Abstract: A system and method for providing a service of playing an accompaniment/musical performance is disclosed. In order to embody the system and method for providing the service of playing the accompaniment/musical performance, virtual orchestra system (VOS) files, which is converted from digital music files, e.g., musical instrument digital interface (MIDI) files and includes play order notes and sound data for each musical instrument capable of being played, are used. A server provides the VOS files through a network, e.g., a local area network (LAN), an Intranet, a value added network (VAN), an Internet or a public switched telephone network. A music is selected by a user through at least a client terminal. The play order note for each musical instrument is provided and the sound data for each musical instrument is played based on the play order note, thereby playing in solo or in concert. (At this time, sound for the others musical instrument is silent or used as a background music.

Abstract: The invention includes a method to determine time location of at least one audio segment in an original audio file comprising: (a) receiving the original audio file; (b) transcribing a current audio segment from the original audio file using speech recognition software; (c) extracting a transcribed element and a binary audio stream corresponding to the transcribed element from the speech recognition software; (d) saving an association between the transcribed element and the corresponding binary audio stream; (e) repeating (b) through (d) for each audio segment in the original audio file; (f) for each transcribed element, searching for the associated binary audio stream in the original audio file, while tracking an end time location of that search within the original audio file; and (g) inserting the end time location for each binary audio stream into the transcribed element-corresponding binary audio stream association.

Abstract: An audio reproducing apparatus comprises: audio decoding means for decoding an input audio signal frame by frame; data expanding/compressing means for subjecting data in a decoded frame to time-scale modification process; a frame sequence table which contains a sequence determined according to a given speed rate in which respective frames are expanded/compressed; frame counting means for counting the number of frames of the input audio signal; and data expansion/compression control means for instructing the dalta expanding/compressing means to subject the frame to one of time-scale compression process, time-scale expansion process, and process without time-scale modification process, with reference to the frame sequence table based on a count value output from the frame counting means, the data expanding/compressing means subjecting the audio signal to time-scale modification process in accordance with an instruction signal from the data expansion/compression control means.

Abstract: The invention provides an information recording medium, such as an optical disk, having a large capacity and being capable of performing read/write operations at high speeds. The recording medium includes an audio stream prepared for after-recording data, and a audio attribute information having a bit rate information to the recorded audio stream as a management information. A recorder according to the invention has a check unit for checking, in advance, the possibility of after-recording operation of the recorder to the audio stream to be after-recorded with reference to the bit rate information of the audio attribute information.

Abstract: An audio/video reproducing apparatus is connectable to a communications network for selectively reproducing items of audio/video material from a recording medium in response to a request received via the communications network. The audio/video reproducing apparatus may comprise a control processor operable in use to receive data representing the request for the audio/video material item via the communications network. A reproducing processor is operable in response to signals identifying the audio/video material items from the control processor to reproduce the audio/video material items. The data identifying the audio/video material items includes meta data indicative of the audio/video material items. The meta data may be one of UMID, tape ID and time codes, and a Unique Material Identifier the material items.

Abstract: A multi-source input and playback utility that accepts inputs from various sources, transcribes the inputs as text, and plays aloud user-selected portions of the text is disclosed. The user may select a portion of the text and request audio playback thereof. The utility examines each transcribed word in the selected text. If stored audio data is associated with a given word, that audio data is retrieved and played. If no audio data is associated, then a textto-speech entry or series of entries is retrieved and played instead.

Abstract: The method and a system is for locating and recording time-limited signal sequences in media channels that may contain undesirable signal components, e.g., recording music in radio transmissions. The signals are continuously buffered in a memory. The user identifies a desired source material. Out of this desired source material a section may be taken as a search key. The device may also select search keys automatically. If a second instance of the search key is detected, signal sequences that in time are connected to the search keys are compared. The signal sequences that by comparison are substantially identical are identified as belonging to the same, wanted, source material. The next step is an iteration of the above procedure results in a longer and higher quality segment of source material than the initial common segment.

