Abstract: An audio decoder is provided with a programmable and re-configurable downmixing process. In one embodiment, the audio decoder includes a control module and a data path. The data path is configured to read, scale, add, and write audio samples to and from various audio channel frame buffers. The control module implements state diagrams which specify various control signals for directing the operations of the data path. The control module implements state diagrams for directing windowing and downmixing operations. The order in which these operations are performed may be reconfigurable, i.e. downmixing may be performed before or after windowing. This reconfigurability advantageously permits the system designer to trade a slight audio quality enhancement for a decreased memory requirement for some speaker configurations. The downmixing operation requires scaling coefficients which are provided by the control module.

Abstract: A listener's ability to recognize and identify similar sounds is improved by forming exaggerated sounds which differ in spectral/temporal energy distribution from one another more than the similar sounds differ from one another and in generally the same way in which the similar sounds differ and testing the listener's ability to recognize and identify the exaggerated sounds. The exaggerated sounds are formed as follows: The similar sounds are spectrally transformed into a coordinate space and a linear function is projected between a pair of points representing the similar sounds in the coordinate space. Between the pair of points, the linear function represents a continuum of sounds between the similar sounds. To exaggerate the sounds, points are extrapolated outward from the pair of points along the linear function.

Abstract: A method for generating animated sequences of talking heads in text-to-speech applications wherein a processor samples a plurality of frames comprising image samples. Representative parameters are extracted from the image samples and stored in an animation library. The processor also samples a plurality of multiphones comprising images together with their associated sounds. The processor extracts parameters from these images comprising data characterizing mouth shapes, maps, rules, or equations, and stores the resulting parameters and sound information in a coarticulation library. The animated sequence begins with the processor considering an input phoneme sequence, recalling from the coarticulation library parameters associated with that sequence, and selecting appropriate image samples from the animation library based on that sequence. The image samples are concatenated together, and the corresponding sound is output, to form the animated synthesis.

Abstract: Audio sources are coded by recognizing different classes of audio such as speech and music. The classes are used to select between coding algorithms and to provide object definitions. Objects have abstract and concrete classes which may further rely on parameters produced by linear prediction and subband filters to provide a frame-based bit stream of information. Each object in the bit stream has layers of information such as basic bit rate, coding parameters and enhancement parameters. The layers of information in each object allow altering selected parameters to manipulate audio signals.

Abstract: For use in a synthesizer having a wave source that produces a periodic wave, frequency shifting circuitry for frequency-shifting the periodic wave and waveshaping circuitry for transforming the periodic wave into a waveform containing a formant, the frequency-shifting causing displacement of the formant, a circuit for, and method of, compensating for the displacement and a synthesizer employing the circuit or the method. In one embodiment, the circuit includes bias circuitry, coupled to the wave source and the frequency shifting circuitry, that introduces a bias into the periodic wave based on a degree to which the frequency shifting circuitry frequency shifts the periodic wave, the bias reducing a degree to which the formant is correspondingly frequency-shifted.

Abstract: A voice recording/reproducing apparatus includesan A/D converting circuit for converting an input analog voice signal into a digital signal;a digital signal processing circuit for coding the converted digital signal in a predetermined format and decoding the coded voice data;a recording medium for storing the coded voice data;a buffer memory;a D/A converter for converting the decoded digital signal stored in the recording medium into an analog voice signal;an operation unit for setting a predetermined mode; anda control circuit for performing a predetermined process for reducing noise in a reproducing operation to the voice data obtained from time a predetermined period of time before the operation unit recognizes that a recording stop operation is performed by the operation unit during a recording operation to time when the operation unit recognizes that the recording stop operation is performed whereby the noise due to a stop operation is reduced when the voice data is in said buffer memory and then transfe

Abstract: The present invention provides a text-to-speech conversion system (TTS) for interlocking with multimedia and a method for organizing input data of the TTS which can enhance the natural of synthesized speech and accomplish the synchronization of multimedia with TTS by defining additional prosody information, the information required to interlock TTS with multimedia, and interface between these information and TTS for use in the production of the synthesized speech.

