Abstract: A method of funneling user responses in a voice portal system to determine a desired item or service includes (a) querying a user for an attribute value associated with a first particular attribute of the desired item or service; and (b) determining if the attribute value given by the user satisfies an end state. If the end state is not satisfied, steps (a) and (b) are performed with a new particular attribute.

Abstract: A parametric audio system that permits greater control over the bandwidth of a modulated signal. The system includes a carrier signal generator for generating a carrier signal, at least one audio signal source for generating at least one audio signal, and a modulation component for generating an envelope signal based on the at least one audio signal, modulating the phase of the carrier signal based on a predetermined function to generate a first modulated signal, and multiplying the envelope signal and the first modulated signal to generate a second modulated signal. By selection of the predetermined function, the modulation component can alter the spectrum of the second modulated signal, thereby permitting greater control over the bandwidth of the second modulated signal.

Abstract: A method, a system, and computer readable medium comprising instructions for analyzing data of a speech application are provided. The method comprises defining a set of data collection objects for a call flow in a speech application, collecting data using the set of data collection objects during execution of the speech application, and analyzing the data using a benchmarking and bootstrapping engine, storing the data in a repository, and presenting the data for analysis.

Abstract: A sound processing apparatus includes a target sound emphasizing unit configured to acquire a sound frequency component by emphasizing target sound in input sound in which the target sound and noise are included, a target sound suppressing unit configured to acquire a noise frequency component by suppressing the target sound in the input sound, a gain computing unit configured to compute a gain value to be multiplied by the sound frequency component using a gain function that provides a gain value and has a slope that are less than predetermined values when an energy ratio of the sound frequency component to the noise frequency component is less than or equal to a predetermined value, and a gain multiplier unit configured to multiply the sound frequency component by the gain value computed by the gain computing unit.

Abstract: Adaptive time/frequency-based audio encoding and decoding apparatuses and methods. The encoding apparatus includes a transformation & mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal, an encoding unit to encode each frequency-domain signal in the respective encoding mode, and a bitstream output unit to output encoded data, division information, and encoding mode information for each respective frequency-domain signal. In the apparatuses and methods, acoustic characteristics and a voicing model are simultaneously applied to a frame, which is an audio compression processing unit. As a result, a compression method effective for both music and voice can be produced, and the compression method can be used for mobile terminals that require audio compression at a low bit rate.

Abstract: A method, medium, and apparatus encoding and/or decoding a multichannel audio signal. The method includes detecting the type of spatial extension data included in an encoding result of an audio signal, if the spatial extension data is data indicating a core audio object type related to a technique of encoding core audio data, detecting the core audio object type; decoding core audio data by using a decoding technique according to the detected core audio object type, if the spatial extension data is residual coding data, decoding the residual coding data by using the decoding technique according to the core audio object type, and up-mixing the decoded core audio data by using the decoded residual coding data. According to the method, the core audio data and residual coding data may be decoded by using an identical decoding technique, thereby reducing complexity at the decoding end.

Abstract: A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

Abstract: A multi-channel synthesizer includes a post processor for determining post processed reconstruction parameters or quantities derived from the reconstruction parameter for an actual time portion of the input signal so that the post processed reconstruction parameter or the post processed quantity is different from the corresponding quantized and inversely quantized reconstruction parameter in that the value of the post processed reconstruction parameter or the derived quantity is not bound by the quantization step size. A multi-channel reconstructor uses the post-processed reconstruction parameter for reconstructing the multi-channel output signal.

Abstract: A method of communicating digitized speech from a transmitting forum participant comprises the step of receiving a data structure that includes said digitized speech. The data structure is analyzed to determine whether the digitized speech is redundantly represented in a plurality of forms in the data structure. A portion of the data structure is forwarded to a receiving forum participant, thereby communicating the digitized speech from the transmitting forum participant. In this method, when the digitized speech is redundantly represented in the data structure in a plurality of forms, the forwarding step includes a step of selecting one or more forms, based on a function, from the plurality of forms in the data structure. Furthermore, the portion of the data structure that is forwarded to the receiving forum participant includes data in the data structure that corresponds to each of the selected one or more forms.

Abstract: Methods and systems for non-negative hidden Markov modeling of signals are described. For example, techniques disclosed herein may be applied to signals emitted by one or more sources. The modeling may be constrained according to high level information. In some embodiments, methods and systems may enable the separation of a signal's various components. As such, the systems and methods disclosed herein may find a wide variety of applications. In audio-related fields, for example, these techniques may be useful in music recording and processing, source separation/extraction, noise reduction, teaching, automatic transcription, electronic games, audio search and retrieval, and many other applications.

