Entries |
Document | Title | Date |
20080208572 | High-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 |
20080215315 | METHODS 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 |
20080243492 | VOICE-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 |
20080255830 | Method 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 |
20080262835 | Encoding 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 |
20080270124 | METHOD 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 |
20080270125 | METHOD 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 |
20080300866 | METHOD AND SYSTEM FOR CREATION AND USE OF A WIDEBAND VOCODER DATABASE FOR BANDWIDTH EXTENSION OF VOICE - The invention concerns a system ( | 12-04-2008 |
20090012779 | Sound 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 |
20090043567 | VARIABLE 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 |
20090048827 | METHOD 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 |
20090083030 | Method 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 |
20090112579 | SPEECH 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 |
20090112580 | Speech 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 |
20090157394 | SYSTEM 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 |
20090198489 | METHOD 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 |
20090204394 | DECODING 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 |
20090240489 | Voice 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 |
20090254338 | SYSTEM 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 |
20090271182 | COMPUTER-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 |
20090281795 | SPEECH 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 |
20090281796 | ENHANCED 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 |
20090306973 | Sound 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 |
20100017197 | VOICE 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 |
20100017198 | ENCODING 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 |
20100017199 | ENCODING 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 |
20100030554 | SIGNAL 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 (C | 02-04-2010 |
20100036656 | AUDIO 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 |
20100042408 | SYSTEM 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 M | 02-18-2010 |
20100057446 | ENCODING 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 |
20100063803 | Spectrum 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 |
20100138218 | Encoder, 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 |
20100145685 | Regeneration 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 |
20100145686 | INFORMATION 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 |
20100161322 | ENCODING 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 |
20100174532 | Speech 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 |
20100174533 | AUTOMATIC 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 |
20100198587 | Bandwidth 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 |
20100198588 | SIGNAL 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 |
20100198589 | AUDIO 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 |
20100211382 | Dereverberation 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 |
20100223052 | REGENERATION 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 |
20100250245 | METHOD 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 |
20100262422 | Device 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 |
20110040556 | METHOD 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 |
20110082691 | BROADCASTING 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 |
20110093261 | SYSTEM 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 |
20110106529 | APPARATUS 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 |
20110112829 | APPARATUS 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 |
20110119055 | APPARATUS 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 |
20110125489 | METHOD 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 |
20110125490 | NOISE SUPPRESSOR AND VOICE DECODER - A processed component calculating unit | 05-26-2011 |
20110131039 | COMPLEX 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 |
20110137644 | Decoding 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 |
20110153316 | Acoustic 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 |
20110172993 | Single 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 |
20110178795 | TIME 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 |
20110178796 | Signal 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 |
20110184731 | SIGNAL 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 |
20110191101 | Apparatus 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 |
20110208516 | INFORMATION 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 |
20110224976 | SPEECH 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 |
20110295598 | SYSTEMS, 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 |
20110307248 | ENCODER, 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 |
20110313758 | Method 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 |
20110320194 | Decoder 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 |
20120004906 | METHOD 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 |
20120010880 | APPARATUS, 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 |
20120016669 | APPARATUS 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 |
20120046942 | TERMINAL 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 |
20120046943 | APPARATUS 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 |
20120095755 | AUDIO 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 |
20120116754 | NOISE 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 |
20120116755 | APPARATUS 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 |
20120136654 | Apparatus 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 |
20120143599 | WARPED 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 |
20120166186 | Dual-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 |
20120173230 | SPEECH 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 |
20120185243 | SPEECH 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 |
20120203546 | ENCODING 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 |
20120209597 | ENCODING 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 |
20120215525 | METHOD 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 |
20120215526 | ENCODER, 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 |
20120215527 | ENCODER 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 |
20120221325 | CONTEXT-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 |
20120221326 | Methods 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 (S | 08-30-2012 |
20120226495 | DEVICE 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 |
20120239388 | EXCITATION SIGNAL BANDWIDTH EXTENSION - An apparatus for generating a high band extension of a low band excitation signal (e | 09-20-2012 |
20120245928 | GATEWAY 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 |
20120278067 | VECTOR 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 |
20130013300 | BAND 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 |
20130024190 | SYSTEMS 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 |
20130024191 | AUDIO 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 |
20130030796 | AUDIO 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 |
20130030797 | EFFICIENT 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 |
20130085751 | VOICE 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 |
20130103394 | DEVICE 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 |
20130110506 | Audio Encoder and Decoder and Methods for Encoding and Decoding an Audio Signal | 05-02-2013 |
20130124199 | PULSE 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 |
20130138432 | SPEECH 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 |
20130151244 | HARMONICITY-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 |
20130166286 | VOICE 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 |
20130179158 | Speech 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 |