Abstract: A record playback system includes a display showing elapsed time of a record playback operation together with symbols indicating occurrences of certain sequences of sound during the playback operation, the symbols positioned to indicate times at which respective sequences of sounds occur. In a preferred application, the records reproduced in the system are audible voice-mail messages, the specific sequences of sounds are numbers or sets of numbers spoken consecutively during the message, and the symbols representing such numbers are printed characters corresponding to respective numbers. In the preferred application, the messages are centrally recorded at a server of a computer network and distributed to individual client computers via the network.

Abstract: An audio/video input/output (I/O) port apparatus for acquiring digital audio samples from one or multiple channels of input audio and synthesizing digital audio samples into one or multiple channels of output audio. The apparatus comprises a video I/O port, a frequency synthesizer, and an audio I/O port. The video I/O port generates a video-rate clock, and is configured to digitize input video into digital video, and to synthesize output video from digital video. The frequency synthesizer is configured to derive an audio sampling clock based on the video-rate clock. The audio I/O port is configured to sample input audio and convert it into digital audio samples according to the sampling clock, and to synthesize digital audio samples into output audio according to the sampling clock. The apparatus ensures that the video and audio data track together, both when inputting the information from an external source and when outputting the audio/video data streams.

Abstract: The present invention discloses a method for organizing audio events recorded on a storage medium to facilitate efficient storage and fast access. The method comprises the steps of: (1) maintaining attributes of the audio events in a list; (2) determining if a subset of the audio events causes a medium fragmentation based on the attributes, the subset including originally sequential audio segments; (3) if the medium fragmentation occurs, copying the subset of said audio events from the storage medium into a buffer memory of a predetermined size; and (4) copying the buffer memory onto the storage medium.

Abstract: A code recording apparatus code images voice data, records and prints the voice data, and prints and records the image data or text data on a recording sheet. A recording time arithmetic section calculates the inputtable time of the voice data, and displays the inputtable time in a recording time display section.

Abstract: A method for configuring an audio interface for a speech recognition application in a computer system comprises the steps of: (a) determining from a data storage registry of said speech recognition application whether a microphone operatively associated with said computer system has been previously successfully set up for operation with said speech recognition application; (b) displaying a first graphical user interface having a first user activatable icon for setting up said microphone and a second user activatable icon for adjusting an audio recording level for said microphone; (c) in the event said microphone is not registered as having been previously successfully set up, altering said second icon in appearance and disregarding attempted activation of said second icon; and, (d) in the event said microphone is registered as having been previously successfully set up, respectively initiating said setting up of said microphone and said adjusting of said audio recording level responsive to activation of said

Abstract: A voice data processing device for at least one of recording and reproducing a voice comprises a digital signal processor; a rewritable program memory, integrated with the digital signal processor, for storing a program; and a receiver, such as an earphone jack, for receiving a program rewrite signal to rewrite the program without exposing the program memory.

Abstract: A listener's ability to recognize and identify similar sounds is improved by testing the listener's ability to recognize and identify exaggerated sounds that are formed from pairs of similar sounds. The exaggerated sounds differ from one another in spectral/temporal energy distribution more that the similar sounds differ from one another and in generally the same way in which the similar sounds differ. The exaggerated sounds are formed as follows: the similar sounds are spectrally transformed into a coordinate space and a linear function is projected between a pair of points. To exaggerate the sounds, points are extrapolated outward from the pair of points along a linear function. The listener is challenged to repeatedly identify each of the exaggerated sounds. When the listener has demonstrated proficiency, the degree to which the exaggerated sounds differ from the similar sounds is reduced and the listener is again challenged.

