Abstract: A hearing aid apparatus includes a frequency analysis device configured to determine an instantaneous fundamental frequency value of a speech signal for a time portion of the speech signal. A statistical evaluation device is configured to determine an average fundamental frequency value of the speech signal over several time portions. A hearing aid apparatus further includes a fundamental frequency modifier that is configured to modify the instantaneous fundamental frequency value to a modified fundamental frequency value such that a difference or a quotient of the instantaneous fundamental frequency value is changed to the average fundamental frequency value according to a specific function. Thereby, a frequency range may be modified within which the fundamental frequency value varies. The hearing aid apparatus further includes a speech signal generator that is configured to generate, on the basis of the modified fundamental frequency value, a speech signal modified with regard to the fundamental frequency.

Abstract: Method, system and product for automatic performance of user interaction operations on a computing device. A method comprising: obtaining an identifier of an operations sequence; obtaining the operations sequence by searching a repository of operations sequences using the identifier, wherein the repository of operation sequences comprises operations sequences defined based on a previous execution of one or more operations by another computing device other than the computing device on behalf of another user other than the user; and automatically executing the operations sequence or portion thereof on the computing device. Another method comprises: identifying elements in a layout of a GUI, displaying in visible proximity to each of the elements an assigned unique label; recognizing speech by a user vocally indicating a selected element by referring to the assigned label; and, automatically performing a user interaction operation on the selected element.

Abstract: According to an embodiment, a transliteration work support device includes an analysis unit, a storage unit, an estimation unit, a construction unit, a correction unit, and an update unit. The analysis unit performs language analysis on document data and creates transliteration auxiliary information representing a way of transliteration of a word or a phrase in the document data. The storage unit stores a correction history representing a way of transliteration corrected in the past of the word or the phrase. The estimation unit estimates a correction place and a correction candidate of the document data or the transliteration auxiliary information from the history. The construction unit constructs work list information including work items corresponding to types of corrections according to the correction candidate and progress information. The correction unit corrects the document data or the transliteration auxiliary information.

Abstract: A voice signal processing method and apparatus, which are used to process a voice signal collected by a microphone of a terminal in order to meet requirements of the terminal in different application modes for the voice signal generated after the processing. The method includes collecting at least two voice signals, determining a current application mode of a terminal, determining, according to the current application mode from the voice signals, voice signals corresponding to the current application mode, and performing, in a preset voice signal processing manner that matches the current application mode, beamforming processing on the corresponding voice signals.

Abstract: Described are techniques for performing system management such as a system including one or more data storage systems or appliances. User input may be received by a chat processing engine. The user input may be a natural language input to perform a request. The user input may be received from a management application used to perform data storage management. The chat processing engine may perform first processing of the user input to determine a response to the request. The response may be sent to the management application. The natural language input from a user may be in the form of text or voice that is converted to text for processing by the chat processing engine.

Abstract: According to one embodiment, a method performed by an electronic device includes: receiving an audio signal comprising voice and background sound via a microphone; receiving a user's operation to set a loudness of the voice or the background sound; setting a balance between a first gain of the voice and a second gain of the background sound according to the user's operation; separating the input audio signal into a first signal of the voice and a second signal of the background sound; amplifying the first signal according to the first gain; amplifying the second signal according to the second gain; and outputting the first signal and the second signal at least partially overlapping each other via a speaker.

Abstract: One-to-many comparisons of callers' voice prints with known voice prints to identify any matches between them. When a customer communicates with a particular entity, such as a customer service center, the system makes a recording of the real-time call including both the customer's and agent's voices. The system segments the recording to extract at least a portion of the customer's voice to create a customer voice print, and it formats the segmented voice print for network transmission to a server. The server compares the customer's voice print with multiple known voice prints to determine any matches, meaning that the customer's voice print and one of the known voice prints are likely from the same person. The identification of any matches can be used for a variety of purposes, such as determining whether to authorize a transaction requested by the customer.

Abstract: A global speech user interface (GSUI) comprises an input system to receive a user's spoken command, a feedback system along with a set of feedback overlays to give the user information on the progress of his spoken requests, a set of visual cues on the television screen to help the user understand what he can say, a help system, and a model for navigation among applications. The interface is extensible to make it easy to add new applications.

Abstract: An audio encoding device includes a processor; and a memory which stores a plurality of instructions, which when executed by the processor, cause the processor to execute: calculating a similarity in phase of a first channel signal and a second channel signal contained in a plurality of channels of an audio signal; and selecting, based on the similarity, a first output that outputs one of the first channel signal and the second channel signal, or a second output that outputs both of the first channel signal and the second channel signal.

Abstract: An acoustic signal coding apparatus includes a subband classifier that classifies subbands obtained by dividing a frequency-domain spectrum into a plurality of perceptually important first-category subbands and the other subbands referred to as second-category subbands according to at least one of measures in terms of energy and peak property, a subband peak-algebraic vector quantization (SBP-AVQ) vector generator that generates an SBP-AVQ vector by collecting a maximum peak from each first-category subband, outputs the generated SBP-AVQ vector, and outputs peak position information indicating the positions of the maximum peaks, a bit distributor that distributes bits for AVQ coding to the SBP-AVQ vector and the second-category subband vector, and an AVQ coder that performs AVQ coding on the SBP-AVQ vector and the second-category subband vector.

