013
20100042408SYSTEM FOR BANDWIDTH EXTENSION OF NARROW-BAND SPEECH - 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 M02-18-2010
20090157394SYSTEM AND METHOD FOR FREQUENCY DOMAIN AUDIO SPEED UP OR SLOW DOWN, WHILE MAINTAINING PITCH - Presented herein are system(s) and method(s) for frequency domain audio speed up or slow down, while maintaining pitch. An encoded audio signal is received. Frames from the encoded audio signal are retrieved. The frames of the audio signal are transformed into a frequency domain, wherein each of said frames are associated with a plurality of initial phases, and a corresponding plurality of ending phases. The initial phases of at least one of the frames are replaced with the ending phases of another frame.06-18-2009
20120185243SPEECH FEATURE EXTRACTION APPARATUS, SPEECH FEATURE EXTRACTION METHOD, AND SPEECH FEATURE EXTRACTION PROGRAM - A speech feature extraction apparatus, speech feature extraction method, and speech feature extraction program. A speech feature extraction apparatus includes: first difference calculation module to: (i) receive, as an input, a spectrum of a speech signal segmented into frames for each frequency bin; and (ii) calculate a delta spectrum for each of the frame, where the delta spectrum is a difference of the spectrum within continuous frames for the frequency bin; and first normalization module to normalize the delta spectrum of the frame for the frequency bin by dividing the delta spectrum by a function of an average spectrum; where the average spectrum is an average of spectra through all frames that are overall speech for the frequency bin; and where an output of the first normalization module is defined as a first delta feature.07-19-2012
20090306973Sound Source Separation Apparatus and Sound Source Separation Method - A sound source separation apparatus, includes: a plurality of sound input means into which a plurality of mixed sound signals in which sound source signals from a plurality of sound sources superimpose each other are input; first sound source separating means for separating and extracting SIMO signals corresponding to at least one sound source signal from the plurality of mixed sound signals by means of a sound source separation process of a blind source separation system based on an independent component analysis method; intermediate processing executing means for obtaining a plurality of intermediately processed signals by carrying out a predetermined intermediate processing including one of a selection process and a synthesizing process to a plurality of specified signals which is at least a part of the SIMO signals, for each of frequency components divided into a plurality; and second sound source separating means for obtaining separation signals corresponding to the sound source signals by applying a binary masking process to the plurality of intermediately processed signals or a part of the SIMO signals and the plurality of intermediately processed signals.12-10-2009
20090271182COMPUTER-IMPLEMENTED METHODS AND SYSTEMS FOR MODELING AND RECOGNITION OF SPEECH - 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.10-29-2009
20120226495DEVICE AND METHOD FOR FILTERING OUT NOISE FROM SPEECH OF CALLER - A device and a method for filtering out noise from speech of caller are disclosed. The method is applied to the device, includes: inputting a speech sound of a caller; converting the speech sound to digital signals by an analyzing-to-digital converting unit; analyzing the digital signals to identify a pure speech of the caller and filtering out an extraneous noise thus obtaining pure speech signals of the caller; encoding the pure speech signals by a coder and decoder unit, and submitting the encoded speech signals to the receiver.09-06-2012
20130166286VOICE PROCESSING APPARATUS AND VOICE PROCESSING METHOD - A voice processing apparatus includes: a phase difference calculation unit which calculates for each frequency band a phase difference between first and second frequency signals obtained by applying a time-frequency transform to sounds captured by two voice input units; a detection unit which detects a frequency band for which the percentage of the phase difference falling within a first range that the phase difference can take for a specific sound source direction, the percentage being taken over a predetermined number of frames, does not satisfy a condition corresponding to a sound coming from the direction; a range setting unit which sets, for the detected frequency band, a second range by expanding the first range; and a signal correction unit which makes the amplitude of the first and second frequency signals larger when the phase difference falls within the second range than when the phase difference falls outside the second range.06-27-2013
20090012779Sound source separation apparatus and sound source separation method - A sound source separation apparatus includes: an SIMO-ICA process unit, separating and generating an SIMO signal by the BSS method based on the ICA method; a sound source direction estimation unit, estimating a sound source direction based on a separating matrix, computed by a learning calculation of the BSS method based on the ICA method; a beamformer process unit, performing, on each SIMO signal, a beamformer process of enhancing, according to each frequency bin, a sound component from each sound source direction; an intermediate process unit, performing an intermediate process that includes performing a selection process, etc., according to each frequency bin on signals other than a specific signal among the beamformer processed sound signals; and an untargeted signal component elimination unit, eliminating noise signal components by comparing for one signal in the specific SIMO signal, volumes of the specific beam former processed sound signal and the intermediate processed signal according to each frequency bin.01-08-2009
20120010880APPARATUS, METHOD AND COMPUTER PROGRAM FOR GENERATING A REPRESENTATION OF A BANDWIDTH-EXTENDED SIGNAL ON THE BASIS OF AN INPUT SIGNAL REPRESENTATION USING A COMBINATION OF A HARMONIC BANDWIDTH-EXTENSION AND A NON-HARMONIC BANDWIDTH-EXTENSION - An apparatus for generating a representation of a bandwidth-extended signal on the basis of an input signal representation includes a phase vocoder configured to obtain values of a spectral domain representation of a first patch of the bandwidth-extended signal on the basis of the input signal representation. The apparatus also includes a value copier configured to copy a set of values of the spectral domain representation of the first patch, which values are provided by the phase vocoder, to obtain a set of values of a spectral domain representation of a second patch, wherein the second patch is associated with higher frequencies than the first patch. The apparatus is configured to obtain the representation of the bandwidth-extended signal using the values of the spectral domain representation of the first patch and the values of the spectral domain representation of the second patch.01-12-2012
20100198588SIGNAL BANDWIDTH EXTENDING APPARATUS - A signal bandwidth extending apparatus including: a bandwidth extending section configured to extend a frequency bandwidth of a target signal, the target signal included in an input signal; a calculating section configured to calculate a degree of the target signal included in the input signal; and a controller configured to change a method of extending the frequency bandwidth by the bandwidth extending section according to a result of the calculating section.08-05-2010
20110295598SYSTEMS, METHODS, APPARATUS, AND COMPUTER PROGRAM PRODUCTS FOR WIDEBAND SPEECH CODING - Methods of audio coding are described in which an excitation signal for a first frequency band of the audio signal is used to calculate an excitation signal for a second frequency band of the audio signal that is separated from the first frequency band.12-01-2011
20100145685Regeneration of wideband speech - A method and system for regenerating wideband speech from narrowband speech. The method comprises: receiving samples of a narrowband speech signal in a first range of frequencies; modulating received samples of the narrowband speech signal with a modulation signal having a modulating frequency adapted to upshift each frequency in the first range of frequencies by an amount determined by the modulating frequency wherein the modulating frequency is selected to translate into a target band a selected frequency band within the first range of signals; filtering the modulated samples using a high pass filter to form a regenerated speech signal in the target band, wherein the lower limit of the high pass filter defines the lowermost frequency in the target band; and combining the narrow band speech signal with the regenerated speech signal in the target band to regenerate a wideband speech signal.06-10-2010
