| Entries |
| Document | Title | Date |
|---|
| 20090076805 | METHOD AND DEVICE FOR PERFORMING FRAME ERASURE CONCEALMENT TO HIGHER-BAND SIGNAL - The present invention discloses a method for performing a frame erasure concealment to a higher-band signal, including: calculating a periodic intensity of a higher-band signal with respect to a lower-band signal; judging whether the periodic intensity of the higher-band signal is higher than or equal to a preconfigured threshold; if the periodic intensity of the higher-band signal is higher than or equal to the preconfigured threshold, using a pitch period repetition method to perform the frame erasure concealment to the higher-band signal of a current lost frame; and if the periodic intensity of the higher-band signal is lower than the preconfigured threshold, using a previous frame data repetition method to perform the frame erasure concealment to the higher-band signal of the current lost frame. The present invention further discloses a device for performing a frame erasure concealment to a higher-band signal and a speech decoder. The problem that the quality of the voice signal is lowered is avoided. | 03-19-2009 |
| 20090076806 | EMPHASIS OF SHORT-DURATION TRANSIENT SPEECH FEATURES - A sound processor including a microphone ( | 03-19-2009 |
| 20090192788 | Sound Processing Device and Program - In a sound processing device, a modulation spectrum specifier specifies a modulation spectrum of an input sound for each of a plurality of unit intervals. An index calculator calculates an index value corresponding to a magnitude of components of modulation frequencies belonging to a predetermined range of the modulation spectrum. A determinator determines whether the input sound of each of the unit intervals is a vocal sound or a non-vocal sound based on the index value. The modulation spectrum specifier analyzes the input sound to obtain a cepstrum or a logarithmic spectrum of the input sound for each of a sequence of frames defined within the unit interval, then specifies a temporal trajectory of a specific component in the cepstrum or the logarithmic spectrum along the sequence of the frames for the unit interval, and performs a Fourier transform on the temporal trajectory throughout the unit interval to thereby specify the modulation spectrum of the unit interval as the result of the Fourier transform of the temporal trajectory. | 07-30-2009 |
| 20090192789 | METHOD AND APPARATUS FOR ENCODING/DECODING AUDIO SIGNALS - Provided are a method and apparatus for effectively encoding/decoding remaining difference signals excluding sinusoidal components, from input audio signals. In the method and apparatus for encoding audio signals, sinusoidal analysis is performed on low frequency signals of less than a predetermined critical frequency in order to extract sinusoidal signals and then, an encoding operation is performed on the remaining difference signals excluding the sinusoidal signals, from input audio signals, by using linear prediction coding (LPC) analysis. | 07-30-2009 |
| 20090204395 | STRAINED-ROUGH-VOICE CONVERSION DEVICE, VOICE CONVERSION DEVICE, VOICE SYNTHESIS DEVICE, VOICE CONVERSION METHOD, VOICE SYNTHESIS METHOD, AND PROGRAM - A strained-rough-voice conversion unit ( | 08-13-2009 |
| 20090326929 | Methods for Improving High Frequency Reconstruction - The present invention proposes a new method and a new apparatus for enhancement of audio source coding systems utilising high frequency reconstruction (HFR). It utilises a detection mechanism on the encoder side to assess what parts of the spectrum will not be correctly reproduced by the HFR method in the decoder. Information on this is efficiently coded and sent to the decoder, where it is combined with the output of the HFR unit. | 12-31-2009 |
| 20100010809 | Method, apparatus, and medium for bandwidth extension encoding and decoding - Provided are a method, apparatus, and medium for encoding/decoding a high frequency band signal by using a low frequency band signal corresponding to an audio signal or a speech signal. Accordingly, since the high frequency band signal is encoded and decoded by using the low frequency band signal, encoding and decoding can be carried out with a small data size while avoiding deterioration of sound quality. | 01-14-2010 |
| 20100036657 | SPEECH ESTIMATION SYSTEM, SPEECH ESTIMATION METHOD, AND SPEECH ESTIMATION PROGRAM - The speech estimation system of the present invention includes a transmitter ( | 02-11-2010 |