Abstract: A representation of an audio signal having a first, a second and a third frame is derived by estimating first warp information for the first and second frames and second warp information for the second and third frames, the warp information describing pitch information of the audio signal. First or second spectral coefficients for first and second frames or second and third frames are derived using first or second warp information and a first or second weighted representation of the first and second frames or second and third frames, the first or second weighted representation derived by applying a first or second window function to the first and second frames or second and third frames, wherein the first or second window function depends on the first or second warp information. The representation of the audio signal is generated including the first and the second spectral coefficients.

Abstract: In particular embodiments, a method includes receiving data sets, constructing a first binary decision diagram (BDD) representing the data sets, iteratively adding data from the data sets to the first BDD until a compression rate of the first BDD reaches a threshold compression rate, constructing a second BDD representing data from the data sets received after the compression rate of the first BDD equals a threshold compression rate, and iteratively adding data from the data sets to the second BDD.

Abstract: A voice input device includes: a mastery level identifying device identifying a mastery level of a user with respect to voice input; and an input mode setting device switching a voice input mode between a guided input mode and an unguided input mode. In the guided input mode, preliminary registered contents of the voice input are presented to the user. The input mode setting device sets the voice input mode to the unguided input mode at a starting time when the voice input device starts to receive the voice input. The input mode setting device switches the voice input mode from the unguided input mode to the guided input mode at a switching time. The input mode setting device sets a time interval between the starting time and the switching time in proportion to the mastery level.

Abstract: The invention provides a system for controlling flame to produce a music-reactive fire display. This system comprises a digital signal analyzer, electronically-controlled burner elements that allow variable control of fuel flow rate, an automatic ignition system, flame detection, and a means of communication between the signal analyzer and the burner elements.

Abstract: A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.

Abstract: A method for suppressing ambient noise using multiple audio signals may include providing at least two audio signals captured by at least two electro-acoustic transducers. The at least two audio signals may include desired audio and ambient noise. The method may also include performing beamforming on the at least two audio signals in order to obtain a desired audio reference signal that is separate from a noise reference signal.

Abstract: An audio encoder has a common preprocessing stage, an information sink based encoding branch such as spectral domain encoding branch, a information source based encoding branch such as an LPC-domain encoding branch and a switch for switching between these branches at inputs into these branches or outputs of these branches controlled by a decision stage. An audio decoder has a spectral domain decoding branch, an LPC-domain decoding branch, one or more switches for switching between the branches and a common post-processing stage for post-processing a time-domain audio signal for obtaining a post-processed audio signal.

Abstract: Aspects of a method and system for an asynchronous pipeline architecture for multiple independent dual/stereo channel PCM processing are provided. Asynchronously pipeline processing of audio information comprised within a decoded PCM frame may be based on metadata information generated from the decoded PCM frame and an output decoding rate. The asynchronously pipeline processing may comprise mixing a primary audio information portion and a secondary audio information, portion, sample rate converting the audio information, and buffering the audio information. The asynchronously pipeline processing may comprise multiple pipeline stages. Feeding back an output of one of the pipeline stages to an input of a previous one of the pipeline stages may be enabled. The metadata information may comprise a frame start indicator associated with the decoded PCM frame and/or a plurality of mixing coefficients.

Abstract: Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment.

Abstract: Methods and devices to enable efficient beat-matched, DJ-style crossfading are provided. For example, such a method may involve determining beat locations of a first audio stream and a second audio stream and crossfading the first audio stream and the second audio stream such that the beat locations of the first audio stream are substantially aligned with the beat locations of the second audio stream. The beat locations of the first audio stream or the second audio stream may be determined based at least in part on an analysis of frequency data unpacked from one or more compressed audio files.

Abstract: A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit.

Abstract: Differential dynamic content delivery including providing a session document for a presentation, wherein the session document includes a session grammar and a session structured document; selecting from the session structured document a classified structural element in dependence upon user classifications of a user participant in the presentation; presenting the selected structural element to the user; streaming presentation speech to the user including individual speech from at least one user participating in the presentation; converting the presentation speech to text; detecting whether the presentation speech contains simultaneous individual speech from two or more users; and displaying the text if the presentation speech contains simultaneous individual speech from two or more users.