20130231923 | Voice Signal Enhancement - Implementations include systems, methods and/or devices operable to enhance the intelligibility of a target speech signal by targeted voice model based processing of a noisy audible signal. In some implementations, an amplitude-independent voice proximity function voice model is used to attenuate signal components of a noisy audible signal that are unlikely to be associated with the target speech signal and/or accentuate the target speech signal. In some implementations, the target speech signal is identified as a near-field signal, which is detected by identifying a prominent train of glottal pulses in the noisy audible signal. Subsequently, in some implementations systems, methods and/or devices perform a form of computational auditory scene analysis by converting the noisy audible signal into a set of narrowband time-frequency units, and selectively accentuating the time-frequency units associated with the target speech signal and deemphasizing others using information derived from the identification of the glottal pulse train. | 09-05-2013 |
20130246055 | System and Method for Post Excitation Enhancement for Low Bit Rate Speech Coding - In accordance with an embodiment, a method of decoding an audio/speech signal includes decoding an excitation signal based on an incoming audio/speech information, determining a stability of a high frequency portion of the excitation signal, smoothing an energy of the high frequency portion of the excitation signal based on the stability of the high frequency portion of the excitation signal, and producing an audio signal based on smoothing the high frequency portion of the excitation signal. | 09-19-2013 |
20130246056 | SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD AND SIGNAL PROCESSING PROGRAM - The purpose of the present invention is to achieve a high-quality signal processing performance. A signal processing device provided with a suppression unit for suppressing a second signal by processing a mixed signal in which a first signal and the second signal are present. The signal processing device is provided with an analysis unit for analyzing, per frequency component, the importance of the first signal contained in the mixed signal, and an inhibition unit for inhibiting the suppression of the second signal of a frequency component having a high importance over a frequency component having a low importance on the basis of the analysis result of the analysis means. | 09-19-2013 |
20130275126 | METHODS AND SYSTEMS TO MODIFY A SPEECH SIGNAL WHILE PRESERVING AURAL DISTINCTIONS BETWEEN SPEECH SOUNDS - Methods and systems to modify an audio signal, such as a speech signal, while preserving aural distinctions between sounds of the audio signal. Methods and systems disclosed herein may be implemented with respect to cellular telephones and portable music devices, such as to reduce and/or prevent hearing loss due. Audio modification may include sweeping through one or more frequency ranges of a speech signal and modifying frequencies within the frequency range as a function of a pattern, such as an infinite rising wave pattern. Speech modification may include adding and subtracting one or more equalization curves to and from the speech signal to vary amplitudes substantially without lateral movement in pitch. | 10-17-2013 |
20130275127 | APPARATUS AND METHOD FOR CONCEALING FRAME ERASURE AND VOICE DECODING APPARATUS AND METHOD USING THE SAME - An apparatus and method for concealing frame erasure and a voice decoding apparatus and method using the same. The frame erasure concealment apparatus includes: a parameter extraction unit determining whether there is an erased frame in a voice packet, and extracting an excitement signal parameter and a line spectrum pair parameter of a previous good frame; and an erasure frame concealment unit, if there is an erased frame, restoring the excitement signal and line spectrum pair parameter of the erased frame by using a regression analysis from the excitement signal and line spectrum pair parameter of the previous good frame. According to the method and apparatus, by predicting and restoring the parameter of the erased frame through the regression analysis, the quality of the restored voice signal can be enhanced and the algorithm can be simplified. | 10-17-2013 |
20130297298 | SOURCE SEPARATION USING INDEPENDENT COMPONENT ANALYSIS WITH MIXED MULTI-VARIATE PROBABILITY DENSITY FUNCTION - Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals by way of independent component analysis. Source separation described herein involves mixed multivariate probability density functions that are mixtures of component density functions having different parameters corresponding to frequency components of different sources, different time segments, or some combination thereof. | 11-07-2013 |
20130304458 | BANDWIDTH DEPENDENT AUDIO QUALITY ADJUSTMENT - A system includes a controller that provides an output control signal based on two or more control inputs. The controller determines an indication of radio frequency (RF) bandwidth availability based on a given one of the control inputs. The output control signal can correspond to the RF bandwidth availability and status of at least one other wireless condition affecting RF bandwidth. An audio quality adjuster can adjust quality parameters used to encode an audio stream for one or more audio sinks based on the output control signal. | 11-14-2013 |
20130332149 | METHOD AND AN APPARATUS FOR PROCESSING AN AUDIO SIGNAL - An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving, by an audio processing apparatus, an audio signal including a first data of a first block encoded with rectangular coding scheme and a second data of a second block encoded with non-rectangular coding scheme; receiving a compensation signal corresponding to the second block; estimating a prediction of an aliasing part using the first data; and, obtaining a reconstructed signal for the second block based on the second data, the compensation signal and the prediction of aliasing part. | 12-12-2013 |
20130332150 | ENCODING DEVICE, DECODING DEVICE, AND METHOD THEREOF FOR SPECIFYING A BAND OF A GREAT ERROR - Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. A first position identifier uses a first layer error conversion coefficient indicating an error of a decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band. A second position identifier searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identifier and generates second position information indicating the identified target frequency band. An encoder encodes a first layer decoding error conversion coefficient contained in the target frequency band. | 12-12-2013 |
20140019125 | LOW BAND BANDWIDTH EXTENDED - Apparatus comprising: an input amplitude and phase calculator configured to determine at least one amplitude value and phase value dependent on a first audio signal; a synthesis amplitude calculator configured to synthesize a further amplitude value associated with each amplitude value dependent on a determined harmonic shaping function; a synthesis phase calculator configured to synthesize a further phase value associated with each phase value; and a signal synthesizer configured to generate a bandwidth extension signal dependent the further amplitude value and the further phase values. | 01-16-2014 |
20140067382 | COMPUTING DEVICE AND METHOD FOR DETECTING EVENT IN MONITORING AREA - In a method for detecting an event occurred in a monitoring area, the method defines a reference characteristic parameter for a specific event, and stores the reference characteristic parameter in a storage device of the computing device. The method further obtains a voice stream of the specific event from an IP camera through a wireless network in real time, and extracts a characteristic parameter from the voice stream using a predefined algorithm. The method compares the extracted characteristic parameter with the reference characteristic parameter, and determines that the specific event occurs in the monitoring area if the extracted characteristic parameter accords with the reference characteristic parameter. | 03-06-2014 |
20140067383 | ADJUSTMENT APPARATUS AND METHOD - A disclosed adjustment apparatus includes: a calculation unit that calculates a ratio between a first frequency characteristic in a first frequency bandwidth of voice signals and a second frequency characteristic in a second frequency bandwidth of the voice signals, which is higher than the first frequency bandwidth, and calculates an adjustment amount for adjusting at least a portion of a frequency characteristic of the voice signals so that the calculated ratio approaches a predetermined reference, when the calculated ratio does not satisfy the predetermined reference; and a modification unit that modifies at least the portion of the frequency characteristic of the voice signals according to the adjustment amount. | 03-06-2014 |
