Abstract: A voice emphasizing device emphasizes in a speech a “strained rough voice” at a position where a speaker or user of the speech intends to generate emphasis or musical expression. Thereby, the voice emphasizing device can provide the position with emphasis of anger, excitement, tension, or an animated way of speaking, or musical expression of Enka (Japanese ballad), blues, rock, or the like. As a result, rich vocal expression can be achieved. The voice emphasizing device includes: an emphasis utterance section detection unit (12) detecting, from an input speech waveform, an emphasis section that is a time duration having a waveform intended by the speaker or user to be converted; and a voice emphasizing unit (13) increasing fluctuation of an amplitude envelope of the waveform in the detected emphasis section.

Abstract: A voice conversion apparatus stores, in a parameter memory, target speech spectral parameters of target speech, stores, in a voice conversion rule memory, a voice conversion rule for converting voice quality of source speech into voice quality of the target speech, extracts, from an input source speech, a source speech spectral parameter of the input source speech, converts extracted source speech spectral parameter into a first conversion spectral parameter by using the voice conversion rule, selects target speech spectral parameter similar to the first conversion spectral parameter from the parameter memory, generates an aperiodic component spectral parameter representing from selected target speech spectral parameter, mixes a periodic component spectral parameter included in the first conversion spectral parameter with the aperiodic component spectral parameter, to obtain a second conversion spectral parameter, and generates a speech waveform from the second conversion spectral parameter.

Abstract: The invention relates to speech speed conversion, and provides a speech speed converting device and a speech speed converting method for changing a speed of voice without degrading the voice quality, without changing characteristics, regarding a signal containing voice. The speech speed converting device includes: a voice classifying unit that is input with voice waveform data and a voice code based on a linear prediction, and that classifies the input signal based on the characteristic of the input signal; and a speed adjusting unit that selects either one of or both a speed conversion processing using the voice waveform and a speed conversion processing using the voice code, based on the classification, and that changes a speech speed of the input signal using the selected speed converting method.

Abstract: A timeline-based approach for selecting and manipulating audio tracks is presented. This is accomplished via a graphical user interface that provides users with a series of visual cues and enhancements when selecting a particular area of an audio track depicted within the interface. These visual cues are rendered as a display region having multiple other display areas, components or interface components that provide the user with a location for initiating actions upon the file. User input provided to the timeline component generates a selection overlay that indicates a selected area of the audio file. The user can perform numerous actions with that audio file, such as copying and pasting. The user can do this more quickly and efficiently because the user is not required to switch tools. Everything is accomplished “modelessly.” Multiple instances of the selection overlay applied, for example, across multiple audio tracks may achieve even more powerful results.

Abstract: A system receives a spoken utterance, identifies at least one keyword within the spoken utterance, and identifies a function using the identified at least one keyword. The system further performs the identified function on at least a portion of the spoken utterance to create a voice file.

Abstract: A communication system includes a data base storing voice applications corresponding to programs executable by user equipment and user profiles. The voice applications are arranged to provide assistance in relation to the programs. A communication handler causes execution of a voice application from the data base responsive to user identification data.

Abstract: A programmed “Stutter Edit” creates, stores and triggers combinations of effects to be used on a repeated short sample (“slice”) of recorded audio. The combination of effects (“gesture”) act on the sample over a specified duration (“gesture length”), with the change in parameters for each effect over the gesture length being dictated by user-defined curves. Such a system affords wide manipulation of audio recorded on-the-fly, perfectly suited for live performance. These effects preferably include not only stuttering but also imposing an amplitude envelope on the slice being triggered, sample rate and bit rate manipulation, panning (interpolation between pre-defined spatial positions), high- and low-pass filters and compression. Destructive edits, such as reversing, pitch shifting, and fading may also alter the way the Stutter Edit is heard. More advanced techniques, include using filters, FX processors, and other plug-ins, can increase the detail and uniqueness of a particular Stutter Edit effect.

