Class / Patent application number | Description | Number of patent applications / Date published |
704221000 | Pattern matching vocoders | 59 |
20080208574 | Name synthesis - An automated method of providing a pronunciation of a word to a remote device is disclosed. The method includes receiving an input indicative of the word to be pronounced. The method further includes searching a database having a plurality of records. Each of the records has an indication of a textual representation and an associated indication of an audible representation. At least one output is provided to the remote device of an audible representation of the word to be pronounced. | 08-28-2008 |
20090281799 | TANDEM-FREE VOCODER OPERATIONS BETWEEN NON-COMPATIBLE COMMUNICATION SYSTEMS - Tandem-free vocoder operations (TFO) between non-compatible communication systems may be enabled through hardware modifications at communication elements within each system. In one aspect, each infrastructure entity in System 1 comprises an intra-system TFO Frame Generator G | 11-12-2009 |
20100010812 | SPEECH CODECS - A method and apparatus include a voice activity detection module configured to detect silent frames, and a codec mode selection module configured to determine a codec mode. The voice activity detection module includes a receiver configured to receive a frame, a first determiner configured to determine a first set of parameters from the frame, and a providing unit configured to provide the first set of parameters to the codec mode selection module. The codec mode selection module includes a second determiner configured to determine a second set of parameters in dependence on the first set of parameters, and a selector configured to select a codec mode in dependence on the second set of parameters. | 01-14-2010 |
20100049511 | CODING METHOD, DECODING METHOD, CODER AND DECODER - A coding method, a decoding method, a coder, and a decoder are disclosed herein. A coding method includes: obtaining the pulse distribution, on a track, of the pulses to be encoded on the track; determining a distribution identifier for identifying the pulse distribution according to the pulse distribution; and generating a coding index that includes the distribution identifier. A decoding method includes: receiving a coding index; obtaining a distribution identifier from the coding index, wherein the distribution identifier is configured to identify the pulse distribution, on a track, of the pulses to be encoded on the track; determining the pulse distribution, on a track, of all the pulses to be encoded on the track according to the distribution identifier; and reconstructing the pulse order on the track according to the pulse distribution. | 02-25-2010 |
20100082338 | VOICE PROCESSING APPARATUS AND VOICE PROCESSING METHOD - A voice processing apparatus, which processes a first voice signal, includes: an acoustic analysis part which analyzes a feature quantity of an input second voice signal; a reference range calculation part which calculates a reference range based on the feature quantity; a comparing part which compares the feature quantity and the reference range and outputs a comparison result; and a voice processing part which processes and outputs the input first voice signal based on the comparison result. | 04-01-2010 |
20100274559 | Fixed Codebook Search Method and Searcher - A fixed codebook search method includes initializing a counter, searching for pulses and calculating the value of a cost function Qk, initializing the counter if the Qk value increases, increasing the value of the counter if the Qk value does not increase, judging whether the value of the counter is greater than the threshold value, continuing the search process if the value of the counter is not greater than the threshold value, and ending the whole search process if the value of the counter is greater than the threshold value. | 10-28-2010 |
20120035919 | VOICE RECORDING DEVICE AND METHOD THEREOF - A voice recording method is applied in a recording device that includes a voice receiving unit and a storage unit. The voice receiving unit receives voice signals. The storage unit stores voice models and personal information associated with each voice model. The recording method includes: recording voice signals received by the voice receiving unit and storing the recorded voice signals to the storage unit. Extracting speaker voice features from the recorded speaker's voice. Comparing the extracted features with the voice models to find a match. Obtaining the speaker personal information associated with the voice model when a match is found. Obtaining the storage path of the voice signals stored in the storage unit, then generating an index document according to the obtained voice model and the obtained storage path of the voice signals. | 02-09-2012 |
