Abstract: A speech-signal-processing-circuit configured to receive a time-frequency-domain-reference-speech-signal and a time-frequency-domain-degraded-speech-signal. The time-frequency-domain-reference-speech-signal comprises: an upper-band-reference-component with frequencies that are greater than a frequency-threshold-value; and a lower-band-reference-component with frequencies that are less than the frequency-threshold-value. The time-frequency-domain-degraded-speech-signal comprises: an upper-band-degraded-component with frequencies that are greater than the frequency-threshold-value; and a lower-band-degraded-component with frequencies that are less than the frequency-threshold-value.

Abstract: A method for recognizing time information from speech data includes: obtaining text information corresponding to speech data; identifying a first time feature contained in the speech data based on the text information; searching within a respective configuration file corresponding to the first time feature to obtain a corresponding time identifier for the first time feature; assigning the corresponding time identifier to a field that is in intermediate time data and that corresponds to the respective configuration file in which the first time feature is located, the intermediate time data comprising multiple fields, and each field being corresponding to a respective configuration file of a plurality of configuration files, each of the plurality of configuration files corresponding a respective category related to time; and obtaining, according to content of one or more fields in the intermediate time data, system time data corresponding to the time information contained in the speech data.

Abstract: Techniques are disclosed for initiating an electronic file transfer between users based on conversational context. A cognitive service monitors a communication between a given user and another user. The communication is exchanged by a mobile device associated with the given user and a mobile device of the other user. The cognitive service determines a context of the communication. The cognitive service detects, based on the context of the communication, a request between the users to share a file maintained by the given user. Upon successfully validating the request, a transfer of the file to the second user is performed.

Abstract: An approach is provided for global voice-based entry of location information. The approach involves partitioning a global speech decoding graph into spatial partitions. The approach also involves determining key entities occurring in each spatial partitions to construct a combined set of key entities. The approach further involves creating a retrieval index to map the key entities in the combined set of key entities to a corresponding partition. A first partition, the combined set of key entities, and the retrieved index are stored in a memory of a device for processing a voice input signal. A second partition that is not in the memory of the device is retrieved based on the combined set of key entities and the retrieval index to automatically re-process the voice input signal when an out-of-vocabulary result is obtained from the first partition.

Abstract: Disclosed herein is a method of enhancing speech. The method includes calculating a far-end speech spectrum by performing fast Fourier transformation of a signal received by a far-end user, calculating a background noise spectrum collected by a microphone provided to a mobile device of a near-end user; calculating a gain from the far-end speech spectrum and the background noise spectrum using a speech intelligibility index-based module, and deriving an enhanced far-end speech spectrum by applying the gain to the far-end speech spectrum, wherein, in calculating a gain using a speech intelligibility index-based module, a power budget used for transmitting and receiving a speech signal is set to vary with the background noise spectrum.

Abstract: Various embodiments provide a communication system that monitors environmental noise at a computing device. The communication system additionally monitors call audio received at the computing device. Based on the environmental noise and the received call audio, the communication system determines that a user of the computing device is unlikely to hear the received call audio when played back by the computing device. In response to determining that the user of the computing device is unlikely to hear the received call audio, the communication system visually displays a transcription of at least a portion of the received call audio.

Abstract: An acoustic interference cancellation system that combines acoustic echo cancellation and an adaptive beamformer to cancel acoustic interference from an audio output. The system uses a fixed beamformer to generate a target signal in a look direction and an adaptive beamformer to generate noise reference signals corresponding to non-look directions. The noise reference signals are used to estimate acoustic noise using an acoustic interference canceller (AIC), while reference signals associated with loudspeakers are used to estimate an acoustic echo using a multi-channel acoustic echo canceller (MC-AEC). The system cancels the acoustic echo and the acoustic noise simultaneously by adding the estimate of the acoustic noise and the estimate of the acoustic echo to generate an interference reference signal and cancelling the interference reference signal from the target signal. The system jointly updates adaptive filters for the AIC and the MC-AEC logic to improve a robustness of the system.

Abstract: The invention relates to a system and method for integrating domain information into state transitions of a Finite State Transducer (“FST”) for natural language processing. A system may integrate semantic parsing and information retrieval from an information domain to generate an FST parser that represents the information domain. The FST parser may include a plurality of FST paths, at least one of which may be used to generate a meaning representation from a natural language input. As such, the system may perform domain-based semantic parsing of a natural language input, generating more robust meaning representations using domain information. The system may be applied to a wide range of natural language applications that use natural language input from a user such as, for example, natural language interfaces to computing systems, communication with robots in natural language, personalized digital assistants, question-answer query systems, and/or other natural language processing applications.

Abstract: Disclosed herein are various embodiments of methods and systems for constructing a first language model for use by a first Language Processing (LP) application of a plurality of LP applications. Each LP application of the plurality of LP applications receives one or more of a language based input, a derivative of the language based input, a response to the language based input and a derivative of the response. The method includes processing at least one input by a second LP application of the plurality of LP applications. Based on the processing of the second LP application, at least one output is generated. Subsequently, at least a portion of the first language model is constructed based on the at least one output.

