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: Described herein are methods, systems, apparatuses and products for reconstruction of a smooth speech signal from a stuttered speech signal. One aspect provides for accessing a stored speech signal having stuttering; identifying at least one stuttered region in the stored speech signal; modifying the at least one stuttered region in the stored speech signal; and responsive to modifying the at least one stuttered region, reconstructing a smooth speech signal corresponding to the stored speech signal. Other embodiments are disclosed.

Abstract: Disclosed is a method of processing a signal, which includes receiving at least one of a first signal and a second signal, receiving mode information, and decoding the at least one of the first signal and the second signal using at least one of a first coding scheme and a second coding scheme according to the mode information. The mode information is information for indicating that a prescribed mode corresponds to one of at least three modes. The method includes detecting when a restricted mode change occurs and changing at least one mode when detecting a restricted mode change.

Abstract: A system, apparatus, and method is disclosed for receiving user input at a client device, interpreting the user input to identify a selection of at least one of a plurality of web interaction modes, producing a corresponding client request based in part on the user input and the web interaction mode; and sending the client request to a server via a network.

Abstract: A method and apparatus for separating and extracting main sound sources from a mixed musical sound signal are provided. A musical sound source separation apparatus may include an prior information signal compressor to compress an prior information signal including a characteristic of a predetermined sound source, a mixed signal divider to divide a mixed signal including a plurality of sound sources into a plurality of segments, a Nonnegative Matrix Partial Co-Factorization (NMPCF) analyzer to acquire common information shared by the plurality of segments, by applying an NMPCF algorithm to the prior information signal, and a target musical instrument signal separator to separate a target musical instrument signal corresponding to the predetermined sound source from the mixed signal, based on the common information.

Abstract: Techniques for employing improved prompts in a speech-to-speech translation system are disclosed. By way of example, a technique for use in indicating a dialogue turn in an automated speech-to-speech translation system comprises the following steps/operations. One or more text-based scripts are obtained. The one or more text-based scripts are synthesizable into one or more voice prompts. At least one of the one or more voice prompts is synthesized for playback from at least one of the one or more text-based scripts, the at least one synthesized voice prompt comprising an audible message in a language understandable to a speaker interacting with the speech-to-speech translation system, the audible message indicating a dialogue turn in the automated speech-to-speech translation system.

Abstract: Provided are an apparatus and method for visualizing multichannel audio signals. The apparatus includes a spatial audio decoding unit for receiving a downmix signal of a time domain, converting the downmix signal into a signal of a frequency domain to output a frequency domain downmix signal, and synthesizing a multichannel audio signal based on the spatial parameter and the downmix signal; and a multichannel visualizing unit for creating visualization information of the multichannel audio signal based on the frequency domain downmix signal and the spatial parameter.

Abstract: A signal processing apparatus for performing signal processing including a plurality of steps in data units by software signal processing includes signal processing modules performing the steps, a circuit configuration information storing and managing unit storing the signal processing modules and circuit configuration information, a signal processing order determining unit determining a signal processing order by performing path routing, a signal processing executing unit executing the signal processing in the determined order, and a circuit configuration changing unit changing circuit configuration information and causing the signal processing order determining unit to re-execute path routing to determine a signal processing order for the changed circuit configuration information during a period from the end of the software signal processing in the data unit to the beginning of the subsequent data unit.

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. 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 extraction, noise reduction, teaching, automatic transcription, electronic games, audio search and retrieval, and many other applications.

Abstract: Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context. Particularly, certain embodiments contemplate suppressing the context component from the digital audio signal to obtain a context-suppressed signal; generating an audio context signal that is based on a first filter and a first plurality of sequences, each of the first plurality of sequences having a different time resolution and mixing a first signal that is based on the generated audio context signal with a second signal that is based on the context-suppressed signal to obtain a context-enhanced signal, wherein generating an audio context signal includes applying the first filter to each of the first plurality of sequences.

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 system and method may be configured to process an audio signal. The system and method may track pitch, chirp rate, and/or harmonic envelope across the audio signal, may reconstruct sound represented in the audio signal, and/or may segment or classify the audio signal. A transform may be performed on the audio signal to place the audio signal in a frequency chirp domain that enhances the sound parameter tracking, reconstruction, and/or classification.