Abstract: At an audio source, pause information is added to audio data, the combination of which is subsequently packetized. The resulting packets are transmitted to an audio destination via a network in which different packets may be subjected to varying levels of delay. At the audio destination, the pause information may be used to insert pauses at appropriate times to accommodate the occurrence of delays in packet delivery. In one embodiment, pauses are inserted based on a hierarchy of pause types. During pauses, audio filler information may be injected. In this manner, the effects of variable network delays upon reconstructed audio may be mitigated.

Abstract: A sound device includes a silent state detecting unit for detecting a silent state in a sound signal supplied by a personal computer; and a sound production preventing unit for preventing a sound from being produced from the sound signal supplied by the personal computer when the silent state is detected by the silent state detecting unit. By halting the production of a sound from the sound signal supplied from the personal computer when the silent state is detected, production of noise in a silent state is prevented so that the quality of sound in the expansion station is improved.

Abstract: The present invention relates to a method and apparatus for directing a pre-recorded audio file to a speech recognition program that does not normally accept such files, such as IBM Corporation's Via Voice™ speech recognition program. The method includes: (a) launching the speech recognition program to accept speech as if the speech recognition program were receiving live audio from a microphone; (b) finding a mixer utility associated with the sound card; (c) opening the mixer utility, the mixer utility having settings that determine an input source and an output path; (d) changing the settings of the mixer utility to specify a line-in input source and a wave-out output path; (e) activating a microphone input of the speech recognition software; and (f) initiating a media player associated with the computer to play the pre-recorded audio file into the line-in input source. The method may additionally save and restore the original configuration settings of the mixer utility.

Abstract: A computer-animated image of a video model is stored for synchronized outputting with an audio wave. When receiving the audio wave representation, the model is dynamically varied under control of the audio wave, and outputted together with the audio wave. In particular, an image parameter is associated to the model. By measuring an actual audio wave amplitude, and mapping the amplitude in a multivalued or analog manner on the image parameter the outputting is synchronized.

Abstract: A background sound sending side multiplexes and sends, in a multiplexer, uttered encoded speech data generated in a speech sending section and encoded background sound data outputted from a background sound storing section. Simultaneously, a background sound reproducing section, reproduces encoded background sound data and reproduced background sound signal is superposed on received speech in a receiving section and outputted from a receiver. A background sound receiving side demultiplexes, in a demultiplexer, received multiplexed data into received encoded speech data and encoded background sound data which are decoded in the receiving section and the background sound reproducing section respectively, and in the receiving section, a sound in which received speech and background sound are superposed is outputted from a receiver.

Abstract: Audio associated with a video program, such as an audio track or live or recorded commentary, may be analyzed to recognize or detect one or more predetermined sound patterns, such as words or sound effects. The recognized or detected sound patterns may be used to enhance video processing, by controlling video capture and/or delivery during editing, or to facilitate selection of clips or splice points during editing.

Abstract: An electronic processing device for producing digitally processed audio-signal effects is provided. The electronic processing device comprises, an audio-signal input circuitry for receiving an audio input signal from a peripheral audio device, an audio-signal output circuitry for outputting the received audio-signal, the signal comprising a throughput signal after signal processing, a digital signal processor for applying audio-signal effects to the throughput audio-signal, one or more memory slots for receiving one or more modular memory components and an input control mechanism for controlling parameters of the throughput audio-signal. The one or more modular memory components are used as storage for externally sourced audio-signal effects such that when the one or more memory components are plugged into the electronic processing device, the processing device may utilize the effects applications stored on the one or more memory components in the processing of the throughput audio-signal.

Abstract: A system for associating a prerecorded audio snippet with a photograph, includes: an audio data base containing a plurality of audio snippets, each snippet having a corresponding identification code. A scene identification display includes an identification code associated with the scene. A camera having a sensor for sensing the identification code on the display includes a memory for storing the identification code in association with a photograph of the scene taken by the camera. The audio snippet corresponding to the identification code is retrieved from the audio data base and reproduced in conjunction with the display of the photograph.

