Abstract: Methods, systems and apparatus for editing audio clips. A computer-implemented method includes displaying in a user interface, a first audio clip including a first plurality of time instants and a second audio clip including a second plurality of time instants; displaying a first transition point identifier associated with the first audio clip to designate a portion from a beginning of the first audio clip to the first transition point identifier that is playable; displaying a second transition point identifier associated with the second audio clip to designate a portion from the second transition point identifier to an end of the second audio clip that is playable; and generating a combined audio clip comprising the portion from the beginning of the first audio clip to the first transition point identifier and the portion from the second transition point identifier to the end of the second audio clip.

Abstract: A dictation system that allows using trainable code phrases is provided. The dictation system operates by receiving audio and recognizing the audio as text. The text/audio may contain code phrases that are identified by a comparator that matches the text/audio and replaces the code phrase with a standard clause that is associated with the code phrase. The database or memory containing the code phrases is loaded with matched standard clauses that may be identified to provide a hierarchal system such that certain code phrases may have multiple meanings depending on the user.

Abstract: A method and a system for identity authentication are presented. In one example embodiment, audio data (e.g. a sound wave) may be received from a user. The audio data may be used to establish an identity of a first entity to the user. The audio data may be stored at a storage location; and be presented to the user to establish the identity of the first entity when the first entity participates in an electronic communication with the user. In another example embodiment, a server (e.g., a web client or client application server) may present a plurality of audio data instances to a user; receive the user selection of selected audio data from the plurality of audio data instances; responsive to the user selection, the server may communicate, via a network, the selected audio data to another server. The selected audio data may be used as an identity authentication.

Abstract: A mobile replacement-dialogue recording system enables the creation of replacement-dialogue items by mobile users not at a media recording studio. Studio-users prepare guide media video, audio and text data which are made available to mobile users through a media server. A mobile user's mobile replacement-dialogue recording device obtains guide media and allows the user to view the guide media in rehearsal mode. The mobile replacement-dialogue recording device then records the mobile user's dialogue performance while presenting the mobile user with synchronized guide media. The mobile user can review, delete, and rerecord the resulting potential replacement dialogue, as well as create feedback media characterizing the replacement dialogue. Selected replacement dialogue items can be transmitted to the media server. A studio-module can then obtain the selected replacement dialogue items and feedback media from the media server so that they may be used in media-replacement.

Abstract: Apparatus having corresponding methods and computer-readable media comprise: a muter configured to pass or block an audio signal; a voice activity detector configured to detect voice activity in the audio signal; and a vibrator configured to produce a mechanical vibration responsive to the contemporaneous occurrence of i) the voice activity detector detecting the voice activity in the audio signal; and ii) the muter being configured to block the audio signal.

Abstract: Systems and associated methods for editing telecom web applications through a voice interface are described. Systems and methods provide for editing telecom web applications over a connection, as for example accessed via a standard phone, using speech and/or DTMF inputs. The voice based editing includes exposing an editing interface to a user for a telecom web application that is editable, dynamically generating a voice-based interface for a given user for accomplishing editing tasks, and modifying the telecom web application to reflect the editing commands entered by the user.

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: Methods, graphical user interfaces, computer apparatus and computer readable medium for producing media content are disclosed. For example, a user of a computing device can utilize the methods, graphical user interfaces, computer apparatus, and computer readable medium to edit the media content. In one embodiment, the media content pertains to media tracks, such as audio or video tracks. The media content can be a plurality of individual media tracks that can be segmented and the resulting segments from different media tracks can be combined into a composite media track.

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: An encoder includes a precoder for encoding an input information object according to a preset encoding scheme and storing the encoded information object in a precoder buffer, a sample number/address generation unit for generating a sample number of each sample and an address, which corresponds to each bit of each sample and the address of the precoder buffer, a multiplexer for selecting a bit of the precoder buffer corresponding to the address generated by the sample number/address generation module, a sampling buffer for storing a bit of each sample output from the multiplexer, a control packet generation module for generating a control packet including information on the sample number generated by the sample number/address generation module, a packet assembling unit for assembling the sample stored in the sampling buffer with the control packet generated by the control data generation module, and a modulation module for modulating the packet output from the packet assembling unit into a sound signal accordi

Abstract: A method for message translation and a Messaging Translation Service Application Server (MTS AS) are provided for translating messages exchanged with, and among, social network services alike Facebook™ and Tweeter™. According to the invention, a message written in a first language by a user is received by a first social media network, which further obtains from other social media network(s) information related to a language used by therein. Then, the first social media network requests translation of the message from the first language into the language used by the other social network systems, and further sends the translated message to the other social network systems.