Abstract: This invention relates to generation of a sample error coefficient suitable for use in an audio signal quality assessment system. The invention provides a method of determining a sample error coefficient between a first signal and a similar second signal comprising the steps of: determining a first periodicity measure from the first signal; determining a second periodicity measure from the second signal; generating a ratio in dependence upon said first periodicity measure and said second periodicity measure; and determining a sampling rate error coefficient in dependence upon said ratio.

Abstract: Disclosed is a method of improving a sound quality, including: receiving a transmission signal of a first user equipment; removing noise in the transmission signal using noise information of the first user equipment side; performing speech reinforcement with respect to the noise removed transmission signal using noise information of a second user equipment side; and transmitting the speech reinforced transmission signal to the second user equipment.

Abstract: A multi-channel signal enhancement system reinforces signal content and improves the signal-to-noise ratio of a multi-channel signal. The system detects, tracks, and reinforces non-stationary periodic signal components of a multi-channel signal. The periodic signal components of the signal may represent vowel sounds or other voiced sounds. The system may detect, track, or attenuate quasi-stationary signal components in the multi-channel signal.

Abstract: Enhancing speech components of an audio signal composed of speech and noise components includes controlling the gain of the audio signal in ones of its subbands, wherein the gain in a subband is reduced as the level of estimated noise components increases with respect to the level of speech components, wherein the level of estimated noise components is determined at least in part by (1) comparing an estimated noise components level with the level of the audio signal in the subband and increasing the estimated noise components level in the subband by a predetermined amount when the input signal level in the subband exceeds the estimated noise components level in the subband by a limit for more than a defined time, or (2) obtaining and monitoring the signal-to-noise ratio in the subband and increasing the estimated noise components level in the subband by a predetermined amount when the signal-to-noise ratio in the subband exceeds a limit for more than a defined time.

Abstract: Techniques described herein include the use of equalization techniques to improve intelligibility of a reproduced audio signal (e.g., a far-end speech signal).

Abstract: An automated emotional recognition system is adapted to determine emotional states of a speaker based on the analysis of a speech signal. The emotional recognition system includes at least one server function and at least one client function in communication with the at least one server function for receiving assistance in determining the emotional states of the speaker. The at least one client function includes an emotional features calculator adapted to receive the speech signal and to extract therefrom a set of speech features indicative of the emotional state of the speaker. The emotional state recognition system further includes at least one emotional state decider adapted to determine the emotional state of the speaker exploiting the set of speech features based on a decision model. The server function includes at least a decision model trainer adapted to update the selected decision model according to the speech signal.

Abstract: A telephone system comprising switching circuitry configured for coupling a call to a telephone extension coupled to the system, voice processing circuitry configured for automatically interacting with the call, a microprocessor, a first data bus connected between the microprocessor and the switching circuitry, and a second data bus connected between the microprocessor and the voice processing circuitry.

Abstract: An audio decoder for decoding a multi-audio-object signal having an audio signal of a first type and an audio signal of a second type encoded therein is described, the multi-audio-object signal having a downmix signal and side information, the side information having level information of the audio signals of the first and second types in a first predetermined time/frequency resolution, and a residual signal specifying residual level values in a second predetermined time/frequency resolution, the audio decoder having a processor for computing prediction coefficients based on the level information; and an up-mixer for up-mixing the downmix signal based on the prediction coefficients and the residual signal to obtain a first up-mix audio signal approximating the audio signal of the first type and/or a second up-mix audio signal approximating the audio signal of the second type.

Abstract: A method of receiving an audio signal includes measuring a periodicity of the audio signal to determine a checked periodicity. At least one best available subband is determined. At least one extended subband is composed, wherein composing includes reducing a ratio of composed harmonic components to composed noise components if the checked periodicity is lower than a threshold, and scaling a magnitude of the at least one extended subband based on a spectral envelope on the audio signal.

Abstract: In one embodiment, a method of transceiving an audio signal is disclosed. The method includes providing low band spectral information having a plurality of spectrum coefficients and predicting a high band extended spectral fine structure from the low band spectral information for at least one subband, where the high band extended spectral fine structure are made of a plurality of spectrum coefficients. The predicting includes preparing the spectrum coefficients of the low band spectral information, defining prediction parameters for the high band extended spectral fine structure and index ranges of the prediction parameters, and determining possible best indices of the prediction parameters, where determining includes minimizing a prediction error between a reference subband in high band and a predicted subband that is selected and composed from an available low band. The possible best indices of the prediction parameters are transmitted.