20100036656AUDIO SWITCHING DEVICE AND AUDIO SWITCHING METHOD - There is disclosed a speech switching device capable of improving quality of a decoded signal. In the device, a weighted addition unit (02-11-2010
20100030554SIGNAL SEPARATION REPRODUCTION DEVICE AND SIGNAL SEPARATION REPRODUCTION METHOD - A first matrix (W(k)) indicating frequency characteristics of a separation filter is calculated from input signals of a plurality of channels. A second matrix (Ws(k)) is calculated by using the restriction coefficients (C02-04-2010
20110125489METHOD AND APPARATUS TO REMOVE NOISE FROM AN INPUT SIGNAL IN A NOISY ENVIRONMENT, AND METHOD AND APPARATUS TO ENHANCE AN AUDIO SIGNAL IN A NOISY ENVIRONMENT - A method of removing noise includes detecting a frequency spectrum of a noise signal around the transmitting terminal, when an input signal which is a mixture of a voice signal and the noise signal is received, detecting a frequency spectrum of the input signal and an energy level of the voice signal, multiplying the frequency spectrum of the noise signal by a weight value that is determined based on the energy level of the voice signal to obtain a weighted noise frequency spectrum, and subtracting the weighted noise frequency spectrum from the frequency spectrum of the input signal.05-26-2011
20090048827METHOD AND SYSTEM FOR AUDIO FRAME ESTIMATION - The disclosed systems and methods relate to estimating an audio frame. Aspects of the present invention may improve audio quality at the client side when a section of voice data is corrupted or delayed during transmission. The present invention may be suitable for decoding in, for example, circuit switched and packet switched digital voice applications.02-19-2009
20090281795SPEECH ENCODING APPARATUS, SPEECH DECODING APPARATUS, SPEECH ENCODING METHOD, AND SPEECH DECODING METHOD - There is provided an audio encoding device for correcting the component having insufficient encoding capability in the core layer by an extended layer. In this device, a core layer encoding unit (11-12-2009
20080208572High-frequency bandwidth extension in the time domain - A system extends the high-frequency spectrum of a narrow band audio signal in the time domain. The system extends the harmonics of vowels by introducing a non linearity in a narrow band signal. Extended consonants are generated by a random-noise generator. The system differentiates the vowels from the consonants by exploiting predetermined features of a speech signal.08-28-2008
20110172993Single channel EVRCx, ISLP and G.711 transcoding in packet networks - An apparatus in one example has: a receiver configured to receive an input signal in a first encoding format, the input signal having an input payload; and a transcoder operatively coupled to the receiver, the transcoder structured to transcode in a single channel the first encoding format to a second encoding format, the transcoder configured to generate an output signal in the second encoding format based on the input signal, the output signal having an output payload; and wherein the transcoder is configured to switch between providing encrypted data in the output payload and non-encrypted data in the output payload.07-14-2011
20090281796ENHANCED CONVERSION OF WIDEBAND SIGNALS TO NARROWBAND SIGNALS - Wideband speech signals must be converted to narrowband speech signals if the transmission medium or the destination terminal is constructed with narrowband constraints. A typical wideband-to-narrowband conversion method is the elimination of frequencies above 3400 Hz using a low pass filter and a down sampler. However, this method produces a muffled speech sound since the resulting narrowband signal has a flat frequency response. Methods and apparatus are presented herein to enhance the acoustic quality of a wideband-to-narrowband converted signal. A bandwidth switching filter is used to emphasize a mid-range frequency portion of the wideband signal so that the resulting narrowband signal has a non-flat frequency spectrum.11-12-2009
20100138218Encoder, Decoder and Methods for Encoding and Decoding Data Segments Representing a Time-Domain Data Stream - An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.06-03-2010
20100145686INFORMATION PROCESSING APPARATUS CONVERTING VISUALLY-GENERATED INFORMATION INTO AURAL INFORMATION, AND INFORMATION PROCESSING METHOD THEREOF - In an information processing apparatus, the information of a webpage acquired by a page information reception unit is analyzed for a tag and the like by a page information analysis unit, and a character string is extracted under an extraction condition that is set in advance. Multiple character string groups are extracted so that multiple character strings are concurrently perceived in an aural manner. The extracted character strings are converted into respective audio signals by a text-to-sound conversion unit. The multiple audio signals thus generated are processed and synthesized by an audio processing unit based on the allocation pattern set by a frequency band allocation unit, the localization set by a localization allocation unit, and the difference in time at which the audio signals are output set by a time allocation unit. The output unit outputs the synthesized sounds.06-10-2010
20120046943APPARATUS AND METHOD FOR IMPROVING COMMUNICATION QUALITY IN MOBILE TERMINAL - An apparatus and a method for voice communication of a mobile terminal are provided. More particularly, an apparatus and a method for clearly receiving a counterpart user's voice signal in a mobile terminal positioned at a place where a noise occurs are provided. The apparatus includes an input unit, an extension signal generator, and an adder. The input unit receives a voice signal. The extension signal generator generates, based on a voice signal received via the input unit, a harmonics signal corresponding to a frequency band that represents a reaction sensitive to a sense of hearing. The adder merges the generated harmonics signal with the received voice signal.02-23-2012
20080270124METHOD AND APPARATUS FOR ENCODING AND DECODING AUDIO/SPEECH SIGNAL - Provided is a method of encoding an audio/speech signal, the method including determining a variable length of a frame, that is, a processing unit of an input signal in accordance with a position of an attack in the input signal; transforming each frame of the input signal to a frequency domain and dividing the frame into a plurality of sub frequency bands; and, if a signal of a sub frequency band is determined to be encoded in the frequency domain, encoding the signal of the sub frequency band in the frequency domain, and if the signal of the sub frequency band is determined to be encoded in a time domain, inverse transforming the signal of the sub frequency band to the time domain and encoding the inverse transformed signal in the time domain. According to the present invention, the audio/speech signal may be efficiently encoded by controlling time resolution and frequency resolution.10-30-2008
20120143599WARPED SPECTRAL AND FINE ESTIMATE AUDIO ENCODING - A warped spectral estimate of an original audio signal can be used to encode a representation of a fine estimate of the original signal. The representation of the warped spectral estimate and the representation of the fine estimate can be sent to a speech recognition system. The representation of the warped spectral estimate can be passed to a speech recognition engine, where it may be used for speech recognition. The representation of the warped spectral estimate can also be used along with the representation of the fine estimate to reconstruct a representation of the original audio signal.06-07-2012