20140195224 | METHOD AND DEVICE FOR IDENTIFYING AND OBTAINING AMBE ENCODING AND DECODING RATE INFORMATION IN SDP - The present invention provides a method and a device for identifying and obtaining AMBE encoding and decoding rate information in an SDP. The method for identifying the AMBE encoding and decoding rate information in the SDP includes: setting an SDP, which specifically includes: in an m attribute line of the SDP, using a payload type PT value to describe AMBE encoding and decoding, and in an fmtp attribute of an a attribute line of the SDP, identifying, in a format field, a PT value same as that of the m attribute line, and identifying AMBE encoding and decoding rate information in a format specific parameter field; and sending the set SDP to a peer end. The method and the device for identifying and obtaining the AMBE encoding and decoding rate information in the SDP provided by the present invention can implement automatic rate switching in a bearer plane. | 07-10-2014 |
20140330556 | LOW COMPLEXITY REPETITION DETECTION IN MEDIA DATA - Low complexity detection of a time-wise position of a representative segment in media data is described. A subset of offset values is located in a set of offset values in media data using a first type of one or more types of features, which are extractable from (e.g., derivable from components of) the media data. The subset of offset values comprise values that are selected from the set of offset values based on one or more selection criteria. A set of candidate seed time points is identified based on the subset of offset values using a second type of the one or more types of features. | 11-06-2014 |
20150340027 | VOICE RECOGNITION SYSTEM - A voice recognition system includes: a storage unit for storing a voice model of at least one user; a voice acquiring and preprocessing unit for acquiring a voice signal to be recognized, performing a format conversion on the voice signal to be recognized and encoding it; a feature extracting unit for extracting a voice feature parameter from the encoded voice signal to be recognized; a mode matching unit for matching the extracted voice feature parameter with at least one voice model and determining the user that the voice signal to be recognized belongs to. The voice recognition system analyzes the characteristics of the voice starting from the generating principle of the voice, and establishing the voice feature mode of the speaker by using the MFCC parameter to realize the feature recognition algorithm of the speaker, through which the purpose of increasing the speaker detection reliability can be achieved, so that the function of recognizing the speaker can finally be implemented on the electronic products. | 11-26-2015 |
20160071503 | Continuous Score-Coded Pitch Correction - Vocal musical performances may be captured and continuously pitch-corrected at a mobile device for mixing and rendering with backing tracks in ways that create compelling user experiences. In some cases, the vocal performances of individual users are captured in the context of a karaoke-style presentation of lyrics in correspondence with audible renderings of a backing track. Such performances can be pitch-corrected in real-time at the mobile device in accord with pitch correction settings. In some cases, such pitch correction settings code a particular key or scale for the vocal performance or for portions thereof. In some cases, pitch correction settings include a score-coded melody sequence of note targets supplied with, or for association with, the lyrics and/or backing track. In some cases, pitch correction settings are dynamically variable based on gestures captured at a user interface. | 03-10-2016 |
20160118047 | METHOD AND SYSTEM FOR USING CONVERSATIONAL BIOMETRICS AND SPEAKER IDENTIFICATION/VERIFICATION TO FILTER VOICE STREAMS - A method and system for using conversational biometrics and speaker identification and/or verification to filter voice streams during mixed mode communication. The method includes receiving an audio stream of a communication between participants. Additionally, the method includes filtering the audio stream of the communication into separate audio streams, one for each of the participants. Each of the separate audio streams contains portions of the communication attributable to a respective participant. Furthermore, the method includes outputting the separate audio streams to a storage system. | 04-28-2016 |
20160171978 | VOICE RECOGNITION SYSTEM AND CONSTRUCTION METHOD THEREOF | 06-16-2016 |
704222000 | Vector quantization | 28 |
20080262838 | Method, apparatus and computer program product for providing voice conversion using temporal dynamic features - An apparatus for providing voice conversion using temporal dynamic features includes a feature extractor and a transformation element. The feature extractor may be configured to extract dynamic feature vectors from source speech. The transformation element may be in communication with the feature extractor and configured to apply a first conversion function to a signal including the extracted dynamic feature vectors to produce converted dynamic feature vectors. The first conversion function may have been trained using at least dynamic feature data associated with training source speech and training target speech. The transformation element may be further configured to produce converted speech based on an output of applying the first conversion function. | 10-23-2008 |