Abstract: A system and method for locating a preferable playback start location after a winding or rewinding action in an audio playing device. In response to an adjustment of the playing location for audio content to a desired playing position, the system determines whether at least one non-speech or silent period of at least a predetermined duration exists within the vicinity of the desired playing position. If at least one such non-speech or silent period exists within the vicinity of the desired playing position, the system adjusts the playing position to fall within one of the at least one non-speech period or silent period.

Abstract: A computer-implemented system and method for identifying and masking special information within recorded speech is provided. A field for entry of special information is identified. Movement of a pointer device along a trajectory towards the field is also identified. A correlation of the pointer device movement and entry of the special information is determined based on a location of the trajectory in relation to the field. A threshold is applied to the correlation. The special information is received as verbal speech. A recording of the special information is rendered unintelligible when the threshold is satisfied.

Abstract: Systems, methods and computer program products are provided for translating a natural language into music. Through systematic parsing, music compositions can be created. These compositions can be created by one or more persons who do not speak the same natural language.

Abstract: A recording and/or reproducing apparatus includes a microphone, a semiconductor memory, an operating section and a controller. An output signal from the microphone is written in the semiconductor memory and the written signals are read out from the semiconductor memory. The operating section performs input processing for writing a digital signal outputted by an analog/digital converter, reading out the digital signal stored in the semiconductor memory and for erasing the digital signal stored in the semiconductor memory. The control section controls the writing of the microphone output signal in the semiconductor memory based on an input from the operating section and the readout of the digital signal stored in the semiconductor memory.

Abstract: Indexing digitized speech with words represented in the digitized speech, with a multimodal digital audio editor operating on a multimodal device supporting modes of user interaction, the modes of user interaction including a voice mode and one or more non-voice modes, the multimodal digital audio editor operatively coupled to an ASR engine, including providing by the multimodal digital audio editor to the ASR engine digitized speech for recognition; receiving in the multimodal digital audio editor from the ASR engine recognized user speech including a recognized word, also including information indicating where, in the digitized speech, representation of the recognized word begins; and inserting by the multimodal digital audio editor the recognized word, in association with the information indicating where, in the digitized speech, representation of the recognized word begins, into a speech recognition grammar, the speech recognition grammar voice enabling user interface commands of the multimodal digital au

Abstract: Subject matter described herein relates to providing to a mobile device a version of content (e.g., music, video, text message, live call, etc.) that is consistent with a user's filter setting. That is, a user is allowed to specify content elements (e.g., words or images) that are proscribed from being presented on the mobile device, and the user's preferences are stored by a mobile telecommunications network. When the network receives content to be provided to the mobile device, the network edits the content in real time to prevent proscribed elements from being presented on the mobile device.

Abstract: Automatic correcting of user's speech impairment in speech may include obtaining the audio signal of a given user's speech, and analyzing the obtained audio signal to identify artifacts caused by the user's impairment. The obtained audio signal may be modified by eliminating the identified artifacts from it. The modified audio signal may be provided, e.g., to be played or broadcast or transmitted.

Abstract: Techniques have been developed to facilitate the capture performances on handheld or other portable computing devices and, in some cases, the pitch-correction and mixing of such vocal performances with backing tracks for audible rendering on such devices. Captivating visual animations and/or facilities for listener comment and ranking are provided in association with an audible rendering of a performance, e.g., a vocal performance captured and pitch-corrected at another similarly configured mobile device and mixed with backing instrumentals and/or vocals. Geocoding of captured vocal performances and/or listener feedback may facilitate animations or display artifacts in ways that are suggestive of a performance or endorsement emanating from a particular geographic locale on a user manipulable globe. In this way, implementations of the described functionality can transform otherwise mundane mobile devices into social instruments that foster a unique sense of global connectivity and community.

Abstract: An apparatus and a method for improving communication sound quality in a mobile terminal in order to remove a neighboring noise that occurs together with a user's voice signal in a mobile terminal by discriminating signals occurring at different distances using two microphones and removing a noise. The mobile terminal preferably includes a first microphone, a second microphone, and a voice processor. The first microphone receives a voice signal occurring at a closer distance from the mobile terminal and a voice signal occurring at a longer distance from the mobile terminal. The second microphone receives only a voice signal occurring at the long distance. The voice processor discriminates between the signal occurring at the long distance and the signal occurring at the close distance by receiving voice signals received via the first microphone and the second microphone at different phases.

