323results about How to "Improve speech recognition performance" patented technology

A method for speech recognition includes providing a source of geographical information within a vehicle. The geographical information pertains to a current location of the vehicle, a planned travel route of the vehicle, a map displayed within the vehicle, and/or a gesture marked by a user on a map. Words spoken within the vehicle are recognized by use of a speech recognition module. The recognizing is dependent upon the geographical information.

Apparatus with accompanying subsystems and methods for developing conversational computer applications. As a user interface, the apparatus allows for a user to initiate the conversation. The apparatus also answers simple and complex questions, understands complex requests, pursues the user for further information when the request is incomplete, and in general provides customer support with a human like conversation while, at the same time, it is capable to interact with a company's proprietary database. As a development tool, the apparatus allows a software developer to implement a conversational system much faster than takes, with current commercial systems to implement basic dialog flows. The apparatus contains three major subsystems: a state transition inference engine, a heuristic answer engine and a parser generator with semantic augmentations. A main process broker controls the flow and the interaction between the different subsystems. The state transition inference engine handles requests that require processing a transaction or retrieving exact information. The heuristic answer engine answers questions that do not require exact answers but enough information to fulfill the user's request. The parser generator processes the user's natural language request, that is, it processes the syntactical structure of the natural language requests and it builds a conceptual structure of the request. After the parser generator processes the user's request, a main process broker feeds the conceptual structure to either the heuristic answer engine or to the state transition inference engine. The interaction between the main process broker and the subsystems creates a conversational environment between the user and the apparatus, while the apparatus uses information from proprietary databases to provide information, or process information, during the course of the conversation. The apparatus is equipped with a programming interface that allows implementers to declare and specify transactions based requests and answers to a multiplicity of questions. The apparatus may be used with a speech recognition interface, in which case, the apparatus improves the recognition results through the context implicitly created by the apparatus.

A speech recognition method and system for use in a vehicle navigation system utilize grammar weighted based upon geographical information regarding the locations corresponding to the tokens in the grammars and/or the location of the vehicle for which the vehicle navigation system is used, in order to enhance the performance of speech recognition. The geographical information includes the distances between the vehicle location and the locations corresponding to the tokens, as well as the size, population, and popularity of the locations corresponding to the tokens.

The present invention relates to a method for detecting a voice section in time-space by using audio and video information. According to an embodiment of the present invention, a method for detecting a voice section from time-space by using audio and video information comprises the steps of: detecting a voice section in an audio signal which is inputted into a microphone array; verifying a speaker from the detected voice section; sensing the face of the speaker by using a video signal which is inputted into a camera if the speaker is successfully verified, and then estimating the direction of the face of the speaker; and determining the detected voice section as the voice section of the speaker if the estimated face direction corresponds to a reference direction which is previously stored.

A speech recognition apparatus includes a speech recognition dictionary and a speech recognition unit. The speech recognition dictionary includes comparison data used to recognize a voice input. The speech recognition unit is adapted to calculate the score for each comparison data by comparing voice input data generated based on the voice input with each comparison data, recognize the voice input based on the score, and produce the recognition result of the voice input. The speech recognition apparatus further includes data indicating score weights associated with particular comparison data, used to weight the scores calculated for the particular comparison data. After the score is calculated for each comparison data, the score weights are added to the scores of the particular comparison data, and the voice input is recognized based on total scores including the added score weights.

A method and system for providing natural language processing in a communication system is disclosed. A voice request is generated with a remote terminal that is transmitted to a base station. A voice recognition application is then used to identify a plurality of words that are contained in the voice request. After the words are identified, a grammar associated with each word is also identified. Once the grammars have been identified, each word is categorized into a respective grammar category. A structured response is then generated to the voice request with a response generation application.