Abstract: An apparatus and a method for generating a multi-channel synthesizer control signal, a multi-channel synthesizer, a method of generating an output signal from an input signal and a machine-readable storage medium are provided. On an encoder-side, a multi-channel input signal is analyzed for obtaining smoothing control information, which is to be used by a decoder-side multi-channel synthesis for smoothing quantized transmitted parameters or values derived from the quantized transmitted parameters for providing an improved subjective audio quality in particular for slowly moving point sources and rapidly moving point sources having tonal material such as fast moving sinusoids.

Abstract: A scalable speech and audio codec is provided that implements combinatorial spectrum encoding. A residual signal is obtained from a Code Excited Linear Prediction (CELP)-based encoding layer, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal is transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum having a plurality of spectral lines. The transform spectrum spectral lines are transformed using a combinatorial position coding technique. The combinatorial position coding technique includes generating a lexicographical index for a selected subset of spectral lines, where each lexicographic index represents one of a plurality of possible binary strings representing the positions of the selected subset of spectral lines. The lexicographical index represents non-zero spectral lines in a binary string in fewer bits than the length of the binary string.

Abstract: Provided are methods and systems of managing handoffs in a wireless communication system having different types of vocoders. Some embodiments include translating state memory of a first vocoder to a second vocoder using a state memory transcoder. The state memory may be delayed to align differences in algorithmic delays between the first vocoder and the second vocoder. In one embodiment, a speech signal may be decoded from the first vocoder, delayed, and encoded to the second vocoder. Furthermore, for a period of time during and/or adjacent to the handoff, the first vocoder may output with decreasing amplitude while the second vocoder outputs with increasing amplitude. Such techniques may be used alone or in combination.

Abstract: Embodiments of methods, apparatuses, devices and systems associated with encoding and/or decoding audio data are disclosed. More particularly, the claimed subject matter relates, at least in part, to a data compression/decompression method or technique, such as a lossless, approximately lossless, and/or relatively lossless data compression/decompression method or technique, for example, along with systems or apparatuses that may relate to method or technique. The disclosed techniques and methods may achieve audio data compression ratios that may be comparable to lossless compression processes. In addition, under certain circumstances, such compression ratios may be achieved while also reducing or simplifying the computational complexity of the compression and/or decompression method or technique.

Abstract: There is provided an audio coding device which appropriately sets the quantization bit number by a small calculation amount in each stage when coding an input audio signal by performing multi-stage normalization/quantization. A quantization information calculation section determines total quantization information idwl0, based on normalization information idsf, and allocates the total quantization information idwl0 for quantization information idwl1 and quantization information idwl2. At this time, the quantization information calculation section limits the quantization information idwl1 by a limiter lim1, and allocates the total quantization information idwl0 for quantization information idwl1. If the quantization information idwl1 exceeds the limiter lim1, the excess is allocated for the quantization information idwl2. A first normalization section and a first quantization section normalizes and quantizes a frequency spectrum mdspec1 in the first stage.

Abstract: Codebook indices for a scalable speech and audio codec may be efficiently encoded based on anticipated probability distributions for such codebook indices. A residual signal from a Code Excited Linear Prediction (CELP)-based encoding layer may be obtained, where the residual signal is a difference between an original audio signal and a reconstructed version of the original audio signal. The residual signal may be transformed at a Discrete Cosine Transform (DCT)-type transform layer to obtain a corresponding transform spectrum. The transform spectrum is divided into a plurality of spectral bands, where each spectral band having a plurality of spectral lines. A plurality of different codebooks are then selected for encoding the spectral bands, where each codebook is associated with a codebook index. A plurality of codebook indices associated with the selected codebooks are then encoded together to obtain a descriptor code that more compactly represents the codebook indices.

Abstract: The example embodiment of the present invention provides an acoustic model adaptation method for enhancing recognition performance for a non-native speaker's speech. In order to adapt acoustic models, first, pronunciation variations are examined by analyzing a non-native speaker's speech. Thereafter, based on variation pronunciation of a non-native speaker's speech, acoustic models are adapted in a state-tying step during a training process of acoustic models. When the present invention for adapting acoustic models and a conventional acoustic model adaptation scheme are combined, more-enhanced recognition performance can be obtained. The example embodiment of the present invention enhances recognition performance for a non-native speaker's speech while reducing the degradation of recognition performance for a native speaker's speech.