Abstract: Spectral encoding methods are more robust when used with improved weak signal detection and synchronizations methods. Further robustness gains are achieved by using informed embedding, error correction and embedding protocols that enable signal to noise enhancements by folding and pre-filtering the received signal.

Abstract: Method and systems are provided for interpreting speech data. A method and system for recognizing speech involving a filter module to generate a set of processed audio data based on raw audio data; a translation module to provide a set of translation results for the raw audio data; and a decision module to select the text data that represents the raw audio data. A method for minimizing noise in audio signals received by a microphone array is also described. A method and system of automatic entry of data into one or more data fields involving receiving a processed audio data; and operating a processing module to: search in a trigger dictionary for a field identifier that corresponds to the trigger identifier; identify a data field associated with a data field identifier corresponding to the field identifier; and providing content data associated with the trigger identifier to the identified data field.

Abstract: Analyzing textual data is disclosed, including by: receiving textual data; determining that the textual data is a candidate for analogy analysis based at least in part on at least a portion of the textual data matching an analogical question template; extracting a source substantive from the textual data; using the source substantive to determine a target substantive from a word vector model that is trained on a set of training data; and generating an answer including the target substantive based at least in part on an analogical answer template corresponding to the analogical question template.

Abstract: Embodiments of the present disclosure provide a method of on-site speech broadcasting. The method includes receiving text signal, wherein the text signal is generated in response to a parameter reaching a predetermined value sensed by an on-site sensor (110) arranged in the field; converting the text signal to a speech signal by a speech converting module (210) disposed within an on-site speaker device (200); playing the converted speech signal by using a speaker (220) of the on-site speaker device (200). Embodiments of the present disclosure also provide an on-site speaker device and an on-site speech broadcasting system. With the embodiments of the present disclosure, an effective text-to-speech conversion is achieved, pre-created speech data is not necessary and the workload of design is reduced, making it more cost effective. Meanwhile, speech messages being broadcasted allow operators instantaneously knowing the content of the alarm.

Abstract: A method may include receiving, by a computing device associated with a user, a message from an origination source and receiving, by the computing device, an audio input. The method may also include determining, by the computing device and based at least in part on the audio input and contextual information, a probability that the user intends to send a response message to the origination source. The method may further include, responsive to determining that the probability the user intends to send the response message to the origination source satisfies a threshold probability, determining, by the computing device, that the user intends to send the response message to the origination source. The method may also include, responsive to determining that the user intends to send the response message to the origination source, generating, by the computing device and based on the audio input, the response message, and sending, by the computing device, the response message to the origination source.

Abstract: A method for adjusting a voice recognition system and a voice recognition system is disclosed, wherein the voice recognition system comprises a speaker and a microphone, and wherein the method comprises the steps of: memorizing an audio frequency signal, playing back the audio frequency signal by means of the speaker, generating a detection signal by detecting the audio frequency signal by means of the microphone, and adjusting parameters of the voice recognition system dependent on the detection signal.

Abstract: According to an embodiment, an information notification supporting device includes an analyzer configured to analyze an input voice so as to identify voice information indicating information related to speech; a storage unit configured to store therein a history of the voice information; an output controller configured to determine, using the history of the voice information, whether a user is able to listen to a message of which the user should be notified; and an output unit configured to output the message when it is determined that the user is in a state in which the user is able to listen to the message.

Abstract: Method and systems are provided for interpreting speech data. A method and system for recognizing speech involving a filter module to generate a set of processed audio data based on raw audio data; a translation module to provide a set of translation results for the raw audio data; and a decision module to select the text data that represents the raw audio data. A method for minimizing noise in audio signals received by a microphone array is also described. A method and system of automatic entry of data into one or more data fields involving receiving a processed audio data; and operating a processing module to: search in a trigger dictionary for a field identifier that corresponds to the trigger identifier; identify a data field associated with a data field identifier corresponding to the field identifier; and providing content data associated with the trigger identifier to the identified data field.

Abstract: System, method, and computer product embodiments for processing audio commands are described. A multimedia device receives a digitized audio sequence from a control device. The digitized audio sequence is associated with a display command for controlling a display device. Then, the digitized audio sequence is converted into an audio signature that corresponds to the digitized audio sequence. The multimedia device sends the audio signature to a platform server configured to use the audio signature to process the display command. After, the multimedia device receives content information from the platform server resulting from processing the display command. Then, the multimedia device causes the display device to provide media content related to the content information.

Abstract: Method and systems are provided for interpreting speech data. A method and system for recognizing speech involving a filter module to generate a set of processed audio data based on raw audio data; a translation module to provide a set of translation results for the raw audio data; and a decision module to select the text data that represents the raw audio data. A method for minimizing noise in audio signals received by a microphone array is also described. A method and system of automatic entry of data into one or more data fields involving receiving a processed audio data; and operating a processing module to: search in a trigger dictionary for a field identifier that corresponds to the trigger identifier; identify a data field associated with a data field identifier corresponding to the field identifier; and providing content data associated with the trigger identifier to the identified data field.

Abstract: An information processing apparatus may include a control device to determine a plurality of candidate texts to correct a target text which is from an input text string, based on preceding text located at a position preceding the target text and succeeding text located at a position succeeding the target text.