| 20100082336 | APPARATUS AND METHOD FOR CALCULATING A FUNDAMENTAL FREQUENCY CHANGE - A logarithmic frequency spectrum within a predetermined time range is calculated from a speech signal. The logarithmic frequency spectrum has a frequency element at equal intervals along a logarithmic frequency axis. A logarithmic frequency spectrogram is calculated by connecting a plurality of logarithmic frequency spectrums. A value of the frequency element along a straight line on the logarithmic frequency spectrogram is voted onto a Hough plane. The Hough plane has a voted value in correspondence with a gradient of the straight line. The voted value above a threshold and the gradient corresponding to the voted value are extracted from the Hough plane. A fundamental frequency change is calculated using the voted value and the gradient extracted. | 04-01-2010 |
| 20100094619 | AUDIO FREQUENCY REMAPPING - An exemplary system and method are directed at receiving an audio signal and process the audio signal into a remapped audio signal based on a plot profile. The plot profile may include at least one of an identified range of audio frequencies. The processing may comprise retrieving an identified range of audio frequencies from the plot profile; determining a range of impaired audio frequencies in the audio signal based on the identified range of audio frequencies; shifting the frequency of at least a portion of the impaired audio frequencies to outside of the identified range; and continuing to retrieve identified ranges of audio frequencies from the plot profile. The shifting of the impaired audio frequencies of the audio signal may be performed until no further identified ranges of audio frequencies are available for consideration. | 04-15-2010 |
| 20100145687 | REMOVING NOISE FROM SPEECH - Method for removing noise from a digital speech waveform, including receiving the digital speech waveform having the noise contained therein, segmenting the digital speech waveform into one or more frames, each frame having a clean portion and a noisy portion, extracting a feature component from each frame, creating an nonlinear speech distortion model from the feature components, creating a statistical noise model by making a Piecewise Linear Approximation (PLA) of the nonlinear speech distortion model, determining the clean portion of each frame using the statistical noise model, a log power spectra of each frame, and a model of a digital speech waveform recorded in a noise controlled environment, and constructing a clean digital speech waveform from each clean portion of each frame. | 06-10-2010 |
| 20100191523 | Method and apparatus for recovering line spectrum pair parameter and speech decoding apparatus using same - A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame. The method and apparatus can improve the quality of a recovered speech signal, be applied to a variety of technologies, and provide a method of recovering an LSP parameter for development of an algorithm for speech decoding. | 07-29-2010 |
| 20100217584 | SPEECH ANALYSIS DEVICE, SPEECH ANALYSIS AND SYNTHESIS DEVICE, CORRECTION RULE INFORMATION GENERATION DEVICE, SPEECH ANALYSIS SYSTEM, SPEECH ANALYSIS METHOD, CORRECTION RULE INFORMATION GENERATION METHOD, AND PROGRAM - A speech analysis device which accurately analyzes an aperiodic component included in speech in a practical environment where there is background noise includes: a frequency band division unit which divides, into bandpass signals each associated with a corresponding one of frequency bands, an input signal representing a mixed sound of background noise and speech; a noise interval identification unit which identifies a noise interval and a speech interval of the input signal; an SNR calculation unit which calculates an SN ratio; a correlation function calculation unit which calculates an autocorrelation function of each bandpass signal; a correction amount determination unit which determines a correction amount for an aperiodic component ratio, based on the calculated SN ratio; and an aperiodic component ratio calculation unit which calculates, for each frequency band, an aperiodic component ratio of the aperiodic component, based on the determined correction amount and the calculated autocorrelation function. | 08-26-2010 |
| 20100241423 | SYSTEM AND METHOD FOR FREQUENCY TO PHASE BALANCING FOR TIMBRE-ACCURATE LOW BIT RATE AUDIO ENCODING - Embodiments of a system and method for encoding audio data have been described. In one embodiment, the method includes transforming frequency domain data in a plurality of signal windows of an audio dataset from a cosine/sine format to a magnitude/cosine/sine format. The magnitude/cosine/sine format disproportionately represents a magnitude of the frequency domain data over a phase of the frequency domain data. The above transformation may be a pre-processing stage of vector quantization usable to produce a codebook. | 09-23-2010 |
| 20100286981 | Method for Estimating a Fundamental Frequency of a Speech Signal - The invention provides a method for estimating a fundamental frequency of a speech signal comprising the steps of receiving a signal spectrum of the speech signal, filtering the signal spectrum to obtain a refined signal spectrum, determining a cross-power spectral density using the refined signal spectrum and the signal spectrum, transforming the cross-power spectral density into the time domain to obtain a cross-correlation function, and estimating the fundamental frequency of the speech signal based on the cross-correlation function. | 11-11-2010 |