Abstract: A method of encoding stereo audio that minimizes a number of pieces of side information required for parametric-encoding and parametric-decoding of the stereo audio. The side information may include parameters about interchannel intensity difference (IID), interchannel correlation (IC), overall phase difference (OPD), and interchannel phase difference (IPD), which are required to restore the mono audio to the stereo audio.

Abstract: A system and method are disclosed for tracking image and audio data over time to automatically identify a person based on a correlation of their voice with their body in a multi-user game or multimedia setting.

Abstract: A Yule-Walker based, low-complexity voice activity detector (VAD) is disclosed. An input signal is typically noisy speech (i.e., corrupted with, for example, babble noise). In one embodiment, a first initialization stage of the VAD computes an occurrence of a silent period within the input signal and the AR parameters. The VAD could accordingly compute a tentative adaptive threshold and output hypothesis H1 (which means speech is present) during this stage. During the second initialization stage, the VAD generally builds a database of associated values and computes the adaptive threshold accordingly. The second initialization stage could also output tentative VAD decisions based on the tentative threshold computed in the first initialization stage. Finally, the VAD periodically retrains or updates AR parameters, threshold values and/or the database and outputs VAD decisions accordingly.

Abstract: A sound encoder for efficiently encoding stereophonic sound. A prediction parameter analyzer determines a delay difference D and an amplitude ratio g of a first-channel sound signal with respect to a second-channel sound signal as channel-to-channel prediction parameters from a first-channel decoded signal and a second-channel sound signal. A prediction parameter quantizer quantizes the prediction parameters, and a signal predictor predicts a second-channel signal using the first decoded signal and the quantization prediction parameters. The prediction parameter quantizer encodes and quantizes the prediction parameters (the delay difference D and the amplitude ratio g) using a relationship (correlation) between the delay difference D and the amplitude ratio g attributed to a spatial characteristic (e.g., distance) from a sound source of the signal to a receiving point.

Abstract: Systems and methods are disclosed for encoding audio in a set-top box that is invoked by a user when listening to a broadcast audio signal from a radio, TV, streaming or other audio device. A detection and identification system comprising an audio encoder is integrated in a set-top box, where detection and identification of media is realized. The encoding automatically identifies characteristics of the media (e.g., the source of a particular piece of material) by embedding an inaudible code within the content. This code contains information about the content that can be decoded by a machine, but is not detectable by human hearing. The embedded code may be used to provide programming information to the view or audience measurement date to the provider.

Abstract: A method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: Provided are an audio encoding method and apparatus and an audio decoding method and apparatus in which audio signals can be encoded or decoded so that sound images can be localized at any desired position for each object audio signal. The audio decoding method generating a third downmix signal by combining a first downmix signal extracted from a first audio signal and a second downmix signal extracted from a second audio signal; generating third object-based side information by combining first object-based side information extracted from the first audio signal and second object-based side information extracted from the second audio signal; converting the third object-based side information into channel-based side information; and generating a multi-channel audio signal using the third downmix signal and the channel-based side information.

Abstract: An improved technique tailors a biometric challenge activity to a particular user. The particular user submits electronic input from which an authentication system extracts information concerning traits of the particular user; such traits can include keystroke and swiping patterns, handheld device positions, and place of origin. An authentication server maps values of user attributes such as place of origin, age, and UI device to the extracted traits. The authentication server then selects biometric challenges for the particular user based on user attributes having values which deviate most from a mean value of that attribute taken across a population of users. That is, the authentication server bases biometric challenges on the most distinguishing traits of the particular user.

Abstract: Methods for the automatic creation of a category tree with respect to the contents of a data stock, wherein a taxonomy of the data stock will be created on the base of co-occurrences. Another object of the present invention is furthermore a data processing system comprising data which represent information in at least one data stock which is accessible via at least one data source, which is designed and/or adapted to at least partially carry out a method according to the invention. Another object of the present invention is furthermore a data processing device for the electronic processing of data, comprising a control and/or computer unit, an input unit and an output unit, which is designed and/or adapted to at least partially carry out a method according to the invention, preferably using at least a part of a data processing system according to the invention.