20110208516INFORMATION PROCESSING APPARATUS AND OPERATION METHOD THEREOF - An information processing apparatus includes an acquisition unit configured to acquire a first sound recorded from a first recording apparatus and a second sound recorded from a second recording apparatus that is different from the first recording apparatus, a determination unit configured to determine a frequency band representing a voice by analyzing a frequency of the first sound, and a change unit configured to, from among frequency components representing the second sound, change a frequency component in the frequency band.08-25-2011
20090198489METHOD AND APPARATUS FOR FREQUENCY ENCODING, AND METHOD AND APPARATUS FOR FREQUENCY DECODING - Provided are a method and apparatus for encoding the frequency of a continuation sinusoidal signal and a method and apparatus for decoding the same. In the encoding method, a continuation sinusoidal signal successive to a sinusoidal signal in a previous section is extracted from a current section; a frequency of the continuation sinusoidal signal at the boundary between the current and previous sections is changed to a first frequency, based on representative frequencies of the continuation sinusoidal signal and at least one sinusoidal signal that belongs to a section adjacent to the current section and is successive to the continuation sinusoidal signal; and the first frequency is encoded.08-06-2009
20090240489Voice band expander and expansion method, and voice communication apparatus - A band-limited voice signal is processed to reduce its spectral envelope or harmonic structure, or both. The resulting reduced signal is moved into a frequency band above the upper limit frequency of the band-limited voice signal, and then combined with the band-limited voice signal to form a band expanded signal with improved quality and comprehensibility, free of unnatural high-frequency resonances and unnaturally strong high-frequency harmonics.09-24-2009
20090083030Method and apparatus for transmitting wideband speech signals - A method and an apparatus for transmitting a speech signal are provided. A speech signal transmitter includes a quadrature mirror filter, a base sub-band encoder, an enhancement sub-band encoder, and a network connector. The quadrature mirror filter receives a speech signal, divides the speech signal into an enhancement band speech signal and a base band speech signal, and outputs the enhancement band speech signal and the base band speech signal. The base sub-band encoder receives and encodes the base band speech signal. The enhancement sub-band encoder receives and encodes the enhancement band speech signal. The network connector multiplexes the encoded enhancement band speech signal and the encoded base band speech signal based on the kinds of networks over which speech signals are transmitted, and transmits the multiplexed signals to the networks. A speech signal is multiplexed and transmitted by various methods based on the kinds of networks. Thus, the speech signal can be efficiently transmitted.03-26-2009
20080262835Encoding Device, Decoding Device, and Method Thereof - There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units (10-23-2008
20110224976SPEECH INTELLIGIBILITY PREDICTOR AND APPLICATIONS THEREOF - The application relates to a method of providing a speech intelligibility predictor value for estimating an average listener's ability to understand of a target speech signal when said target speech signal is subject to a processing algorithm and/or is received in a noisy environment. The application further relates to a method of improving a listener's understanding of a target speech signal in a noisy environment and to corresponding device units. The object of the present application is to provide an alternative objective intelligibility measure, e.g. a measure that is suitable for use in a time-frequency environment. The invention may e.g. be used in audio processing systems, e.g. listening systems, e.g. hearing aid systems.09-15-2011
20090204394DECODING METHOD AND DEVICE - A decoding method and device are provided. The spectrum parameter of a current bad data frame is determined. Specifically, a number of continuous bad frames that occur currently is determined. A spectrum parameter of a good data frame before the current bad data frame is determined. And a constant mean value of a spectrum parameter is determined. Then, the spectrum parameter of the good data frame is adaptively shifted towards the constant mean value of the spectrum parameter according to the number of the continuous bad data frames to calculate and obtain spectrum parameter information of the current bad frame. When the continuous bad data frames occur, the relevance between the spectrum parameter of the nearest good frame and the spectrum parameter of the current bad frame is gradually reduced, so that more accurate spectrum parameter of the current bad data frame can be obtained, thereby obtaining a better speech quality under a same code rate and a same frame error rate.08-13-2009
20090254338SYSTEM AND METHOD FOR GENERATING A SEPARATED SIGNAL - The present invention relates to blind source separation. More specifically it relates to the blind source separation using frequency domain processes.10-08-2009
20100262422Device and method for improving communication through dichotic input of a speech signal - The device and method of the present invention improves electronic communication which have behavioral consequences, including for example, flight communication, two-way closed circuit communication such as for fire, police, miners, scuba divers and other heath and safety workers, and even for mobile communication which happens during activities such as cellular or mobile conversations during driving. Dichotic listening techniques are altered to enhance dyadic (involving two people) interactions with a partner. The speech of at least the first member of the dyad is filtered to isolate the component below 0.5 Khz, which will be input with a gain to the left ear of the second person (provided that they are right-handed), and thus their right cerebral hemispheres, and the component with a frequency above 0.5 Khz. will be input to their right ears, and thus their left cerebral hemispheres. The apparatus of the invention includes a communication source, which could include live and simultaneous broadcast, or pre-recorded communication. This constitutes the communication input which is directed to a filter to split off the speech fundamental frequency, i.e. the SFF. The post filtered communication signal, or “SFF augmented signal” is fed to a differentiation device which differentiates two signals, one with an enhanced SFF, and one without the enhancement subsequently, a delivery device delivers the now differentiated left and right signals to the appropriate ears.10-14-2010
20080215315METHODS AND APPRATUS FOR CHARACTERIZING MEDIA - Methods and apparatus for characterizing media are described. In one example, a method of characterizing media includes capturing a block of audio; converting at least a portion of the block of audio into a frequency domain representation including a plurality of complex-valued frequency components; defining a band of complex-valued frequency components for consideration; determining a decision metric using the band of complex-valued frequency components; and determining a signature bit based on a value of the decision metric. Other examples are shown and described.09-04-2008
20110112829APPARATUS AND METHOD FOR ENCODING AND DECODING OF INTEGRATED SPEECH AND AUDIO - Provided are an apparatus and a method for integrally encoding and decoding a speech signal and a audio signal. The encoding apparatus may include: an input signal analyzer to analyze a characteristic of an input signal; a first conversion encoder to convert the input signal to a frequency domain signal, and to encode the input signal when the input signal is a audio characteristic signal; a Linear Predictive Coding (LPC) encoder to perform LPC encoding of the input signal when the input signal is a speech characteristic signal; and a bitstream generator to generate a bitstream using an output signal of the first conversion encoder and an output signal of the LPC encoder.05-12-2011
20100211382Dereverberation Method, Apparatus, and Program for Dereverberation - A dereverberation device includes a reverberation estimation unit for estimating a later reflection component by using information on an impulse response from a signal source to an observation point, a noise estimation unit, and a mixing unit. As a result, it is possible to obtain a high-quality dereverberated signal with a small calculation amount even in a noisy environment.08-19-2010