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: A system and method for providing previews of media files from a user's media collection to an associated portable media player are provided. In general, media files from the user's media collection are selected based on a play history of the user and optionally a user profile of the user. Once the media files are selected, previews of the media files are generated. The previews may then be transferred to the portable media player during a docking, or synchronization, process. Thereafter, the previews may be played by the portable media player and, if desired, selected by the user for transfer to the portable media player. The media files corresponding to the selected previews are then transferred to the portable media player during a subsequent synchronization process.

Abstract: A speech output apparatus is disclosed, which can allow the user to easily catch synthetic speech when the synthetic speech is output upon being superposed on a music output. The apparatus output can output a music and synthetic speech that indicates contents of information such as an e-mail and is superposed on the music. When the synthetic speech is output to be superposed on the music during output, the apparatus gradually decreases a tone volume of the music.

Abstract: The present invention provides an apparatus having a microphone, an analog to digital converting circuit, a semiconductor memory, input device, and a controller. 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. The input device at least carry out input of a record start and a record end. The controller, according to the input from the input device, carries out operation control for start and stop of 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 controls to start writing the digital signal from the analog/digital conversion circuit into the semiconductor memory.

Abstract: A system and method are provided for creating shorter more natural sounding voice messages and prompts from a plurality of pre-recorded sound segments, the prerecorded sound segments are dynamically cross faded in order to produce a more natural blended sound, various cross fade parameters such as the fade length and the shape of the cross fade amplitude envelopes are determined based on characteristics of the various sound segments being combined.

Abstract: Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context.

Abstract: Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context.

Abstract: Configurations disclosed herein include systems, methods, and apparatus that may be applied in a voice communications and/or storage application to remove, enhance, and/or replace the existing context.

Abstract: An audio information processing system, which when incorporated in home audio video systems, provides independent volume control capability, independent equalization setting capability and independent special effects capability of voice and background sound, to the home audio-video system. The audio information processing system receives an audio signal and extracts there from a voice signal and a background signal based upon correlation of language tracks, correlation of a center channel with surround sound channels, via a voice detection circuit, or via other means. Once the voice signal and background signal are determined, separate processing is performed, and combining of the separately processed voice and background signals may be performed.

Abstract: A system is disclosed for facilitating speech recognition and transcription among users employing incompatible protocols for generating, transcribing, and exchanging speech. The system includes a system transaction manager that receives a speech information request from at least one of the users. The speech information request includes formatted spoken text generated using a first protocol. The system also includes a speech recognition and transcription engine, which communicates with the system transaction manager. The speech recognition and transcription engine receives the speech information request from the system transaction manager and generates a transcribed response, which includes a formatted transcription of the formatted speech. The system transmits the response to the system transaction manager, which routes the response to one or more of the users. The latter users employ a second protocol to handle the response, which may be the same as or different than the first protocol.

Abstract: The present invention includes systems and methods for altering a cellular phone user's speech so that the speech can be less bothersome to third parties in the surrounding area and so that the user has more privacy. Sound cancellation can be used to cancel, reduce, or modify the user's voice so third parties cannot hear the voice as easily or so that the user's voice cannot be understood. Furthermore, the user device can encourage the user to speak in a lower voice. The user device can accomplish this encouragement by indicating to the user their level of speech. In this manner, the user knows when he may lower his voice and yet still provide an adequate volume of speech for the cellular phone. Additionally, the user device can encourage the user to speak in a lower voice by audibly playing back the user's voice in real time.

Abstract: A method of reproduction by a reproduction apparatus for reproducing audio documents forming part of a set of documents. The method includes a prior step of partitioning of the documents of the set into groups of documents whose audio parameters exhibit a similitude, making it possible to determine at least one document representing each group by taking into account its audio parameters. Then, an identifier of a document representing the group is reproduced graphically and/or in a sound manner. In this way, the user can take note of the type of music involved and can select this group by virtue of the graphical identifier. A command may be activated making it possible to go from one group to another; a group may be selected and reproduce the documents of this group. The invention also relates to a reproduction apparatus furnished with a user interface allowing reproduction.

Abstract: A system and method is disclosed for detecting and repairing audio recordings that contain busy signals and extended periods of silence by searching for clusters of silence by reviewing the amplitude in an audio recording sample and listing each silence and sample time.