20080270126 | Apparatus for Vocal-Cord Signal Recognition and Method Thereof - Provided are a vocal-cord recognition apparatus and a method thereof. The vocal-cord signal recognition apparatus includes a vocal-cord signal extracting unit for analyzing a feature of a vocal-cord signal inputted through a throat microphone, and extracting a vocal-cord feature vector from the vocal-cord signal using the analyzing data; and a vocal-cord signal recognition unit for recognizing the vocal-cord signal by extracting the feature of the vocal-cord signal using the vocal-cord signal feature vector extracted at the vocal-cord signal extracting means. | 10-30-2008 |
20080275698 | EXCITATION VECTOR GENERATOR, SPEECH CODER AND SPEECH DECODER - A speech encoder includes an adaptive codebook that generates an adaptive codevector representing a pitch component, a random codebook that generates a random codevector representing a random component, and a synthesis filter that generates a synthetic speech signal by being excited by the adaptive codevector and the random codevector. The random codebook includes an input vector provider configured to provide an input vector, and an excitation vector generator configured to generate an excitation vector as the random codevector by dispersing the input vector by using a fixed pattern. A length of the fixed pattern is shorter than a length of a sub-frame. | 11-06-2008 |
20080288248 | VOICE MODEL FOR SPEECH PROCESSING BASED ON ORDERED AVERAGE RANKS OF SPECTRAL FEATURES - Methods and arrangements for generating a voice model in speech processing. Upon accepting at least two input vectors with spectral features, vectors of ranks are created via ranking values of the spectral features of each input vector, ordered vectors are created via arranging the values of each input vector according to rank, and a vector of ordered average values is created via determining the average of corresponding values of the ordered vectors. Thence, a vector of ordered average ranks is created via determining the sum or average of the vectors of ranks, a vector of ordered ranks is created via ranking the values of the ordered average ranks and a spectral feature vector is created via employing the rank order represented by the vector of ordered ranks to reorder the vector of ordered average ranks. | 11-20-2008 |
20080294429 | Adaptive tilt compensation for synthesized speech - There is provided a method of using an adaptive tilt compensation by a speech decoder. The method comprises receiving a bit stream including a plurality of parameters representative of a speech signal; identifying an adaptive code vector and a fixed code vector using the plurality of parameters; scaling the adaptive code vector and the fixed code vector to generate a scaled adaptive code vector and a scaled fixed code vector; summing the scaled adaptive code vector and the scaled fixed code vector to generate a synthesized output; calculating a first reflection coefficient based on the plurality of parameters representative of the speech signal; multiplying the first reflection coefficient by a factor to generate a tilt factor; and applying the tilt factor to the synthesized output based on an encoding bit rate. | 11-27-2008 |
20090018825 | LOW-COMPLEXITY, NON-INTRUSIVE SPEECH QUALITY ASSESSMENT - A non-intrusive signal quality assessment apparatus includes a feature vector calculator that determines parameters representing frames of a signal and extracts a collection of per-frame feature vectors (φ;(n)) representing structural information of the signal from the parameters. A frame selector preferably selects only frames (Ω\with a feature vector (φ;(n)) lying within a predetermined multi-dimensional window (⊖). Means determine a global feature set (ψ) over the collection of feature vectors (φ;(n)) from statistical moments of selected feature vector components ((1̂,01, . . . O11). A quality predictor predicts a signal quality measure (Qj from the global feature set (ψ)). | 01-15-2009 |
20090037169 | METHOD AND APPARATUS FOR IMPLEMENTING FIXED CODEBOOKS OF SPEECH CODECS AS COMMON MODULE - A method and apparatus for implementing fixed codebooks as a common module are provided. In the method of implementing fixed codebooks of a plurality of speech codecs as a common module, it is possible to include only a part excluding fixed codebooks in a communication terminal or communication system, support various speech codecs without using a chip with high price and high performance, and reduce a memory space that is occupied by the speech codecs by generating a track of a fixed codebook corresponding to a speech codec based on information on the speech codec among the plurality of speech codecs and selecting a codebook vector corresponding to a target signal among codebook vectors constructed with combinations of pulses represented by the generated track. In addition, it is possible to reduce processing complexity as compared with a case of embodying the common fixed codebook module in software by embodying the common fixed codebook module in hardware. In addition, it is possible to improve the entire voice processing performance by applying the latest fixed codebook searching algorithm only to the common fixed codebook, thereby easily applying the latest fixed codebook searching algorithm to the entire voice codec. | 02-05-2009 |