20090112580Speech processing apparatus and method of speech processing - The speech processing apparatus configured to split a first speech waveform and a second speech waveform into a plurality of frequency bands respectively to generate a first band speech waveform and a second band speech waveform each being a component of each frequency band; determine an overlap-added position between the first band speech waveform and the second band speech waveform by the each frequency band so that a high cross correlation between the first band speech waveform and the second band speech waveform is obtained; and overlap-add the first band speech waveform and the second band speech waveform by the each frequency band on the basis of the overlap-added position and integrates overlap-added band speech waveforms in the plurality of frequency bands over all the plurality of frequency bands to generate a concatenated speech waveform.04-30-2009
20090112579SPEECH ENHANCEMENT THROUGH PARTIAL SPEECH RECONSTRUCTION - A system improves speech intelligibility by reconstructing speech segments. The system includes a low-frequency reconstruction controller programmed to select a predetermined portion of a time domain signal. The low-frequency reconstruction controller substantially blocks signals above and below the selected predetermined portion. A harmonic generator generates low-frequency harmonics in the time domain that lie within a frequency range controlled by a background noise modeler. A gain controller adjusts the low-frequency harmonics to substantially match the signal strength to the time domain original input signal.04-30-2009
20130138432SPEECH ENCODING/DECODING DEVICE - A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is transformed. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a band extension technique in the frequency domain represented by SBR.05-30-2013
20100057446ENCODING DEVICE AND ENCODING METHOD - Provided is an encoding device which can obtain a sound quality preferable for auditory sense even if the number of information bits is small. The encoding device includes a shape quantization unit (03-04-2010
20100063803Spectrum Harmonic/Noise Sharpness Control - A transmitted data that includes audio data and a transmitted spectral sharpness parameter representing a spectral harmonic/noise sharpness of a plurality of subbands are received. A measured spectral sharpness parameter is estimated from received audio data. The transmitted spectral sharpness parameter is compared with the measured spectral sharpness parameter. A main sharpness control parameter is formed for each of the decoded subbands. The main sharpness control parameter for each of the decoded subbands is analyzed. Ones of the decoded subbands are sharpened if the corresponding main sharpness control indicates that a corresponding subband is not sharp enough, wherein sharpened subbands are formed. Likewise, ones of the decoded subbands are flattened if the corresponding main sharpness control indicates that a corresponding subband is not flat enough, wherein flattened subbands are formed. An energy level of each sharpened subband and each flattened subband is normalized to keep an energy level of each sharpened and/or flattened subband substantially unchanged.03-11-2010
20110082691BROADCASTING SYSTEM INTERWORKING WITH ELECTRONIC DEVICES - Provided is a technology of controlling an electronic device using a broadcasting system. A control signal may be modulated to an audible frequency band and thereby be transmitted from a transmission apparatus to a reception apparatus using the broadcasting system. The reception apparatus may reproduce the control signal of the audible frequency band. A controller may perform a control operation based on the control signal.04-07-2011
20110040556METHOD AND APPARATUS FOR ENCODING AND DECODING RESIDUAL SIGNAL - A method and apparatus for encoding and decoding a residual signal are provided. The encoding method includes generating a residual signal indicating a difference between a multi-channel audio signal, and an audio signal downmixed from the multi-channel audio signal and then upmixed by using additional information from the downmixed audio signal; and performing a parametric encoding method on the residual signal. The decoding method includes decoding a sinusoidal component; restoring a sine wave by using the sinusoidal component; dividing the sine wave into a plurality of sub-bands in a frequency domain; transforming the plurality of sub-bands from the frequency domain into a time domain by applying a window to each of the plurality of sub-bands; and synthesizing the plurality of domain-transformed sub-bands to restore a residual signal.02-17-2011
20120203546ENCODING DEVICE, DECODING DEVICE AND METHODS THEREFOR - Disclosed is an encoding device, wherein the energy information of a given layer is efficiently encoded using a scalable encoding method in which the band to be encoded is selected in each layer, and the quality of decoded signals can be enhanced. An encoding device (08-09-2012
20080243492VOICE-SCRAMBLING-SIGNAL CREATION METHOD AND APPARATUS, AND COMPUTER-READABLE STORAGE MEDIUM THEREFOR - Original voice uttered in a first space is acquired via a microphone and a series of digital waveform data of the acquired original voice are obtained. The waveform data are sequentially segmented into plural frames and the waveform data of the individual frames are written into a memory. In parallel with writing, into the memory, of the waveform data, individual ones of the frames already written in the memory are sequentially or randomly selected and read out in a direction opposite to a direction the waveform data of the frame have been written so that a reverse-reproduced voice signal is generated. As the original voice is transmitted, as a leaked voice from the first space to a second space near the first space, a scrambling voice based on the reverse-reproduced voice signal is spatially mixed with the leaked voice in the second space.10-02-2008
20080270125METHOD AND APPARATUS FOR ENCODING AND DECODING HIGH FREQUENCY BAND - Provided is a method and apparatus for encoding or decoding a signal corresponding to a high frequency band in an audio signal. The method and apparatus for encoding a high frequency band detects and encodes frequency component(s) according to a pre-set criterion from a signal corresponding to a frequency band higher than a pre-set frequency and encodes energy value(s) of a signal to reconstruct band(s) in which the detected frequency component(s) are included. The method and apparatus for decoding a high frequency band decodes the signal by adjusting a signal to reconstruct a band in which important frequency component(s) are included by considering an energy value of the important frequency component(s). Accordingly, even though encoding or decoding is performed using a small number of bits, there is no degradation in sound quality of a signal corresponding to a high frequency band, and thus coding efficiency can be maximized.10-30-2008
20080255830Method and device for modifying an audio signal - A method of modifying acoustic characteristics of an original audio signal as a function of modification instructions relating at least to the fundamental frequency and the spectral envelope of the original signal. The method comprises a first modification operation applied to the original signal to deliver an intermediate audio signal, the first modification operation being intended to deform the spectral envelope of the original signal in application of said spectral envelope modification instruction; and a second modification operation applied to the intermediate signal to deliver a final audio signal, the second modification operation being intended to modify at least the fundamental frequency of the intermediate signal, in application of a modification factor that is determined so as to take account of the effects of the first modification operation on the fundamental frequency of the original audio signal, so that the fundamental frequency obtained for the final signal conforms to said instruction relating to fundamental frequency.10-16-2008
20110125490NOISE SUPPRESSOR AND VOICE DECODER - A processed component calculating unit 05-26-2011
