Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19 or 20 of unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: Embodiments of the present invention provide a multiple description audio coding and decoding method, apparatus, and system. The audio coding method includes: dividing residual signals indicating current audio signal information into multiple frequency band parts having different frequencies; respectively coding the multiple frequency band parts by using multiple description coding (MDC) methods with different speech quality; and combining each of description signal parts that are generated after coding is performed by using different MDC methods to form multiple description bit streams of the residual signals. According to the present invention, multiple description coding and decoding methods with different speech quality are used for different frequency bands, which reduces the bit rate of multiple description coding and decoding, improves the effect of multiple description coding and decoding, and hence enhances the quality of audio transmission.

Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19, or 20 unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: A data processing apparatus including a voice input unit adapted to input voice, an operating unit adapted to be operated by a user, a data processing unit adapted to process data according to voice inputted in the voice input unit or an operation of the operating unit, and a determination unit adapted to determine whether a subsequent process is a voice input prohibited process in the case where the subsequent process is to be conducted based on voice inputted into the voice input unit, wherein the data processing unit switches from processing according to voice inputted in the voice input unit to processing according to an operation of the operating unit in the case where the determination unit determines that the subsequent process is a voice input prohibited process.

Abstract: Systems and methods are provided for using automatic speech recognition to analyze a voice interaction and verify compliance of an agent reading a script to a client during the voice interaction. In one aspect of the invention, a method may include conducting the voice interaction between the agent and a client, wherein the agent follows the script via a plurality of panels. From there, the voice interaction is evaluated via the plurality of panels employing panel-by-panel playback with an automatic speech recognition component adapted to analyze the voice interaction. As such, it may be determined, via generating a score using confidence level thresholds of an automatic speech recognition component such that confidence level thresholds are assigned to each of the plurality of panels and evaluating the score against at least one of a static standard and a varying standard, whether the agent has adequately followed the script.

Abstract: A speech recognition client sends a speech stream and control stream in parallel to a server-side speech recognizer over a network. The network may be an unreliable, low-latency network. The server-side speech recognizer recognizes a first portion of the speech stream and, if a predetermined criterion is satisfied by the speech recognition result, waits until the speech recognizer has been reconfigured before recognizing a second portion of the speech stream. The speech recognition client receives recognition results from the server-side recognizer in response to requests from the client. The client may remotely reconfigure the state of the server-side recognizer during recognition.

Abstract: An automated movable barrier control system component (11) as comprises a part of a movable barrier control system (10) has an audible speech output device (21) operably coupled thereto. These elements can share a common housing (22). The audible speech output device (21) can audiblize one or more items of stored speech content relating to one or more aspects of the component itself, other components within the system, and/or the system itself.

Abstract: An audio signal decoder includes a transform domain path configured to obtain a time-domain representation of a portion of an audio content on the basis of a first set of spectral coefficients, a representation of an aliasing-cancellation stimulus signal and a plurality of linear-prediction-domain parameters. The transform domain path applies a spectrum shaping to the first set of spectral coefficients to obtain a spectrally-shaped version thereof. The transform domain path obtains a time-domain representation of the audio content on the basis of the spectrally-shaped version of the first set of spectral coefficients. The transform domain path includes an aliasing-cancellation stimulus filter to filter the aliasing-cancellation stimulus signal in dependence on at least a subset of the linear-prediction-domain parameters.

Abstract: An embodiment provides (a) a method and system for representing natural languages in a common machine-readable form, including the thorough design of the lexicon and grammar, the resulting representation called interlingua, (b) a method and system for using a computer to convert a text of a natural language into and out of a coded text of said interlingua representation, including a programming framework which is independent of other languages, said system is called interlingua engine, and (c) a method and system of machine translation using said interlingua engine, said system called interlingua machine translation system. Alternative embodiments are described.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: The present technology provides adaptive noise reduction of an acoustic signal using a sophisticated level of control to balance the tradeoff between speech loss distortion and noise reduction. The energy level of a noise component in a sub-band signal of the acoustic signal is reduced based on an estimated signal-to-noise ratio of the sub-band signal, and further on an estimated threshold level of speech distortion in the sub-band signal. In embodiments, the energy level of the noise component in the sub-band signal may be reduced to no less than a residual noise target level. Such a target level may be defined as a level at which the noise component ceases to be perceptible.

Abstract: An interactive voice response system is described that supports full duplex data transfer to enable the playing of a voice prompt to a user of telephony system while the system listens for voice barge-in from the user. The system includes a speech detection module that may utilize various criteria such as frame energy magnitude and duration thresholds to detect speech. The system also includes an automatic speech recognition engine. When the automatic speech recognition engine recognizes a segment of speech, a feature extraction module may be used to subtract a prompt echo spectrum, which corresponds to the currently playing voice prompt, from an echo-dirtied speech spectrum recorded by the system. In order to improve spectrum subtraction, an estimation of the time delay between the echo-dirtied speech and the prompt echo may also be performed.