In an information processing system having an equipment controller for enabling the digitization of complex work processes conducted during testing and/or operation of machinery and equipment, a local wireless communications network is implemented using at least one fixed point wireless communications access station and one or more voice-responsive wireless mobile computing/communications devices that are capable of accurate vocal command recognition in noisy industrial environments. The computing/communications device determines spectral characteristics of ambient non-speech background noises and provides active noise cancellation through the use of a directional microphone and an adaptive noise tracking and subtraction process. The device is carried by a user and is responsive to one or more vocal utterances of the user for communicating data to the information processing system and/or generating operational control commands to provide to the equipment controller for controlling the machinery or equipment.

The invention discloses a speech recognition device and a speech recognition method. The speech recognition device comprises a unit which is configured to obtain speech inputted by a current user, a unit which is configured to split obtained speech and output at least two voice command segments, a unit which is configured to recognize a first predefined voice command from the voice command segments through using an acoustic model unrelated to a speaker, a unit which is configured to calculate a transformation matrix for the current user based on a voice command segment which is recognized as the first predefined voice command, a unit which is configured to select an acoustic model for the current user from acoustic models registered in the speech recognition device based the calculated transformation matrix, and a unit which is configured to recognize a second voice command from the voice command segments through using the selected acoustic model. According to the speech recognition device and the speech recognition method of the invention adopted, speech recognition performance can be improved through using the selected acoustic model (AM).

In accordance with a present invention speech recognition is disclosed. It uses a microphone to receive audible sounds input by a user into a first computing device having a program with a database consisting of (i) digital representations of known audible sounds and associated alphanumeric representations of the known audible sounds and (ii) digital representations of known audible sounds corresponding to mispronunciations resulting from known classes of mispronounced words and phrases. The method is performed by receiving the audible sounds in the form of the electrical output of the microphone. A particular audible sound to be recognized is converted into a digital representation of the audible sound. The digital representation of the particular audible sound is then compared to the digital representations of the known audible sounds to determine which of those known audible sounds is most likely to be the particular audible sound being compared to the sounds in the database. A speech recognition output consisting of the alphanumeric representation associated with the audible sound most likely to be the particular audible sound is then produced. An error indication is then received from the user indicating that there is an error in recognition. The user also indicates the proper alphanumeric representation of the particular audible sound. This allows assistant to determine whether the error is a result of a known type or instance of mispronunciation. In response to a determination of error corresponding to a known type or instance of mispronunciation, the system presents an interactive training program from the computer to the user to enable the user to correct such mispronunciation.

A speech recognition method comprises model selection step which selects a recognition model based on characteristic information of input speech and speech recognition step which translates input speech into text data based on the selected recognition model.

The present invention relates to implementing new ways of automatically and robustly evaluating agent performance, customer satisfaction, campaign and competitor analysis in a call-center and it is comprising; analysis consumer server, call pre-processing module, speech-to-text module, emotion recognition module, gender identification module and fraud detection module.

A virtual vocabulary database is provided for use with a with a particular user database as part of a speech recognition system. Vocabulary elements within the virtual database are imported from the user database and are tagged to include numerical data corresponding to the historical use of the vocabulary element within the user database. For each speech input, potential vocabulary element matches from the speech recognition system are provided to the virtual database software which creates virtual sub-vocabularies from the criteria according to predefined criteria templates. The software then applies vocabulary element weighting adjustments according to the virtual sub-vocabulary weightings and applies the adjustment to the default weighting provided by the speech recognition system. The modified weightings are returned with the associated vocabulary elements to the speech engine for selection of an appropriate match to the input speech.

The invention discloses a man-machine voice interaction method. The method includes the following steps: dividing a recording signal and reference signal echo cancellation process into multiple sub-processes in a system state, splicing the sub-processes, and obtaining an output result of recording signal and reference signal echo cancellation, the reference signal being a voice signal emitted by a machine player; inputting the output result of recording signal and reference signal echo cancellation to a VAD model, and outputting a voice recognition result in real time when a user voice signal is detected; stopping a system from playing the voice signal when determining, based on the real-time output voice recognition result, the intention of a user and interrupting output of the system voice signal, entering a user state, and monitoring and recognizing the user voice signal; and sending feedback to the user according to the final voice recognition result when detecting that the output of the user voice signal is finished, and completing the interruption and responding process of one round of users. Interruption behaviors of users during an interaction process can be effectively detected, talking opportunities and contents of a machine can be decided, and thus the machine becomes smarter.

