Abstract: In an encoding process, a CPU transforms an audio signal from the real-time domain to the frequency domain, and transforms the signal into spectra consisting of MDCT coefficients. The CPU separates the audio signal into several frequency bands, and performs bit shifting in each band such that the MDCT coefficients can be expressed with pre-configured numbers of bits. The CPU re-quantizes the MDCT coefficients at a precision differing for each band, and transmits the values acquired thereby and shift bit numbers as encoded data. Meanwhile, in a decoding process, a CPU receives encoded data and inverse re-quantizes and inverse bit shifts the data, thereby restoring the MDCT coefficients. Furthermore, the CPU transforms the data from frequency domain to the real-time domain by using the inverse MDCT, and restores and outputs the audio signal.

Abstract: An encoder comprising an input for inputting frames of an audio signal in a frequency band, at least a first excitation block for performing a first excitation for a speech like audio signal, and a second excitation block for performing a second excitation for a non-speech like audio signal. The encoder further comprises a filter for dividing the frequency band into a plurality of sub bands each having a narrower bandwidth than the frequency band. The encoder also comprises an excitation selection block for selecting one excitation block among the at least first excitation block and the second excitation block for performing the excitation for a frame of the audio signal on the basis of the properties of the audio signal at least at one of the sub bands. The invention also relates to a device, a system, a method and a storage medium for a computer program.

Abstract: A method, system and computer program for encoding speech according to a source-filter model. The method comprises deriving a spectral envelope signal representative of a modelled filter and a first remaining signal representative of a modelled source signal, and deriving a second remaining signal from the first remaining signal by, at intervals during the encoding: exploiting a correlation between approximately periodic portions in the first remaining signal to generate a predicted version of a later portion from a stored version of an earlier portion, and using the predicted-version of the later portion to remove an effect of said periodicity from the first remaining signal. The method further comprises, once every number of intervals, transforming the stored version of the earlier portion of the first remaining signal prior to generating the predicted version of the respective later portion.

Abstract: A codebook generation system and associated methods are generally described herein. For instance, a codebook generation agent (CGA) may implement techniques for generating one or more matrix codebooks from vector codebooks. The CGA may be implemented in mobile devices (e.g., stations, subscriber units, handsets, laptops, etc.). In this regard, the dynamic generation of matrix codebooks rather than having them stored on the mobile device enables the mobile device to utilize the memory normally consumed by the matrix codebooks in support of other features and/or services.

Abstract: A codebook generation system and associated methods are generally described herein. For instance, a codebook generation agent (CGA) may implement techniques for generating one or more matrix codebooks from vector codebooks. The CGA may be implemented in mobile devices (e.g., stations, subscriber units, handsets, laptops, etc.). In this regard, the dynamic generation of matrix codebooks rather than having them stored on the mobile device enables the mobile device to utilize the memory normally consumed by the matrix codebooks in support of other features and/or services.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: In an embodiment, a method of frequency domain post-processing is disclosed. The method includes applying adaptive modification gain factor to each frequency coefficient, and determining gain factors based on Local Masking Magnitude and Local Masked Magnitude.

Abstract: A method of encoding one or more parent blocks of values, the number of values being the length of each block, the method comprising for each parent block: (a) determining a first sum of values in the parent block; (b) splitting the parent block into smaller subblocks; (c) for at least one of the subblocks, determining a second sum of the values in the subblock, selecting a likelihood table from the plurality of likelihood tables based on said first sum of values in the parent block and encoding the second sum using the likelihood table; (d) designating each subblock a parent block; (e) carrying out steps (a), (b), (c) and (d) until at least one parent block reaches a predetermined condition.

Abstract: A method and device for improving coding efficiency in audio coding. From the pitch values of a pitch contour of an audio signal, a plurality of simplified pitch contour segments are generated to approximate the pitch contour, based on one or more pre-selected criteria. The contour segments can be linear or non-linear with each contour segment represented by a first end point and a second end point. If the contour segments are linear, then only the information regarding the end points, instead of the pitch values, are provided to a decoder for reconstructing the audio signal. The contour segment can have a fixed maximum length or a variable length, but the deviation between a contour segment and the pitch values in that segment is limited by a maximum value.