20090125302 | Stabilization and Glitch Minimization for CCITT Recommendation G.726 Speech CODEC During Packet Loss Scenarios by Regressor Control and Internal State Updates of the Decoding Process - This invention decoded encoded speech using alternative parameters upon detection of a lost packet. Upon detection of a first good packet following packet loss, this invention uses second alternative parameters intermediate between the default parameters and the alternative parameters for a predetermined interval. Thereafter the invention reverts to the default parameters. This minimizes glitches in the decoded speech upon packet loss. This invention is suitable for use in decoding speech data encoded in the CCITT Recommendation G.726 ADPCM based speech coding standard. | 05-14-2009 |
20090138261 | SPEECH CODER USING AN ORTHOGONAL SEARCH AND AN ORTHOGONAL SEARCH METHOD - Speech is coded using an orthogonal search by calculating a search reference value. An adaptive codevector representing a pitch component is generated. A random codevector representing a random component is also generated. The orthogonal search further includes generating a synthetic speech signal by a synthesis filter being excited by the adaptive codevector and the random codevector. A distortion between the input speech signal and the synthetic speech signal is calculated. One random codevector that minimizes the distortion is selected. | 05-28-2009 |
20090299738 | VECTOR QUANTIZING DEVICE, VECTOR DEQUANTIZING DEVICE, VECTOR QUANTIZING METHOD, AND VECTOR DEQUANTIZING METHOD - A vector quantizing device for dividing a sequence of vectors and quantizing them with an enhanced performance of vector quantization by using information on the correlation between the high and low order that the vector sequence has. The vector quantizing device ( | 12-03-2009 |
20090326932 | Reducing Computational Complexity in Determining the Distance from Each of a Set of Input Points to Each of a Set of Fixed Points - An aspect of the present invention takes advantage of the fact that the coordinates of fixed points do not change, and thus the energy (sum of squares of the coordinates defining the vector) of each fixed point is computed and stored. The energy of each variable input point may also be computed. The distance between each pair of fixed and input points is computed based on the respective energies and the dot product. | 12-31-2009 |
20100017204 | ENCODING DEVICE AND ENCODING METHOD - Provided is a voice encoding device which can accurately encode a spectrum shape of a signal having a strong tonality such as a vowel. The device includes: a sub-band constituting unit ( | 01-21-2010 |
20100036658 | Speech compression and decompression apparatuses and methods providing scalable bandwidth structure - A speech compression apparatus including: a first band-transform unit transforming a wideband speech signal to a narrowband low-band speech signal; a narrowband speech compressor compressing the narrowband low-band speech signal and outputting a result of the compressing as a low-band speech packet; a decompression unit decompressing the low-band speech packet and obtaining a decompressed wideband low-band speech signal; an error detection unit detecting an error signal that corresponds to a difference between the wideband speech signal and the decompressed wideband low-band speech signal; and a high-band speech compression unit compressing the error signal and a high-band speech signal of the wideband speech signal and outputting the result of the compressing as a high-band speech packet. | 02-11-2010 |
20100057448 | MULTICODEBOOK SOURCE-DEPENDENT CODING AND DECODING - A method for coding data, includes: grouping data into frames; classifying the frames into classes; for each class, transforming the frames belonging to the class into filter parameter vectors, which are extracted from the frames by applying a first mathematical transformation; for each class, computing a filter codebook based on the filter parameter vectors belonging to the class; segmenting each frame into subframes; for each class, transforming the subframes belonging to the class into source parameter vectors, which are extracted from the subframes by applying a second mathematical transformation based on the filter codebook computed for the corresponding class; for each class, computing a source codebook based on the source parameter vectors belonging to the class; and coding the data based on the computed filter and source codebooks. | 03-04-2010 |
20100174539 | METHOD AND APPARATUS FOR VECTOR QUANTIZATION CODEBOOK SEARCH - A vector quantization codebook search method and apparatus use support vector machines (“SVMs”) to compute a hyperplane, where the hyperplane is used to separate codebook elements into a plurality of bins. During execution, a controller determines which of the plurality of bins contains a desired codebook element, and then searches the determined bin. Codebook search complexity is reduced and an exhaustive codebook search is selectively avoided. | 07-08-2010 |