A system and method for collaborating multiple ASR (automatic speech recognition) systems. The system and method analyzes voice data on various computers having speech recognition residing thereon. The speech recognition residing on the various computers may be different systems. The speech recognition systems detect voice data and recognize their respective masters. The master computer as well as those computers which did not recognize their master may analyze the voice data (evaluate) and then integrate this analyzed voice data into a single decoded output. In this manner, many different speakers, utilizing the system and method for collaborating multiple ASR systems, may have their voice data analyzed and integrated into a single decoded output, regardless of ASR systems.

The invention provides a method for adjusting a confidence coefficient threshold of voice recognition and an electronic device. The method for adjusting the confidence coefficient threshold of voice recognition is applied to the electronic device which supports voice recognition. The current confidence coefficient threshold of voice recognition serves as a first value. The method for adjusting the confidence coefficient threshold of voice recognition comprises the steps of detecting N parameters to obtain N detection results, wherein the N is an integer larger than or equal to one; adjusting the confidence coefficient threshold on the basis of at least one of the N detection results so that the confidence coefficient threshold can be changed from the first value into a second value, wherein the second value is equal to or different from the first value.

A method for providing participants to a multiparty meeting with a transcript of the meeting, comprising the steps of: establishing an meeting among two or more participants; exchanging during said meeting voice data as well as documents; uploading at least a part of said voice data and at least a part of said documents to a remote speech recognition server (1), using an application programming interface of said remote speech recognition server; converting at least a part of said voice data to text with an automatic speech recognition system (13) in said remote speech recognition server, wherein said automatic speech recognition system uses said documents to improve the quality of speech recognition; building in said remote speech recognition server a computer object (120) embedding at least a part of said voice data, at least a part of said documents, and said text; making said computer object (120) available to at least one of said participant.

The invention provides an artificial cochlea speech processing method based on frequency modulation information and an artificial cochlea speech processor. The artificial cochlea speech processing method comprises the following steps: pre-emphasizing a speech signal; decomposing the speech signal by an analysis filter into a plurality of sub frequency bands; then, extracting the time-domain envelope information of each sub frequency band signal; adopting a Hilbert transform method to extract the frequency modulation information of a low-frequency part to multiple by time-domain envelopes so asto acquire a synthetic time-domain envelope containing the frequency modulation information; utilizing various acquired time-domain envelopes of the sub frequency bands to modulate a pulse sequence by a pulse generator; adding modulated pulses of various sub frequency bands to acquire a finally synthesized stimulus signal; and sending the stimulus signal to an electrode to generate an electric pulse to stimulate the auditory nerve. The artificial cochlea speech processor is suitable for deafness patients speaking Chinese as a native language to recognize speeches in a noise environment and has noise robustness, thereby enabling the deafness patients to feel more fine speech structure information, enhancing the speech recognition abilities of the deafness patients in the noise environmentand benefiting the tone recognition.

To improve the performance and the recognition rate of a method for recognizing speech in a dialogue system, or the like, it is suggested to derive emotion information data (EID) from speech input (SI) being descriptive for an emotional state of a speaker or a change thereof based upon which a process of recognition is chosen and/or designed.

A voice controlled medical system with improved speech recognition includes a first microphone array, a second microphone array, a controller in communication with the first and second microphone arrays, and a medical device operable by the controller. The controller includes a beam module that generates a first beamed signal using signals from the first microphone array and a second beamed signal using signals from the second microphone array. The controller also includes a comparison module that compares the first and second beamed signals and determines a correlation between the first and second beamed signals. The controller also includes a voice interpreting module that identifies commands within the first and second beamed signals if the correlation is above a correlation threshold. The controller also includes an instrument control module that executes the commands to operate said medical device.