Abstract: Disclosed are systems, methods, and computer readable media having programs for classifying sports video. In one embodiment, a method includes: extracting, from an audio stream of a video clip, a plurality of key audio components contained therein; and classifying, using at least one of the plurality of key audio components, a sport type contained in the video clip. In one embodiment, a computer readable medium having a computer program for classifying ports video includes: logic configured to extract a plurality of key audio components from a video clip; and logic configured to classify a sport type corresponding to the video clip.

Abstract: A method and apparatus for analyzing and segmenting a vocal interaction captured in a test audio source, the test audio source captured within an environment. The method and apparatus first use text and acoustic features extracted from the interaction with tagging information, for constructing a model. Then, at production time, text and acoustic features are extracted from the interactions, and by applying the model, tagging information is retrieved for the interaction, enabling analysis, flow visualization or further processing of the interaction.

Abstract: A computer-implemented technique for transcribing audio data includes generating, along a vertical axis on a display of a client device, an image representing audio content. The technique further includes receiving, from a user of the client device, a selection of a portion of the image; and generating, via an audio module of the client device, an audio output corresponding to the selected portion of the image. The technique further includes receiving, from the user, a selection indicating a position along the vertical axis on the display to enter a text portion representing the audio output, wherein the position is aligned to the selected portion of the image. The technique further includes receiving, from the user, the text portion representing the audio output; and displaying, on the display, the text portion at the position, wherein the text portion extends along a horizontal axis on the display.

Abstract: Systems and associated methods for editing telecom web applications through a voice interface are described. Systems and methods provide for editing telecom web applications over a connection, as for example accessed via a standard phone, using speech and/or DTMF inputs. The voice based editing includes exposing an editing interface to a user for a telecom web application that is editable, dynamically generating a voice-based interface for a given user for accomplishing editing tasks, and modifying the telecom web application to reflect the editing commands entered by the user.

Abstract: A speech masking apparatus includes a microphone and a speaker. The microphone can detect a human voice. The speaker can output a masking language which can include phonemes resembling human speech. At least one component of the masking language can have a pitch, a volume, a theme, and/or a phonetic content substantially matching a pitch, a volume, a theme, and/or a phonetic content of the voice.

Abstract: The disclosure includes a system and method for generating audio snippets from a subset of audio tracks. In some embodiments an audio snippet is an audio summary of a group or collection of songs.

Abstract: The present invention is a system and method for generating a personal voice font including, monitoring voice segments automatically from phone conversations of a user by a voice learning processor to generate a personalized voice font and delivering the personalized voice font (PVF) to the a server.

Abstract: An apparatus includes a plurality of applications and an integrator having a voice recognition module configured to identify at least one voice command from a user. The integrator is configured to integrate information from a remote source into at least one of the plurality of applications based on the identified voice command. A method includes analyzing speech from a first user of a first mobile device having a plurality of applications, identifying a voice command based on the analyzed speech using a voice recognition module, and incorporating information from the remote source into at least one of a plurality of applications based on the identified voice command.

Abstract: A computer implemented method, system, and/or computer program product generates an audio cohort. Audio data from a set of audio sensors is received by an audio analysis engine. The audio data, which is associated with a plurality of objects, comprises a set of audio patterns. The audio data is processed to identify audio attributes associated with the plurality of objects to form digital audio data. This digital audio data comprises metadata that describes the audio attributes of the set of objects. A set of audio cohorts is generated using the audio attributes associated with the digital audio data and cohort criteria, where each audio cohort in the set of audio cohorts is a cohort of accompanied customers in a store, and where processing the audio data identifies a type of zoological creature that is accompanying each of the accompanied customers.

Abstract: An automatic marking method for Karaoke vocal accompaniment is provided. In the method, pitch, beat position and volume of a singer are compared with the original pitch, beat position and volume of the theme of a song to generate a score of pitch, a score of beat and a score of emotion respectively, so as to obtain a weighted total score in a weighted marking method. By using the method, the pitch, beat position and volume error of each section of the song sung by the singer can be exactly worked out, and a pitch curve and a volume curve can be displayed, so that the singer can learn which part is sung incorrectly and which part needs to be enhanced. The present invention also has the advantages of dual effects of teaching and entertainment, high practicability and technical advancement.

Abstract: Sounds are inserted into audio content according to a pattern. A library stores humanly perceptible voice sounds. Pattern control information is received that is associated with a device recording the audio content. A pattern is retrieved and washing machine sounds are inserted into the audio content according to the pattern. The humanly perceptible voice sounds are inserted into the audio content according to the pattern to generate a signed audio recording.