Abstract: An audio dubbing system which is composed of an MD recorder 5 capable of recording digital data obtained by converting audio signals of respective tracks from one or plural CD(s) onto MD, and a personal computer 1 for causing the MD recorder 5 to record the digital audio signals, and in which the personal computer 1 is connected to a CD-ROM drive 12 for reading the audio signals from the CD(s) as digital data and to a hard disk drive 13 for storing the digital data read by the CD-ROM drive 12.

Abstract: In a speech communications network, continuous play of audio packets is achieved using a jitter buffer in a receiver. Audio packets are stored in the jitter buffer before decoding the audio packets into an audible output. When the level of stored audio packets approaches the full capacity of the jitter buffer, the rate at which the audio packets are played out of the jitter buffer is increased signaling a compression operation in the decoder. When the level of stored audio packets approaches an empty level of the jitter buffer, the rate which the audio packets are played out of the jitter buffer is reduced signaling an expansion operation in the decoder. Audio packets are not modified when the level of stored audio packets is within a predetermined range. A speed controller is provided to instruct the decoder to decode the audio packets according to either a compressed, expanded or normal audio packet status.

Abstract: A method transforms non-speech input signals into the temporal, spectral and redundancy patterns resembling that of human speech.

Abstract: Temporarily storing audio data to be reproduced in the block form, temporarily storing synthesized audio data in the block form, generating and supplying a reference signal, and calculating the first address of the block of audio data, the editing system of the present invention enables identifying the buffering position of the recording signal and then matching the position to the reproduced signal, resulting in quick editing.

Abstract: A system for generating scripts having verbal content. The system includes a computer having a user input receiver operative to receive a user's definition of a script for at least one computer-controllable animated physical figure. The script includes a plurality of interconnected script elements each representing an action performable by the computer-controllable animated figure. The script comprises at least one verbal script element representing a verbal action performable by the computer-controllable animated figure. A graphics interface is operative to generate a pictorial image of the script as the script is generated by the user. The graphics interface including a drag and drop facility and a flowchart generating facility.

Abstract: The invention enables the inclusion of voice and remaining audio information at different parts of the audio production process. In particular, the invention embodies special techniques for VRA-capable digital mastering and accommodation of VRA by those classes of audio compression formats that sustain less losses of audio data as compared to any codecs that sustain comparable net losses equal or greater than the AC3 compression format. The invention facilitates an end-listener's voice-to-remaining audio (VRA) adjustment upon the playback of digital audio media formats by focusing on new configurations of multiple parts of the entire digital audio system, thereby enabling a new technique intended to benefit audio end-users (end-listeners) who wish to control the ratio of the primary vocal/dialog content of an audio program relative to the remaining portion of the audio content in that program.

Abstract: An integrated receiver mixer system is disclosed wherein the plurality of wireless receivers is remotely controlled, and retained in synchronism, via reference and control signals outputted by the system mixer. Further, pairs of receivers are connected to each other, and to the mixer, in a manner which minimizes the requisite cabling therebetween.

Abstract: A method for editing audio data includes the steps of creating a header portion containing at least information for indicating a start of an audio unit to be decoded and having composite elements whose values are equal to those of the audio data to which dummy data is to be added, and creating the audio data composed of the dummy data to be ignored during a decoding time. The system for editing audio data is also provided for executing the editing method.

Abstract: Audio associated with a video program, such as an audio track or live or recorded commentary, may be analyzed to recognize or detect one or more predetermined sound patterns, such as words or sound effects. The recognized or detected sound patterns may be used to enhance video processing, by controlling video capture and/or delivery during editing, or to facilitate selection of clips or splice points during editing.

Abstract: A sound effects affixing device which enables sound effects and background music to be affixed in relation to inputted sentences automatically. A keyword extraction device is provided with a onomatopoeias extraction measure, a sound source extraction measure, and a subjective words extraction measure, which measures extract keyword of the onomatopoeias, the sound source names, or the subjective words within inputted sentences. A sound retrieval device selects sound effects and music by these keywords, thus selected sound effects and music are outputted by an output sound control device synchronized with synthesized speech.