Abstract: An audio restoration apparatus is provided which restores an audio to be restored having a missing audio part and being included in a mixed audio. The audio restoration apparatus includes: a mixed audio separation unit which extracts the audio to be restored included in the mixed audio; an audio structure analysis unit which generates at least one of a phoneme sequence, a character sequence and a musical note sequence of the missing audio part; an unchanged audio characteristic domain analysis unit which segments the extracted audio to be restored into time domains in each of which an audio characteristic remains unchanged; an audio characteristic extraction unit which identifies a time domain where the missing audio part is located, and extracts audio characteristics of the identified time domain in the audio to be restored; and an audio restoration unit which restores the missing audio part in the audio to be restored.

Abstract: An apparatus for processing a speech signal includes a receiver, a speech signal decoder, a speech rate conversion information detector, and a speech rate converting processor. The receiver receives multiplexed signal of information concerning controls and programs, including speech packets through a transmission line. The decoder decodes the speech signal of packets out of the received signals. The detector detects speech rate conversion execution information in the received signals. The processor subjects the decoded speech signal to a speech rate conversion process if the speech rate conversion execution information indicates that the speech signal has not been subjected to the speech rate conversion process on the transmitting end, and which does not subject the decoded speech signal to the speech rate conversion process if the speech rate conversion execution information indicates that the speech signal has been subjected to the speech rate conversion process on the transmitting end.

Abstract: A simple and efficient method for producing an obfuscated speech signal which may be used to mask a stream of speech, is disclosed. A speech signal representing the speech stream to be masked is obtained. The speech signal is then temporally partitioned into segments, preferably corresponding to phonemes within the speech stream. The segments are then stored in a memory, and some or all of the segments are subsequently selected, retrieved, and assembled into an obfuscated speech signal representing an unintelligible speech stream that, when combined with the speech signal or reproduced and combined with the speech stream, provides a masking effect. While the presently preferred embodiment finds application most readily in an open plan office, embodiments suitable for use in restaurants, classrooms, and in telecommunications systems are also disclosed.

Abstract: A system capable of reducing the influence of sound reverberation or reflection to improve sound-source separation accuracy. An original signal X(?,f) is separated from an observed signal Y(?,f) according to a first model and a second model to extract an unknown signal E(?,f). According to the first model, the original signal X(?,f) of the current frame f is represented as a combined signal of known signals S(?,f?m+1) (m=1 to M) that span a certain number M of current and previous frames. This enables extraction of the unknown signal E(?,f) without changing the window length while reducing the influence of reverberation or reflection of the known signal S(?,f) on the observed signal Y(?,f).

Abstract: An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L?R signals.

Abstract: In a speech processor incorporated into a communication terminal device, an extractor extracts speech characteristics data (e.g. voiceprint data) from speech signals input thereto; then, a speech signal processing module processes input speech signals in accordance with signal processing parameters, which are stored in a memory in relation to preset speech characteristics data in advance. A parameter setting device selects one of preset speech characteristics data having a similarity with the extracted speech characteristics data so as to set the corresponding signal processing parameters stored in the memory to the speech signal processing module. Thus, the communication terminal device is capable of appropriately processing input speech signals so as to enhance specific ranges or to adjust the volume of input speech.

Abstract: According to some embodiments, content filtering is provided for a digital audio signal.

Abstract: A method for time aligning audio signal, wherein one signal has been derived from the other or both have been derived from another signal, comprises deriving reduced-information characterizations of the audio signals, auditory scene analysis. The time offset of one characterization with respect to the other characterization is calculated and the temporal relationship of the audio signals with respect to each other is modified in response to the time offset such that the audio signals are coicident with each other. These principles may also be applied to a method for time aligning a video signal and an audio signal that will be subjected to differential time offsets.

Abstract: Method of processing a transmitted encoded media data stream is received. If a data element arrives prior to, or at, a predetermined playout deadline, the data element is decoded, the media represented by the decoded data element is played, and the data element is provided to a decoder state machine to update a decoder state. If a data element arrives after the predetermined playout deadline, the data element is provided to the decoder state machine to update the decoder state. In one embodiment, if the specified data element fails to arrive by the playout deadline, a subsequently received data element is saved in memory. Then, if the specified data element arrives after the predetermined playout deadline, the specified data element and the saved, subsequently received, data element are provided to the decoder state machine to update the decoder state.