Abstract: A voice service system which includes a database storing information and is connected with a plurality of terminals by a communication network, comprising a call connection unit for transmitting-receiving a signal, on a voice service to/from each of the terminals, a voice processing unit for interpreting a voice instruction received from the terminal and converting an internally-created voice service document into a voice message, a recognition management unit for extracting a grammar, which is expected to be spoken by a user, from the database, a document management unit for creating the voice service document including the extracted grammar as a voice anchor, and a control unit for controlling information related to the voice anchor matching with the voice instruction to be provided.

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: The present invention relates to co-channel audio source separation. In one embodiment a first frequency-related representation of plural regions of the acoustic signal is prepared over time, and a two-dimensional transform of plural two-dimensional localized regions of the first frequency-related representation, each less than an entire frequency range of the first frequency related representation, is obtained to provide a two-dimensional compressed frequency-related representation with respect to each two dimensional localized region. For each of the plural regions, at least one pitch is identified. The pitch from the plural regions is processed to provide multiple pitch estimates over time. In another embodiment, a mixed acoustic signal is processed by localizing multiple time-frequency regions of a spectrogram of the mixed acoustic signal to obtain one or more acoustic properties.

Abstract: An apparatus (100, 200) and method are disclosed for managing voicemail messages. A system that incorporates teachings of the present disclosure may include, for example, a voicemail system (100) having a communications interface (110) and a controller (102) for managing operations of the communications interface. The controller is programmed to store (310) voicemail messages corresponding to a communication device (102), transmit (316) a log of the voicemail messages to the communication device, and receive (318) from the communication device a request to delete a voicemail message selectively chosen from a user interface corresponding to the voicemail log. The present disclosure further describes embodiments for the communication device.

Abstract: Method and apparatus for environment detection and adaptation in hearing assistance devices. Performance of feature extraction and environment detection to perform adaptation to hearing assistance device operation for a number of hearing assistance environments. The system detecting various noise sources independent of speech. The system determining adaptive actions to take place based on predicted sound class. The system providing individually customizable response to inputs from different sound classes. In various embodiments, the system employing a Bayesian classifier to perform sound classifications using a priori probability data and training data for predetermined sound classes. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: Conference bridge (1) for managing an audio scene comprising two or more participants, the conference bridge comprising a mixer (2) and several user channels (3a, 3b, 3N). The conference bridge is arranged to continuously create a 3D positional audio environment signal for each participant as a listening participant, by rendering the speech of each participant as a 3D positioned virtual sound source and excluding the speech of the listening participant, and to distribute each created 3D positional audio environment signal to the corresponding listening participant. Further, the conference bridge is arranged to place the virtual sound source corresponding to each participant at the same spatial position relative the listening participant in every created 3D positional audio environment.

Abstract: The present invention relates to an improved scheme for coding of audio. In particular, the present invention relates to an encoder device and a method for coding an input signal in an encoder system. The method comprises applying a first mode to the input signal to form a first output and applying a second mode to the input signal to form a second output. A first processed output is then formed from at least a part of the first output, and a second processed output is formed from at least a part of the second output. Forming a second processed output comprises estimating a part of the input signal from at least a part of the second output. Then, an optimum mode is determined based on the first processed output and the second processed output, and the output according to the optimum mode is selected.

Abstract: The APPARATUSES, METHODS AND SYSTEMS FOR SPARSE SINUSOIDAL AUDIO PROCESSING AND TRANSMISSION (hereinafter “SS-Audio”) provides a platform for encoding and decoding audio signals based on a sparse sinusoidal structure. In one embodiment, the SS-Audio encoder may encode received audio inputs based on its sparse representation in the frequency domain and transmit the encoded and quantized bit streams. In one embodiment, the SS-Audio decoder may decode received quantized bit streams based on sparse reconstruction and recover the original audio input by reconstructing the sinusoidal parameters in the frequency domain.

Abstract: The present invention teaches a new audio coding system that can code both general audio and speech signals well at low bit rates. A proposed audio coding system comprises linear prediction unit for filtering an input signal based on an adaptive filter; a transformation unit for transforming a frame of the filtered input signal into a transform domain; and a quantization unit for quantizing the transform domain signal. The quantization unit decides, based on input signal characteristics, to encode the transform domain signal with a model-based quantizer or a non-model-based quantizer. Preferably, the decision is based on the frame size applied by the transformation unit.