Abstract: When a point on a tablet is designated by a marker, a recording indication signal is on, and recording of sound data is started. When the marker is released from the tablet, the recording indication signal is off, the recording of the sound data is finished, and coordinate data of the point and the sound data are linked. When information, such as characters, is inputted on the tablet using the marker, coordinate data based on the input and the sound data are linked through the coordinate data of the point. When the information such as characters is designated using a marker for playback, the sound data is played back.

Abstract: Audio associated with a video program, such as an audio track or live or recorded commentary, may be analyzed to recognize or detect one or more predetermined sound patterns, such as words or sound effects. The recognized or detected sound patterns may be used to enhance video processing, by controlling video capture and/or delivery during editing, or to facilitate selection of clips or splice points during editing.

Abstract: A computer-animated image of a video model is stored for synchronized outputting with an audio wave. When receiving the audio wave representation, the model is dynamically varied under control of the audio wave, and outputted together with the audio wave. In particular, an image parameter is associated to the model. By measuring an actual audio wave amplitude, and mapping the amplitude in a multivalued or analog manner on the image parameter the outputting is synchronized.

Abstract: An apparatus having a microphone, an analog to digital converting circuit, a semiconductor memory an, input device, and a controller. Wherein the analog to digital converting circuit converts an output signal from the microphone into a digital signal. The semiconductor memory stores the output signal from the analog to digital converting circuit, and the input device carries out input of a record start and a record end. The controller, according to the input from the input device, carries out control to start and stop writing into the semiconductor memory a digital signal from the analog to digital converting circuit. When the input device is operated and a predetermined time interval has passed, the controller starts writing the digital signal from the analog/digital conversion circuit into the semiconductor memory.

Abstract: An audio information reproducing method and apparatus are provided in which audio information read from an audio information source is at first stored in a buffer memory, the stored audio information is then read out at a preset speed magnification, and reproduced upon receiving a reproducing speed conversion treatment. The method comprises sending a request for reading audio information to the audio information source in accordance with an amount of information accumulated in the buffer memory; reading a predetermined amount of audio information from the buffer memory in accordance with the preset speed magnification, and reproducing the predetermined amount of audio information after performing a reproducing speed conversion treatment on the audio information.

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, for 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: An audio storing and reproducing apparatus is provided that has a semiconductor chip and a control unit. The semiconductor chip includes a semiconductor memory having a plurality of storage areas, a single storage specifying input terminal, a single reproduce specifying input terminal, and a storage medium controller that controls storing of the audio data to the storage areas and readout of the audio data therefrom in accordance with the signal input to the storage specifying and reproduce specifying input terminals. The control unit includes a single storage area specifying switch, a single storage specifying switch, a single reproduce specifying switch and a control circuit.

Abstract: Each of control signal transmitters (21-2n) is provided for each of a plurality of guide objects and transmits a control signal for discrimination of the corresponding guide object. A control signal detector (13) starts execution thereof by operation of a start button (12). A control signal detector receives the control signals, each of which is supplied from the control signal transmitters and detects one of the control signals, which has the maximum level for use of selection of one kind of the voice-data corresponding to the detected control signal transmitter. A controller (14) has received a message class such as a language class of Japanese or English for example from a class selector (19). The controller makes and sends out a selection signal with a detected control signal and a selected language class to a voice-data take-out circuit (15).

Abstract: A method and an apparatus for providing visual aid to a presenter involve converting the spoken words of a presenter into an electronic text format, electronically comparing the converted spoken words to electronically stored reference text to find text string matches, utilizing the text string matches between the converted spoken words and the reference text to determine a current location of the presentation with respect to the reference text, and delivering upcoming portions of the reference text to the presenter as needed to enable a continuous presentation by the presenter. A preferred presentation support system is incorporated into a portable personal computer that includes a speech recognition subsystem. The speech recognition subsystems allows a presentation to be tracked in real-time so that presentation support material can be automatically displayed to the presenter in synchronization with the in-progress presentation.