Abstract: Methods and systems for log mining for grammar-based text processing are provided. A method may comprise receiving, from a device, an activity log. The activity log may comprise one or more of an input instruction, a determined function based at least in part on a match of the input instruction to a grammar-based textual pattern including associations of a given function based on one or more grammars, and a response determination based on an acknowledgement of the determined function. The method may also comprise comparing at least a portion of the activity log with stored activity logs in order to determine a correlation between the activity log and the stored activity logs. The method may also comprise modifying the grammar-based textual pattern based on the determined correlation and providing information indicative of the modification to the device so as to update the grammar-based textual pattern.

Abstract: An apparatus for synthesizing a speech including a waveform memory that stores a plurality of speech unit waveforms, an information memory that correspondingly stores speech unit information and an address of each of the speech unit waveforms, a selector that selects a speech unit sequence corresponding to the input phoneme sequence by referring to the speech unit information, a speech unit waveform acquisition unit that acquires a speech unit waveform corresponding to each speech unit of the speech unit sequence from the waveform memory by referring to the address, a speech unit concatenation unit that generates the speech by concatenating the speech unit waveform acquired.

Abstract: An adaptive noise modeling speech recognition system improves speech recognition by modifying an activation of the system's grammar rules or models based on detected noise characteristics. An adaptive noise modeling speech recognition system includes a sensor that receives acoustic data having a speech component and a noise component. A processor analyzes the acoustic data and generates a noise indicator that identifies a characteristic of the noise component. An integrating decision logic processes the noise indicator and generates a noise model activation data structure that includes data that may be used by a speech recognition engine to adjust the activation of associated grammar rules or models.

Abstract: A voice reproduction apparatus includes an ambient sound analysis unit to analyze a characteristic of an ambient sound, a characteristic analysis unit to analyze an acoustic characteristic of a signal for reproduction, a reproduction timing adjusting unit to record the signal for reproduction and to read the signal for reproduction at a reproduction timing of follow-up reproduction, a reproduction speed changing unit to change a reproduction speed of the read signal for reproduction, and a control unit to control the reproduction timing adjusting unit so that the signal for reproduction is reproduced at the reproduction timing corresponding to an analysis result of the ambient sound analysis unit and to control the reproduction speed changing unit so that the signal for reproduction is reproduced at the reproduction speed corresponding to the analysis result of the ambient sound analysis unit and the acoustic characteristic obtained by the characteristic analysis unit.

Abstract: Methods and apparatus for characterizing media are described. In one example, a method of characterizing media includes capturing a block of audio; converting at least a portion of the block of audio into a frequency domain representation; dividing the frequency domain representation into a plurality of bands; determining a characteristic difference of a first band of the plurality of bands based on a comparison of a characteristic of the first band and a characteristic of a second band different from the first band, wherein the characteristic of the first band occurs at a time that is different than a time at which the characteristic of the second band takes place; and determining a signature bit based on a characteristic difference. Other examples are shown and described.

Abstract: Many embodiments may comprise logic such as hardware and/or code to implement user interface for traversal of long sorted lists, via audible mapping of the lists, using sensor based gesture recognition, audio and tactile feedback and button selection while on the go. In several embodiments, such user interface modalities are physically small in size, enabling a user to be truly mobile by reducing the cognitive load required to operate the device. For some embodiments, the user interface may be divided across multiple worn devices, such as a mobile device, watch, earpiece, and ring. Rotation of the watch may be translated into navigation instructions, allowing the user to traverse the list while the user receives audio feedback via the earpiece to describe items in the list as well as audio feedback regarding the navigation state. Many embodiments offer the user a simple user interface to traverse the list without visual feedback.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a device including a controller to detect a plurality of users engaging in a voice conference related to a presentation of a media program, convert speech dialog detected in the voice conference to textual dialog, detect from the textual dialog a behavioral profile of at least one of the plurality of users, and identify at least one of advertisement content and marketable media content based on the behavioral profile of the at least one user. Other embodiments are disclosed.

Abstract: A technique is disclosed for disambiguating speech input for multimodal systems by using a combination of speech and visual I/O interfaces. When the user's speech input is not recognized with sufficiently high confidence, a the user is presented with a set of possible matches using a visual display and/or speech output. The user then selects the intended input from the list of matches via one or more available input mechanisms (e.g., stylus, buttons, keyboard, mouse, or speech input). These techniques involve the combined use of speech and visual interfaces to correctly identify user's speech input. The techniques disclosed herein may be utilized in computer devices such as PDAs, cellphones, desktop and laptop computers, tablet PCs, etc.