Abstract: Automatic correcting of user's speech impairment in speech may include obtaining the audio signal of a given user's speech, and analyzing the obtained audio signal to identify artifacts caused by the user's impairment. The obtained audio signal may be modified by eliminating the identified artifacts from it. The modified audio signal may be provided, e.g., to be played or broadcast or transmitted.

Abstract: A system for controlling digital effects in live performances with vocal improvisation is described. The system features a controller that utilizes several switches attached to clothing that is worn by an artist during a live performance. The switches activate a digital vocal processor unit that provides a dual mode, multi-channel phrase looping capability wherein individual channels can be selected for recording and replay during the performance. This combination of features allows a sequence of digital audio and video effects to be controlled by the artist during a performance while maintaining the freedom of movement desired to enhance the performance.

Abstract: A method, apparatus, and computer-readable medium for editing a data stream based on a corpus are provided. The data stream includes stream words. A sequence includes a predetermined number of sequential words of the stream words. The method, apparatus, and computer-readable medium determine whether the sequence exists in the corpus at least at a predetermined minimum frequency. When the sequence exists in the corpus at least at the predetermined minimum frequency, the sequence is edited in the data stream.

Abstract: Provided is a method of canceling a vocal signal, wherein the method includes obtaining a difference signal between two audio signals; and smoothing the frequency of the difference signal. Also provided is a device for canceling a vocal signal, the device including a subtracter which obtains a difference signal between two audio signals; and a frequency smoothing unit which smoothes a frequency of the difference signal.

Abstract: Disclosed is a recording and reproducing apparatus comprising: an apparatus main body; and a remote controller to perform remote control of the apparatus main body, wherein the remote controller comprises: a key operating section to receive a key operation by a user; a sound information inputting section to input sound information; and a transmitting section to transmit sound data based on the sound information to the apparatus main body, and the apparatus main body comprises: a recording section to record input content data on a recording medium; a reproducing section to reproduce the content data; a receiving section to receive the sound data; a sound information recording section to record the sound data so as to be associated with a piece of the content data; and a sound information outputting section to reproduce the sound data to output the reproduced sound data.

Abstract: Disclosed are Multipurpose Media Players that enable users to create transcriptions, closed captions, and/or logs of digitized recordings, that enable the presentation of transcripts, closed captions, logs, and digitized recordings in a correlated manner to users, that enable users to compose one or more scenes of a production, and that enable users to compose storyboards for a production. The multipurpose media players can be embodied within Internet browser environments; thereby providing high availability of the multipurpose players across software platforms, networks, and physical locations.

Abstract: A method and apparatus for transmitting an audio signal over a communication channel comprising encoding the audio signal with an encoder 204 using a first sampling rate, filtering the audio signal using a first cut off frequency, the first cut off frequency being chosen in dependence upon the first sampling rate, and transmitting the encoded and filtered audio signal over the communication channel. The presence of a condition in which the sampling rate of the encoder 204 is to be switched to a second sampling rate at a switching time is determined and if the condition has been determined to be present, the cut off frequency used in the filtering step is gradually changed from the first cut off frequency to a second cut off frequency, the second cut off frequency being chosen in dependence upon the second sampling rate, such that the audio bandwidth of the transmitted signal changes gradually when the sampling rate is switched to the second sampling rate.

Abstract: An apparatus and method for generating an avatar based video message are provided. The apparatus and method are capable of generating an avatar based video message based on speech of a user. The avatar based video message apparatus and method displays information that corresponds to input user speech. The avatar based video message apparatus and method edits the input user speech according to a user input signal with reference to the displayed information, generates avatar animation according to the edited speech, and generates an avatar based video message based on the edited speech and the avatar animation.

Abstract: A method for synchronizing sound data and text data, said text data being obtained by manual transcription of said sound data during playback of the latter. The proposed method comprises the steps of repeatedly querying said sound data and said text data to obtain a current time position corresponding to a currently played sound datum and a currently transcribed text datum, respectively, correcting said current time position by applying a time correction value in accordance with a transcription delay, and generating at least one association datum indicative of a synchronization association between said corrected time position and said currently transcribed text datum. Thus, the proposed method achieves cost-effective synchronization of sound and text in connection with the manual transcription of sound data.

Abstract: The present invention provides for a method and apparatus for segmenting a multi-media program based upon audio events. In an embodiment a method of classifying an audio stream is provided. This method includes receiving an audio stream. Sampling the audio stream at a predetermined rate and then combining a predetermined number of samples into a clip. A plurality of features are then determined for the clip and are analyzed using a linear approximation algorithm. The clip is then characterized based upon the results of the analysis conducted with the linear approximation algorithm.