Abstract: A method according to the invention separates multiple audio signals recorded as a mixed signal via a single channel. The mixed signal is A/D converted and sampled. A sliding window is applied to the samples to obtain frames. The logarithms of the power spectra of the frames are determined. From the spectra, the a posteriori probabilities of pairs of spectra are determined. The probabilities are used to obtain Fourier spectra for each individual signal in each frame. The invention provides a minimum-mean-squared error metho or a soft mask method for making this determination. The Fourier spectra are inverted to obtain corresponding signals, which are concatenated to recover the individual signals.

Abstract: A recorded voice message having a first length and a background sound having a second length longer than the first length are received. The first length of the recorded voice message is determined. The level of a portion of the background sound is lowered, the lowered portion having a length that is substantially the same as the first length. The recorded voice message and the lowered portion of the background sound are interleaved. The length of the background sound is adjusted to the first length plus a third length.

Abstract: A sound processor including a microphone (1), a pre-amplifier (2), a bank of N parallel filters (3), means for detecting short-duration transitions in the envelope signal of each filter channel, and means for applying gain to the outputs of these filter channels in which the gain is related to a function of the second-order derivative of the slow-varying envelope signal in each filter channel, to assist in perception of low-intensity sort-duration speech features in said 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: A system, method and computer program product for performing blind change detection audio segmentation that combines hypothesized boundaries from several segmentation algorithms to achieve the final segmentation of the audio stream. Automatic segmentation of the audio streams according to the system and method of the invention may be used for many applications like speech recognition, speaker recognition, audio data mining, online audio indexing, and information retrieval systems, where the actual boundaries of the audio segments are required.

Abstract: A method and apparatus of mobile phone using a semiconductor device includes a first converter, a second converter, a first digital processing circuit, and a second digital processing circuit. The first converter converts a first digital audio signal sampled with a predetermined audio sampling frequency into a second digital audio signal sampled with a predetermined voice sampling frequency. The second converter converts a first digital voice signal sampled with the predetermined voice sampling frequency into a second digital voice signal sampled with the predetermined audio sampling frequency. The first digital processing circuit performs a predetermined digital computation on the second digital audio signal sampled with the predetermined voice sampling frequency and a third digital voice signal.

Abstract: Some embodiments of the invention provide a computer system for processing an audio track. This system includes at least one DSP for processing the audio track. It also includes an application for editing the audio track. To process audio data in a first interval of the audio track, the application first asks and obtains from the DSP an impulse response parameter related to the DSP's processing of audio data. From the received impulse response parameter, the application identifies a second audio-track interval that is before the first interval. To process audio data in the first interval, the application then directs the DSP to process audio data within the first and second intervals.

Abstract: An apparatus and method for the preparation of a censored recording of an audio source according to a procedure whereby no tangible, durable version of the original audio data is created in the course of preparing the censored record. Further, a method is provided for identifying target speech elements in a primary speech text by iteratively using portions of already identified target elements to locate further target elements that contain identical portions. The target speech elements, once identified, are removed from the primary speech text or rendered unintelligible to produce a censored record of the primary speech text. Copies of such censored primary speech text elements may be transmitted and stored with reduced security precautions.

Abstract: A system and method of processing sound signals are disclosed. In one embodiment, a speech coder applies a first sound signal enhancement process to a first part of a sound signal and applies a second sound signal enhancement process to a second part of the sound signal. The sound signal is then coded using the enhanced first part of the sound signal and the enhanced first part of the sound signal and the enhanced sound part of the sound signal. Examples of the portions of the sound signal that are separately processed include an excitation signal component and a spectral component of the sound signal.

Abstract: When there are missing voice-transmission-signals, a repetition-section calculating unit sets a plurality of repetition sections of different lengths that are determined to be similar to the voice-transmission-signals preceding the missing voice-transmission-signal, the repetition sections being determined with respect to stationary voice-transmission-signals stored in a normal signal storage unit, the stationary voice-transmission-signals being selected from the previously input voice-transmission-signals. A controller generates a concealment signal using the repetition sections.

