Abstract: Textual Geographical Location relates a placename, which is a set of terms, from one to any maximum as defined in an individual language, to a unique point or area (many points) as found on a map or other coordinate system, such as the map of the United States as used in global positioning system (GPS).

Abstract: Apparatus and methods consistent with the present invention measure one or more of the characteristics of a voice recording and use such measurements to create a synthetic voice that approximates the recorded voice and uses such created synthetic voice to verbalize the content of an electronically conveyed written message such as an SMS text message. The vocal characteristics measured may include frequency, timbre, intensity, rhythm, and rate of speech as well as others.

Abstract: A smart playback method for TV programs includes: converting voice data to text data including a plurality of words; selecting a keyword from the words in the text data; providing a TV program according to the keyword; and controlling a screen to play the TV program.

Abstract: A method and apparatus for improving the performance of voice recognition in a mobile device are provided. The method of recognizing a voice includes: monitoring the usage pattern of a user of a device for inputting a voice; selecting predetermined words from among words stored in the device based on the result of monitoring, and storing the selected words; and recognizing a voice based on an acoustic model and predetermined words. In this way, a voice can be recognized by using prediction of whom the user mainly makes a call to. Also, by automatically modeling the device usage pattern of the user and applying the pattern to vocabulary for voice recognition based on probabilities, the performance of voice recognition, as actually felt by the user, can be enhanced.

Abstract: Embodiments herein include receiving a request to modify an audio characteristic associated with a first user for a voice communication system. One or more suggested modified audio characteristics may be provided for the first user, based on, at least in part, one or more audio preferences established by another user. An input of one or more modified audio characteristics may be received for the first user for the voice communication system. A user-specific audio preference may be associated with the first user for voice communications on the voice communication system, the user-specific audio preference including the one or more modified audio characteristics.

Abstract: Provided are an integrated multilingual consonant pattern search method and apparatus for extracting original strings, in correspondence with a number that is small compared to that of a conventional technology, as a search result and displaying the search result, by inputting a consonant pattern which is formed of a plurality of consonants, with respect to an original string list that is pre-stored in a database in a language written with a phonogram in which an initial consonant and a final consonant are distinguished from each other. Provided are also a method and apparatus for generating a character input unit for inputting consonant characters to be searched fast with a low typing error rate, by using the integrated multilingual consonant pattern search method.

Abstract: Voice-operable avionic systems and methods supporting utilization of speech recognition to facilitate control of avionic systems are disclosed. Utilizing speech recognition to control avionic systems may help reduce the head-down time of the flight crew. Safety features may also be implemented to ensure safety-critical commands are carried out as intended when commands are received through speech recognition. In addition, voice-operable avionic systems configured in accordance with embodiments of the inventive concepts disclosed herein may be implemented in manners that can help reduce the complexity and cost associated with obtaining certifications from aviation authorities.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media for modeling phonemes. One method includes receiving an acoustic sequence, the acoustic sequence representing an utterance, and the acoustic sequence comprising a respective acoustic feature representation at each of a plurality of time steps; for each of the plurality of time steps: processing the acoustic feature representation through each of one or more recurrent neural network layers to generate a recurrent output; processing the recurrent output using a softmax output layer to generate a set of scores, the set of scores comprising a respective score for each of a plurality of context dependent vocabulary phonemes, the score for each context dependent vocabulary phoneme representing a likelihood that the context dependent vocabulary phoneme represents the utterance at the time step; and determining, from the scores for the plurality of time steps, a context dependent phoneme representation of the sequence.

Abstract: Techniques are disclosed for adjusting user experience of a software application based on paralinguistic information. One embodiment presented herein includes a computer-implemented method for adjusting a user experience of a software application. The method comprises receiving, at a computing device, an audio stream comprising audio of a user. The method further comprises analyzing the audio stream for paralinguistic information to determine an attribute of the user. The method further comprises identifying content of the audio stream. The method further comprises determining one or more actions based on the content of the audio stream. The method further comprises selecting at least one of the one or more actions based on the attribute of the user.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media for learning pronunciations from acoustic sequences. One method includes receiving an acoustic sequence, the acoustic sequence representing an utterance, and the acoustic sequence comprising a sequence of multiple frames of acoustic data at each of a plurality of time steps; stacking one or more frames of acoustic data to generate a sequence of modified frames of acoustic data; processing the sequence of modified frames of acoustic data through an acoustic modeling neural network comprising one or more recurrent neural network (RNN) layers and a final CTC output layer to generate a neural network output, wherein processing the sequence of modified frames of acoustic data comprises: subsampling the modified frames of acoustic data; and processing each subsampled modified frame of acoustic data through the acoustic modeling neural network.