Abstract: A reproduction speed of speech sound changing apparatus which reproduces speech data at a speed in which essential part thereof can be caught so that the outline of the speech sound can be grasped even when changing the reproduction speed, besides remarkably reduces the whole reproducing time wherein a reproducing speed in each predetermined period is calculated according to a parameter value in every predetermined period of speech data in accordance with such a manner that a part having a high parameter value such as high power, high pitch or the like of speech data is judged to be the part, where important contents are involved, and such part of important contents is reproduced at such a speed that the contents can be caught, while the parts other than that described above are reproduced either at such a speed that the whole reproduction of speech data can be completed within a required time, or reproduced by skipping over the parts if at thus determined reproduction speed, reproduced speech sound cannot be

Abstract: Method envelope-invariant for audio signal synthesis from elementary audio waveforms stored in a dictionary wherein:the waveforms are perfectly periodic, and stored as one of their period,synthesis is obtained by overlap-adding of the waveforms obtained from time-domain repetition of the periodic waveforms with a weighting window whose size is approximately two times the period of the signals to weight, and whose relative position inside of the period is fixed to any value identical for all the periods, each extracted from a reharmonized and thus periodic waveform, obtained by modifying, without changing the spectral envelope, the frequencies and amplitudes of harmonics in the spectrum of a frame of the original continuous speech waveform,whereby the time shift between two successive waveforms obtained by weighting the original signals is set to the imposed fundamental frequency of the signal to synthesize.

Abstract: The table document preparation module prepares a table document containing cells, the read-out attribute setting module sets a read-out attribute specifying a way of reading-out cell data supplied through the setting screen from the table document preparation module being assisted by the setting display module, and voice-generating data generation module generates voice-generating data for the table document according to the way of reading out specified by the read-out attribute, and the voice synthesis module synthesizes voices according to the voice-generating data.

Abstract: A method for configuring an audio mixer in an audio interface for a speech recognition application in a computer system in accordance with an inventive arrangement comprises the steps of: (a) displaying at least one graphical user interface for selecting an audio input and output device from a plurality of audio input and output devices compatible with the speech recognition application, including: a headset with a microphone and an earpiece speaker; a microphone not forming part of a headset; at least one external speaker connected to the computer system; at least one internal speaker connected to the computer system; and, different combinations of the input and output audio devices; (b) selecting and adjusting microphone inputs of the mixer in accordance with an audio input device selected in step (a); and, (c) deselecting audio inputs and outputs of the audio mixer in accordance with the audio input device and an audio output device selected in step (a), whereby unneeded audio inputs and outputs of the mix

Abstract: An electronic notebook assigns properties to digital ink of note marks and keywords to digital pages. The properties indicate the type of note mark and the keywords indicate general content of the note marks on a digital page. The notebook also synchronizes audio data with the note marks and permits audio highlighting. The electronic notebook provides an updatable database of note marks and audio that is capable of being searched based on the properties, keywords, and audio highlights.

Abstract: The system disclosed herein employs a method for facilitating access to supplemental information relating to text items entered into an editing application program by a speech recognition system, e.g. alternate translations of utterances stored in a dictation event database (DEDB). A speech event database (SEDB) associated with the editing program is maintained so as to contain text items entered by said speech recognition system and context information identifying preceding and following text items. In response to a user initiated change in the editing point, a context list is created characterizing a region of text around the new editing point. The speech SEDB event database is searched for text items matching items in said context list thereby to create a mapping of the list to the SEDB. The speech event database is then used as an index into said dictation event database, e.g. for correcting previously misrecognized utterances.

Abstract: This invention relates to converting the characteristics of sounds such as oices, musical tones, natural sounds, and so on, and more specifically to facilitating the conversion operation, and also to sound-label association suitable for the characteristic conversion. Various embodiments of the invention comprise several of the following elements to provide useful results: sound-label data holding means, display control means, conversion means, sound-label dividing means, label-data dividing means, association forming means, data inputting means, and communication means. Other embodiments of the invention may be practiced as processes or articles of manufacture.