Abstract: A device may include logic configured to receive a first speech input from a party, to compare the first speech input to a second speech input to produce a result, and to determine if the party is impaired based on the result.

Abstract: Provided are parametric audio encoding and decoding apparatuses and methods thereof. In the parametric audio encoding method, an audio signal is segmented into a plurality of segments. At least one sine wave is extracted from each of the segments, and the extracted sine waves are connected. It is determined whether an extracted sine wave is a birth sine wave. If the extracted sine wave is a birth sine wave, a bit stream is generated by encoding the phase of the birth sine wave on the basis of the frequency of the birth sine wave, wherein the number of bits allocated to encode the phase of the birth sine wave is adjusted according to the frequency of the birth sine wave.

Abstract: A system for transmitting input data over a speech channel of a network comprising: a modulator arranged to produce a modulated waveform signal transforming the data for transmission over the network; a channel compensation filter arranged to filter the modulated waveform signal after it has been transmitted over the speech channel to compensate for the response of the speech channel; and a demodulator arranged to retrieve the data from the filtered waveform signal.

Abstract: Provided are, among other things, systems, methods and techniques for decoding an audio signal from a frame-based bit stream. At least one frame includes processing information pertaining to the frame and entropy-encoded quantization indexes representing audio data within the frame. The processing information includes: (i) code book indexes, and (ii) code book application information specifying ranges of entropy-encoded quantization indexes to which the code books are to be applied. The entropy-encoded quantization indexes are decoded by applying the identified code books to the corresponding ranges of entropy-encoded quantization indexes.

Abstract: A method of generating a frame of audio data for an audio signal from preceding audio data for the audio signal that precede the frame of audio data, the method comprising the steps of: predicting a predetermined number of data samples for the frame of audio data based on the preceding audio data, to form predicted data samples; identifying a section of the preceding audio data for use in generating the frame of audio data; and forming the audio data of the frame of audio data as a repetition (602) of at least part of the identified section to span the frame of audio data, wherein the beginning of the frame of audio data comprises a combination of a subset of the repetition (602) of the at least part of the identified section and the predicted data samples.

Abstract: A method and an apparatus for processing a signal are provided. The method includes: obtaining an energy average value of each sub-band for a current frame frequency-domain signal; obtaining a current frame modification coefficient of each sub-band for the current frame frequency-domain signal according to a spectral envelope and the energy average value of each sub-band; obtaining a weighted modification coefficient of each sub-band for the current frame frequency-domain signal by using the current frame modification coefficient and a relevant frame modification coefficient; and modifying the spectral envelope of each sub-band for the current frame frequency-domain signal by using the weighted modification coefficient.

Abstract: A method of processing an audio signal is disclosed. The present invention includes receiving downmix information, object information and mix information, generating and transferring multi-channel information using at least one of the downmix information, the object information and the mix information, and selectively generating and transferring either first gain information or extra multi-channel information including second gain information in accordance with a decoding mode using at least one of the object information and the mix information.

Abstract: A method for defining a network of nodes is provided, each representing a unique concept, and making connections between individual concepts through unique relationships to other concepts. Each of the nodes is operable to store a unique identifier in the network and information regarding the concept in addition to the unique relationships.

Abstract: There is disclosed an encoding device capable of improving similarity between the high frequency band spectrum of the original signal and a new spectrum to be generated while realizing a low bit rate when encoding a wide-band signal spectrum. The encoding device has sub-band amplitude calculation units (122, 128) for calculating the amplitude of the respective sub-bands for the high frequency band spectrum obtained from the wide-band signal. A search unit (124) and a gain codebook (125) select some sub-bands from a plurality of sub-bands and only the gain of the selected sub-bands is subjected to encoding. An interpolation unit (126) expresses the gain of the sub-band not selected, by mutually interpolating the selected gains.

Abstract: An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

Abstract: The present invention is a method and system for extracting messages from a person using the body features presented by a user. The present invention captures a set of images and extracts a first set of body features, along with a set of contexts, and a set of meanings. From the first set of body features, the set of contexts, and the set of meanings, the present invention generates a set of words corresponding to the message that the person is attempting to convey. The present invention can also use the body features of the person in addition to the voice of the person to further improve the accuracy of extracting the person's message.

Abstract: A method and associated apparatus for using visual separators to indicate additional character combination choices from a disambiguation function on a handheld electronic device.