20100185442 | ADAPTIVE SOUND SOURCE VECTOR QUANTIZING DEVICE AND ADAPTIVE SOUND SOURCE VECTOR QUANTIZING METHOD - It is an object to disclose an adaptive sound source vector quantizing device, etc. that can be configured to improve quantizing accuracy in adaptive sound source vector quantization to be carried out for every sub-frame. In this device, a pitch period designating unit ( | 07-22-2010 |
20110010169 | Reduced-Complexity Vector Indexing and De-indexing - This invention relates to indexing an input vector contained in a set of vectors of a plurality of sets of vectors. The indexing comprises performing, in case that the input vector is contained in a set of vectors of a pre-defined group of one or more sets of vectors of the plurality of sets of vectors, a specific processing that is adapted to a characteristic of the sets of vectors in the pre-defined group of sets of vectors and is only applicable in case of input vectors contained in sets of vectors with the characteristic. The indexing further comprises performing, in case that the input vector is not contained in a set of vectors of the pre-defined group of sets of vectors, a general processing. The invention further relates to an according determining of a target vector contained in a set of vectors of a plurality of sets of vectors based on an index associated with said target vector. | 01-13-2011 |
20110040558 | SCALABLE ENCODING APPARATUS, SCALABLE DECODING APPARATUS, SCALABLE ENCODING METHOD, SCALABLE DECODING METHOD, COMMUNICATION TERMINAL APPARATUS, AND BASE STATION APPARATUS - A scalable encoding apparatus, a scalable decoding apparatus and the like are disclosed which can achieve a band scalable LSP encoding that exhibits both a high quantization efficiency and a high performance. In these apparatuses, a narrow band-to-wide band converter receives and converts a quantized narrow band LSP to a wide band, and then outputs the quantized narrow band LSP as converted (i.e., a converted wide band LSP parameter) to an LSP-to-LPC converter. The LSP-to-LPC converter converts the quantized narrow band LSP as converted to a linear prediction coefficient and then outputs it to a pre-emphasizer. The pre-emphasizer calculates and outputs the pre-emphasized linear prediction coefficient to an LPC-to-LSP converter. The LPC-to-LSP converter converts the pre-emphasized linear prediction coefficient to a pre-emphasized quantized narrow band LSP as wide band converted, and then outputs it to a prediction quantizer. | 02-17-2011 |
20110137645 | METHOD AND APPARATUS OF COMMUNICATION - The invention pertains to a method and apparatus of efficient encoding and decoding of vector quantized data. The method and system explores and implements sub-division of a quantization vector space comprising class-leader vectors and representation of the class-leader vectors by a set of class-leader root-vectors facilitating faster encoding and decoding, and reduced storage requirements. | 06-09-2011 |
20120078618 | METHOD AND APPARATUS FOR GENERATING LATTICE VECTOR QUANTIZER CODEBOOK - A method and an apparatus for generating a lattice vector quantizer codebook are disclosed. The method includes: storing an eigenvector set that includes amplitude vectors and/or length vectors, where the amplitude vectors and/or length vectors are different from each other and correspond to a root leader of a lattice vector quantizer; storing storage addresses of the amplitude vectors and length vectors, where the amplitude vectors and length vectors correspond to the root leader and are in the eigenvector set; and generating a lattice vector quantizer codebook according to the eigenvector set and the storage addresses. | 03-29-2012 |
20120203547 | SYSTEM AND METHOD FOR IMPROVING ROBUSTNESS OF SPEECH RECOGNITION USING VOCAL TRACT LENGTH NORMALIZATION CODEBOOKS - Disclosed are systems, methods, and computer readable media for performing speech recognition. The method embodiment comprises selecting a codebook from a plurality of codebooks with a minimal acoustic distance to a received speech sample, the plurality of codebooks generated by a process of (a) computing a vocal tract length for a each of a plurality of speakers, (b) for each of the plurality of speakers, clustering speech vectors, and (c) creating a codebook for each speaker, the codebook containing entries for the respective speaker's vocal tract length, speech vectors, and an optional vector weight for each speech vector, (2) applying the respective vocal tract length associated with the selected codebook to normalize the received speech sample for use in speech recognition, and (3) recognizing the received speech sample based on the respective vocal tract length associated with the selected codebook. | 08-09-2012 |