20110178796Signal Classifying Method and Apparatus - A signal classifying method and apparatus are disclosed. The signal classifying method includes: obtaining a spectrum fluctuation parameter of a current signal frame determined as a foreground frame, and buffering the spectrum fluctuation parameter; obtaining a spectrum fluctuation variance of the current signal frame according to spectrum fluctuation parameters of all buffered signal frames, and buffering the spectrum fluctuation variance; and calculating a ratio of signal frames whose spectrum fluctuation variance is above or equal to a first threshold to all the buffered signal frames, and determining the current signal frame as a speech frame if the ratio is above or equal to a second threshold or determining the current signal frame as a music frame if the ratio is below the second threshold. In the embodiments of the present invention, the spectrum fluctuation variance of the signal is used as a parameter for classifying the signals, and a local statistical method is applied to decide the type of the signal. Therefore, the signals are classified with few parameters, simple logical relations and low complexity.07-21-2011
20110191101Apparatus and Method for Processing an Audio Signal for Speech Enhancement Using a Feature Extraction - An apparatus for processing an audio signal to obtain control information for a speech enhancement filter has a feature extractor for extracting at least one feature per frequency band of a plurality of frequency bands of a short-time spectral representation of a plurality of short-time spectral representations, where the at least one feature represents a spectral shape of the short-time spectral representation in the frequency band. The apparatus additionally has a feature combiner for combining the at least one feature for each frequency band using combination parameters to obtain the control information for the speech enhancement filter for a time portion of the audio signal. The feature combiner can use a neural network regression method, which is based on combination parameters determined in a training phase for the neural network.08-04-2011
20110137644Decoding speech signals - A method, terminal and program for processing a speech signal, in which the speech signal is received over a network from a transmitting device, wherein the frequency components in the received speech signal are limited to a predetermined frequency range and the received speech signal has been filtered using a transmitter frequency response over the predetermined frequency range. The received speech signal is decoded. The decoded speech signal is filtered using a receiver frequency response which is complementary to the transmitter frequency response over the predetermined frequency range to thereby reduce distortion in the speech signal introduced over the predetermined frequency range by using said transmitter frequency response.06-09-2011
20110153316Acoustic Perceptual Analysis and Synthesis System - The present invention relates to the field of speech recognition and synthesis, and more specifically to a novel non-phonemically based system for recognizing and synthesizing speech.06-23-2011
20100017199ENCODING DEVICE, DECODING DEVICE, AND METHOD THEREOF - Provided is a decoding device and others which can mitigate the spectrum energy discontinuity and improves the decoded signal quality even when a sub-band is subjected to a spectrum attenuation process in the band extension method. The device includes: a substitution unit (01-21-2010
20100017198ENCODING DEVICE, DECODING DEVICE, AND METHOD THEREOF - Disclosed is a decoding device and others capable of flexibly calculating high-band spectrum data with a high accuracy in accordance with an encoding band selected by an upper-node layer of the encoding side. In this device: a first layer decoding unit (01-21-2010
20100017197VOICE CODING DEVICE, VOICE DECODING DEVICE AND THEIR METHODS - It is an object to disclose a voice coding device, etc. in which the deterioration of a voice quality of a decoded signal can be reduced in the case that low frequency domain components of a spectrum are used for coding high frequency domain components and that no low frequency domain components exist. In this voice coding device, a frequency domain transform unit (01-21-2010
20110119055APPARATUS FOR ENCODING AND DECODING OF INTEGRATED SPEECH AND AUDIO - Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.05-19-2011
20090043567VARIABLE FRAME OFFSET CODING - The present invention relates to an improvement in frame based codecs and in particular to a en encoding/decoding method, an coder/decoder (codec) and a radio communication device. The signal provided at the output of the improved frame based codec comprises frames of regular duration though the start of a frame is time offset in relation to the end of the preceding frame. The time offset varies from frame to frame. The output signal from the improved codec has no fixed framing grid. Time offsets may be positive, in which case two successive frames are separated with a gap in which substitution signals are inserted, or negative in which case two successive frames are overlapping. As substitution signals an extrapolation of the signal in a preceding frame, an interpolation of signals from a preceding frame and a following frame or a directly coded signal may be used. Negative offset makes it possible to capture transients in the signal to be encoded. The present invention relates to an improvement in frame based codecs and in particular to a en encoding/decoding method, an coder/decoder (codec) and a radio communication device. The signal provided at the output of the improved frame based codec comprises frames of regular duration though the start of a frame is time offset in relation to the end of the preceding frame. The time offset varies from frame to frame. The output signal from the improved codec has no fixed framing grid. Time offsets may be positive, in which case two successive frames are separated with a gap in which substitution signals are inserted, or negative in which case two successive frames are overlapping. As substitution signals an extrapolation of the signal in a preceding frame, an interpolation of signals from a preceding frame and a following frame or a directly coded signal may be used. Negative offset makes it possible to capture transients in the signal to be encoded.02-12-2009
20120004906METHOD FOR SEPARATING SIGNAL PATHS AND USE FOR IMPROVING SPEECH USING ELECTRIC LARYNX - In order to improve the speech quality of an electric larynx (EL) speaker, the speech signal of which is digitized by suitable means, the following steps are carried out: a) dividing a single-channel speech signal into a series of frequency channels by transferring it from a time domain into a discrete frequence domain; b) filtering out the modulation frequency of the EL by way of a high-pass or notch filter, in each frequency channel; and c) back-transforming the filtered speech signal from the frequency domain into the time domain and combining it into a single-channel output signal.01-05-2012
20120209597ENCODING APPARATUS, DECODING APPARATUS AND METHODS THEREOF - Disclosed is an encoding apparatus that can efficiently encode a signal that is a broad or extra-broad band signal or the like, thereby improving the quality of a decoded signal. This encoding apparatus includes a band establishing unit (08-16-2012
20120046942TERMINAL TO PROVIDE USER INTERFACE AND METHOD - A terminal and method to determine surrounding circumstances using received sound signals and to automatically control various user interfaces according to the surrounding circumstances. The terminal divides the received sound signals into voice and non-voice signals, analyzes the divide sound signals based on frequencies and determines the circumstances based on the analyzed sound signals. The terminal may further control a user interface based on the determined surrounding circumstances.02-23-2012