20140067384 | METHOD AND APPARATUS FOR CANCELING VOCAL SIGNAL FROM AUDIO SIGNAL - Provided is a method of canceling a vocal signal, wherein the method includes obtaining a difference signal between two audio signals; and smoothing the frequency of the difference signal. Also provided is a device for canceling a vocal signal, the device including a subtracter which obtains a difference signal between two audio signals; and a frequency smoothing unit which smoothes a frequency of the difference signal. | 03-06-2014 |
20140081628 | REPRODUCE A VOICE FOR A SPEAKER BASED ON VOCAL TRACT SENSING USING ULTRA WIDE BAND RADAR - Examples are disclosed for reproducing a voice for a speaker based on vocal tract sensing using ultra wide band (UWB) radar. These examples may include sensing a vocal tract of the speaker during non-sounded speech communication and mapping information associated with the sensed vocal tract to a voice model to generate a simulation of the vocal tract during sounded speech communication. The examples may also include reproducing a voice for the speaker based on the simulation. | 03-20-2014 |
20140095154 | VOICE TRANSMITTING DEVICE, VOICE TRANSMITTING METHOD, VOICE RECEIVING DEVICE, AND VOICE RECEIVING METHOD - There is provided a voice transmitting device, including a band limiting unit that performs band limitation on an input time-series signal, a coding unit that encodes a time-series signal output from the band limiting unit, a transmitting unit that transmits a code string output from the coding unit, and a control unit that controls a band limitation operation in the band limiting unit. | 04-03-2014 |
20140108007 | METHOD AND SYSTEM FOR LOW BIT RATE VOICE ENCODING AND DECODING APPLICABLE FOR ANY REDUCED BANDWIDTH REQUIREMENTS INCLUDING WIRELESS - A voice encoder/decoder (vocoder) may provide receiving a voice sample and generating zero crossings of the voice sample in response to voice excitation in a first formant and creating a corresponding output signal. Additional operations may include dividing the output signal by two, and sampling the output signal at a predefined frequency such that a resulting combination uses half of a bit rate for an excitation and a remainder for short term spectrum analysis. | 04-17-2014 |
20140108008 | METHOD 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. | 04-17-2014 |
20140114652 | AUDIO CODING DEVICE, AUDIO CODING METHOD, AND AUDIO CODING AND DECODING SYSTEM - An audio coding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute, calculating a frequency spectrum characteristic of an input signal; defining a scale factor that is used to quantize a frequency spectrum converted from the input signal, based on the frequency spectrum characteristic, for each of a plurality of bands; and quantizing the frequency spectrum based on the scale factor. | 04-24-2014 |
20140122064 | SIGNAL PROCESSING DEVICE AND METHOD, AND PROGRAM - There is provided a signal processing device including a feature amount extraction unit configured to extract, from a frequency-domain signal obtained by frequency conversion on a voice signal, a feature amount of the frequency-domain signal, and a determination unit configured to determine, based on the extracted feature amount, presence or absence of noise in the voice signal within a predetermined section. The feature amount is composed of a plurality of elements. The plurality of elements contain an element defined based on a correlation value between a feature amount waveform which is a waveform according to the frequency-domain signal in the voice signal within the predetermined section and a feature amount waveform within another section sequential in time to the predetermined section. | 05-01-2014 |
20140122065 | VOICE CODING DEVICE, VOICE DECODING DEVICE, VOICE CODING METHOD AND VOICE DECODING METHOD - A voice coding device capable of preventing overall quality degradation even when the bit rate for coding is lowered. The voice coding device codes a wide band signal in a first layer, and codes an extended band signal whose frequency band is located in higher frequency than the wide band signal in an extended band layer. An adaptive band selection unit ( | 05-01-2014 |
20140122066 | PULSE 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. | 05-01-2014 |
20140163972 | SPEECH 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 shaped. 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 bandwidth extension technique in the frequency domain represented by SBR. | 06-12-2014 |
20140180681 | SPEECH SYNTHESIS APPARATUS AND METHOD - A waveform memory that stores a plurality of speech unit waveforms corresponding to respective speech units, wherein an address order of the speech unit waveforms is determined by a sort order of speech units included in a speech unit sequence corresponding to a phoneme sequence of training data, and the speech units included in the speech unit sequence are selected so as to synthesize a speech of the phone sequence. | 06-26-2014 |
20140200881 | NOISE REDUCTION DEVICES AND NOISE REDUCTION METHODS - A noise reduction device may be provided. The noise reduction device may include: an input configured to receive an input signal including a representation in a frequency domain of an audio signal, wherein the representation includes a plurality of time frames and a plurality of coefficients for each time frame; a noise detection circuit configured to determine a first indicator being indicative of a bandwidth of a coefficient over at least two time; a noise reduction circuit configured to reduce based on the first indicator a noise component in the audio signal; and an output configured to output an output signal including a representation in the frequency domain of the audio signal with the reduced noise component. | 07-17-2014 |
20140200882 | METHOD AND APPARATUS TO TRANSMIT DATA THROUGH TONES - Aspects of the disclosure provide a method for transmitting data. The method includes encoding data, by a first device, symbol-by-symbol into a first electrical signal of frequencies in a specific range, mixing, by the first device, the first electrical signal with a second electrical signal corresponding to captured voices, and transmitting, by the first device, the mixed electrical signal to a second device via a channel being configured for transmitting the captured voices. | 07-17-2014 |
20140200883 | METHOD AND DEVICE FOR SPECTRAL EXPANSION FOR AN AUDIO SIGNAL - A method and device for automatically increasing the spectral bandwidth of an audio signal including generating a “mapping” (or “prediction”) matrix based on the analysis of a reference wideband signal and a reference narrowband signal, the mapping matrix being a transformation matrix to predict high frequency energy from a low frequency energy envelope, generating an energy envelope analysis of an input narrowband audio signal, generating a resynthesized noise signal by processing a random noise signal with the mapping matrix and the envelope analysis, high-pass filtering the resynthesized noise signal, and summing the high-pass filtered resynthesized noise signal with the input narrowband audio signal. Other embodiments are disclosed. | 07-17-2014 |
20140236582 | LOW POWER VOICE DETECTION - Methods of enabling voice processing with minimal power consumption includes recording time-domain audio signal at a first clock frequency and a first voltage, and performing Fast Fourier Transform (FFT) operations on the time-domain audio signal at a second clock frequency to generate frequency-domain audio signal. The frequency domain audio signal may be enhanced to obtain better signal to noise ratio, through one or multiple filtering and enhancing techniques. The enhanced audio signal may be used to generate the total signal energy and estimate the background noise energy. Decision logic may determine from the signal energy and the background noise, the presence or absence of the human voice. The first clock frequency may be different from the second clock frequency. | 08-21-2014 |
