Abstract: A requirements testing system facilitates the review and analysis of requirement statements for software applications. The requirements testing system automatically generates test artifacts from the requirement statements, including test scenarios, test conditions, test hints, and expected results. These test artifacts characterize the requirements statements to provide valuable analysis information that aids understanding what the intentions of the requirement statements are. The automation of the generation of these test artifacts produces numerous benefits, including fewer errors, objectivity, and no dependency on the skills and experience of a creator.

Abstract: The subject matter of this specification can be implemented in, among other things, a computer-implemented method for correcting words in transcribed text including receiving speech audio data from a microphone. The method further includes sending the speech audio data to a transcription system. The method further includes receiving a word lattice transcribed from the speech audio data by the transcription system. The method further includes presenting one or more transcribed words from the word lattice. The method further includes receiving a user selection of at least one of the presented transcribed words. The method further includes presenting one or more alternate words from the word lattice for the selected transcribed word. The method further includes receiving a user selection of at least one of the alternate words. The method further includes replacing the selected transcribed word in the presented transcribed words with the selected alternate word.

Abstract: A speech signal processing apparatus includes an amplitude and phase signal generation section that, based on an analyzing signal expressed by a complex signal generated from a speech signal applied with pitch marks every 1 pitch cycle, generates an amplitude signal and a phase signal on the time axis of the speech signal, a phase signal conversion section that converts the phase signal into a phase signal of a target pitch cycle width for each section of the 1 pitch cycle width based on the pitch marks, and a pitch conversion speech signal generation section that generates a speech signal in which pitch cycle is converted to the target pitch cycle based on an amplitude signal of the target pitch cycle width of a section corresponding to the section of the amplitude signal and based on a phase signal of the target pitch cycle width.

Abstract: Features are disclosed for generating markers for elements or other portions of an audio presentation so that a speech processing system may determine which portion of the audio presentation a user utterance refers to. For example, an utterance may include a pronoun with no explicit antecedent. The marker may be used to associate the utterance with the corresponding content portion for processing. The markers can be provided to a client device with a text-to-speech (“TTS”) presentation. The markers may then be provided to a speech processing system along with a user utterance captured by the client device. The speech processing system, which may include automatic speech recognition (“ASR”) modules and/or natural language understanding (“NLU”) modules, can generate hints based on the marker. The hints can be provided to the ASR and/or NLU modules in order to aid in processing the meaning or intent of a user utterance.

Abstract: Implementations of systems, method and devices described herein enable enhancing the intelligibility of a target voice signal included in a noisy audible signal received by a hearing aid device or the like. In particular, in some implementations, systems, methods and devices are operable to generate a machine readable formant based codebook. In some implementations, the method includes determining whether or not a candidate codebook tuple includes a sufficient amount of new information to warrant either adding the candidate codebook tuple to the codebook or using at least a portion of the candidate codebook tuple to update an existing codebook tuple. Additionally and/or alternatively, in some implementations systems, methods and devices are operable to reconstruct a target voice signal by detecting formants in an audible signal, using the detected formants to select codebook tuples, and using the formant information in the selected codebook tuples to reconstruct the target voice signal.

Abstract: Embodiments of the present invention include a data storage device and a method for storing data in a hash table. The data storage device can include a first memory device, a second memory device, and a processing device. The first memory device is configured to store one or more data elements. The second memory device is configured to store one or more status bits at one or more respective table indices. In addition, each of the table indices is mapped to a corresponding table index in the first memory device. The processing device is configured to calculate one or more hash values based on the one or more data elements.

Abstract: An exemplary system includes a detection facility configured to detect an input sound level of an audio signal presented to an auditory prosthesis patient; and an adaptive gain control (AGC) facility configured to 1) determine whether the detected input sound level is in a quiet region, an intermediate region, or a loud region, and 2) apply a gain to the audio signal in accordance with an AGC gain function that specifies the gain to be substantially equal to or less than a first gain threshold if the detected input sound level is in the quiet region, substantially equal to or less than a second gain threshold if the detected input sound level is in the loud region, and greater than the first and second gain thresholds if the detected input sound level is in the intermediate region. Corresponding systems and methods are also disclosed.

Abstract: A speech analysis system and method for analyzing speech. The system includes: a voice recognition system for converting inputted speech to text; an analytics system for generating feedback information by analyzing the inputted speech and text; and a feedback system for outputting the feedback information.

Abstract: A methodology may be provided that automatically annotates web conference application sharing (e.g., sharing scenes and/or slides) based on voice and/or web conference data. In one specific example, a methodology may be provided that threads the annotations and assigns authorship to the correct resources.