Abstract: A system that synchronously segments a speech waveform using pitch period and a center of the pitch waveform. The pitch waveform center is determined by finding a local minimum of a centroid histogram waveform of the low-pass filtered speech waveform for one pitch period. The speech waveform can then be represented by one or more of such pitch waveforms or segments during speech compression, reconstruction or synthesis. The pitch waveform can be modified by frequency enhancement/filtering, waveform stretching/shrinking in speech synthesis or speech disguise. The utterance rate can also be controlled to speed up or slow down the speech.

Abstract: An audio interactive system including a plurality of terminals each having an audio input means for converting voices into aural signals and an audio output means for converting aural signals into voices, the plurality of terminals being connected to a communication line; and a server connected with the plurality of terminals via the communication line to perform collection and distribution of the aural signals, in which a conversation is made through voices between the plurality of terminals. The server comprises a buffer for temporarily storing the aural signals transmitted from the terminals, and a scheduler for controlling the distribution of the aural signals stored in the buffer.

Abstract: A memory that includes data corresponding to a play list is controlled by an associated circuit that eliminates the problems created during playback by short data segments. The play list is parsed to determine whether any segments in the play list fail to exceed a threshold length. Such a segment is copied along with all or part of the next segment to create a new "virtual" segment that exceeds the threshold length. During playback, the virtual segment is used instead of the original short segment.

Abstract: An analog voice signal is encoded for playback in a form in which the identity of the speaker's voice is disguised. To do this, the analog voice signal is converted to a first digital voice signal which is divided into a plurality of sequential speech segments. A plurality of voice fonts, for different types of voices are stored and one of these is selected as a playback voice font. An encoded voice signal for playback which includes the plurality of sequential speech segments and either the selected font or an identification of the selected font is generated. In addition, the digital voice signal is analyzed to identify characteristics of the voice signal.

Abstract: Prosodic databases hold fundamental frequency templates for use in a speech synthesis system. Prosodic database templates may hold fundamental frequency values for syllables in a given sentence. These fundamental frequency values may be applied in synthesizing a sentence of speech. The templates are indexed by tonal pattern markings. A predicted tonal marking pattern is generated for each sentence of text that is to be synthesized, and this predicted pattern of tonal markings is used to locate a best-matching template. The templates are derived by calculating fundamental frequencies on a pursuable basis for sentences that are spoken by a human trainer for a given unlabeled corpus.

Abstract: A book database stores at least phonetic signal information including phoneme information and rhythm information as document data, a central system transmits phonetic signal information stored on the book database to a terminal and the terminal receives the phonetic signal information is then carried out at the terminal and the document is then recited via synthesized sounds.

Abstract: The presentation of audio information, particularly audio information generated by a voice synthesizer from text using a text or screen reader, is controlled using a style sheet. The style sheet permits default presentation styles, such as voice-family, voice-pitch, voice-variant, voice speed and volume to be set, and then varied based on embedded text presentation commands such as those found in hypertext markup language and in desktop publishing.

Abstract: An acoustic signal processing device used in, for example, karaoke systems, includes a signal processor, a digital/analog converter and a mixer. The signal processor converts the input analog acoustic signal into a digital signal and processes the converted digital signal by signal processing for generating a digital signal added to with pre-set acoustic characteristics. The digital/analog converter converts the digital signal generated by the signal processor into an analog signal. The mixer mixes the input analog signal with the analog signal outputted by the digital/analog converter.

Abstract: A singing sound-synthesizing apparatus sequentially synthesizes vocal sounds based on singing data including lyric data of a lyric formed of a plurality of phonemes and sounding data designating a sounding time period over which the lyric data is sounded. A designating device designates a predetermined voiced phoneme from the plurality of phonemes of the lyric data. A sounding control device carries out sounding control such that sounding of the predetermined voiced phoneme designated by the designating device is started within the sounding time period designated for the plurality of phonemes by the sounding data and continued until the sounding time period designated for the plurality of phonemes elapses. In another form, ones of phoneme parameter sets and ones of coarticulation parameter sets corresponding to signing data are read from a phoneme data storing the phoneme parameter sets and the coarticulation parameter sets.