20140236583 | SYSTEMS AND METHODS FOR DETERMINING AN INTERPOLATION FACTOR SET - A method for determining an interpolation factor set by an electronic device is described. The method includes determining a value based on a current frame property and a previous frame property. The method also includes determining whether the value is outside of a range. The method further includes determining an interpolation factor set based on the value and a prediction mode indicator if the value is outside of the range. The method additionally includes synthesizing a speech signal. | 08-21-2014 |
20140249805 | Variable-Resolution Processing of Frame-Based Data - Provided are systems, methods and techniques for processing frame-based data. A frame of data, an indication that a transient occurs within the frame, and a location of the transient within the frame are obtained. Based on the indication of the transient, a block size is set for the frame, thereby effectively defining a plurality of equal-sized blocks within the frame. In addition, different window functions are selected for different ones of the plurality of equal-sized blocks based on the location of the transient, and the frame of data is processed by applying the selected window functions. | 09-04-2014 |
20140257798 | CONVERSION OF LINEAR PREDICTIVE COEFFICIENTS USING AUTO-REGRESSIVE EXTENSION OF CORRELATION COEFFICIENTS IN SUB-BAND AUDIO CODECS - Disclosed are systems and methods for the efficient conversion of linear predictive coefficients. This method is usable, for example, in the conversion of full band linear predictive coding (“LPC”) coefficients to sub-band LPCs of a sub-band speech codec. The sub-bands may or may not be down-sampled. In an embodiment, the LPC coefficients of the sub-bands are obtained from the correlation coefficients, which are in turn obtained by filtering the auto-regressive extended auto-correlation coefficients of the full band LPCs. The method also allows the generation of an LPC approximation of a pole-zero weighted synthesis filter. | 09-11-2014 |
20140278380 | Spectral and Spatial Modification of Noise Captured During Teleconferencing - In some embodiments, a method for modifying noise captured at endpoints of a teleconferencing system, including steps of capturing noise at each endpoint, and modifying the captured noise to generate modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set. In other embodiments, a teleconferencing method including steps of: at endpoints of a teleconferencing system, determining audio frames indicative of audio captured at each endpoint, each of a subset of the frames indicative of noise but not a significant level of speech; and at each endpoint, generating modified frames indicative of modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set, and generating encoded audio including by encoding the modified frames. Other aspects are systems configured to perform any embodiment of the method. | 09-18-2014 |
20140303967 | METHOD AND DEVICE FOR QUANTIZING VOICE SIGNALS IN A BAND-SELECTIVE MANNER - The present invention relates to a method and device for quantizing voice signals in a band-selective manner. A voice decoding method may include inversely quantizing voice parameter information produced from a selectively quantized voice band and performing inverse transform on the basis of the inversely quantized voice parameter information. Thus, according to the present invention, coding/decoding efficiency in voice coding/decoding may be increased by selectively coding/decoding important information. | 10-09-2014 |
20140337018 | METHOD AND DEVICE FOR ADAPTIVELY ADJUSTING SOUND EFFECT - A method and device for adaptively adjusting sound effect, and the method comprises: obtaining an energy value of the current ambient noise; receiving a first trigger instruction and adjusting the current output volume based on the energy value of the current ambient noise; while judging that the energy value of the current ambient noise is bigger than a first threshold, processing treble enhancement; while judging that the energy value of the current ambient noise is less than a second a sound threshold, processing bass enhancement. By collecting the voice data and detecting the speech activity on the voice data, when the first trigger instruction is received, the method can adjust the current volume and adjust the frequency response by the treble enhancement or the bass enhancement based on the energy value of the current ambient noise, thereby obtaining the better sound effect and easy to achieve. | 11-13-2014 |
20140343932 | SPEECH DECODING DEVICE AND SPEECH DECODING METHOD - The present invention pertains to a speech decoding device that is capable of preventing degradation in sound quality associated with an adjustment of the slope of a spectrum of an output signal (a decoding signal), making it less likely that a loss of bandwidth sensitivity due to the attenuation of a higher band region is perceived. For each frame of the bandwidth extension layer decoding signal, a filter assessment unit ( | 11-20-2014 |
20140350922 | SPEECH PROCESSING DEVICE, SPEECH PROCESSING METHOD AND COMPUTER PROGRAM PRODUCT - According to an embodiment, a speech processing device includes an extractor, a detector, a generator, a converter, and a compensator. The extractor is configured to extract a speech parameter from a spectral envelope of input speech. The detector is configured to detect a missing band in which a component is missed in the spectral envelope. The generator is configured to generate a parameter for the missing band on the basis of a position of the missing band, statistical information created by using a parameter extracted from a spectral envelope of speech with no missing component, and the extracted speech parameter. The converter is configured to convert the generated parameter to a spectral envelope of the missing band. The compensator is configured to generate a spectral envelope supplemented with the missing band by combining the spectral envelopes of the missing band and of the input speech. | 11-27-2014 |
20140358528 | Electronic Apparatus, Method for Outputting Data, and Computer Program Product - According to one embodiment, an electronic apparatus includes a receiver and a processor. The receiver is configured to receive a signal of multiplexed sound comprising data of main sound and sub data. The data of main sound is multiplexed in an audible frequency band. The sub data is multiplexed in a non-audible frequency band. The multiplexed sound is output by an audio speaker of another device and is collected by a microphone of the electronic apparatus. The processor is configured to acquire the sub data from the signal of the multiplexed sound, and to output the sub data. | 12-04-2014 |
20140365211 | ADAPTIVE EQUALIZATION SYSTEM - An adaptive equalization system that adjusts the spectral shape of a speech signal based on an intelligibility measurement of the speech signal may improve the intelligibility of the output speech signal. Such an adaptive equalization system may include a speech intelligibility measurement module, a spectral shape adjustment module, and an adaptive equalization module. The speech intelligibility measurement module is configured to calculate a speech intelligibility measurement of a speech signal. The spectral shape adjustment module is configured to generate a weighted long-term speech curve based on a first predetermined long-term average speech curve, a second predetermined long-term average speech curve, and the speech intelligibility measurement. The adaptive equalization module is configured to adapt equalization coefficients for the speech signal based on the weighted long-term speech curve. | 12-11-2014 |
20140372107 | AUDIO PROCESSING - A technique for creating audio objects on basis of a source audio signal is provided. According to an example embodiment, the technique comprises obtaining a plurality of frequency sub-band signals, each representing a directional component of a source audio signal in the respective frequency sub-band, obtaining an indication of dominant sound source direction for one or more of said frequency sub-band signals, and creating one or more audio objects on basis of said plurality of frequency sub-band signals and said indications, said creating comprising deriving one or more audio object signals, each audio object signal comprising a respective directional signal determined on basis of frequency sub-band signals for which dominant sound source direction falls within a respective predetermined range of source directions and deriving one or more audio object direction indications for said one or more audio object signals, each audio object direction indication derived on basis of dominant sound source directions for the frequency sub-band signals used for determining the respective directional signal. | 12-18-2014 |