20130173261 | AUDIO QUANTIZATION CODING AND DECODING DEVICE AND METHOD THEREOF - The present disclosure provides an audio quantization coding and decoding device and a method thereof. In the method, before a quantization coding process is performed on a digital signal, the signal is pre-processed, the digital signal is split into multiple frames based on positive and negative half periods of the signal, and all audio data between two adjacent zero-crossing points belongs to the same positive and negative half periods, so as to have the same sign-bit. A pre-processing module groups the numeric data belonging to the same positive and negative half periods into the same frame. When coding, an audio quantization coding module only needs to record a sign-bit of the frame at a head of the frame, so the sign-bit of each batch of voice data in the frame may be omitted to reduce a data amount or improve a resolution of each batch of voice data. | 07-04-2013 |
20130325457 | ENCODING APPARATUS, DECODING APPARATUS, ENCODING METHOD AND DECODING METHOD - An encoding apparatus includes a first layer encoder that encodes a signal, a first layer decoder that decodes first layer encoded data, a first layer error transform coefficient calculator that transforms a first layer error signal into a frequency domain and a second layer encoder that encodes the first layer error transform coefficient to acquire second layer encoded data. The second layer encoder includes a band determiner that determines a band to be encoded by the second layer encoder, and a first shape vector encoder that refers the first layer error transform coefficient included in the band to generate a first shape vector and first shape encoded information, a target gain calculator calculates target gain per subband, a gain vector generator generates a gain vector using a plurality of target gains, and a gain vector encoder encodes the gain vector to acquire gain encoded information. | 12-05-2013 |
20150317992 | VECTOR QUANTIZATION OF ALGEBRAIC CODEBOOK WITH HIGH-PASS CHARACTERISTIC FOR POLARITY SELECTION - 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 ( | 11-05-2015 |
20150332690 | CODING VECTORS DECOMPOSED FROM HIGHER-ORDER AMBISONICS AUDIO SIGNALS - In general, techniques are described for coding of vectors decomposed from higher order ambisonic coefficients. A device comprising a processor and a memory may perform the techniques. The processor may be configured to obtain from a bitstream data indicative of a plurality of weight values that represent a vector that is included in a decomposed version of the plurality of HOA coefficients. Each of the weight values may correspond to a respective one of a plurality of weights in a weighted sum of code vectors that represents the vector and that includes a set of code vectors. The processor may further be configured to reconstruct the vector based on the weight values and the code vectors. The memory may be configured to store the reconstructed vector. | 11-19-2015 |
20150332691 | DETERMINING BETWEEN SCALAR AND VECTOR QUANTIZATION IN HIGHER ORDER AMBISONIC COEFFICIENTS - In general, techniques are described for coding of vectors decomposed from higher-order ambisonic coefficients. A device comprising a memory and a processor may perform the techniques. The memory may be configured to store audio data. The processor may be configured to determine whether to perform vector dequantization or scalar dequantization with respect to a decomposed version of the plurality of HOA coefficients. | 11-19-2015 |
20150332692 | SELECTING CODEBOOKS FOR CODING VECTORS DECOMPOSED FROM HIGHER-ORDER AMBISONIC AUDIO SIGNALS - In general, techniques are described for performing codebook selection when coding vectors decomposed from higher-order ambisonic coefficients. A device comprising a memory and a processor may perform the techniques. The memory may be configured to store a plurality of codebooks to use when performing vector dequantization with respect to a vector quantized spatial component of a soundfield. The vector quantized spatial component may be obtained through application of a decomposition to a plurality of higher order ambisonic coefficients. The processor may be configured to select one of the plurality of codebooks. | 11-19-2015 |
20150380004 | DERIVATION OF PROBABILISTIC SCORE FOR AUDIO SEQUENCE ALIGNMENT - A match score provides a semantically-meaningful quantification of the aural similarity of two chromae from two corresponding audio sequences. The match score can be applied to the chroma pairs of two corresponding audio sequences, and is independent of the lengths of the sequences, thereby permitting comparisons of matches across subsequences of different length. Accordingly, a single cutoff match score to identify “good” audio subsequence matches can be determined and has both good precision and good recall metrics. A function for determining the match score is determined by establishing a function P | 12-31-2015 |
704223000 | Excitation patterns | 18 |
20080281587 | Audio Encoding Apparatus, Audio Decoding Apparatus, Communication Apparatus and Audio Encoding Method - An audio encoding apparatus and the like are disclosed which can improve the sound quality of encoded audio signals even in a case of scalable CELP encoding the audio signals in sections that vary with time. In this apparatus, an enhancement layer extended adaptive codebook generating part ( | 11-13-2008 |