| 20110099004 | DETERMINING AN UPPERBAND SIGNAL FROM A NARROWBAND SIGNAL - A method for determining an upperband speech signal from a narrowband speech signal is disclosed. A list of narrowband line spectral frequencies (LSFs) is determined from the narrowband speech signal. A first pair of adjacent narrowband LSFs that have a lower difference between them than every other pair of adjacent narrowband LSFs in the list is determined. A first feature that is a mean of the first pair of adjacent narrowband LSFs is determined. Upperband LSFs are determined based on at least the first feature using codebook mapping. | 04-28-2011 |
| 20110106530 | APPARATUS AND METHOD FOR IMPROVING VOICE QUALITY IN PORTABLE TERMINAL - An apparatus and method for improving a sound quality of a portable terminal are provided. Particularly, an apparatus and method for tuning to a voice quality that a user desires using the portable terminal, in case where the user determines an abnormal sound quality, are provided. The apparatus includes a sound quality improving unit for selecting a Digital Signal Processing (DSP) filter value corresponding to a voice quality that a user desires among a plurality of DSP filters for controlling the voice quality according to a user's voice quality. | 05-05-2011 |
| 20120101813 | Coding Generic Audio Signals at Low Bitrates and Low Delay - A mixed time-domain/frequency-domain coding device and method for coding an input sound signal, wherein a time-domain excitation contribution is calculated in response to the input sound signal. A cut-off frequency for the time-domain excitation contribution is also calculated in response to the input sound signal, and a frequency extent of the time-domain excitation contribution is adjusted in relation to this cut-off frequency. Following calculation of a frequency-domain excitation contribution in response to the input sound signal, the adjusted time-domain excitation contribution and the frequency-domain excitation contribution are added to form a mixed time-domain/frequency-domain excitation constituting a coded version of the input sound signal. In the calculation of the time-domain excitation contribution, the input sound signal may be processed in successive frames of the input sound signal and a number of sub-frames to be used in a current frame may be calculated. | 04-26-2012 |
| 20120296641 | SYSTEMS, METHODS, AND APPARATUS FOR WIDEBAND ENCODING AND DECODING OF INACTIVE FRAMES - Speech encoders and methods of speech encoding are disclosed that encode inactive frames at different rates. Apparatus and methods for processing an encoded speech signal are disclosed that calculate a decoded frame based on a description of a spectral envelope over a first frequency band and the description of a spectral envelope over a second frequency band, in which the description for the first frequency band is based on information from a corresponding encoded frame and the description for the second frequency band is based on information from at least one preceding encoded frame. Calculation of the decoded frame may also be based on a description of temporal information for the second frequency band that is based on information from at least one preceding encoded frame. | 11-22-2012 |
| 20120323569 | SPEECH PROCESSING APPARATUS, A SPEECH PROCESSING METHOD, AND A FILTER PRODUCED BY THE METHOD - According to one embodiment, a speech processing apparatus includes a histogram calculation unit, a cumulative frequency calculation unit, and a filter production unit. The histogram calculation unit is configured to calculate a first histogram from a first speech feature extracted from speech data, and to calculate a second histogram from a second speech feature different from the first speech feature. The cumulative frequency calculation unit is configured to calculate a first cumulative frequency by accumulating a frequency of the first histogram, and to calculate a second cumulative frequency by accumulating a frequency of the second histogram. The filter production unit is configured to produce a filter having a characteristic to get the second cumulative frequency near to the first cumulative frequency. | 12-20-2012 |
| 20120330650 | METHODS, SYSTEMS, AND COMPUTER READABLE MEDIA FOR FRICATIVES AND HIGH FREQUENCIES DETECTION - Methods, systems, and computer readable media for fricatives and high frequencies detection are disclosed. According to one method, the method includes receiving a narrowband signal. The method also includes detecting, using one or more autocorrelation coefficients, a high frequency speech component associated with the narrowband signal. | 12-27-2012 |