Abstract: A method, device and system for signal encoding and decoding are disclosed. The method includes: encoding a core layer signal to obtain a core layer signal code; selecting an enhancement sample point that requires enhancement layer signal encoding according to the core layer signal code and the number of bits that can be used by an enhancement layer; obtaining an enhancement layer signal code of the enhancement sample point; and outputting a bit stream, where the bit stream includes the core layer signal code and the enhancement layer signal code. In embodiments of the present invention, according to the number of bits that can be used by the enhancement layer, the enhancement sample point that requires enhancement layer signal encoding is selected; the enhancement layer signal of the selected enhancement sample point is encoded and decoded; when no sufficient bits are available for the enhancement layer, the enhancement quality of the core layer can be improved.

Abstract: Disclosed is a code conversion method to convert a first code sequence conforming to a first speech coding scheme into a second code sequence conforming to a second speech coding scheme. The method includes the following steps. The first step discriminates whether the first code sequence corresponds to a speech part or to a non-speech part, and generates a numerical value that indicates the discrimination result as a control flag. The second step converts the first code sequence into the second code sequence and outputs said second code sequence, when the value of the control flag corresponds to the speech part. The third step outputs the second code sequence that corresponds to the value of the control flag, when the value of the control flag corresponds to the non-speech part.

Abstract: A system for coding a hierarchical audio signal, comprising, at least, a core layer using parametric coding by analysis by synthesis in a first frequency band, a band extension layer for widening said first frequency band into a second frequency band, or wideband. The system also comprises a wideband audio coding quality enhancement layer based on transform coding using a spectral parameter obtained from said band extension layer. Application to transmitting speech and/or audio signals over packet networks.

Abstract: A wideband speech encoder according to one embodiment includes a filter bank having a lowband processing path and a highband processing path. The processing paths have overlapping frequency responses. A first encoder is configured to encode a speech signal produced by the lowband processing path according to a first coding methodology. A second encoder is configured to encode a speech signal produced by the highband processing path according to a second coding methodology that is different than the first coding methodology.

Abstract: Provided is an audio encoding device which can detect an optimal pitch pulse when using pitch pulse information as redundant information.

Abstract: Encoding a sequence of digital speech samples into a bit stream includes dividing the digital speech samples into one or more frames, computing model parameters for a frame, and quantizing the model parameters to produce pitch bits conveying pitch information, voicing bits conveying voicing information, and gain bits conveying signal level information. One or more of the pitch bits are combined with one or more of the voicing bits and one or more of the gain bits to create a first parameter codeword that is encoded with an error control code to produce a first FEC codeword that is included in a bit stream for the frame. The process may be reversed to decode the bit stream.

Abstract: A system and method disclosed for using and updating a database of template responses for a live agent in response to user communications. The method includes computing an average string distance between each response from a live agent and a template, use to generate the response, modifying the computed average string distance based on a customer satisfaction score associated with each response and selecting a response that minimizes the computed average string distance and maximizes customer satisfaction. Upon receiving a further communication on a certain issue, the system presents a prototype response that has been added to the template database to the live agent for use in generating a response to the further communication that reduces handling time and increases customer satisfaction.

Abstract: A fixed code book (FCB) search device simplifies an error minimizing process and reduces a calculation amount so as to prevent deterioration of a coding performance. The FCB search device includes a pulse shape convolution inverse filter having an inverse feature of a pulse diffusion filter and supplied with an ideal residual signal; a pulse candidate preparatory selector that pre-selects a plurality of pulse candidates from the ideal residual signal to which the inverse filter is applied; and a pulse candidate final selector that finally selects one pulse from the selected candidates. Using this configuration, a search is made for an algebra code book to which the pulse diffusion is applied.

Abstract: A high quality speech is reproduced with a small data amount in speech coding and decoding for performing compression coding and decoding of a speech signal to a digital signal. In speech coding method according to a code-excited linear prediction (CELP) speech coding, a noise level of a speech in a concerning coding period is evaluated by using a code or coding result of at least one of spectrum information, power information, and pitch information, and various excitation codebooks are used based on an evaluation result.

Abstract: A codebook generation system and associated methods are generally described herein. For instance, a codebook generation agent (CGA) may implement techniques for generating one or more matrix codebooks from vector codebooks. The CGA may be implemented in mobile devices (e.g., stations, subscriber units, handsets, laptops, etc.). In this regard, the dynamic generation of matrix codebooks rather than having them stored on the mobile device enables the mobile device to utilize the memory normally consumed by the matrix codebooks in support of other features and/or services.

Abstract: Provided are a method and apparatus for encoding and decoding an audio signal. According to the present application, a signal of a high frequency band above a preset frequency band is adaptively encoded or decoded in the time domain or in the frequency domain by using a signal of a low frequency band below the preset frequency band. As such, the sound quality of a high frequency signal is not deteriorate even when an audio signal is encoded or decoded by using a small number of bits and thus coding efficiency may be maximized.