Abstract: An apparatus for processing and editing an analog audio recording. An analog audio recording device comprises a data receiver for receiving analog audio data and a plurality of analog markers, an analog output for outputting the analog audio data with the plurality of analog markers, and a processing unit coupled to the analog output. The processing unit comprises digitizing circuitry for digitizing the analog audio data with the plurality of analog markers to create a digital audio file comprising digital audio data and a plurality of digital markers. The processing unit also comprises integrated circuit memory storing editing software for editing the digital audio file using the plurality of digital markers to create an edited digital audio file.

Abstract: A conduct-along system that can give expressions to sounds and/or images, in which expressions are added to sounds and/or images following the playback of sounds and/or images in real-time based on any one or any combination of parameters, such as tempo, intensity, beat timing and accent, detected from the movement of an input device. The conduct-along system detects any one or any combination of parameters, such as tempo, intensity, beat timing and accent, from the movements of the input device, and plays back voices and/or images in real-time following any one or any combination of detected parameters, such as tempo, intensity, beat timing and accent.

Abstract: A light-weight adhesive note recording device (12) records sounds (22) and written text (20) and attaches to a plurality of different types of surfaces (14). Recording circuitry (34) receives and records sounds (22) as a plurality of digital signals. Memory circuitry (38) stores digital signals, and speaker circuitry (40) plays back digital signals as recordings of sounds (22). Control circuitry (36) controls operation of recording circuitry (34), memory circuit (38), and speaker circuitry (40). Power source (44) provides power to operate recording device (12). Text surface (26) is used for writing or printing written text or message (20). Adhering mechanism (54) adheres recording device (12) to a wide variety of surfaces (14). Recording device (12) has sufficiently small size and a flat writing or text surface (26) for repeated use as an adhesive note recording device.

Abstract: A processing system time compresses a voice message before transmission, and processing system time expands the message after reception. To process the message the processing systems perform at least one of: (a) randomizing the order of a sequence of samples form a silent portion of the message after reception thereof before blending the sequence with a last portion of the expanded message; (b) selecting the sequence of samples from the silent portion of the message after reception thereof, the sequence selected being poorly correlated with the last portion of the expanded message, before blending the sequence with the last portion of the message; and (c) compressing the dynamic range of the message before transmission, by an amount dependent upon the signal-to-noise ratio of the message, and aggressively expanding the dynamic range of the message after reception, by a fixed amount.

Abstract: A voice-generating information making apparatus comprises: a talking way data storing section for storing therein talking way data comprising character string information grouped according to the character string information, a character string input unit for inputting a character string (consisting of a control section, an application storing section, a key entry section, and a display section), a retrieving unit for retrieving a group having the same character string information as the inputted character string, a voice tone data storing section for storing therein a plurality of voice tone data, a voice synthesizing section for synthesizing a voice, a voice selecting unit for selecting a desired voice from the synthesized voice, and a voice-generating document storing section for storing therein talking way data corresponding to the selected voice as a voice-generating document in correlation to the inputted character string.

Abstract: An Audio Editor operates with files capable of storing text and voice data in separate regions, and provides functions for entering data as voice data, and also for fully editing the entered voice data. Files can be uploaded from the Audio Editor to a PC application for converting the file entirely to text, providing a system wherein all variable entry and editing can be done verbally, and conversion to text left as a final chore. In an alternative embodiment the Audio Editor is implemented as a PC application wherein a user can enter and fully edit variable input as voice, and then communicate the resulting file to another for final conversion. In yet another embodiment the Audio Editor is implemented as additional functionality to a high-end word processor application. In further embodiments computerized natural data editors are provided for reviewing and editing natural data streams of all sorts, such as video streams, musical works, and the like.