20140372108 | METHOD AND APPARATUS FOR ENCODING AND DECODING HIGH FREQUENCY SIGNAL - Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal. | 12-18-2014 |
20150012265 | Enhanced Speech Transmission Index measurements through combination of indirect and direct MTF estimation - The Speech Transmission Index (STI) is an objective measure that predicts speech intelligibility. Test signals are played back through a channel under test. An STI analyzer at the channel output calculates an index between 0 and 1, indicating intelligibility. Over the years, various test signals have been used for measuring the STI, all based on the same principles but differing in the modulation frequencies tested and octave bands covered by the signal. The invention disclosed is a new method for constructing test signals: not just the transfer of modulations on a carrier signal is analyzed, but also the carrier signal itself This apprach leads to more accurate STI measurements that correspond more closely to the ideal (theoretical) STI and to subjective speech intelligibility. Shorter measuring times are achieved without compromising the accuracy of the measurement, while extending the range of systems to which the STI can be reliable applied. | 01-08-2015 |
20150051904 | AUDIO DECODING DEVICE, AUDIO CODING DEVICE, AUDIO DECODING METHOD, AUDIO CODING METHOD, AUDIO DECODING PROGRAM, AND AUDIO CODING PROGRAM - An objective of the present invention is to correct a temporal envelope shape of a decoded signal with a small information volume and to reduce perceptible distortions. An audio decoding device which decodes a coded audio signal and outputs an audio signal comprises: a coded series analysis unit that analyzes a coded series which contains the coded audio signal; an audio decoding unit that receives from the coded series analysis unit the coded series which contains the coded audio signal and decodes same, obtaining an audio signal; a temporal envelope shape establishment unit that receives information from the coded series analysis unit and/or the audio decoding unit, and, on the basis of the information, establishes a temporal envelope shape of the decoded audio signal; and a temporal envelope correction unit that, on the basis of the temporal envelope shape which is established with the temporal envelope shape establishment unit, corrects the temporal envelope shape of the decoded audio signal and outputs same. | 02-19-2015 |
20150066488 | TIME 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. | 03-05-2015 |
20150066489 | TIME 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. | 03-05-2015 |
20150066490 | TIME 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. | 03-05-2015 |
20150073780 | METHOD FOR NON-INTRUSIVE ACOUSTIC PARAMETER ESTIMATION - A system and method for non-intrusive acoustic parameter estimation is included. The method may include receiving, at a computing device, a first speech signal associated with a particular user. The method may include extracting one or more short-term features from the first speech signal. The method may also include determining one or more statistics of each of the one or more short-term features from the first speech signal. The method may further include classifying the one or more statistics as belonging to one or more acoustic parameter classes. | 03-12-2015 |
20150081283 | HARMONICITY ESTIMATION, AUDIO CLASSIFICATION, PITCH DETERMINATION AND NOISE ESTIMATION - Embodiments are described for harmonicity estimation, audio classification, pitch determination and noise estimation. Measuring harmonicity of an audio signal includes calculation a log amplitude spectrum of audio signal. A first spectrum is derived by calculating each component of the first spectrum as a sum of components of the log amplitude spectrum on frequencies. In linear frequency scale, the frequencies are odd multiples of the component's frequency of the first spectrum. A second spectrum is derived by calculating each component of the second spectrum as a sum of components of the log amplitude spectrum on frequencies. In linear frequency scale, the frequencies are even multiples of the component's frequency of the second spectrum. A difference spectrum is derived subtracting the first spectrum from the second spectrum. A measure of harmonicity is generated as a monotonically increasing function of the maximum component of the difference spectrum within predetermined frequency range. | 03-19-2015 |
20150088494 | VOICE PROCESSING APPARATUS AND VOICE PROCESSING METHOD - A voice processing apparatus calculates a phase difference between first and second frequency signals obtained by transforming first and second voice signals generated by two voice input units for each frequency, calculates, for each extension range set outside or inside a reference range, a presence ratio based on the number of frequencies with the phase difference between the first and second frequency signals falling within the extension range, the reference range representing a range of the phase difference between the first and second voice signals for each frequency and corresponding to a direction in which a target sound source is assumed to be located, and sets, as a non-suppression range, a first extension range having the presence ratio higher than a predetermined value and a second extension range closer to the phase difference at the center of the reference range than the first extension range is within the reference range. | 03-26-2015 |
20150088495 | ENCODING APPARATUS AND METHOD FOR ENCODING SOUND CODE, DECODING APPARATUS AND METHDO FOR DECODING THE SOUND CODE - A decoding apparatus includes a sound code input unit that receives a sound code output from an encoding apparatus through a sound wave reception device; a frame division unit that divides the sound code depending on a predetermined time interval to generate a plurality of frames; a frequency identification unit that identifies a frequency corresponding to each of the plurality of the frames through frequency analysis for each of the plurality of the frames; and an information generation unit that determines a frequency band, to which each of the identified frequencies corresponds, from an audible sound wave frequency band and a non-audible sound wave frequency band, and a plurality of partial information based on the frequency band and each of the identified frequencies, and generates information corresponding to the sound code based on the plurality of the partial information. | 03-26-2015 |
20150095023 | APPARATUS 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. | 04-02-2015 |
20150134327 | LINGERING CANCELLATION FOR DATA ENCODED AUDIO SIGNALS - Presented herein are techniques to reduce error probability and eliminate the lingering signal that causes problems during audio signal data transmission. An electroacoustic transducer of a first electronic device receives a data encoded audio signal that includes modulated digital data therein. The data encoded audio signal is demodulated to produce a present input signal that includes a target frequency and a lingering effect of a previously received data encoded audio signal. Lingering cancellation is performed on the present input signal to produce a present output signal from which the lingering effect of the previously received data encoded audio signal has been removed. | 05-14-2015 |
20150149156 | SELECTIVE PHASE COMPENSATION IN HIGH BAND CODING - A method includes determining, at an encoder, phase adjustment parameters based on a high-band residual signal. The method also includes inserting the phase adjustment parameters into an encoded version of the audio signal to enable phase adjustment during reconstruction of the audio signal from the encoded version of the audio signal. | 05-28-2015 |
20150149157 | FREQUENCY DOMAIN GAIN SHAPE ESTIMATION - A method includes determining, at a speech encoder, frequency domain gain shape parameters. The frequency domain gain shape parameters are based on a second signal associated with an audio signal. The method further includes adjusting a first signal based on the frequency domain gain shape parameters. The first signal is associated with the audio signal. The method also includes inserting the frequency domain gain shape parameters into an encoded version of the audio signal to enable gain adjustment during reproduction of the audio signal from the encoded version of the audio signal. | 05-28-2015 |