Abstract: The present invention relates to means and methods of classifying speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.

Abstract: A wideband speech decoding apparatus has means for producing an excitation signal from coded data, means for producing a synthesis filter, and means for decoding a speech signal from the excitation signal and the synthesis filter. The wideband speech decoding apparatus comprises acquisition means for acquiring identification information which identifies the speech signal to be decoded is narrowband. The wideband speech decoding apparatus further comprises control means for controlling decoding means based on the identification information.

Abstract: A method and apparatus for expanding a bandwidth of an input narrowband voice signal is provided. The narrowband voice signal is analyzed separately for each frame, and a Degree of Voicing (DV) and a Degree of Stationary (DS) are calculated depending on the analysis. A Degree of Difficulty of Bandwidth Expansion (DDBWE) of the narrowband voice signal is calculated based on DV and DS. Bandwidth expansion is controlled according to DDBWE.

Abstract: An encoding device reduces the encoding distortion as compared to the conventional technique and obtains a preferable sound quality for auditory sense. In the encoding device, a shape quantization unit quantizes the shape of an input spectrum with a small number of pulse positions and polarities. The shape quantization unit sets a pulse amplitude width to be searched later upon search of the pulse position to a value not greater than the pulse amplitude width which has been searched previously. A gain quantization unit calculates a gain of a pulse searched by the shape quantization unit for each of bands.

Abstract: A method of encoding one or more parent blocks of values, the number of values being the length of each block, the method comprising for each parent block: (a) determining a first sum of values in the parent block; (b) splitting the parent block into smaller subblocks; (c) for at least one of the subblocks, determining a second sum of the values in the subblock, selecting a likelihood table from the plurality of likelihood tables based on said first sum of values in the parent block and encoding the second sum using the likelihood table; (d) designating each subblock a parent block; (e) carrying out steps (a), (b), (c) and (d) until at least one parent block reaches a predetermined condition.

Abstract: 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.

Abstract: Methods, and corresponding codec-containing devices are provided that have source coding schemes for encoding a component of an excitation. In some cases, the source coding scheme is an enumerative source coding scheme, while in other cases the source coding scheme is an arithmetic source coding scheme. In some cases, the source coding schemes are applied to encode a fixed codebook component of the excitation for a codec employing codebook excited linear prediction, for example an AMR-WB (Adaptive Multi-Rate-Wideband) speech codec.

Abstract: The invention aims at constructing improved dictionaries of CELP excitation vectors for coding/decoding digital audio signals. Usually, each vector of dimension N comprises pulses capable of occupying N valid positions. The invention concerns the construction of dictionaries with particular structure by: providing a common sequence of pulses forming a base pattern; and assigning the base pattern to each excitation vector of the dictionary, based on one or more occurrences at one or more respective positions among said N valid positions. The invention also concerns a combination of dictionaries thus constructed with optionally standard multipulse dictionaries, by union or summation or cascading.

Abstract: A scalable encoding device capable of reducing an encoding rate to reduce a circuit scale while preventing sound quality deterioration of a decoded signal. 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 predictor for processing the first channel predicts a drive sound source signal of the first channel from a drive sound source signal of a monaural signal, and outputs the predicted drive sound source signal through a multiplier to a first CELP encoder. A sound source predictor for processing the second channel predicts the drive sound source signal of the second channel from the drive sound source signal of the monaural signal and the output from the first CELP encoder, and outputs the predicted drive sound source signal through a multiplier to a second CELP encoder.

Abstract: Provides is an embedded code-excited linear prediction speech coding/decoding apparatus and method that can deal with the capacity change of speech transmission channel by modeling an error signal not coded at a core speech coder based on a transmission rate in a multiple pulse search mode or gain compensation mode and then transmitting it in an optimum mode. The apparatus includes a core speech coding unit for coding an input speech signal with spectral envelop and an excitation signal, a transmission rate determination unit for allocating the number of bits additionally allowed depending on a capacity of a transmission channel, and an embedded excitation signal coding unit for coding a residual excitation signal that is not coded in the core speech coding unit based on the number of additionally allowed bits using one of a multiple pulse excitation coding mode and a gain compensation mode.