Abstract: A scanning apparatus for scanning a video recording for objectionable content is disclosed that includes an audio processor for analyzing the recording and recognizing the objectionable audio, and audio control gate for manipulating the objectionable audio. The scanning apparatus also includes an amplifier that amplifies the signal and converters to convert the signal from analog to digital format prior to analysis and digital to analog format after analysis. The method of scanning the video recording for objectionable content includes the steps of analyzing the recording and manipulating the recording. The method first includes the steps of amplifying the recording signal, separating the audio portion of the recording from the composite recording, digitizing the portions of the recording, and storing the composite portion of the recording in a loop while analysis is performed on the audio portion of the recording.

Abstract: A method, system and product are provided for mixing of encoded audio signals. The method includes receiving a first encoded audio signal, the first signal having a plurality of frequency subbands, each subband having an audio level, and receiving a second encoded audio signal, the second signal having a plurality of frequency subbands, each subband having an audio level. The method also includes mixing the first and second signals to create a mixed encoded audio signal having a plurality of frequency subbands, each subband having an audio level based on a masking effect relationship between the audio levels and sample data of corresponding subbands of the first and second signals. The system includes control logic for performing the method. The product includes a storage medium having computer readable programmed instructions recorded thereon for performing the method.

Abstract: A method, system and product are provided for harmonic enhancement of an encoded audio signal. The method includes receiving the encoded audio signal, the encoded audio signal having multiple frequency subbands, selecting one of the subbands having a data sample associated therewith, and generating a frequency doubled copy the data sample associated with the subband. The method also includes generating a new data sample for a second subband using the frequency doubled copied data sample, the second subband having a frequency greater than the first subband by one octave, and modifying the encoded audio signal to create an enhanced encoded audio signal having the new data sample associated with the second subband. The system includes control logic for performing the method. The product includes a storage medium having computer readable programmed instructions for performing the method.

Abstract: A data transfer system for transferring between a plurality of data storage devices is provided. A data transfer request system requests scheduled and unscheduled data transfers between respective ones of the data storage devices. A first database stores entries corresponding to scheduled data transfer request and a second database stores entries corresponding to unscheduled data transfer requests. A scheduler updates the entries in the first and second databases on a predetermined interval. A file transfer system accesses the entries in the first and second databases and transfer data between the data storage devices according to information in the database entries.

Abstract: An audio effects processing system and method performs audio effects processing on a subband-by-subband basis using less memory and computational resources than audio effects produced through post processing of a fullband audio signal. The input fullband audio signal is split into subbands and the compressed subbands may be stored in memory of the system as compressed audio subband data or may be generated by real time processing of a fullband audio signal. The system stores in memory a set of prototype subband effects filters, the set including echo, flange, chorus, and reverberation effects filters. A subband filter customization procedure allows a sound system designer to build customized versions of the prototype subband effects filters. One or more of the customized subband effects filters are applied to one or more of the subbands comprising the compressed audio subband data.

Abstract: A read/write control circuit for a memory uses a read/write judging circuit which receives read and write request signals and a directional signal indicating whether a sequence of sound data inputted into the memory is in forward (normal) sequence or reversed sequence. The read/write judging circuit refers to the directional signal when both the read and write request signals simultaneously become active such that the read/write judge circuit gives priorty to a write operation in the memory when the sequence is normal by asserting a write enable signal and gives priorty to a read operation in the memory when the sequence is reversed by asserting a read enable signal.

Abstract: A method and system for editing messages within a recording system includes a handheld recorder having flash memory and a plurality of memory address pointers for storing messages. The flash memory has a plurality of sequential memory blocks, each memory block has a plurality of memory locations, and the messages are stored as message segments. The plurality of memory address pointers stores an ordered sequence of addresses of said message segments within the flash memory. Upon an edit point being selected within a message at which the message is to be edited, the edit point corresponding to a first location within the message defining an end point of a first message segment, an edited message comprising a second message segment is generated.