Abstract: Technology is disclosed for snippet pre-translation and dynamic selection of translation systems. Pre-translation uses snippet attributes such as characteristics of a snippet author, snippet topics, snippet context, expected snippet viewers, etc., to predict how many translation requests for the snippet are likely to be received. An appropriate translator can be dynamically selected to produce a translation of a snippet either as a result of the snippet being selected for pre-translation or from another trigger, such as a user requesting a translation of the snippet. Different translators can generate high quality translations after a period of time or other translators can generate lower quality translations earlier. Dynamic selection of translators involves dynamically selecting machine or human translation, e.g., based on a quality of translation that is desired.

Abstract: A method and an apparatus for processing a temporal envelope of an audio signal, and an encoder are disclosed. When multiple temporal envelopes are solved, continuity of signal energy can be well maintained, and in addition, complexity of calculating a temporal envelope is reduced. The method includes: obtaining a high-band signal of the current frame audio signal according to the received current frame audio signal; dividing the high-band signal of the current frame signal into M subframes according to a predetermined temporal envelope quantity M, where M is an integer, M is greater than or equal to 2; calculating a temporal envelope of each of the subframes; performing windowing on the first subframe of the M subframes and the last subframe of the M subframes by using an asymmetric window function; and performing windowing on a subframe except the first subframe and the last subframe of the M subframes.

Abstract: In some embodiments, a method of creating an automatic language characteristic recognition system. The method can include receiving a plurality of audio recordings. The method also can include segmenting each of the plurality of audio recordings to create a plurality of audio segments for each audio recording. The method additionally can include clustering each audio segment of the plurality of audio segments according to audio characteristics of each audio segment to form a plurality of audio segment clusters. Other embodiments are provided.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for receiving audio data encoding an utterance and environmental data, obtaining a transcription of the utterance, identifying an entity using the environmental data, submitting a query to a natural language query processing engine, wherein the query includes at least a portion of the transcription and data that identifies the entity, and obtaining one or more results of the query.

Abstract: A bit allocating method is provided that includes determining the allocated number of bits in decimal point units based on each frequency band so that a Signal-to-Noise Ratio (SNR) of a spectrum existing in a predetermined frequency band is maximized within a range of the allowable number of bits for a given frame; and adjusting the allocated number of bits based on each frequency band.

Abstract: A system for carrying out voice pattern recognition and a method for achieving same. The system includes an arrangement for acquiring an input voice, a signal processing library for extracting acoustic and prosodic features of the acquired voice, a database for storing a recognition dictionary, at least one instance of a prosody detector for carrying out a prosody detection process on extracted respective prosodic features, communicating with an end user application for applying control thereto.

Abstract: An audio signal decoding apparatus is provided that includes a receiver that receives an encoded information, a memory, and a processor that demultiplexes low-band encoding parameters, index information, and scale factor information from the encoded information. The processor also decodes the low-band encoding parameters to obtain a synthesized low frequency spectrum, replicates a high frequency subband spectrum based on the index information using the synthesized low frequency spectrum, and adjusts an amplitude of the replicated high frequency subband spectrum using the scale factor information. The processor further estimates a frequency of a harmonic component in the synthesized low frequency spectrum, adjusts a frequency of a harmonic component in the high frequency subband spectrum using the estimated harmonic frequency spectrum, and generates an output signal using the synthesized low frequency spectrum and the high frequency subband spectrum.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for receiving audio data corresponding to an utterance, determining that the audio data corresponds to a hotword, generating a hotword audio fingerprint of the audio data that is determined to correspond to the hotword, comparing the hotword audio fingerprint to one or more stored audio fingerprints of audio data that was previously determined to correspond to the hotword, detecting whether the hotword audio fingerprint matches a stored audio fingerprint of audio data that was previously determined to correspond to the hotword based on whether the comparison indicates a similarity between the hotword audio fingerprint and one of the one or more stored audio fingerprints that satisfies a predetermined threshold, and in response to detecting that the hotword audio fingerprint matches a stored audio fingerprint, disabling access to a computing device into which the utterance was spoken.

Abstract: A method for estimating the reliability of a result of a speaker recognition system concerning a testing audio and a speaker model, which is based on one, two, three or more model audios, the method using a Bayesian Network to estimate whether the result is reliable. In estimating the reliability of the result of the speaker recognition system one, two, three, four or more than four quality measures of the testing audio and one, two, three, four or more than four quality measures of the model audio(s) are used.

Abstract: A method and an apparatus for decoding a speech/audio bitstream are disclosed, where the method for decoding a speech/audio bitstream includes determining whether a current frame is a normal decoding frame or a redundancy decoding frame, obtaining a decoded parameter of the current frame by means of parsing when the current frame is a normal decoding frame or a redundancy decoding frame, performing post-processing on the decoded parameter of the current frame to obtain a post-processed decoded parameter of the current frame, and using the post-processed decoded parameter of the current frame to reconstruct a speech/audio signal.