20120116755APPARATUS FOR ENHANCING INTELLIGIBILITY OF SPEECH AND VOICE OUTPUT APPARATUS USING THE SAME - An apparatus for enhancing intelligibility of speech and a voice output apparatus using the same are provided. The apparatus detects a level of a voice frame of an input signal through an input envelope detection unit, provides the level to a cutoff frequency estimation unit, detects a level of a voice frame of an output signal through an output envelope detection unit, provides the level to the cutoff frequency estimation unit, determines a difference value between a level of an N-th voice frame that is received from the input envelope detection unit and a level of an (N−1)st voice frame that is received from the output envelope detection unit in the cutoff frequency estimation unit, calculates a cutoff frequency for amplifying a consonant component of the input signal with the difference value, and a shelving filter filters the input signal according to a cutoff frequency that is calculated by the cutoff frequency estimation unit and selectively amplifies a portion that is estimated as a consonant component of the input signal. Accordingly, the shelving filter dynamically changes a cutoff frequency according to more or less of a consonant component in the input signal and according to less being changed, amplifies a high frequency component corresponding to a consonant component of the input signal and attenuates a low frequency component and thus the consonant component is selectively emphasized, whereby speech intelligibility is enhanced.05-10-2012
20120116754NOISE SUPPRESSION IN A MEL-FILTERED SPECTRAL DOMAIN - Techniques are described herein that suppress noise in a Mel-filtered spectral domain. For example, a window may be applied to a representation of a speech signal in a time domain. The windowed representation in the time domain may be converted to a subsequent representation of the speech signal in the Mel-filtered spectral domain. A noise suppression operation may be performed with respect to the subsequent representation to provide noise-suppressed Mel coefficients.05-10-2012
20120016669APPARATUS AND METHOD FOR VOICE PROCESSING AND TELEPHONE APPARATUS - A voice processing apparatus includes a voice signal acquiring unit that acquires a voice signal converted to plural frequency bands from an input signal having a narrowed band; an expanding unit that generates based on a narrowband component of the voice signal acquired by the voice signal acquiring unit, an expansion band component expanding the band of the voice signal; a correcting unit that corrects the power of the expansion band component by a correction amount determined based on a noise component included in the voice signal acquired by the voice signal acquiring unit; and an output unit that outputs the voice signal of which the band has been expanded based on the expansion band component corrected by the correcting unit and based on the narrowband component of the voice signal acquired by the voice signal acquiring unit.01-19-2012
20110093261SYSTEM AND METHOD FOR VOICE RECOGNITION - Systems and methods are operable to associate each of a plurality of stored audio patterns with at least one of a plurality of digital tokens, identify a user based on user identification input, access a plurality of stored audio patterns associated with a user based on the user identification input, receive from a user at least one audio input from a custom language made up of custom language elements wherein the elements include at least one monosyllabic representation of a number, letter or word, select one of the plurality of stored audio patterns associated with the identified user, in the case that the audio input received from the identified user corresponds with one of the plurality of stored audio patterns, determine the digital token associated with the selected one of the plurality of stored audio patterns, and generate the output signal for use in a device based on the determined digital token.04-21-2011
20110106529APPARATUS AND METHOD FOR CONVERTING AN AUDIOSIGNAL INTO A PARAMETERIZED REPRESENTATION, APPARATUS AND METHOD FOR MODIFYING A PARAMETERIZED REPRESENTATION, APPARATUS AND METHOD FOR SYNTHESIZING A PARAMETERIZED REPRESENTATION OF AN AUDIO SIGNAL - Apparatus for converting an audio signal into a parameterized representation, has a signal analyzer for analyzing a portion of the audio signal to obtain an analysis result; a band pass estimator for estimating information of a plurality of band pass filters based on the analysis result, wherein the information on the plurality of band pass filters has information on a filter shape for the portion of the audio signal, wherein the band width of a band pass filter is different over an audio spectrum and depends on the center frequency of the band pass filter; a modulation estimator for estimating an amplitude modulation or a frequency modulation or a phase modulation for each band of the plurality of band pass filters for the portion of the audio signal using the information on the plurality of band pass filters; and an output interface for transmitting, storing or modifying information on the amplitude modulation, information on the frequency modulation or phase modulation or the information on the plurality of band pass filters for the portion of the audio signal.05-05-2011
20100250245METHOD AND APPARATUS FOR CODING OR DECODING WIDEBAND SPEECH - A wideband speech coding method comprising identifying whether an input speech signal is a narrowband signal or a wideband signal, and coding the input speech signal by controlling a predetermined parameter of a wideband speech coding process based on the identification result.09-30-2010
20120166186Dual-Band Speech Encoding - This document describes various techniques for dual-band speech encoding. In some embodiments, a first type of speech feature is received from a remote entity, an estimate of a second type of speech feature is determined based on the first type of speech feature, the estimate of the second type of speech feature is provided to a speech recognizer, speech-recognition results based on the estimate of the second type of speech feature are received from the speech recognizer, and the speech-recognition results are transmitted to the remote entity.06-28-2012
20100223052REGENERATION OF WIDEBAND SPEECH - A method of regenerating wideband speech from narrowband speech, the method comprising: receiving samples of a narrowband speech signal in a first range of frequencies; modulating received samples of the narrowband speech signal with a modulation signal having a modulating frequency adapted to upshift each frequency in the first range of frequencies by an amount determined by the modulating frequency wherein the modulating frequency is selected to translate into a target band a selected frequency band within the first range of signals; filtering the modulated samples using a target band filter to form a regenerated speech signal in the target band; and combining the narrow band speech signal with the regenerated speech signal in the target band to regenerate a wideband speech signal, the method comprising the step of controlling the modulated samples to lie in a second range of frequencies identified by determining a signal characteristic of frequencies in the first range of frequencies.09-02-2010
20120215527ENCODER APPARATUS, DECODER APPARATUS AND METHODS OF THESE - There is disclosed an encoder apparatus whereby, when a band expanding technique for encoding, based on the spectral data of a lower frequency portion, the spectral data of a higher frequency portion is applied to a lower layer in a hierarchical encoding/decoding system, an efficient encoding can be performed in an upper layer as well, thereby improving the decoded-signal quality. In an encoder apparatus (08-23-2012
20120215525METHOD AND APPARATUS FOR MIXED DIMENSIONALITY ENCODING AND DECODING - A method and apparatus for mixed dimensionality encoding and decoding are provided in embodiments of the present invention. The method includes: obtaining at least one variable collection through calculation according to a processed spectral coefficient, determining a processing dimension for a spectral coefficient to be processed, according to a relationship between the at least one variable collection and a corresponding threshold collection, and performing, according to a selected dimension, encoding or decoding under the dimension on the spectral coefficient to be processed. Through the preceding technical means, different processing dimensions are used for different spectral coefficients, improving the encoding and decoding efficiency.08-23-2012
20110178795TIME WARP ACTIVATION SIGNAL PROVIDER, AUDIO SIGNAL ENCODER, METHOD FOR PROVIDING A TIME WARP ACTIVATION SIGNAL, METHOD FOR ENCODING AN AUDIO SIGNAL AND COMPUTER PROGRAMS - An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.07-21-2011
