Abstract: A voice decoding apparatus includes an MBE-type decoder, a sampling convertor, a non-linear components generator and an adder. The decoder decodes digital voice-encoded information to generate a first decoded voice signal. The convertor converts the first decoded voice signal to a second decoded voice signal with a higher sampling frequency. The generator performs a non-linear process to the first or second decoded voice signal to generate an additional voice signal with the same sampling frequency as the second decoded voice signal. The additional voice signal has components in a frequency band in which the first decoded voice signal has no component and continuing to another frequency band of the first decoded voice signal. The adder adds the second decoded voice signal to the additional voice signal.

Abstract: A voice retrieval apparatus includes a display, a memory, and a processor executing a voice recording process of storing recorded voices in the memory, an accepting process of accepting a retrieval term, a retrieval process of retrieving, from the recorded voices, a candidate segment where an utterance of the accepted retrieval term is estimated, a replay process of replaying voices in the candidate segment retrieved in the retrieval process, and a display control process of adding a marking to display information indicating a transition of the recorded voices in time based on the replay result of the voices in the candidate segment in the replay process, and displaying the display information with the marking on the display, the marking specifying an utterance location of the voices in the candidate segment.

Abstract: A method for voice modification during a telephone call comprising receiving a source audio signal associated with at least one participant, wherein the source audio signal comprises a voice of the at least one participant, detecting a source dialect of the at least one participant, selecting a target dialect based on at least a characteristic of a target participant and creating a modulated audio signal based on the source audio signal, the source dialect, and the target dialect and transmitting the modulated audio signal to the target participant.

Abstract: A system and method for automatically generating a narrative story receives data and information pertaining to a domain event. The received data and information and/or one or more derived features are then used to identify a plurality of angles for the narrative story. The plurality of angles is then filtered, for example through use of parameters that specify a focus for the narrative story, length of the narrative story, etc. Points associated with the filtered plurality of angles are then assembled and the narrative story is rendered using the filtered plurality of angles and the assembled points.

Abstract: Methods, apparatus, and computer-readable media are described herein related to a user interface (UI) that can be implemented on a head-mountable device (HMD). The UI can include a voice-navigable UI. The voice-navigable UI can include a voice navigable menu that includes one or more menu items. The voice-navigable UI can also present a first visible menu that includes at least a portion of the voice navigable menu. In response to a first utterance comprising one of the one or more menu items, the voice-navigable UI can modify the first visible menu to display one or more commands associated with the first menu item. In response to a second utterance comprising a first command, the voice-navigable UI can invoke the first command. In some embodiments, the voice-navigable UI can display a second visible menu, where the first command can be displayed above other menu items in the second visible menu.

Abstract: Described herein are systems, methods, and apparatus for determining audio context between an audio source and an audio sink and selecting signal profiles based at least in part on that audio context. The signal profiles may include noise cancellation which is configured to facilitate operation within the audio context. Audio context may include user-to-user and user-to-device communications.

Abstract: In an embodiment, bitstream elements of sub-frames are encoded differentially to a global gain value so that a change of the global gain value results in an adjustment of an output level of the decoded representation of the audio content. Concurrently, the differential coding saves bits. Even further, the differential coding enables the lowering of the burden of globally adjusting the gain of an encoded bitstream. In another embodiment, a global gain control across CELP coded frames and transform coded frames is achieved by co-controlling the gain of the codebook excitation of the CELP codec, along with a level of the transform or inverse transform of the transform coded frames. In another embodiment, the gain value determination in CELP coding is performed in the weighted domain of the excitation signal.

Abstract: In some embodiments, the invention involves context based search engine using a user selected term within a media experience. A natural language processor module is configured to provide context based keywords related to the search term and from within the media experience. In some embodiments, a proximity based statistical analysis is used to derive the keywords. The keywords are provided to at least one content browser or other search engine(s) to effect the search. In some embodiments, a machine learning module is communicatively coupled to the natural language processer to further refine the context for selecting relevant keywords. The context search engine, natural language processor module, machine learning module and search engine may reside on the same computing device or be distributed among a variety of local, remote and cloud devices for processing. Other embodiments are described and claimed.

Abstract: A system and method for an integrated, multi-modal, multi-device natural language voice services environment may be provided. In particular, the environment may include a plurality of voice-enabled devices each having intent determination capabilities for processing multi-modal natural language inputs in addition to knowledge of the intent determination capabilities of other devices in the environment. Further, the environment may be arranged in a centralized manner, a distributed peer-to-peer manner, or various combinations thereof. As such, the various devices may cooperate to determine intent of multi-modal natural language inputs, and commands, queries, or other requests may be routed to one or more of the devices best suited to take action in response thereto.