Abstract: A content processing service may analyze an item of original content and identify several objects, attributes of those objects, and relationships between those objects present in the item of original content. The content processing service may also analyze a source graph, such as a social graph or supplemental graph, and identify several objects, attributes of those objects, and relationships between objects present in the source graph. The content processing service may customize the item of original content by selecting an original object and selecting a source graph object. One or more of the attributes or relationships of the selected original object in the item of original content may be replaced by one or more of the attributes or relationships of the selected source graph object. To customize items of audio content, audio content associated with the source graph object may replace audio content associated with the target graph object.

Abstract: Techniques are described to allow a user of a signal editing tool to “stretch” or “shrink” a selected portion of a recorded signal to change the length of the selected portion of the signal relative to a particular domain, without stretching or shrinking other parts of the signal. In the context of audio signals, techniques are provided to allow a user to “time stretch” an audio signal file to change the duration of the stretched portion of the audio. The user may select how the change affects the total duration of the audio signal. Options are provided for “shifting” the non-selected portion of the signal, or for not shifting the non-selected portion of the signal. When the non-selected portion is not shifted, the signal editing tool automatically generates audio for the gap (for shrinking operations), and automatically deletes audio that overlaps with the stretched portion (for stretching operations).

Abstract: Sounds are inserted into audio content according to a pattern. A library stores humanly perceptible voice sounds. Pattern control information is received that is associated with a device recording the audio content. A pattern is retrieved and washing machine sounds are inserted into the audio content according to the pattern. The humanly perceptible voice sounds are inserted into the audio content according to the pattern to generate a signed audio recording.

Abstract: An audible post-it system includes a post-it note printed with an index and an optical reading and recording device having an optical module, a switch, a storage device, an audio recording device, an audio playing device and a processor. The optical reading and recording device reads an image of the index. When the optical reading and recording device is at a recoding state, the processor receives the image of the index and obtains the index, then receives a digital audio outputted by the audio recording device to match the index with the digital audio, and stores the digital audio based on the index. When the optical reading and recording device is at a playing state, the processor receives the image of an index and retrieves the index, then reads a digital audio based on the index, and sends the digital audio to the audio playing device for playing.

Abstract: Methods and systems for sculpting synthesized speech using a graphic user interface are disclosed. An operator enters a stream of text that is used to produce a stream of target phonetic-units. The stream of target phonetic-units is then submitted to a unit-selection process to produce a stream of selected phonetic-units, each selected phonetic-unit derived from a database of sample phonetic-units. After the stream of sample phonetic-units is selected, an operator can remove various selected phonetic-units from the stream of selected phonetic-units, prune the sample phonetic-database and edit various cost functions using the graphic user interface. The edited speech information can then be submitted to the unit-selection process to produce a second stream of selected phonetic-units.

Abstract: A system and method of creating a customized multi-media message to a recipient is disclosed. The multi-media message is created by a sender and contains an animated entity that delivers an audible message. The sender chooses the animated entity from a plurality of animated entities. The system receives a text message from the sender and receives a sender audio message associated with the text message. The sender audio message is associated with the chosen animated entity to create the multi-media message. The multi-media message is delivered by the animated entity using as the voice the sender audio message wherein the mouth movements of the animated entity conform to the sender audio message.

Abstract: An input signal is converted to a feature-space representation. The feature-space representation is projected onto a discriminant subspace using a linear discriminant analysis transform to enhance the separation of feature clusters. Dynamic programming is used to find global changes to derive optimal cluster boundaries. The cluster boundaries are used to identify the segments of the audio signal.

Abstract: A biochemical analyzer having a microprocessing apparatus with expandable voice capacity is characterized in that a driving module is installed in a data processor and a voice carrier is replaceable. Thereby, increase or decrease of voice files can be easily done by replacing the current voice carrier with an alternative voice carrier storing desired voice files, without the need of replacing the driving module together with the voice carrier, thereby saving costs and reducing processing procedures.

Abstract: This specification describes technologies relating to recognition of text in various media. In general, one aspect of the subject matter described in this specification can be embodied in methods that include receiving an input signal including data representing one or more words and passing the input signal to a text recognition system that generates a recognized text string based on the input signal. The methods may further include receiving the recognized text string from the text recognition system. The methods may further include presenting the recognized text string to a user and receiving a corrected text string based on input from the user. The methods may further include checking if an edit distance between the corrected text string and the recognized text string is below a threshold. If the edit distance is below the threshold, the corrected text string may be passed to the text recognition system for training purposes.