20150149158 | Efficient Combined Harmonic Transposition - The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank ( | 05-28-2015 |
20150302865 | SYSTEM AND METHOD FOR AUDIO CONFERENCING - The present disclosure is directed towards an audio conferencing method. Some embodiments may include receiving, at a first mixing device, an audio signal from a first user associated with an audio conference. Embodiments may further include processing the audio signal at the first mixing device to generate a processed audio signal and transmitting the processed audio signal to a second mixing device, wherein the first mixing device and the second mixing device are distributed over a network in a cascaded configuration. Embodiments may also include receiving, at the second mixing device, a third audio signal from a second user associated with the audio conference and processing the third audio signal at the second mixing device to generate a second processed audio signal. | 10-22-2015 |
20150310873 | SYSTEM AND METHOD FOR IMPROVING SOUND QUALITY OF VOICE SIGNAL IN VOICE COMMUNICATION - Disclosed are a voice communication system and a voice communication method which set a subtraction weight for each of a plurality of frequency subbands split based on a particular frequency response characteristic set to the system, calculate a gain function for each frequency subband according to the particular frequency response characteristic based on the subtraction weight for each of the frequency subbands, and improve sound quality of a voice signal by reflecting the calculated gain function. | 10-29-2015 |
20150310874 | ADAPTIVE AUDIO SIGNAL FILTERING - An apparatus comprising: an audio signal analyser configured to analyse an audio signal; an audio signal processor configured to signal process the audio signal to enhance the speech component of the audio signal dependent on determining the audio signal comprises speech components; and signal processing the audio signal to enhance a loudness of the audio signal otherwise. | 10-29-2015 |
20150317986 | Processing of Audio Signals During High Frequency Reconstruction - The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described. The system comprises means for receiving the plurality of low frequency subband signals; means for receiving a set of target energies, each target energy covering a different target interval within the high frequency interval and being indicative of the desired energy of one or more high frequency subband signals lying within the target interval; means for generating the plurality of high frequency subband signals from the plurality of low frequency subband signals and from a plurality of spectral gain coefficients associated with the plurality of low frequency subband signals, respectively; and means for adjusting the energy of the plurality of high frequency subband signals using the set of target energies. | 11-05-2015 |
20150317991 | VOICE AUDIO ENCODING DEVICE, VOICE AUDIO DECODING DEVICE, VOICE AUDIO ENCODING METHOD, AND VOICE AUDIO DECODING METHOD - Provided are a voice audio encoding device, voice audio decoding device, voice audio encoding method, and voice audio decoding method that efficiently perform bit distribution and improve sound quality. Dominant frequency band identification unit identifies a dominant frequency band having a norm factor value that is the maximum value within the spectrum of an input voice audio signal. Dominant group determination units and non-dominant group determination unit group all sub-bands into a dominant group that contains the dominant frequency band and a non-dominant group that contains no dominant frequency band. Group bit distribution unit distributes bits to each group on the basis of the energy and norm variance of each group. Sub-band bit distribution unit redistributes the bits that have been distributed to each group to each sub-band in accordance with the ratio of the norm to the energy of the groups. | 11-05-2015 |
20150325250 | METHOD AND APPARATUS FOR PRE-PROCESSING SPEECH TO MAINTAIN SPEECH INTELLIGIBILITY - An audio system processes a speech signal to maintain a target value of the speech intelligibility index (SII) while minimizing the overall speech level so that speech intelligibility is preserved across different environmental sound levels while possible distortions and overall loudness are mitigated. In one embodiment, a hearing aid processes a speech signal received from another device to maintain a target value of the SII while minimizing the overall speech level before mixing the speech signal with a microphone signal. | 11-12-2015 |
20150332697 | APPARATUS AND METHOD FOR GENERATING A FREQUENCY ENHANCED SIGNAL USING TEMPORAL SMOOTHING OF SUBBANDS - An apparatus for generating a frequency enhancement signal has: a signal generator for generating an enhancement signal from a core signal, the enhancement signal having an enhancement frequency range not included in the core signal, wherein a current time portion of the enhancement signal or the core signal has subband signals for a plurality of subbands; a controller for calculating the same smoothing information for the plurality of subband signals of the enhancement frequency range or the core signal, and wherein the signal generator is configured for smoothing the plurality of subband signals of the enhancement frequency range or the core signal using the same smoothing information. | 11-19-2015 |
20150332699 | Method for Predicting High Frequency Band Signal, Encoding Device, and Decoding Device - A method includes obtaining a signal type of an audio signal and a low frequency band signal of the audio signal, where the audio signal includes the low frequency band signal and a high frequency band signal; obtaining a frequency envelope of the high frequency band signal according to the signal type; predicting an excitation signal of the high frequency band signal according to the low frequency band signal; and restoring the high frequency band signal according to the frequency envelope of the high frequency band signal and the excitation signal of the high frequency band signal. By using the technical solutions of the embodiments of the present invention, an error existing between a high frequency band signal obtained by prediction and an actual high frequency band signal can be effectively reduced, and an accuracy rate of the predicted high frequency band signal can be increased. | 11-19-2015 |
20150332706 | APPARATUS AND METHOD FOR GENERATING A FREQUENCY ENHANCED SIGNAL USING SHAPING OF THE ENHANCEMENT SIGNAL - An apparatus for generating a frequency enhancement signal has: a calculator for calculating a value describing an energy distribution with respect to frequency in a core signal; and a signal generator for generating an enhancement signal having an enhancement frequency range not included in the core signal, from the core signal, wherein the signal generator is configured for shaping the enhancement signal or the core signal so that a spectral envelope of the enhancement signal or of the core signal depends on the value describing the energy distribution with respect to frequency in the core signal. | 11-19-2015 |
20150332707 | APPARATUS AND METHOD FOR GENERATING A FREQUENCY ENHANCEMENT SIGNAL USING AN ENERGY LIMITATION OPERATION - An apparatus for generating a frequency enhancement signal, includes: a signal generator for generating an enhancement signal from a core signal, the enhancement signal including an enhancement frequency range not included in the core signal, wherein a time portion of the enhancement signal includes subband signals for a plurality of subbands; a synthesis filterbank for generating the frequency enhanced signal using the enhancement signal, wherein the signal generator is configured for performing an energy limitation in order to make sure that the frequency enhanced signal obtained by the synthesis filterbank is so that an energy of a higher band is, at the most, equal to an energy in a lower band or is greater than an energy of a higher band, at the most, by a predefined threshold. | 11-19-2015 |