Abstract: The exemplary embodiments described herein are related to techniques for automatically generating narratives about data based on communication goal data structures that are associated with configurable content blocks. The use of such communication goal data structures facilitates modes of operation whereby narratives can be generated in real-time and/or interactive manners.

Abstract: Methods and apparatus for audio watermarking are disclosed. Example methods disclosed herein include obtaining a watermarked noise signal having energy in a first frequency band, the watermarked noise signal being associated with a first operating state of a device. Disclosed example methods also include adjusting the watermarked noise signal based on an ambient audio level sensed by an audio sensor. Disclosed example methods further include emitting the adjusted watermarked noise signal via a speaker in response to determining that the device is in the first operating state.

Abstract: The exemplary embodiments described herein are related to techniques for automatically generating narratives about data based on communication goal data structures that are associated with configurable content blocks. The use of such communication goal data structures facilitates modes of operation whereby narratives can be generated in real-time and/or interactive manners.

Abstract: A system and method is presented for performing dual mode speech recognition, employing a local recognition module on a mobile device and a remote recognition engine on a server device. The system accepts a spoken query from a user, and both the local recognition module and the remote recognition engine perform speech recognition operations on the query, returning a transcription and confidence score, subject to a latency cutoff time. If both sources successfully transcribe the query, then the system accepts the result having the higher confidence score. If only one source succeeds, then that result is accepted. In either case, if the remote recognition engine does succeed in transcribing the query, then a client vocabulary is updated if the remote system result includes information not present in the client vocabulary.

Abstract: A system and method are presented for selectively biased linear discriminant analysis in automatic speech recognition systems. Linear Discriminant Analysis (LDA) may be used to improve the discrimination between the hidden Markov model (HMM) tied-states in the acoustic feature space. The between-class and within-class covariance matrices may be biased based on the observed recognition errors of the tied-states, such as shared HMM states of the context dependent tri-phone acoustic model. The recognition errors may be obtained from a trained maximum-likelihood acoustic model utilizing the tied-states which may then be used as classes in the analysis.

Abstract: Systems, methods and apparatus for learning parsing rules and argument identification from crowdsourcing of proposed command inputs. Crowdsourcing techniques to generate rules for parsing input sentences. A parse is used to determine whether the input sentence invokes a specific action, and if so, what arguments are to be passed to the invocation of the action.

Abstract: Contextual, focus-based language translation within a broadcast radio automation software application allows the user interface to enable interaction with users having various written or graphical language requirements. The software does not require special training or learning to enable the translation of text within the application. When the user enters a translation mode and selects a text string in the original language, the program displays a corresponding text string in a second language based on the specific display area, e.g., child window. The translated text string can be displayed proximate to original first text string within the specific display area, and can further be displayed as an entry in a table of a translation window which also displays additional text strings in the second language corresponding to additional text string fields for the specific display area.

Abstract: Systems and methods for controlling a flight deck multifunction control display unit are disclosed. In various embodiments, the systems may comprise a flight management system or other MCDU driven devices, a command database that stores a plurality of voice commands and a plurality of multifunction control display unit commands. In various embodiments, each voice command is associated with one of the plurality of multifunction control display unit commands. The systems may further comprise a pilot voice interface configured to receive a voice command from a pilot and transmit the voice command to the multifunction control display unit. The multifunction control display unit can receive the voice command from the pilot voice interface and, in response, access the command database to identify a multifunction control display unit command in the command database that is associated with the voice command.

Abstract: In general, techniques are described for coding a number of code vectors for independent frame of higher order ambisonic coefficients. An audio decoding device comprising a memory and a processor may perform the techniques. The memory may store a first frame of a bitstream and a second frame of the bitstream. The processor may extract, from the first frame, one or more bits indicative of whether the first frame is an independent frame that includes information specifying a number of code vectors to be used when performing vector dequantization with respect to the vector. The processor may also extract, from the first frame without referencing the second frame, the information specifying the number of code vectors.

Abstract: An audio encoder has a window function controller, a windower, a time warper with a final quality check functionality, a time/frequency converter, a TNS stage or a quantizer encoder, the window function controller, the time warper, the TNS stage or an additional noise filling analyzer are controlled by signal analysis results obtained by a time warp analyzer or a signal classifier. Furthermore, a decoder applies a noise filling operation using a manipulated noise filling estimate depending on a harmonic or speech characteristic of the audio signal.

Abstract: The disclosure proposes a smart conversation method and an electronic device using the same method. According to one of the exemplary embodiments, an electronic device may receive via a receiver a first communication in a first communication type and determining a recipient status. The electronic device may determine a second communication type as an optimal communication type based on the recipient status. The electronic device may convert the first communication into a second communication that is suitable for the second communication type. The electronic device may transmit via a transmitter the second communication in the second communication type.