Abstract: A fast and economical method for speeding up an audio signal without changing pitch can be accomplished by eliminating unneeded information from an audio signal. First, the signal is divided into chunks (frames or subframes), on which a mathematical manipulation such as a Fourier transformation is performed to identify the amplitudes of the componenet sinusoids (sines and cosines). These absolute values of the sine and cosine amplitudes for each frequency are averaged together, and the highest value(s) represents the signature, or dominant frequency/frequencies. The dominant frequency/frequencies or signatures from one chunk are compared to those of the next, and when identical the latter unit is marked as redundant. The final step consists of discarding redundant chunks from the original data, thus providing a shortened signal for replay. The pitch will not change because the only modification to the original signal was the elimination of redundant data.

Abstract: Automated methods and apparatus for synchronizing audio and text data, e.g., in the form of electronic files, representing audio and text expressions of the same work or information are described. A statistical language model is generated from the text data. A speech recognition operation is then performed on the audio data using the generated language model and a speaker independent acoustic model. Silence is modeled as a word which can be recognized. The speech recognition operation produces a time indexed set of recognized words some of which may be silence. The recognized words are globally aligned with the words in the text data. Recognized periods of silence, which correspond to expected periods of silence, and are adjoined by one or more correctly recognized words are identified as points where the text and audio files should be synchronized, e.g., by the insertion of bi-directional pointers.

Abstract: The invention relates to a portable apparatus or an apparatus mounted on a vehicle, for reading sounds recorded on supports, either according to an analog process on magnetically recorded supports, which is to say on cassettes, or according to a digital process on compact discs, comprising at least one reading assembly for the recordation supports and a broadcast assembly of sounds comprising an amplifier and at least one restitution element of sounds such as a loudspeaker.

Abstract: A speech-rate converter slowing down input speech regularly monitors the data length of the input speech and the previously estimated extended output data length for the current rate scaling factor, computing new output data length estimates. The conversion rate is adaptively modified depending on the time lag between input and output speech so as to make input and output data lengths consistent without skipping any spoken input portions. Input signal power is monitored to discriminate speech and non-speech intervals, and the portions of input non-speech intervals exceeding a conversion-rate-dependent duration are deleted.

Abstract: Non-looped continuous sound made up of random sequencing of digital sound segments is generated by taking several short segments of an otherwise continuous sound and forming independent records of those short segments. The stored segments are re-assembled into a sound sequence of arbitrary length based on selecting the next sound segment according to some statistical algorithm. The selected algorithm may be simply a random or pseudo-random selection, or it may provide a probability weighting to emphasize some sound records over others, or some combination of factors also affected by external stimuli such as light, heat or operator input. Apparatus for generating random sequenced digital sound are disclosed. Another aspect of the invention is logical sequence sound in which the selection of sound segments proceeds according to a logical sequence which is programmable.

Abstract: A reading system includes a computer and a mass storage device and software including instructions for causing a computer to accept an image file generated from optically scanning an image of a document. The software converts the image file into a converted text file that includes text information, and positional information associating the text with the position of its representation in the image file. The software records the voice of an operator of the reading machine as a series of voice samples in synchronization with a highlighting indicia applied to a displayed representation of the document and stores the series of voice samples in a data structure that associates the voice samples with displayed representation. The reading machine plays back the stored, recorded voice samples corresponding to words in the document as displayed by the monitor while highlighting is applied to the words in the displayed document.

Abstract: A method for adding a spoken language for output generated by a messaging program including a voice messaging program and a voice messaging program running without re-compiling the messaging program includes providing the voice messaging program configured to generate an output message, providing the language server to receive the output message, to receive an ordered plurality of phrase references, to use phrase references from the ordered plurality of phrase references to identify a plurality of spoken phrases, and to output the plurality of spoken phrases, installing a set of language configuration data in a directory in the memory, the set of language configuration data configured to specify an ordered plurality of phrase references to the language server in response to the output message, installing a set of phrase files in a second directory in the memory, each phrase file in the set having an associated phrase reference and configured to store a unique spoken phrase, the set of language configuration d

Abstract: A system for non-linearly editing video and audio information, uses a device for recognizing speech in the audio information and for generating a character sequence, particularly an ASCII character sequence, to produce an edit decision list (EDL). The generated character sequence is displayed on the display screen of an indicator. With reference to marked parts of the character sequence displayed on the display screen of the indicator, editing data is derived for the EDL.