| 20130013301 | AUDIO ENCODER, AUDIO DECODER, METHOD FOR ENCODING AND AUDIO INFORMATION, METHOD FOR DECODING AN AUDIO INFORMATION AND COMPUTER PROGRAM USING A HASH TABLE DESCRIBING BOTH SIGNIFICANT STATE VALUES AND INTERVAL BOUNDARIES - An audio decoder includes an arithmetic decoder for providing a plurality of decoded spectral values on the basis of an arithmetically encoded representation of the spectral values, and a frequency-domain-to-time-domain converter for providing a time-domain audio representation using the decoded spectral values. The arithmetic decoder selects a mapping rule describing a mapping of a code value onto a symbol code in dependence on a context state described by a numeric current context value. The arithmetic decoder determines the numeric current context value in dependence on a plurality of previously decoded spectral values. The arithmetic decoder evaluates a hash table, entries of which define both significant state values and boundaries of intervals of numeric context values, in order to select the mapping rule. A mapping rule index value is individually associated to a numeric context value being a significant state value. | 01-10-2013 |
| 20130035933 | AUDIO SIGNAL PROCESSING APPARATUS AND AUDIO SIGNAL PROCESSING METHOD - Likelihood calculation means extracts audio features expressing features of a voice signal and a non-voice signal from an acquired audio signal, and calculates likelihood expressing probability that the voice signal is included in the audio signal using the audio features. Spectral feature extraction means performs a frequency analysis to the audio signal to extract a spectral feature. Using the spectral feature, first basis matrix producing means produces a first basis matrix expressing the feature of the non-voice signal. Second basis matrix producing means specifies a component having a high association with the voice signal in the first basis matrix using the likelihood, and excludes the component to produce a second basis matrix. Spectral feature estimation means estimates a spectral feature of the voice signal or a spectral feature of the non-voice signal by performing nonnegative matrix factorization to the spectral feature using the second basis matrix. | 02-07-2013 |
| 20140149111 | SPEECH ENHANCEMENT APPARATUS AND SPEECH ENHANCEMENT METHOD - A speech enhancement apparatus includes: a noise estimating unit which estimates a noise component contained in a speech signal for each frequency band; a signal-to-noise ratio computing unit which computes, for each frequency band, a signal-to-noise ratio; a gain computing unit which selects a frequency band whose computed signal-to-noise ratio indicates that the signal component contained in the speech signal for the frequency band is recognizable, and which determines a gain indicating the degree of enhancement to be applied to the speech signal in accordance with the signal-to-noise ratio of the selected frequency band; and an enhancing unit which amplifies an amplitude component of a frequency domain signal in each frequency band in accordance with the gain, and which corrects the amplitude component of the frequency domain signal by subtracting the noise component from the amplitude component in each frequency band. | 05-29-2014 |
| 20140188464 | APPARATUS AND METHOD FOR GENERATING BANDWIDTH EXTENSION SIGNAL - An apparatus for generating a bandwidth extended signal includes an anti-sparseness processing unit to perform anti-sparseness processing on a low-frequency spectrum; and a frequency domain high-frequency extension decoding unit to perform high-frequency extension encoding in the frequency domain on the low-frequency spectrum on which the anti-sparseness processing is performed. | 07-03-2014 |
| 20140200884 | TELECOMMUNICATIONS METHODS AND SYSTEMS PROVIDING USER SPECIFIC AUDIO OPTIMIZATION - Systems and methods for applying user specific acoustic adjustment parameters are provided. The intelligibility of speech for a particular user is determined and a set of acoustic adjustment parameters is determined. The set or template of acoustic adjustment parameters for the user is placed in central store, for example provided as or in association with a server. The template can be obtained from the server for application in connection with a communication involving the user by providing an identification of the template. | 07-17-2014 |
| 20140200885 | AUDIO VISUAL SIGNATURE, METHOD OF DERIVING A SIGNATURE, AND METHOD OF COMPARING AUDIO-VISUAL DATA BACKGROUND - The invention relates to the analysis of characteristics of audio and/or video signals for the generation of audio-visual content signatures. To determine an audio signature a region of interest for example of high entropy—is identified in audio signature data. This region of interest is then provided as an audio signature with offset information. A video signature is also provided. | 07-17-2014 |