Abstract: A double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

Abstract: In one embodiment, a method of generating a highband excitation signal includes harmonically extending the spectrum of a signal that is based on a lowband excitation signal; calculating a time-domain envelope of a signal that is based on the lowband excitation signal; and modulating a noise signal according to the time-domain envelope. The method also includes combining (A) a harmonically extended signal based on a result of the harmonically extending and (B) a modulated noise signal based on a result of the modulating. In this method, the highband excitation signal is based on a result of the combining.

Abstract: A wideband speech coding apparatus comprises a wideband speech coding unit configured to code an input speech signal at a bit rate which is set in advance in accordance with a wideband speech signal. The apparatus further comprises an identification unit configured to identify whether the input speech signal is a wideband speech signal or a narrowband speech signal. The apparatus further comprises a control unit configured to cause the wideband speech coding unit to code the input speech signal, when the identification unit identifies that the input speech signal is the wideband speech signal, and to cause the wideband speech coding unit to raise the bit rate to code the input speech signal, when the identification unit identifies that the input speech signal is the narrowband speech signal. The apparatus further comprises an output unit to output the input speech signal coded by the wideband speech coding unit.

Abstract: A communication device includes memory, an input interface, a processing module, and a transmitter. The processing module receives a digital signal from the input interface, wherein the digital signal includes a desired digital signal component and an undesired digital signal component. The processing module identifies one of a plurality of codebooks based on the undesired digital signal component. The processing module then identifies a codebook entry from the one of the plurality of codebooks based on the desired digital signal component to produce a selected codebook entry. The processing module then generates a coded signal based on the selected codebook entry, wherein the coded signal includes a substantially unattenuated representation of the desired digital signal component and an attenuated representation of the undesired digital signal component. The transmitter converts the coded signal into an outbound signal in accordance with a signaling protocol and transmits it.

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

Abstract: A method and apparatus are provided for processing a set of communicated signals associated with a set of muscles, such as the muscles near the larynx of the person, or any other muscles the person use to achieve a desired response. The method includes the steps of attaching a single integrated sensor, for example, near the throat of the person proximate to the larynx and detecting an electrical signal through the sensor. The method further includes the steps of extracting features from the detected electrical signal and continuously transforming them into speech sounds without the need for further modulation. The method also includes comparing the extracted features to a set of prototype features and selecting a prototype feature of the set of prototype features providing a smallest relative difference.

Abstract: A sound decoding device is capable of improving the lost frame compensation performance and improving quality of the decoded sound. A rise frame sound source compensation unit generates a compensation sound source signal when the current frame is a lost frame and a rise frame. An average sound source pattern update unit updates the average sound source pattern held in an average sound source pattern holding unit over a plurality of frames. When a frame is lost, an LPC synthesis unit performs LPC synthesis on a decoded sound source signal by using the compensation sound source signal inputted via a switching unit and a decoded LPC parameter from an LPC decoding unit and outputs the compensation decoded sound signal.

Abstract: A method and device for concealing frame erasures caused by frames of an encoded sound signal erased during transmission from an encoder to a decoder and for recovery of the decoder after frame erasures comprise, in the encoder, determining concealment/recovery parameters including at least phase information related to frames of the encoded sound signal. The concealment/recovery parameters determined in the encoder are transmitted to the decoder and, in the decoder, frame erasure concealment is conducted in response to the received concealment/recovery parameters. The frame erasure concealment comprises resynchronizing, in response to the received phase information, the erasure-concealed frames with corresponding frames of the sound signal encoded at the encoder. When no concealment/recovery parameters are transmitted to the decoder, a phase information of each frame of the encoded sound signal that has been erased during transmission from the encoder to the decoder is estimated in the decoder.

Abstract: A speech decoding method which generates an excitation signal and a synthesis filter from coded data and which obtains a speech signal based on the excitation signal and the synthesis filter. The method includes acquiring identification information used for determining whether the speech signal to be decoded is a narrowband signal or a wideband signal; and modifying the excitation signal based on the identification information by controlling strength or presence of emphasis of pitch periodicity with respect to the excitation signal generated from the coded data, so as to generate the speech signal by use of the modified excitation signal and the synthesis filter.

Abstract: A system and method for detection of rate determination algorithm errors in variable rate communications system receivers. The disclosed embodiments prevent rate determination algorithm errors from causing audible artifacts such as screeches or beeps. The disclosed system and method detects frames with incorrectly determined data rates and performs frame erasure processing and/or memory state clean up to prevent propagation of distortion across multiple frames. Frames with incorrectly determined data rates are detected by checking illegal rate transitions, reserved bits, validating unused filter type bit combinations and analyzing relationships between fixed code-book gains and linear prediction coefficient gains.