20080281588 | SPEECH PROCESSING METHOD AND APPARATUS, STORAGE MEDIUM, AND SPEECH SYSTEM - A speech processing apparatus includes a spectrum envelope extracting unit which extracts the spectrum envelope of an input speech signal, a spectrum envelope deforming unit which applies deformation to the spectrum envelope to generate a deformed spectrum envelope, a spectrum fine structure extracting unit which extracts the spectrum fine structure of the input speech signal, a deformed spectrum generating unit which generates a deformed spectrum by combining the deformed spectrum envelope with the spectrum fine structure, and a speech generating unit which generates an output speech signal on the basis of the deformed spectrum. This apparatus emits a disrupting sound based on the output speech signal to prevent a third party from eavesdropping on a conversation. | 11-13-2008 |
20090006085 | AUTOMATED CALL CLASSIFICATION AND PRIORITIZATION - An automated voice message or caller prioritization system that extracts words, prosody, and/or metadata from a voice input. The data extracted is classified with a statistical classifier into groups of interest. These groups could indicate the likelihood that a call is urgent versus nonurgent, from someone the user knows well versus someone that the user only knows casually or not at all, from someone using a mobile phone versus a landline, or a business call versus a personal calls. The system then can determine an action based on results of the groups, including the display of likely category labels on the message. Call handling and display actions can be defined by user preferences. | 01-01-2009 |
20090012782 | Method and Arrangements for Coding Audio Signals - According to the invention, an excitation signal is generated as a result of sampled excitation values in order to excite an audio synthesis filter, the generated sampled excitation values being continuously stored in an adaptive codebook. A noise generator is provided which continuously generates random sampled values. A sequence of the stored sampled excitation values is selected from the adaptive codebook based on a fed audio fundamental frequency parameter by means of which a time gap between the sequence that is to be selected and the actual time reference is predefined. The excitation signal is generated by mixing the selected sequence with a random sequence encompassing actual random sampled valued of the noise generator. | 01-08-2009 |
20090018826 | Methods, Systems and Devices for Speech Transduction - Methods, systems, and devices for speech transduction are disclosed. One aspect of the invention involves a computer-implemented method in which a computer receives far-field acoustic data acquired by one or more microphones. The far-field acoustic data are analyzed. The far-field acoustic data are modified to reduce characteristics of the far-field acoustic data that are incompatible with human speech characteristics of near-field acoustic data. | 01-15-2009 |
20090043572 | PULSE ALLOCATING METHOD IN VOICE CODING - A pulse allocating method capable of coding stereophonic voice signals efficiently. In the fixed code note retrievals (ST | 02-12-2009 |
20090043573 | METHOD AND APPARATUS FOR RECOGNIZING A SPEAKER IN LAWFUL INTERCEPTION SYSTEMS - A method and apparatus for identifying a speaker within a captured audio signal from a collection of known speakers. The method and apparatus receive or generate voice representations for each known speakers and tag the representations according to meta data related to the known speaker or to the voice. The representations are grouped into one or more groups according to the indices. When a voice to be recognized is introduced, characteristics are determined according to which the groups are prioritized, so that the representations participating only in part of the groups are matched against the o voice to be identified, thus reducing identification time and improving the statistical significance. | 02-12-2009 |
20090043574 | Speech coding system and method using bi-directional mirror-image predicted pulses - There is provided a method of decoding speech data generated from a speech signal. The method comprises receiving the speech data having at least one main pulse in a subframe of the speech data; generating a first predicted pulse, based on the at least one main pulse, on one side of the main pulse in the subframe of the speech data, wherein the first predicted pulse has a lower gain than the main pulse; generating a second predicted pulse, as a mirror image of the first predicted pulse on a reverse time scale, on the other side of the main pulse in the subframe of the speech data; reconstructing the speech signal using the at least one main pulse, the first predicted pulse and the second predicted pulse. | 02-12-2009 |
20090094026 | METHOD OF DETERMINING AN ESTIMATED FRAME ENERGY OF A COMMUNICATION - A method of processing a communication includes determining an estimated excitation energy component of a subframe of a coded frame. A filter energy component of the subframe is also estimated. Determining an estimated energy of the subframe is based upon the estimated excitation energy component and the estimated filter energy component. This technique allows for estimating frame energy of a communication such as a voice communication without having to fully decode the communication. | 04-09-2009 |