Abstract: A speech signal processing device is equipped with a power acquisition unit, a probability distribution acquisition unit, and a correspondence degree determination unit. The power acquisition unit accepts an inputted speech signal and, based on the accepted speech signal, acquires power representing the intensity of a speech sound represented by the speech signal. The probability distribution acquisition unit acquires a probability distribution using the intensity of the power acquired by the power acquisition unit as a random variable. The correspondence degree determination unit determines whether a correspondence degree representing a degree that power acquired by the power acquisition unit in a case that a predetermined reference speech signal is inputted into the power acquisition unit corresponds with predetermined reference power is higher than a predetermined reference correspondence degree, based on the probability distribution acquired by the probability distribution acquisition unit.

Abstract: A system for and method of speech processing for a vehicle. Speech is received from at least one vehicle occupant via a plurality of microphones corresponding to the plurality of zones in the vehicle, wherein the microphones convert the speech into speech signals. At least one active communication zone is determined in which the at least one vehicle occupant corresponding to the active communication zone is speaking Speech processing is modified in response to the determined active communication zone.

Abstract: Systems and methods for processing encoded digital data for programming a device to be re-programmed in an audio playback system. The system includes an audio media source containing digital data having audio data or encoded data in an audio data format. An audio media reader reads the digital data from the audio media source. A stream detector receives the digital data from the audio media reader and detects whether the received digital data includes encoded data formatted as audio data or audio data. An audio receiver device receives the audio data and processes the audio data for playback. A device to be re-programmed uses the encoded data formatted as audio data.

Abstract: A coding method, a decoding method, a coding-decoding (codec) method, a codec system and relevant apparatuses are disclosed. The coding method includes: obtaining an amplitude vector and a length vector corresponding to a vector to be coded; sorting elements of the amplitude vector and elements of the length vector; and obtaining a position index value according to the sorted amplitude vector and the sorted length vector. A decoding method, a codec system, and relevant apparatuses are also provided.

Abstract: In frame-based bit stream formats the data required for decoding a current frame are usually stored within the data section for that frame. One exception is the mp3 bit stream where data for a current frame is stored in previous frames. If the decoder did not receive the required previous frame, decoding of the current mp3 frame is skipped. The invention can be applied for such bit streams, in an archival mode, a streaming mode and a sample-exact cutting of an archival mode. In the streaming and cutting modes, new headers are established. The number of frames required for initializing the decoder status is signalized in the header, as well as a consistency check value in the streaming mode. These frames are used for decoder initialization but not for decoding samples or coefficients. For a sample-exact cutting, for the frame at which the cut shall occur, the number of samples or coefficients to be muted is also indicated in the header.

Abstract: A method and apparatus for estimating speech intelligibility in a mobile communications network component handling two-way communication between two ends of a signal path. Test signals adapted for speech intelligibility measurements are inserted into the signal path to simulate two-way communication. Double-talk is detected during the communication, and speech intelligibility measurements are performed only during periods of double-talk. This enables the effect of echo to be taken into account while avoiding undesirable effects from non-linear processing, and comfort noise if present, in the signal path. Voice enhancement devices may then be adjusted in response to the estimated speech intelligibility.

Abstract: An echo canceller system includes a first echo canceller having a first voltage divider and an adaptable second voltage divider that is configured to generate a first replica of an echo. A second echo canceller is configured to generate a second replica of an echo and has tap values that are generated in response to an error signal. A controller is coupled to the first and second echo cancellers and includes a selection algorithm that responds to the tap values of the second echo canceller and selects a voltage divider value for the adaptable second voltage divider.

Abstract: An apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information includes a parameter adjuster. The parameter adjuster is configured to receive one or more input parameters and to provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured to provide the one or more adjusted parameters in dependence on the one or more input parameters and the object-related parametric information, such that a distortion of the upmix signal representation caused by the use of non-optimal parameters is reduced at least for input parameters deviating from optimal parameters by more than a predetermined deviation.

Abstract: Methods and systems are provided for gathering research data that includes information pertaining to audio signals received on a portable device, such as a cell phone. Frequency domain data is received or produced, a signature is extracted from the frequency domain data and an ancillary code is read from the frequency domain data.

Abstract: An approach to performing inverse quantization on a quantized integral value is described. This approach involves determining whether a quantized integral value lies within a first range or a second range of possible values. An interpolated inverse quantization value is calculated from the quantized integral value, using a predetermined bit shifting operation, depending on whether the quantized integral value was in the first or the second range.

Abstract: The present invention relates to methods and devices for forward time-domain aliasing cancellation in a coded signal transmitted from a coder to a decoder. Information related to correction of the time-domain aliasing in the coded signal is calculated at the coder and added in a bitstream sent from the coder to the decoder. The decoder receives the bitstream and cancels the time-domain aliasing in the coded signal in response to the information comprised in the bitstream. The information may be representative of a difference between a frame of audio signal to be encoded in a first coding mode and a decoded signal from the frame including time-domain aliasing effects.