20120136654Apparatus And Method For Cancelling Echo In Joint Time Domain And Frequency Domain - Disclosed in the present invention is a method for cancelling echo in joint time domain and frequency domain. The method includes: receiving an input receiver signal and an input transmitter signal; implementing echo cancellation on the received transmitter signal, based on the received receiver signal, by using a first echo canceller which is either a time domain echo canceller or a frequency domain echo canceller, to obtain a first echo-cancelled signal; implementing echo cancellation again on the first echo-cancelled signal, based on the received receiver signal, by using a second echo canceller which is the other one of the time domain echo canceller and the frequency domain echo canceller, to obtain a second echo-cancelled signal; wherein, the filter parameters of the second filter of the second echo canceller is updated based on the second echo-cancelled signal, and the first and second echo canceller respectively include the corresponding first and second filters. By using said method in the present invention, fast response to echo reflecting environment can be achieved with little residual echo, thus the effect of echo cancellation is entirely improved.05-31-2012
20110184731SIGNAL PROCESSING METHOD AND APPARATUS FOR AMPLIFYING SPEECH SIGNALS - A signal processing method is provided. The signal processing method includes extracting a first signal having a first frequency band from a sum signal of a left signal and a right signal, generating a second signal having a second frequency band by using the first signal, generating a third signal by using the first signal and the second signal, and applying a gain, generated by using a rate of a center signal included in the sum signal, to the third signal.07-28-2011
20100174533AUTOMATIC MEASUREMENT OF SPEECH FLUENCY - Techniques are described for automatically measuring fluency of a patient's speech based on prosodic characteristics thereof. The prosodic characteristics may include statistics regarding silent pauses, filled pauses, repetitions, or fundamental frequency of the patient's speech. The statistics may include a count, average number of occurrences, duration, average duration, frequency of occurrence, standard deviation, or other statistics. In one embodiment, a method includes receiving an audio sample that includes speech of a patient, analyzing the audio sample to identify prosodic characteristics of the speech of the patient, and automatically measuring fluency of the speech of the patient based on the prosodic characteristics. These techniques may present several advantages, such as objectively measuring fluency of a patient's speech without requiring a manual transcription or other manual intervention in the analysis process.07-08-2010
20100174532Speech encoding - A method, system and program for encoding and decoding speech according to a source-filter model whereby speech is modelled to comprise a source signal filtered by a time-varying filter. The method comprises: receiving a speech signal comprising successive frames, for each of a plurality of frames of the speech signal, deriving a first line spectral frequency vector for a first portion of the frame, and a second line spectral frequency vector for a second portion of the frame, and determining a transmit line spectral frequency vector and an interpolation factor based on the first and second line spectral frequency vectors, and on the transmit line spectral frequency vector for a preceding one of the frames.07-08-2010
20120215526ENCODER, DECODER AND METHODS THEREOF - An encoder whereby the bit efficiency of encoding can be improved, thereby improving the qualities of signals as decoded. In the encoder: a time-frequency converting unit (08-23-2012
20120221326Methods and Arrangements for Loudness and Sharpness Compensation in Audio Codecs - In a method of improving perceived loudness and sharpness of a reconstructed speech signal delimited by a predetermined bandwidth, performing the steps of providing (S08-30-2012
20120221325CONTEXT-BASED ARITHMETIC ENCODING APPARATUS AND METHOD AND CONTEXT-BASED ARITHMETIC DECODING APPARATUS AND METHOD - Disclosed are a context-based arithmetic encoding apparatus and method and a context-based arithmetic decoding apparatus and method. The context-based arithmetic decoding apparatus may determine a context of a current N-tuple to be decoded, determine a Most Significant Bit (MSB) context corresponding to an MSB symbol of the current N-tuple, and determine a probability model using the context of the N-tuple and the MSB context. Subsequently, the context-based arithmetic decoding apparatus may perform a decoding on an MSB based on the determined probability model, and perform a decoding on a Least Significant Bit (LSB) based on a bit depth of the LSB derived from a process of decoding on an escape code.08-30-2012
20080300866METHOD AND SYSTEM FOR CREATION AND USE OF A WIDEBAND VOCODER DATABASE FOR BANDWIDTH EXTENSION OF VOICE - The invention concerns a system (12-04-2008
20120239388EXCITATION SIGNAL BANDWIDTH EXTENSION - An apparatus for generating a high band extension of a low band excitation signal (e09-20-2012
20110131039COMPLEX ACOUSTIC RESONANCE SPEECH ANALYSIS SYSTEM - A method and apparatus are provided for determining an instantaneous frequency and an instantaneous bandwidth of a speech resonance of a speech signal. The method includes receiving a speech signal having a real component; filtering the speech signal so as to generate a plurality of filtered signals such that the real component and an imaginary component of the speech signal are reconstructed; and generating a first estimated frequency and a first estimated bandwidth of a speech resonance of the speech signal based on both a first filtered signal of the plurality of filtered signals and a single-lag delay of the first filtered signal.06-02-2011
20120278067VECTOR QUANTIZATION DEVICE, VOICE CODING DEVICE, VECTOR QUANTIZATION METHOD, AND VOICE CODING METHOD - Provided are a vector quantization device, a voice coding device, a vector quantization method, and a voice coding method which enable a reduction in the calculation amount of voice codec without deterioration of voice quality. In the vector quantization device, a first reference vector calculation unit (11-01-2012
20120095755AUDIO SIGNAL PROCESSING SYSTEM AND AUDIO SIGNAL PROCESSING METHOD - An audio signal processing system including a time-frequency conversion unit which converts an audio signal in time domain into frequency domain in frame units so as to calculate a frequency spectrum of the audio signal, a spectral change calculation unit which calculates an amount of change between a frequency spectrum of a first frame and a frequency spectrum of a second frame before the first frame based on the frequency spectrum of the first frame and the frequency spectrum of the second frame, and a judgment unit which judges the type of the noise which is included in the audio signal of the first frame in accordance with the amount of spectral change.04-19-2012
20110320194Decoder with embedded silence and background noise compression - There is provided a method for use by a speech encoder to encode an input speech signal. The method comprises receiving the input speech signal; determining whether the input speech signal includes an active speech signal or an inactive speech signal; low-pass filtering the inactive speech signal to generate a narrowband inactive speech signal; high-pass filtering the inactive speech signal to generate a high-band inactive speech signal; encoding the narrowband inactive speech signal using a narrowband inactive speech encoder to generate an encoded narrowband inactive speech; generating a low-to-high auxiliary signal by the narrowband inactive speech encoder based on the narrowband inactive speech signal; encoding the high-band inactive speech signal using a wideband inactive speech encoder to generate an encoded wideband inactive speech based on the low-to-high auxiliary signal from the narrowband inactive speech encoder; and transmitting the encoded narrowband inactive speech and the encoded wideband inactive speech.12-29-2011
20100198589AUDIO CODING APPARATUS, AUDIO DECODING APPARATUS, AUDIO CODING AND DECODING APPARATUS, AND TELECONFERENCING SYSTEM - The delay in a multi-channel audio coding apparatus and a multi-channel audio decoding apparatus is reduced. The audio coding apparatus includes: a downmix signal generating unit (08-05-2010