20090164211 | SPEECH ENCODING APPARATUS AND SPEECH ENCODING METHOD - Provided is a voice encoding device for acquiring a satisfactory sound quality by making sufficient use of a tendency according to the noisiness or noiselessness of an input signal to be encoded. In this voice encoding device, a weight adding unit ( | 06-25-2009 |
20090240494 | VOICE ENCODING DEVICE AND VOICE ENCODING METHOD - Provided is a voice encoding device which performs voice encoding by a fixed code book effectively using a bit. In the voice encoding device, a position/polarity calculation unit ( | 09-24-2009 |
20090248406 | CODING METHOD, ENCODER, AND COMPUTER READABLE MEDIUM - A coding method is adapted to select different codebook search algorithms according to varied types of input signals. An encoder using the coding method is also provided. As appropriate search algorithms may be selected according to all possible structural features of the input signals, certain types of signals for which satisfactory results may be obtained through simple computations may match with search algorithms suitable for these signal types and having low computation complexities, so as to achieve better performance with fewer system resources. Meanwhile, other types of signals that need complicated computations may be processed by more sophisticated search algorithms, thereby ensuring the coding quality. | 10-01-2009 |
20090271184 | SCALABLE ENCODING DEVICE, AND SCALABLE ENCODING METHOD - Disclosed is a scalable encoding device capable of reducing an encoding rate thereby to reduce a circuit scale while preventing sound quality deterioration of a decoded signal. In this device, an extension layer is coarsely divided into a system for processing a first channel and a system for processing a second channel. A sound source prediction unit ( | 10-29-2009 |
20090276211 | METHOD AND DEVICE FOR UPDATING STATUS OF SYNTHESIS FILTERS - A method and device for updating statuses of synthesis filters are provided. The method includes: exciting a synthesis filter corresponding to a first encoding rate by using an excitation signal of the first encoding rate, outputting reconstructed signal information, and updating status information of the synthesis filter and a synthesis filter corresponding to a second encoding rate. In the present disclosure, the status of the synthesis filter corresponding to the current rate and the statuses of the synthesis filters at other rates are updated. Thus, synchronization between the statuses of the synthesis filters corresponding to different rates at the encoding terminal may be realized, thereby facilitating the consistency of the reconstructed signals of the encoding and decoding terminals when the encoding rate is switched, and improving the quality of the reconstructed signal of the decoding terminal. | 11-05-2009 |
20090326933 | Codebook generation system and associated methods - A codebook generation system and associated methods are generally described herein. | 12-31-2009 |
20100114568 | APPARATUS FOR PROCESSING AN AUDIO SIGNAL AND METHOD THEREOF - An apparatus for processing an audio signal and method thereof are disclosed, by which extracting, by an audio processing apparatus, scheme type information indicating either time excitation scheme or frequency excitation scheme for each of a plurality of subframes included in current frame; when the time excitation scheme is applied to at least one subframe among the plurality of subframes according to the scheme type information, extracting mode information representing bit allocation of codebook index for the current frame; and, when the mode information is extracted, decoding the at least one subframe using the mode information according to the time excitation scheme. | 05-06-2010 |
20100179807 | AUDIO ENCODING DEVICE AND AUDIO ENCODING METHOD - Provided is an audio encoding device capable of improving performance of an adaptive codebook and improving quality of a decoded audio. In this audio encoding device, an adaptive codebook ( | 07-15-2010 |
20150073784 | Adaptive Bandwidth Extension and Apparatus for the Same - In one embodiment of the present invention, a method of decoding an encoded audio bitstream and generating frequency bandwidth extension includes decoding the audio bitstream to produce a decoded low band audio signal and generate a low band excitation spectrum corresponding to a low frequency band. A sub-band area is selected from within the low frequency band using a parameter which indicates energy information of a spectral envelope of the decoded low band audio signal. A high band excitation spectrum is generated for a high frequency band by copying a sub-band excitation spectrum from the selected sub-band area to a high sub-band area corresponding to the high frequency band. Using the generated high band excitation spectrum, an extended high band audio signal is generated by applying a high band spectral envelope. The extended high band audio signal is added to the decoded low band audio signal to generate an audio output signal having an extended frequency bandwidth. | 03-12-2015 |