Abstract: A method is disclosed of modification of parameters of audio signals by dividing a digital signal converted from an original analog signal into sound frames, modifying a pitch and a playing rate of the digital signal within a frame and subsequent successive splicing a last modified frame with a first non-modified frame and calculating the mean absolute error to define the best splicing point in terms of producing minimal or no audible noise such that various sections of sound signals can be spliced together to achieve pitch and playing rate modification.An apparatus is also disclosed for implementing the method, the apparatus comprising input and output amplifiers, a low pass filter at the input and a low pass filter at the output, analog-to-digital and digital-to-analog converters, and a pitch shifting processor.

Abstract: A method and device for inputting characters. An answering equipment, Ts, answers a call from a terminal equipment, Te. The answering equipment transmits a sequence of characters which can consist of letters, digits or other characters. The information is transmitted via the telecommunications network to the terminal equipment where the information is read out in a loudspeaker which is listened to by a subscriber. When the subscriber hears the character in question being read, he/she presses a key on the terminal equipment. The key can be one of the keys on the ordinary keypad of twelve keys on the telephone. The signal passes to the answering equipment where it is identified and registers that the character in question shall be stored in a data base, DB. The data base stores the character and combines the other characters which are input, and predicts which possible words or character combinations remain, depending on application.

Abstract: A voice recording/reproducing apparatus comprises a coding parameter extracting section for extracting a coding parameter by use of either past voice data or past parameter. A coding section codes voice data by use of the coding parameter extracted by the coding parameter extracting section. A predicting section predicts a decoding signal by use of either past decoded voice data corresponding to coded voice data from the voice coding means or the past parameter. A voice decoding section decodes the voice data by use of the predicted decoding signal. A voice synthesizing section outputs voice data synthesized based on an output signal from the predicting section and an output signal from the voice decoding section. An initializing section initializes at least one of either a content of the predicting section or a content of the voice synthesizing section in accordance with a reproducing position of recorded voice data.

Abstract: An information processor capable of scanning and printing two-dimensional bar codes representing sounds and moving images. The processor prints text and graphics, the two-dimensional bar codes, and relational information that correlates the bar codes of sounds and moving images to the associated text and graphics printed by the processor. In addition, the processor (or another processor) scans the printed information, bar codes and relational information. Having scanned the printed information, the processor displays the printed text and graphics in a manner that allows a person viewing the display to mark the displayed text and graphics, and hear the bar-coded sounds and images associated with the marked text and graphics.

Abstract: A speech information recording/reproduction apparatus includes a CPU for recording digital speech data, into which a speech signal is converted, in a rewritable solid memory, a partial erasing switch for erasing the speech data recorded in the solid memory, and a reproduction switch for setting an erasing range of the speech data from the solid memory. The CPU delays execution of erasure for several seconds after the erasing range is set, and predetermined information about the execution of erasure is displayed by an LCD and an LED for these several seconds. The CPU does not erase the speech data from the solid memory when a reset switch is operated to cancel the execution of erasure during the display of the information, but erases it after the information is displayed when the reset switch is not operated during the display.

Abstract: A recording and playback device that allows the user to record a desired message and then play back the message in either the order in which the message was recorded or in an order reversed from the order in which the message is recorded. The message is preferably stored in the proper, forward order and reversed only when reverse playback is desired. The message is re-recorded as desired, the previously recorded message being overwritten.

Abstract: A method for producing a voice file for use in an automated concatenated voice system. The words and phrases to be used in the system are scripted in a staged script, and read by a voice talent. The recording of the staged script as read by the voice talent is processed and edited to produce a plurality of naturally sounding words and phrases which may be concatenated into voice messages. The edited words and phrases are stored in a composite voice file for use by an automated concatenated voice system.