Abstract: Presented herein are systems and methods for processing sound signals for use with electronic speech systems. Sound signals are temporally parsed into frames, and the speech system includes a speech codebook having entries corresponding to frame sequences. The system identifies speech sounds in an audio signal using the speech codebook.

Abstract: Frames containing audio data may be received, the audio data having been derived from a microphone array, at least some of the frames containing residual acoustic echo after having acoustic echo partially removed therefrom. Probability distribution functions are determined from the frames of audio data. A probability distribution function comprises likelihoods that respective directions are directions of sources of sounds. An active speaker may be identified in frames of video data based on the video data and based on audio information derived from the audio data, where use of the audio information as a basis for identifying the active speaker is controlled by determining whether the probability distribution functions indicate that corresponding audio data includes residual acoustic echo.

Abstract: Methods and apparatus are disclosed for approximating an MDCT coefficient of a block of windowed sinusoid having a defined frequency, the block being multiplied by a window sequence and having a block length and a block index. A finite trigonometric series is employed to approximate the window sequence. A window summation table is pre-computed using the finite trigonometric series and the defined frequency of the sinusoid. A block phase is computed for each block with the defined frequency, the block length and the block index. An MDCT coefficient is approximated by the dot product of a phase vector computed using the block phase with a corresponding row of the window summation table.

Abstract: A system and method disclosed for using and updating a database of template responses for a live agent in response to user communications. The method includes computing an average string distance between each response from a live agent and a template, use to generate the response, modifying the computed average string distance based on a customer satisfaction score associated with each response and selecting a response that minimizes the computed average string distance and maximizes customer satisfaction. Upon receiving a further communication on a certain issue, the system presents a prototype response that has been added to the template database to the live agent for use in generating a response to the further communication that reduces handling time and increases customer satisfaction.

Abstract: Compression of audio signal data is described herein. In various embodiments, the compression of each unit of the audio signal data includes the employment of a distribution substantially representative of a subblock of residual data of the unit of audio signal data, to reduce the amount of data having to be transmitted to transmit the unit of audio signal data to a recipient.

Abstract: Systems and methods for adaptive processing of a close microphone array in a noise suppression system are provided. A primary acoustic signal and a secondary acoustic signal are received. In exemplary embodiments, a frequency analysis is performed on the acoustic signals to obtain frequency sub-band signals. An adaptive equalization coefficient may then be applied to a sub-band signal of the secondary acoustic signal. A forward-facing cardioid pattern and a backward-facing cardioid pattern are then generated based on the sub-band signals. Utilizing cardioid signals of the forward-facing cardioid pattern and backward-facing cardioid pattern, noise suppression may be performed. A resulting noise suppressed signal is output.

Abstract: A system extends the high-frequency spectrum of a narrowband 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.

Abstract: Synthesizing a set of digital speech samples corresponding to a selected voicing state includes dividing speech model parameters into frames, with a frame of speech model parameters including pitch information, voicing information determining the voicing state in one or more frequency regions, and spectral information. First and second digital filters are computed using, respectively, first and second frames of speech model parameters, with the frequency responses of the digital filters corresponding to the spectral information in frequency regions for which the voicing state equals the selected voicing state. A set of pulse locations are determined, and sets of first and second signal samples are produced using the pulse locations and, respectively, the first and second digital filters. Finally, the sets of first and second signal samples are combined to produce a set of digital speech samples corresponding to the selected voicing state.

Abstract: A codebook spectral envelope may be used to extend the bandwidth of a bandwidth limited signal. A system includes codebooks that list codebook spectral envelopes. A codebook spectral envelope may be selected based on a characteristic of the spectral envelope of the bandwidth limited signal. Modifications of selected codebook spectral envelopes may generate a bandwidth extension signal that may be added to the bandwidth limited signal to improve the quality of the signal.

Abstract: The present research can decrease the amount of computation and enhance speech quality by using a global pulse replacement method in a fixed codebook search. The fixed codebook search method in a speech encoder based upon global pulse replacement, includes the steps of: (a) computing absolute values of the pulse-position likelihood-estimator vectors; (b) temporarily obtaining a codebook vector; (c) computing a mathematical equation by replacing a pulse; (d) determining whether a value computed based upon the mathematical equation is increased after pulse replacement; (e) obtaining a new codebook vector by replacing the pulse; and (f) maintaining a previous codebook vector.