Abstract: There is disclosed an audio decoding device capable of improving audio quality of a decoded signal by considering the energy change of a past signal in eracure concealment processing. In this device, an energy change calculation unit (143) calculates an average energy of an audio source signal of one-pitch cycle from the end of the ACB vector outputted from an adaptive codebook (106). Moreover, the energy change calculation unit (143) calculates a ratio of the average energy of the current sub-frame and the sub-frame immediately before and outputs the ratio to an ACB gain generation unit (135). The ACB gain generation unit (135) outputs a conceal processing ACB gain defined by the ACB gain decoded in the past or information on the energy change ratio outputted from the energy change calculation unit (143) to a multiplier (132).

Abstract: Provided is an encoding device which can obtain a sound quality preferable for auditory sense even if the number of information bits is small. The encoding device includes a shape quantization unit (111) having: a section search unit (121) which searches for a pulse for each of bands into which a predetermined search section is divided; and a whole search unit (122) which performs search for a pulse over the entire search section. The shape of an input spectrum is quantized by a small number of pulse positions and polarities. A gain quantization unit (112) calculates a gain of the pulse searched by the shape quantization unit (111) and quantizes the gain for each of the bands.

Abstract: Embodiments of the present invention provide systems, methods, and computer-readable media for generating a voice characteristic profile based on detected sound components. In embodiments, a call is initiated between a first caller and a second caller. Information communicated during the call is monitored to determine that sound components have been spoken by the first caller. The sound components are determined to be associated with a language dialect. Further, the sound components are stored in association with the first caller. In particular, the sound components are stored in association with the first caller in a voice characteristic profile of the first caller.

Abstract: A system and method for detecting a non-visual code using an application on a mobile device, where the application is capable of associating the non-visual code with at least one item contained in a transmitted presentation and connecting the mobile device to information about the item in a database associated with the transmitted presentation. The non-visual code may comprise a high frequency signal played alone or with another audio or video signal. A mobile device application executing on a processor of the mobile device performs signal processing on the audio signal of the presentation to extract the high frequency signal. Also contemplated is obtaining information about the visual content and presenting the information on the personal device.

Abstract: In one embodiment of a controllable prosody re-estimation system, a TTS/STS engine consists of a prosody prediction/estimation module, a prosody re-estimation module and a speech synthesis module. The prosody prediction/estimation module generates predicted or estimated prosody information. And then the prosody re-estimation module re-estimates the predicted or estimated prosody information and produces new prosody information, according to a set of controllable parameters provided by a controllable prosody parameter interface. The new prosody information is provided to the speech synthesis module to produce a synthesized speech.

Abstract: A sequence is received of time domain digital audio samples representing sound (e.g., a sound generated by a human voice or a musical instrument). The time domain digital audio samples are processed to derive a corresponding sequence of audio pulses in the time domain. Each of the audio pulses is associated with a characteristic frequency. Frequency domain information is derived about each of at least some of the audio pulses. The sound represented by the time domain digital audio samples is transformed by processing the audio pulses using the frequency domain information. The sound transformation utilizes overlapping windows and a computational cost function which depends on a product of the number of the pitch periods and the inverse of the minimum fundamental frequency within the window is determined.

Abstract: A method of encoding samples in a digital signal is provided that includes receiving a frame of N samples of the digital signal, determining L possible distinct data values in the N samples, determining a reference data value in the L possible distinct data values and a coding order of L?1 remaining possible distinct data values, wherein each of the L?1 remaining possible distinct data values is mapped to a position in the coding order, decomposing the N samples into L?1 coding vectors based on the coding order, wherein each coding vector identifies the locations of one of the L?1 remaining possible distinct data values in the N samples, and encoding the L?1 coding vectors.

Abstract: A method of processing a signal, including taking a signal formed from a plurality of source signal emitters and expressed in an original domain, decomposing the signal into a mathematical representation of a plurality of constituent elements in an alternate domain, analyzing the plurality of constituent elements to associate at least a subset of the constituent elements with at least one of the plurality of source signal emitters, separating at least a subset of the constituent elements based on the association and reconstituting at least a subset of constituent elements to produce an output signal in at least one of the original domain, the alternate domain and another domain.