| 20140207443 | AUDIO SIGNAL RESTORATION DEVICE AND AUDIO SIGNAL RESTORATION METHOD - A sound source generating unit | 07-24-2014 |
| 20140214411 | ENCODING DEVICE AND ENCODING METHOD - This encoding device ( | 07-31-2014 |
| 20140249806 | AUDIO ENCODING APPARATUS, AUDIO DECODING APPARATUS, AUDIO ENCODING METHOD, AND AUDIO DECODING METHOD - An audio encoding apparatus capable of reducing the bit rate even if a codebook having a larger codebook number is selected in a split multi-rate lattice vector quantization is provided. Sub-vector determining unit ( | 09-04-2014 |
| 20140257799 | SHOUT MITIGATING COMMUNICATION DEVICE - The present invention is a means to provide a user interface that will naturally cause a person to speak at a normal talking volume. It is based on a mechanism whereby the user's speech is compared to a threshold to determine if the user is speaking too loudly and provides feedback to the user. This mechanism could be incorporated into a headset, a cell phone, a smartphone, or into other communication devices. It is useful for operation with or without a headset. | 09-11-2014 |
| 20140278381 | ACOUSTIC SIGNAL PROCESSING SYSTEM CAPABLE OF DETECTING DOUBLE-TALK AND METHOD - An acoustic signal processing system and method. In accordance with an embodiment, the acoustic signal processing system includes an adaptive filter that filters a signal from a frequency band reservation module and generates a filter signal that is received by a subtractor. The subtractor generates an error signal that is used by a double-talk indicator module to generate a control signal that indicates the presence of double-talk. | 09-18-2014 |
| 20140278382 | SIGNAL DECOMPOSITION, ANALYSIS AND RECONSTRUCTION USING HIGH-RESOLUTION FILTER BANKS AND COMPONENT TRACKING - A system and method for representing quasi-periodic waveforms, for example, representing a plurality of limited decompositions of the quasi-periodic waveform. Each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the quasi-periodic waveform. Data-structure attributes are created and used to reconstruct the quasi-periodic waveform. Features of the quasi-periodic wave are tracked using pattern-recognition techniques. The fundamental rate of the signal (e.g., heartbeat) can vary widely, for example by a factor of 2-3 or more from the lowest to highest frequency. To get quarter-phase representations of a component (e.g., lowest frequency “rate” component) that varies over time (by a factor of two to three) many overlapping filters use bandpass and overlap parameters that allow tracking the component's frequency version on changing quarter-phase basis. | 09-18-2014 |
| 20140288925 | BANDWIDTH EXTENSION OF AUDIO SIGNALS - Audio decoder and method therein for supporting bandwidth extension (BWE) of a received signal. The method involves receiving a first signal representing the lower frequency spectrum of a segment of an original audio signal; receiving a second signal, being a BWE signal, representing a higher frequency spectrum of the segment of the original audio signal. The method further comprises determining a degree of voicing in the lower frequency spectrum of the audio signal, based on the received first signal; and selecting a spectral tilt adaptation filter, out of at least two spectral tilt adaptation filters having different spectral attenuation characteristics, based on the determined degree of voicing. The selected spectral tilt adaptation filter is then applied on the received second signal. Thus, a differentiation of spectral tilt in the higher frequency spectrum of a reconstructed audio signal, based on lower frequency spectrum characteristics of the original audio signal is enabled. | 09-25-2014 |
| 20140297270 | SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD - A signal processing apparatus feeding a frame of a signal in frequency domain of a reception voice signal into a sound echo canceler includes a first reception section for receiving frames of the reception voice signal in frequency domain before having a rate-of-speech change process applied; a second reception section for receiving frames of a signal in time domain having the rate-of-speech change process applied by units of frames; and a frequency-domain frame synthesis section for synthesizing a frame of the signal in frequency domain of the reception voice signal based on the signal in time domain having the rate-of-speech change process applied at a frame currently being processed by the signal processing apparatus, and a frame of the reception voice signal in frequency domain corresponding to the signal in time domain having the rate-of-speech change process applied. | 10-02-2014 |