Abstract: An electronic communication device (104) receives a first set of digital audio signals via radio frequency signaling, translates the first set of digital audio signals from speech of a first language (e.g. English) into translated speech of a second language (e.g., French), and emits the translated speech of the second language via a loudspeaker (226). The electronic communication device (104) also receives a second set of digital audio signals, which it translates from speech of the second language into translated speech of the first language. The electronic communication device (104) transmits, via radio frequency signaling, a third set of digital audio signals. The third set of digital audio signals represents the translated speech of the first language.

Abstract: A computer program product, according to one embodiment, includes a computer readable storage medium having program instructions embodied therewith, the program instructions readable and/or executable by a computer to cause the computer to: receive one or more voice signals from one or more participants of a call, correlate each of the one or more received voice signals with speaker profiles in a database, output speaker identification information from the correlated speaker profiles, and send the speaker identification information at least one of the participants of the call. Other systems, methods, and computer program products are described in additional embodiments.

Abstract: This disclosure describes systems and methods for audibly presenting metadata. Audibly presentable metadata is referred to as audible metadata. Audible metadata may be associated with one or more media objects. In one embodiment, audible metadata is pre-recorded requiring little or no processing before it can be rendered. In another embodiment, audible metadata is text, and a text-to-speech conversion device may be used to convert the text into renderable audible metadata. Audible metadata may be rendered at any point before or after rendering of a media object, or may be rendered during rendering of a media object via a dynamic user request.

Abstract: A voice selection supporting device according to an embodiment of the present invention includes an acceptance unit that accepts input of a text, an analysis knowledge storage unit that stores therein text analysis knowledge to be used for characteristic analysis for the input text, an analysis unit that analyzes a characteristic of the text by referring to the text analysis knowledge, a voice attribute storage unit that stores therein a voice attribute of each voice dictionary, an evaluation unit that evaluates similarity between the voice attribute of the voice dictionary and the characteristic of the text, and a candidate presentation unit that presents, based on the similarity, a candidate for the voice dictionary suitable for the text.

Abstract: In an embodiment, a method of providing an on demand translation service is provided. A subscriber may be charged a reduced fee or no fee for use of the on demand translation service in exchange for displaying commercial messages to the subscriber, the commercial messages being selected based on subscriber information. A multimedia signal including information in a source language may be received. The information may be obtained as text in the source language from the multimedia signal. The text may be translated from the source language to a target language. Translated information, based on the translated text, may be transmitted to a processing device for presentation to the subscriber. The received multimedia signal may be sent to a multimedia device for viewing.

Abstract: Noise filling of a spectrum of an audio signal is improved in quality with respect to the noise filled spectrum so that the reproduction of the noise filled audio signal is less annoying, by performing the noise filling in a manner dependent on a tonality of the audio signal.

Abstract: A device including a speech recognition function which recognizes speech from a user, includes: a loudspeaker which outputs speech to a space; a microphone which collects speech in the space; a first speech recognition unit which recognizes the speech collected by the microphone; a command control unit which issues a command for controlling the device, based on the speech recognized by the first speech recognition unit; and a control unit which prohibits the command issuance unit from issuing the command, based on the speech to be output from the loudspeaker.

Abstract: Disclosed herein are systems, computer-implemented methods, and tangible computer-readable media for handling missing speech data. The computer-implemented method includes receiving speech with a missing segment, generating a plurality of hypotheses for the missing segment, identifying a best hypothesis for the missing segment, and recognizing the received speech by inserting the identified best hypothesis for the missing segment. In another method embodiment, the final step is replaced with synthesizing the received speech by inserting the identified best hypothesis for the missing segment. In one aspect, the method further includes identifying a duration for the missing segment and generating the plurality of hypotheses of the identified duration for the missing segment. The step of identifying the best hypothesis for the missing segment can be based on speech context, a pronouncing lexicon, and/or a language model. Each hypothesis can have an identical acoustic score.

Abstract: A speech recognition capability in which speakers of spoken text are identified based on the contour of sound waves representing the spoken text. Variations in the contour of the sound waves are identified, features are assigned to those variations, and parameters of those features are grouped into predefined characteristics. The predefined characteristics are combined into voice characteristic groups. If a prior voice characteristic group is present, the voice characteristic group from the soundlet is compared to existing voice characteristic groups and, if a match is present, the sound construct is assigned to a speaker identified by the existing voice characteristic group.

Abstract: A system and method for parallel speech recognition processing of multiple audio signals produced by multiple microphones in a handheld portable electronic device. In one embodiment, a primary processor transitions to a power-saving mode while an auxiliary processor remains active. The auxiliary processor then monitors the speech of a user of the device to detect a wake-up command by speech recognition processing the audio signals in parallel. When the auxiliary processor detects the command it then signals the primary processor to transition to active mode. The auxiliary processor may also identify to the primary processor which microphone resulted in the command being recognized with the highest confidence. Other embodiments are also described.