20150340045 | Audio Watermarking via Phase Modification - An audio watermarking system conveys information using an audio channel by modulating an audio signal to produce a modulated signal by embedding additional information into the audio signal. Modulating the audio signal includes segmenting the audio signal into overlapping time segments using a non-rectangular analysis window function produce a windowed audio signal, processing the windowed audio signal for a time segment to produce frequency coefficients representing the windowed time segment and having phase values and magnitude values, selecting one or more of the frequency coefficients, modifying phase values of the selected frequency coefficients using the additional information to map the phase values onto a known phase constellation, and processing the frequency coefficients including the modified phase values to produce the modulated signal. | 11-26-2015 |
20150364145 | SELF-VOICE FEEDBACK IN COMMUNICATIONS HEADSETS - Techniques for providing self-voice feedback in a communications headset include processing signals carrying near-end speech in parallel digital and analog signal processing paths to produce a combined gain-adjusted near-end signal carrying the near-end speech for output to transducers of the communications device. | 12-17-2015 |
20150371648 | ENHANCING PERCEPTION OF FREQUENCY-LOWERED SPEECH - Among other things, a sound processing device system is disclosed to assist a hearing-impaired human listener recognize speech sounds or phonemes. The device system may be configured at least to generate an output audio signal at least by transposing and causing a negative rank ordering of frequency of at least a portion of the input audio signal. Compression also may be performed on the at least the portion of the input audio signal as part of generating the output audio signal. The negative rank ordering may be performed on a high-frequency portion of the input audio signal that becomes a low-frequency portion of the output audio signal by the transposing. The low-frequency portion of the output audio signal may represent an inverted ordering of frequencies or frequency segments present in the high-frequency portion of the input audio signal. | 12-24-2015 |
20150371662 | VOICE PROCESSING DEVICE AND VOICE PROCESSING METHOD - A voice processing device includes a memory; and a processor configured to execute a plurality of instructions stored in the memory, the instructions includes acquiring a transmitted voice; first detecting a first utterance segment of the transmitted voice; second detecting a response segment from the first utterance segment; determining a frequency of the response segment included in the transmitted voice; and estimating an utterance time period of a received voice on a basis of the frequency. | 12-24-2015 |
20150372723 | METHOD AND APPARATUS FOR MITIGATING FEEDBACK IN A DIGITAL RADIO RECEIVER - Embodiments of an acoustic feedback suppressor determine the energy in each of a plurality of frequency bands of frames of an audio signal. The energy in each of the plurality of frequency bands is compared to characteristic of human voice to determine that a present frame contains content that is not likely human voice and exhibits a characteristic of feedback. Upon determining that feedback is occurring, an adaptive gain reduction is applied to the band in which feedback is suspected to be occurring. | 12-24-2015 |
20150380006 | TEMPORAL GAIN ADJUSTMENT BASED ON HIGH-BAND SIGNAL CHARACTERISTIC - The present disclosure provides techniques for adjusting a temporal gain parameter and for adjusting linear prediction coefficients. A value of the temporal gain parameter may be based on a comparison of a synthesized high-band portion of an audio signal to a high-band portion of the audio signal. If a signal characteristic of an upper frequency range of the high-band portion satisfies a first threshold, the temporal gain parameter may be adjusted. A linear prediction (LP) gain may be determined based on an LP gain operation that uses a first value for an LP order. The LP gain may be associated with an energy level of an LP synthesis filter. The LP order may be reduced if the LP gain satisfies a second threshold. | 12-31-2015 |
20150381822 | ECHO CANCELLATION DEVICE - An echo cancellation device includes: a full-band echo canceller that generates a pseudo-echo signal; a downsample processor that downsamples a received signal and extracts a low-band component delayed by a delay amount D | 12-31-2015 |
20160005412 | GENERATION OF A SIGNATURE OF A MUSICAL AUDIO SIGNAL - The invention concerns a method for generating a signature of a musical audio signal of a given duration, the method comprising the following steps: —modelling ( | 01-07-2016 |
20160005413 | Audio Signal Enhancement Using Estimated Spatial Parameters - Received audio data may include a first set of frequency coefficients and a second set of frequency coefficients. Spatial parameters for at least part of the second set of frequency coefficients may be estimated, based at least in part on the first set of frequency coefficients. The estimated spatial parameters may be applied to the second set of frequency coefficients to generate a modified second set of frequency coefficients. The first set of frequency coefficients may correspond to a first frequency range (for example, an individual channel frequency range) and the second set of frequency coefficients may correspond to a second frequency range (for example, a coupled channel frequency range). Combined frequency coefficients of a composite coupling channel may be based on frequency coefficients of two or more channels. Cross-correlation coefficients, between frequency coefficients of a first channel and the combined frequency coefficients, may be computed. | 01-07-2016 |
20160005420 | VOICE EMPHASIS DEVICE - An input signal analyzer determines a boundary frequency within the limit of a range which does not exceed a first frequency from the mode of an input signal. A spectrum compressor compresses a power spectrum of frequencies in a band higher than the first frequency in a frequency direction. A gain corrector performs a gain correction on the compressed power spectrum. A spectrum synthesizer reflects the power spectrum outputted from the gain corrector in a band determined by both the first frequency and the boundary frequency. A frequency-to-time converter converts both a synthesized power spectrum provided by the spectrum synthesizer and a phase spectrum of the input signal into ones in the time domain, and outputs these spectra. | 01-07-2016 |
20160012828 | WIND NOISE REDUCTION FOR AUDIO RECEPTION | 01-14-2016 |
20160027450 | Classification Between Time-Domain Coding and Frequency Domain Coding - A method for processing speech signals prior to encoding a digital signal comprising audio data includes selecting frequency domain coding or time domain coding based on a coding bit rate to be used for coding the digital signal and a short pitch lag detection of the digital signal. | 01-28-2016 |
20160035363 | METHOD AND APPARATUS FOR ENCODING AND DECODING NOISE SIGNAL - Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits. | 02-04-2016 |
20160035364 | METHOD AND DEVICE FOR ENCODING A HIGH FREQUENCY SIGNAL, AND METHOD AND DEVICE FOR DECODING A HIGH FREQUENCY SIGNAL - A method and a device for encoding a high frequency signal, and a method and a device for decoding a high frequency signal are provided, which relate to encoding and decoding technology. The method for encoding a high frequency signal includes: determining a signal type of a high frequency signal of a current frame; smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded, if the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal; and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame. | 02-04-2016 |
20160042742 | Audio Encoder and Decoder for Interleaved Waveform Coding - There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved. | 02-11-2016 |
20160064009 | Adaptively Reducing Noise While Limiting Speech Loss Distortion - The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In various embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible. | 03-03-2016 |