Abstract: Disclosed is a device and method for modifying acoustic characteristics of a speech signal. The method comprises decomposing the signal into a parametric portion and a non-parametric residue; estimating the temporal envelope of the residue; modifying acoustic characteristics of the parametric portion and of the residue in compliance with modification instructions; determining a new temporal envelope for the modified residue using said modification instructions; and synthesizing a modified speech signal from the modified parametric portion and from the residue as modified and with the new temporal envelope.

Abstract: The present invention provides an apparatus having a microphone, an analog to digital converting circuit, a semiconductor memory, input device, and a controller. 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. The input device at least carry out input of a record start and a record end. The controller, according to the input from the input device, carries out operation control for start and stop of 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 controls to start writing the digital signal from the analog/digital conversion circuit into the semiconductor memory.

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: 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: An apparatus for voice modulation in a mobile terminal comprises: a voice input unit being inputted a voice of a subscriber and generating an analog voice signal; a voice modulation unit for modulating the generated analog voice signal; an audio processor for converting the modulated analog voice signal into a digital signal; and an mobile station modem (MSM) for processing the digital signal to be suitable for a wireless transmission. Therefore, the apparatus for voice modulation in a mobile terminal is able to protect the privacy of subscriber by modulating the voice of subscriber during speaking on the phone, and is able to prevent the telephone harassment. Also, the voice of subscriber can be modulated variously as voice in a cave, child voice, devil voice, man's voice, woman's voice, and user defined effect sound, etc., and therefore, the various desires of mobile terminal user can be satisfied.

Abstract: A speech delimiting processing system and method are provided herein.

Abstract: Integrating voice communication into a game console to minimize or eliminate voice data processing by a primary processor. Input voice data from a microphone or a network is processed by a secondary processor and stored in a circular buffer. Drift between storing and reading the processed voice data may result from differing data rates, interrupts, and other latencies. If the circular buffer, accumulates an amount of data that exceeds a predefined threshold corresponding to a human perceptible latency, a pointer in the circular buffer is reset, so that only a portion of the processed voice data is output. A stream of packet contexts each indicate a location and length of voice data in the circular buffer to be output. Preferably, the output voice data is encoded in a standard digital format, such as universal serial bus. The output voice data may be communicated to a network or a sound transducer.

Abstract: This invention generally relates to system, methods and computer program code for editing or modifying speech affect. A speech affect processing system to enable a user to edit an affect content of a speech signal, the system comprising: input to receive speech analysis data from a speech processing system said speech analysis data, comprising a set of parameters representing said speech signal; a user input to receive user input data defining one or more affect-related operations to be performed on said speech signal; and an affect modification system coupled to said user input and to said speech processing system to modify said parameters in accordance with said one or more affect-related operations and further comprising a speech reconstruction system to reconstruct an affect modified speech signal from said modified parameters; and an output coupled to said affect modification system to output said affect modified speech signal.

Abstract: A auto-recording method is disclosed for auto-recording further to user request, via generating user image and voice data, extracting feature points from the image data according to pre-defined user recognition and following by considering the user as an object of following according to extracted feature points, determining whether the image and voice data satisfy a recording reference needed to perform recording. If determined that the image and voice data satisfy the recording reference, editing the image and voice data in a pre-set edit form and generating and storing at least one of recording image and recording voice data.

Abstract: An apparatus for performing stereo adaptation for audio editing includes a stereo decorrelator configured to receive a stereo audio frame in a compressed domain. The stereo audio frame includes a first channel and a second channel. The stereo decorrelator includes a bandwidth limitation element configured to receive a user input defining a desired editing operation to be performed with respect to one of the first and second channels in the compressed domain, and to limit a bandwidth of the other of the first and second channels based on the user input.

Abstract: In one embodiment, the methods and apparatuses detect an original audio signal;detect a sound model wherein the sound model includes a sound parameter; transform the original audio signal based on the parameter whereby forming a transformed audio signal; and compare the transformed audio signal with the original audio signal.