Abstract: The technique is implemented by a vehicular device for performing dialogs with the driver. This device has: a communication portion for communicating with a server; an output portion for outputting speech information to the driver; an input portion for inputting information based on speech uttered by the driver; and a controller for controlling the communication portion, the output portion, and the input portion. When trigger information for starting a dialog process in the vehicular device or in the server is generated, the controller receives information indicative of the type of a first dialog forming a starting point of the dialog process and information indicative of the type of a second dialog forming an ending point of the dialog process from the server and carries out the dialog process based on the received information. The second dialog is different in type from the first dialog.

Abstract: One embodiment of the present invention is a method for increasing the speed of a computer in identifying occurrences of strings in a character stream that match a string pattern involving repetitions of characters of a particular character class. The method uses a parallel bit stream processing module of the computer, which processing module includes a processor equipped with parallel processing instructions, to form a plurality of parallel property bit streams Pj.

Abstract: A system, method and computer readable medium that enhances a speech database for speech synthesis is disclosed. The method may include labeling audio files in a primary speech database, identifying segments in the labeled audio files that have varying pronunciations based on language differences, identifying replacement segments in a secondary speech database, enhancing the primary speech database by substituting the identified secondary speech database segments for the corresponding identified segments in the primary speech database, and storing the enhanced primary speech database for use in speech synthesis.

Abstract: Applications can be localized by localization experts to allow them to be used by a broader customer base. The localization can be done given interface context to produce more applicable results. A localization expert may determine that certain content within the interface context should be localized, however is not localizable by the expert. The localization expert may indicate that the content is localizable and the developer may receive the indication and provide the localization expert with updated interface context allowing localization of the previously un-localizable content.

Abstract: A group of users may be presented with text and a synthesized speech recording of the text. The users can listen to the synthesized speech recording and submit feedback regarding errors or other issues with the synthesized speech. A system of one or more computing devices can analyze the feedback, modify the voice or language rules, and recursively test the modifications. The modifications may be determined through the use of machine learning algorithms or other automated processes.

Abstract: Features are disclosed for generating text-to-speech (TTS) audio programs from textual content received from multiple sources. A TTS system may assemble an audio program from several individual audio presentations of user-selected network-accessible content. Users may configure the TTS system to retrieve personal content as well as publically accessible content. The audio program may include segues, introductions, summaries, and the like. Voices may be selected for individual content items based on user selections or on characteristics of the content or content source.

Abstract: A semiconductor integrated circuit device for speech recognition includes a scenario setting unit that receives a command designating scenario flow information and selects prescribed speech reproduction data in a speech reproduction data storage and a prescribed conversion list, in accordance with the scenario flow information, a standard pattern extraction unit that extracts a standard pattern corresponding to at least part of individual words or sentences included in the prescribed conversion list from a speech recognition database, a speech signal synthesizer that synthesizes an output speech signal, a signal processor that generates a feature pattern representing the distribution state of the frequency component of an input speech signal, and a match detector that compares the feature pattern with the standard pattern and outputs a speech recognition result.

Abstract: A method and system is disclosed for non-parametric speech conversion. A text-to-speech (TTS) synthesis system may include hidden Markov model (HMM) HMM based speech modeling for both synthesizing output speech. A converted HMM may be initially set to a source HMM trained with a voice of a source speaker. A parametric representation of speech may be extract from speech of a target speaker to generate a set of target-speaker vectors. A matching procedure, carried out under a transform that compensates for speaker differences, may be used to match each HMM state of the source HMM to a target-speaker vector. The HMM states of the converted HMM may be replaced with the matched target-speaker vectors. Transforms may be applied to further adapt the converted HMM to the voice of target speaker. The converted HMM may be used to synthesize speech with voice characteristics of the target speaker.

Abstract: A method and system for is disclosed for cross-lingual voice conversion. A speech-to-speech system may include hidden Markov model (HMM) HMM based speech modeling for both recognizing input speech and synthesizing output speech. A cross-lingual HMM may be initially set to an output HMM trained with a voice of an output speaker in an output language. An auxiliary HMM may be trained with a voice of an auxiliary speaker in an input language. A matching procedure, carried out under a transform that compensates for speaker differences, may be used to match each HMM state of the output HMM to a HMM state of the auxiliary HMM. The HMM states of the cross-lingual HMM may be replaced with the matched states. Transforms may be applied to adapt the cross-lingual HMM to the voices of the auxiliary speaker and of an input speaker. The cross-lingual HMM may be used for speech synthesis.

Abstract: System, apparatus and method for determining semantic information from audio, where incoming audio is sampled and processed to extract audio features, including temporal, spectral, harmonic and rhythmic features. The extracted audio features are compared to stored audio templates that include ranges and/or values for certain features and are tagged for specific ranges and/or values. The semantic information may be associated with audio codes to determine changing characteristics of identified media during a time period.

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.