20160064012 | VOICE PROCESSING DEVICE, VOICE PROCESSING METHOD, AND NON-TRANSITORY COMPUTER READABLE RECORDING MEDIUM HAVING THEREIN PROGRAM FOR VOICE PROCESSING - A voice processing device includes a first sound collecting unit for generating a first voice signal; a human-body vibration obtaining unit for generating a human-body vibration signal; a first distance calculating unit for calculating a ratio in power or amplitude between the human-body vibration signal and the first voice signal and for calculating a first distance from the first sound collecting unit to a sound source in accordance with the ratio and distance estimation information; a second distance calculating unit for calculating, for each of a plurality of frequencies, a second distance from the first sound collecting unit to a sound source which produces a component of a frequency of a first frequency signal; a gain determining unit for determining, for each of the plurality of frequencies, a gain based on a comparison result between the first distance and the second distance. | 03-03-2016 |
20160086618 | A METHOD AND APPARATUS FOR SUPPRESSION OF UNWANTED AUDIO SIGNALS - A device for removal of unwanted components in an audio signal, the device comprising a processor, coupled to memory, configured to receive reference and processed inputs into memory where the processed input is a result of a reduction process of unwanted components of the audio signal, estimate envelope values for processed and reference inputs at a plurality of time and frequency instances, for each time and frequency instance: compute a first gain in relation to a ratio of the estimated envelope value of the processed input to the estimated envelope value of the reference input, apply a nonlinear process to said first gain to produce a second gain, compute an output gain as the ratio between second gain and first gain and, apply the output gain to processed input, thereby producing a filtered output with unwanted components suppressed. | 03-24-2016 |
20160118055 | DECODING METHOD AND DECODING APPARATUS - Embodiments of the present disclosure provide a decoding method and a decoding apparatus. The decoding method includes: in a case in which it is determined that a current frame is a lost frame, synthesizing a high frequency band signal; determining subframe gains of multiple subframes of the current frame; determining a global gain of the current frame; and adjusting, according to the global gain and the subframe gains of the multiple subframes, the synthesized high frequency band signal to obtain a high frequency band signal of the current frame. A subframe gain of the current frame is obtained according to a gradient between subframe gains of subframes previous to the current frame, so that transition before and after frame loss is more continuous, thereby reducing noise during signal reconstruction, and improving speech quality. | 04-28-2016 |
20160133264 | Comfort Noise Generation - A system for generating comfort noise for a stream of frames carrying an audio signal includes frame characterizing logic configured to generate a set of filter parameters characterising the frequency content of a frame; an analysis filter adapted using the filter parameters and configured to filter the frame so as to generate residual samples; an analysis controller configured to cause the residual samples to be stored in a store responsive to receiving an indication that the frame does not comprise speech; and a synthesis controller operable to select stored residual samples from the store and cause a synthesis filter, inverse to the analysis filter and adapted using filter parameters generated by the frame characterizing logic for one or more frames not comprising speech, to filter the selected residual samples so as to generate a frame of comfort noise. | 05-12-2016 |
20160133273 | IMPROVED FREQUENCY BAND EXTENSION IN AN AUDIO SIGNAL DECODER - The invention relates to a method for extending the frequency band of an audio signal during a decoding or improvement process comprising a step of decoding or extracting, in a first so-called low frequency band, an excitation signal and coefficients of a linear prediction filter. The method comprises the following steps: —obtaining a signal (U | 05-12-2016 |
20160140970 | AUDIO DATA RECEIPT/EXPOSURE MEASUREMENT WITH CODE MONITORING AND SIGNATURE EXTRACTION - Systems and methods are provided for gathering audience measurement data relating to receipt of and/or exposure to audio data by an audience member. Audio data is monitored to detect a monitoring code. Based on detection of the monitoring code, a signature characterizing the audio data is extracted. | 05-19-2016 |
20160140977 | NOISE CANCELLATION METHOD - An embodiment of the invention provides a noise cancellation method for an electronic device. The method comprises: receiving an audio signal; applying a Fast Fourier Transform operation on the audio signal to generate a sound spectrum; acquiring a first spectrum corresponding to a noise and a second spectrum corresponding to a human voice signal from the sound spectrum; estimating a center frequency according to the first spectrum and the second spectrum; and applying a high pass filtering operation to the sound spectrum according to the center frequency. | 05-19-2016 |
20160140983 | RATE CONVERTOR - Embodiments of the invention may be used to implement a rate converter that includes: 6 channels in forward (audio) path, each channel having a 24-bit signal path per channel, an End-to-end SNR of 110 dB, all within the 20 Hz to 20 KHz bandwidth. Embodiment may also be used to implement a rate converter having: 2 channels in a reverse path, such as for voice signals, 16-bit signal path per channel, an End-to-end SNR of 93 dB, all within 20 Hz to 20 KHz bandwidth. The rate converter may include sample rates such as 8, 11.025, 12, 16, 22.05, 24, 32 44.1, 48, and 96 KHz. Further, rate converters according to embodiments may include a gated clock in low-power mode to conserve power. | 05-19-2016 |
20160180854 | Audio Decoder Having A Bandwidth Extension Module With An Energy Adjusting Module | 06-23-2016 |
20160189707 | SPEECH PROCESSING - A wireless communication device is disclosed. The wireless communication device includes a processor, a memory, a transceiver configured to receive an audio signal, a codec to decode the audio signal, a dynamic range controller and a phoneme processor. The phoneme processor is configured to extract acoustic cues from each frame of the decoded audio signal and to identify a phoneme class in the each frame. The dynamic range controller is configured to apply dynamic range compression on the each frame based on the identified phoneme class. | 06-30-2016 |
20160189720 | Model Based Prediction in a Critically Sampled Filterbank - A method and method of extracting information from a netlist. The netlist for a device under test (DUT) is read and a circuit selected to be transformed. Transformation candidates are identified using transformation specific criteria and verification methods are applied to prove the transformation is equivalent to the circuit being transformed. If the candidate transformation is equivalent to the circuit being transformed, the system commits to the transformation. If the candidate transformation is not equivalent to the circuit being transformed, the transformation is undone. | 06-30-2016 |
20160379655 | High Frequency Regeneration of an Audio Signal with Temporal Shaping - A method for generating a reconstructed audio signal having a baseband portion and a highband portion is disclosed. The method includes extracting temporal envelope information and spectral components of the baseband portion. The method further includes obtaining a decoded baseband audio signal. The obtaining includes filtering in a frequency domain at least some of the spectral components of the baseband portion with the reconstruction filter using the temporal envelope information to shape a temporal envelope of the baseband portion. The method also includes extracting a noise parameter and an estimated spectral envelope of the highband portion and obtaining a plurality of subband signals by filtering the decoded baseband audio signal. The method further includes generating a high-frequency reconstructed signal by copying a number of consecutive subband signals of the plurality of subband signals and obtaining an envelope adjusted high-frequency signal by adjusting, based on the estimated spectral envelope of the highband portion, a spectral envelope of the high-frequency reconstructed signal. | 12-29-2016 |
20180025740 | OPTIMIZATION OF SPEECH INPUT FOR MULTIPLE SPEECH AGENTS USED IN A COMMON APPLICATION ENVIRONMENT | 01-25-2018 |
20180025743 | ESCALATION DETECTION USING SENTIMENT ANALYSIS | 01-25-2018 |