Abstract: The present invention is a method and apparatus for re-recording audio events. Scattered audio events on a first track are determined based on a linked list. The scattered audio events are merged into a combined audio event on a second track. The combined audio event is copied on the second track to the first track.

Abstract: A hand-held data acquisition device includes a display presenting at least one of an address book, a date book, a memo pad, a to-do list, a contact manager, an expense tracker, an e-mail client, and a project manager, at least one of which contains multiple data items. An input device is operatively connected to the device is suitable to receive voice data from the user. The data acquisition device stores the voice data and associates the voice data with at least one of the data items.

Abstract: A method for providing voice dynamics of human utterances converted to and represented by text within a data processing system. A plurality of predetermined parameters for recognition and representation of dynamics in human utterances are selected. An enhanced human speech recognition software program is created implementing the predetermined parameters on a data processing system. The enhanced software program includes an ability to monitor and record human voice dynamics and provide speech-to-text recognition. The dynamics in a human utterance is captured utilizing the enhanced human speech recognition software. The human utterance is converted into a textual representation utilizing the speech-to-text ability of the software. Finally, the dynamics are merged along with the textual representation of the human utterance to produce a marked-up text document on the data processing system.

Abstract: A method for selecting a range of an information signal comprises the steps of detecting the area in which the information signal is specified among a plurality of range-specifying areas displayed on a display device and selecting the information signal range in a unit of the information signal determined in accordance with the range-specifying area in which the specification is executed. The unit of information signal to be selected is different according to which area is selected, so that at least two units of information signal are available for selection.

Abstract: A method for playback of speech in an audio recording. The method comprises performing full word-level recognition of the speech including recognition of silent pauses and filled pauses, suppressing playback of the filled pauses and silent pauses, alerting a listener of the audio recording to locations of suppressed filled pauses and silent pauses during playback of the audio recording, and accepting a user command to disable suppression of any filled pause or silent pause during playback of the audio recording.

Abstract: The subject invention concerns a system for phonological training a sound reception device (1), an operating device (5) for controlling the system, interpreting and processing devices (2), and presentation device (3).The presentation device (3) includes a display screen divided into a plurality of windows (11-17) for simultaneous presentation of a graphic reproduction of the desired sound as well as of the sound produced by the user and received by the sound reception device (1), and of an animated reproduction of speech device (1), and of an animated reproduction of speech organs. The system is adapted to reproduce the sound by fields(s) (41, 42, 51, 52), the longitudinal extension of the field(s) in one direction reflecting the time during which the sound is produced and the graphic display content within each field, such as colours, shading or the like, of the fields denoting the place of formation of the sound in the oral cavity.

Abstract: This invention relates to a method by means of which more than one audio signal can be recorded in compressed form in a memory element, and to a system implementing such a method. In the system according to the invention, audio signal samples are recorded only when voice is detected in the audio signals. The system according to the invention saves memory capacity required by the recording by combining the audio signal samples when voice is detected in samples of more than one audio signal. Furthermore, an audio signal is not recorded when no voice is detected in the signal. The invention also reduces the average computing capacity needed and thus power consumption, since signal combination, or mixing, is advantageously performed only when voice is detected in the samples of more than one audio signal.

Abstract: An apparatus for reproducing recorded signals by using a recorded medium and a method for reproducing recorded signals. The learning of language is done by using the general recorded medium such as cassette tape and video tape with movies or music recorded thereon, thereby improving the learning efficiency. In the method for reproducing recorded signals of a recording medium in a language learning apparatus, the operation is carried out in the following manner. A control section switches a first switch in accordance with a reproduction command of a reproduction key inputting section, so that the audio signals of an audio signal processing section would be supplied to a speaker. Further, the control section turns on a second switch in accordance with a voice recognition command of a voice recognition key inputting section, so that the voices of a voice detecting section would be supplied and stored to a voice recognizing section.