| 20140358529 | Systems, Devices and Methods for Processing Speech Signals - Systems and methods are provided for acquiring a smooth spectrum of speech signals. For example, linear-spectrum-pairs (LSP) parameters of one or more speech signals to be processed are acquired; one or more first cosine values of the LSP parameters are calculated; one or more second cosine values are calculated for one or more predetermined frequency points; one or more first smooth spectrum values of the one or more predetermined frequency points are calculated based on at least information associated with the first cosine values of the LSP parameters and the second cosine values of the predetermined frequency points; and a smooth spectrum of the speech signals is generated based on at least information associated with the first smooth spectrum values of the predetermined frequency points. | 12-04-2014 |
| 20150012266 | Talker Collisions in an Auditory Scene - From a plurality of received voice signals, a signal interval in which there is a talker collision between at least a first and a second voice signal is detected. A processor receives a positive detection result and processes, in response to this, at least one of the voice signals with the aim of making it perceptually distinguishable. A mixer mixes the voice signals to supply an output signal, wherein the processed signal(s) replaces the corresponding received signals. In example embodiments, signal content is shifted away from the talker collision in frequency or in time. The invention may be useful in a conferencing system. | 01-08-2015 |
| 20150294673 | SPEECH AUDIO ENCODING DEVICE, SPEECH AUDIO DECODING DEVICE, SPEECH AUDIO ENCODING METHOD, AND SPEECH AUDIO DECODING METHOD - By the present invention, the number of encoding bits allocated to encoding of extended-band spectrum is reduced while degradation of sound quality in the extended band is suppressed. A band compression unit creates combinations of sub-band spectra in pairs of two samples each in order from a low-range side in a band compression target sub-band, selects a spectrum having a large absolute-value amplitude among the combinations, and arranges the selected spectrum close to the low-range side on a frequency axis. A number-of-units recalculation unit redistributes bits saved in the sub-band for which band compression was performed to a low range outside the extended band, and redistributes the number of units on the basis of the redistributed bits. | 10-15-2015 |
| 20150317995 | MULTI-BAND SIGNAL PROCESSOR FOR DIGITAL AUDIO SIGNALS - A method includes: processing the digital audio input signal to generate M delayed digital audio signal samples; converting the delayed digital audio signal samples to frequency domain representation in N number of frequency bands to compute respective signal spectrum values; determining respective signal level estimates; computing respective frequency domain gain coefficients based on the respective signal level estimates and band gain laws; transforming the frequency domain gain coefficients to time domain representation to produce M time-varying filter coefficients of a processing filter; convolving the M delayed digital audio signal samples with the time-varying filter coefficients to produce the processed digital output signal; and updating the delayed digital audio signal samples in accordance with a sample-by-sample or a predetermined block rate; wherein two of the signal spectrum values for at least two of the frequency bands are updated at different rates; and wherein M and N are positive integer numbers. | 11-05-2015 |
| 20150348562 | APPARATUS AND METHOD FOR IMPROVING AN AUDIO SIGNAL IN THE SPECTRAL DOMAIN - Method of improving audio signal in the spectral domain starts by receiving audio signal that includes signals from sources including speech source and music source. Audio signal is tuned for output by sound output device. Portions of audio signal are analyzed in a spectral domain to determine whether adjustments are required. Analyzing portions of audio signal includes determining whether anomaly is present in frequency band of audio signal in spectral domain by using at least one metric. Metrics include band energy ratios, spectral centroid, spectral tilt, spectral flux, spectral variance, absolute thresholds, and relative thresholds. Audio signal is adjusted to improve audio signal in spectral domain when audio signal is determined to require adjustments. Adjusting audio signal includes adjusting values of the metric in frequency band that is determined to include anomaly to correspond to clustering of metric values for audio signal in spectral domain. Other embodiments are also described. | 12-03-2015 |
| 20150371653 | SYSTEM AND METHOD FOR SPEECH ENHANCEMENT ON COMPRESSED SPEECH - The present disclosure is directed towards a method for speech intelligibility. The method may include receiving, at one or more computing devices, a first speech input from a first user and performing voice activity detection upon the first speech input. The method may also include analyzing a spectral tilt associated with the first speech input, wherein analyzing includes computing an impulse response of a linear predictive coding (“LPC”) synthesis filter in a linear pulse code modulation (“PCM”) domain and wherein the one or more computing devices includes an adaptive high pass filter configured to recalculate one or more linear prediction coefficients. | 12-24-2015 |