20100198587Bandwidth Extension Method and Apparatus for a Modified Discrete Cosine Transform Audio Coder - A method includes defining a transition band for a signal having a spectrum within a first frequency band, where the transition band is defined as a portion of the first frequency band, and is located near an adjacent frequency band that is adjacent to the first frequency band. The method analyzes the transition band to obtain a transition band spectral envelope and a transition band excitation spectrum; estimates an adjacent frequency band spectral envelope; generates an adjacent frequency band excitation spectrum by periodic repetition of at least a part of the transition band excitation spectrum with a repetition period determined by a pitch frequency of the signal; and combines the adjacent frequency band spectral envelope and the adjacent frequency band excitation spectrum to obtain an adjacent frequency band signal spectrum. A signal processing logic for performing the method is also disclosed.08-05-2010
20130013300BAND BROADENING APPARATUS AND METHOD - A band broadening apparatus includes a processor configured to analyze a fundamental frequency based on an input signal bandlimited to a first band, generate a signal that includes a second band different from the first band based on the input signal, control a frequency response of the second band based on the fundamental frequency, reflect the frequency response of the second band on the signal that includes the second band and generate a frequency-response-adjusted signal that includes the second band, and synthesize the input signal and the frequency-response-adjusted signal.01-10-2013
20130024190SYSTEMS AND METHODS FOR SPEECH PROCESSING - Systems and methods described herein modify audio content on an electronic device. Embodiments can be configured to detect a mode of the electronic device to determine whether the device is in a telephone mode; receive a speech signal from a speech source while the device is in the telephone mode; and process the speech signal to improve the perceived quality of the speech at a recipient when the electronic device is in a telephone mode; wherein processing the speech signal to improve the perceived quality of the speech comprises, decreasing the signal level of audio content outside of a determined frequency band relative to the signal level of the audio content within the determined frequency band; and wherein the determined frequency band is a frequency band associated a vocal range of the anticipated speech content.01-24-2013
20130024191AUDIO COMMUNICATION DEVICE, METHOD FOR OUTPUTTING AN AUDIO SIGNAL, AND COMMUNICATION SYSTEM - An audio communication device comprises an input connectable to a narrowband audio signal source. The input can receive a narrowband audio signal having a first bandwidth. An extraction unit is connected to the input and arranged to extract a plurality of narrowband parameters from the narrowband audio signal. An extrapolation unit is connected to receive the plurality of narrowband parameters and arranged to generate a plurality of wideband parameters from the plurality of narrowband parameters. The extrapolation unit comprises one or more adaptive neuro-fuzzy inference system modules. The device further comprises a synthesis unit connected to receive the plurality of wideband parameters and arranged to generate, using the wideband parameters, a synthesized wideband audio signal having a second bandwidth wider than the first bandwidth. And the device comprises an output connectable to an acoustic transducer arranged to output for humans perceptible acoustic signals, for providing said synthesized wideband audio signal to the acoustic transducer.01-24-2013
20130179158Speech Feature Extraction Apparatus and Speech Feature Extraction Method - According to one embodiment, a speech feature extraction apparatus includes an extraction unit and a calculation unit. The extraction unit extracts speech segments over a predetermined period at intervals of a unit time from either an input speech signal or a plurality of subband input speech signals obtained by extracting signal components of a plurality of frequency bands from the input speech signal, to generate either a unit speech signal or a plurality of subband unit speech signals. The calculation unit calculates either each average time of the unit speech signal in each of the plurality of frequency bands or each average time of each of the plurality of subband unit speech signals to obtain a speech feature.07-11-2013
20130124199PULSE ENCODING AND DECODING METHOD AND PULSE CODEC - In a pulse encoding and decoding method and a pulse codec, more than two tracks are jointly encoded, so that free codebook space in the situation of single track encoding can be combined during joint encoding to become code bits that may be saved. Furthermore, a pulse that is on each track and required to be encoded is combined according to positions, and the number of positions having pulses, distribution of the positions that have pulses on the track, and the number of pulses on each position that has a pulse are encoded separately, so as to avoid separate encoding performed on multiple pulses of a same position, thereby further saving code bits.05-16-2013
20110313758Method and Arrangement for Processing of Speech Quality Estimate - Method and arrangement for processing of a speech quality estimate, which involve adaption of a speech quality estimate based on information related to the bandwidth of a reference signal used when determining said speech quality estimate, such that the adapted speech quality estimate is independent of the bandwidth of the reference signal. The method and arrangement enable objective speech quality measurements or assessments to be performed on a unified bandwidth scale, independent of the bandwidth of a reference signal, which allows e.g. a more relevant comparison of communication systems and/or equipment, such as e.g. codecs.12-22-2011
20130151244HARMONICITY-BASED SINGLE-CHANNEL SPEECH QUALITY ESTIMATION - Speech quality estimation technique embodiments are described which generally involve estimating the human speech quality of an audio frame in a single-channel audio signal. A representation of a harmonic component of the frame is synthesized and used to compute a non-harmonic component of the frame. The synthesized harmonic component representation and the non-harmonic component are then used to compute a harmonic to non-harmonic ratio (HnHR). This HnHR is indicative of the quality of a user's speech and is designated as an estimate of the speech quality of the frame. In one implementation, the HnHR is used to establish a minimum speech quality threshold below which the quality of the user's speech is considered unacceptable. Feedback to the user is then provided based on whether the HnHR falls below the threshold.06-13-2013
20110307248ENCODER, DECODER, AND METHOD THEREFOR - Provided is an encoder which can effectively encode/decode spectrum data of a broad frequency signal in a high frequency range, can dramatically reduce the number of the arithmetic operations to be performed, and can improve the quality of the decoded signal. The encoder comprises a first layer coding unit (12-15-2011
20130103394DEVICE AND METHOD FOR EFFICIENTLY ENCODING QUANTIZATION PARAMETERS OF SPECTRAL COEFFICIENT CODING - This invention introduces apparatus and methods to efficiently encode the quantization parameters of split multi-rate lattice vector quantization. In this invention, by doing spectral analysis on the split multi-rate vector quantized spectrum, the spectrum is split to null vectors region and non-null vectors region. For the null vectors region, instead of transmitting series of indication for null vectors, an indication of null vectors region and the quantized value of index of the ending vector in the null vectors region (or the number of the null vectors in the null vectors region) are transmitted. The indication of null vectors region can be designed in many ways, the only requirement is the indication should be distinguishable in the decoder side. The ending index or the number of null vectors can be quantized by an adaptively designed codebook. By applying of the invented method, some bits can be saved from the codebook indications.04-25-2013

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