| 20150371660 | VOICE DATA PLAYBACK SPEED CONVERSION METHOD AND VOICE DATA PLAYBACK SPEED CONVERSION DEVICE - The present invention addresses the problems of enabling a process of converting voice data playback speed even in a voice data playback device alone. The solution is a voice data playback speed conversion method and a voice data playback speed conversion device, comprising: a step of setting a reference zero cross point from any arbitrary zero cross point; a step of selecting a zero cross point temporally after the reference zero cross point within a first predetermined time range; a step of calculating a reference correlation function in a waveform from the reference zero cross point until a second predetermined time; and a step of calculating a correlation function in a waveform from a plurality of previously selected zero cross points until the second predetermined time, wherein a second reference zero cross point is the zero cross point of the waveform having a correlation function in which a concordance rate of the correlation value between the reference correlation function and the correlation function is the highest value, the difference between the reference zero cross point and the second reference zero cross point is calculated as a basic cycle, and the expansion and contraction of voice data is executed in basic cycle units so as to perform a process of converting the playback speed of the voice data. | 12-24-2015 |
| 20150379991 | INTELLIGENT SOUND SYSTEM/MODULE FOR CABIN COMMUNICATION - Speech communication on board a transport device. In particular, a method for adapting speech data serving for speech communication on board a transport device, to a computer program for executing the method, to a device for adapting speech data serving for speech communication on board a transport device, and also to a transport device with a device of such a type. An embodiment of the method for adapting speech data serving for speech communication on board a transport device comprises: obtaining information relating to at least one tone property, the information relating to the at least one tone property having been derived from recorded speech data appertaining to a person; and adapting speech data appertaining to the person and serving for speech communication on board the transport device on the basis of the obtained information about the at least one tone property. | 12-31-2015 |
| 20160100021 | INFORMATION PROCESSING DEVICE, DESTINATION INFORMATION UPDATING METHOD, AND RECORD MEDIUM - An information processing device, a destination information updating method, and a destination information updating program in a push-type distribution system that updates destination information without impairing a real-time nature while ensuring security are provided. In updating destination information including an identifier of an application installed onto a terminal, a push server that transmits a push message to a terminal in response to a push message transmission request report from a Web server in the push-type distribution system stores destination information before and after updating, receives the push message transmission request report specifying the destination information before updating, converts the destination information before the updating into the destination information after updating, and transmits the push message to the terminal. | 04-07-2016 |
| 20160111105 | FREQUENCY ENVELOPE VECTOR QUANTIZATION METHOD AND APPARATUS - Embodiments of the present application proposes a frequency envelope vector quantization method and apparatus, where the method includes: dividing N frequency envelopes in one frame into N1 vectors; quantizing a first vector in the N1 vectors by using a first codebook, to obtain a code word corresponding to the quantized first vector, where the first codebook is divided into 2 | 04-21-2016 |
| 20160140226 | SYSTEMS AND METHODS FOR PROVIDING SEARCHABLE CUSTOMER CALL INDEXES - A system and method is provided for providing searchable customer call indexes. Consistent with disclosed embodiments, a system may receive call information associated with telephone conversations between callers and a vendor, the call information including an audio recording or transcript for each telephone conversation. The system may also identify one or more keywords from the audio recordings or transcripts and index the call information into one or more indexes based on the identified keywords. Finally, the system may determine search results responsive to a search query based on the indexing. In some embodiments, changes to customer service may be identified based on the search results. | 05-19-2016 |
