Abstract: Methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR) are introduced. The problem of insufficient noise contents is addressed in a reconstructed highband, by using Adaptive Noise-floor Addition. New methods are also introduced for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The methods and apparatus used are applicable to both speech coding and natural audio coding systems.

Abstract: Systems, methods, and apparatus for low-bit-rate coding of transitional speech frames are disclosed.

Abstract: Systems, methods, and apparatus for low-bit-rate coding of transitional speech frames are disclosed.

Abstract: An embodiment of an apparatus for generating audio subband values in audio subband channels has an analysis windower for windowing a frame of time-domain audio input samples being in a time sequence extending from an early sample to a later sample using an analysis window function having a sequence of window coefficients to obtain windowed samples. The analysis window function has a first group of window coefficients and a second group of window coefficients. The first group of window coefficients is used for windowing later time-domain samples and the second group of window coefficients is used for windowing an earlier time-domain samples. The apparatus further has a calculator for calculating the audio subband values using the windowed samples.

Abstract: A set-top box device comprises a speech recognition module, a video image recognition module, and a voice over Internet protocol bridge. The speech recognition module is configured to perform speech recognition on a voice command signal to determine an action to take in the set-top box device. The video image recognition module is connected to the speech recognition module, and is configured to recognize a display device image. The voice over Internet protocol bridge is coupled to the video image recognition module, and is configured to connect a voice telephone call from the set-top box device to a call center.

Abstract: A method and system for backward compatible multi-channel audio encoding and decoding in sense of the space information maximum entropy is disclosed. The technical solution according to the invention can adopt any existing stereo channel encoding system to encode the multi-channels audio signals, so as to transmit the multi-channel audio signals at the low bit rate as that of the stereo audio signals. More importantly, the existing stereo channel reproducing systems can also decode the audio format that is encoded utilizing the encoding method according to the invention.

Abstract: A multimedia device can be used to play audio. Speech in an environment proximate to a multimedia device can be detected. The detected speech can be recorded. The playing of the audio can be paused. The recorded speech can be audibly presented. A condition to resume the paused audio can be detected. The paused audio can be resumed from the previously paused position.

Abstract: A sound source separation apparatus, includes: a plurality of sound input means into which a plurality of mixed sound signals in which sound source signals from a plurality of sound sources superimpose each other are input; first sound source separating means for separating and extracting SIMO signals corresponding to at least one sound source signal from the plurality of mixed sound signals by means of a sound source separation process of a blind source separation system based on an independent component analysis method; intermediate processing executing means for obtaining a plurality of intermediately processed signals by carrying out a predetermined intermediate processing including one of a selection process and a synthesizing process to a plurality of specified signals which is at least a part of the SIMO signals, for each of frequency components divided into a plurality; and second sound source separating means for obtaining separation signals corresponding to the sound source signals by applying a bin

Abstract: A system is provided for transmitting information through a speech codec (in-band) such as found in a wireless communication network. A modulator transforms the data into a spectrally noise-like signal based on the mapping of a shaped pulse to predetermined positions within a modulation frame, and the signal is efficiently encoded by a speech codec. A synchronization sequence provides modulation frame timing at the receiver and is detected based on analysis of a correlation peak pattern. A request/response protocol provides reliable transfer of data using message redundancy, retransmission, and/or robust modulation modes dependent on the communication channel conditions.

Abstract: Service architecture for providing to a user terminal of a communications network textual information and relative speech synthesis, the user terminal being provided with a speech synthesis engine and a basic database of speech waveforms includes: a content server for downloading textual information requested by means of a browser application on the user terminal; a context manager for extracting context information from the textual information requested by the user terminal; a context selector for selecting an incremental database of speech waveforms associated with extracted context information and for downloading the incremental database into the user terminal; a database manager on the user terminal for managing the composition of an enlarged database of speech waveforms for the speech synthesis engine including the basic and the incremental databases of speech waveforms.

Abstract: An audio signal quality enhancement apparatus and method. The apparatus includes a pitch calculating unit to extract a pitch period of an audio signal, a frequency domain transforming unit to transform the audio signal to a frequency domain, a frequency band dividing unit to classify the transformed audio signal into audio signals for each of the plurality of frequency bands based on the extracted pitch period, and a pitch enhancement unit to determine a gain based on a volume of the transformed audio signal, and to generate an output signal by multiplying each of the classified audio signals with respect to each of the plurality of frequency bands by the gain, thereby enhancing quality of the audio signal.

Abstract: An audio file generation method and system. A computing system receives a first audio file comprising first speech data associated with a first party. The computing system receives a second audio file comprising second speech data associated with a second party. The first audio file differs from the second audio file. The computing system generates a third audio file from the second audio file. The third audio file differs from the second audio file. The process to generate the third audio file includes identifying a first set of attributes missing from the second audio file and adding the first set of attributes to the second audio file. The process to generate the third audio file additionally includes removing a second set of attributes from the second audio file. The third audio file includes third speech data associated with the second party. The computing system broadcasts the third audio file.

Abstract: A vector quantizing device for dividing a sequence of vectors and quantizing them with an enhanced performance of vector quantization by using information on the correlation between the high and low order that the vector sequence has. The vector quantizing device (100) creates a predicted vector by prediction using a first quantization divided vector, creates the difference between the divided vector and the predicted vector as a predicted residual vector, and determines a second code by converting the predicted residual vector into a quantized vector. A vector dequantizing device (150) creates a predated vector by prediction using a first quantization divided vector, creates a second quantization divided vector by adding the predicted vector and the predicted residual vector, and creates a quantized vector by connecting the first and second quantization divided vectors.

Abstract: A mobile communication device for allowing a user to interact with network or internet based data using only verbal communications. The mobile communication device provides the functionality to browse internet web sites and select menus items and hyperlinks by listing to a web page and speaking the identity of the menu item or the hyperlink. The mobile communication system also provides functionality to listen to email and reply to or forward the email, including adding a response by speaking to the mobile communication device. Security is also provided, when appropriate, by requiring the user to speak a predefined security phrase before listening to data designated as secure.

Abstract: An apparatus for viewing information includes a wireless interactive monitor including a screen for displaying the information and adapted to receive the information wirelessly and a surgeon scrub sink for allowing a surgeon to sterilize the hands of the surgeon, positioned under the wireless interactive monitor.

Abstract: A method including displaying content on a display of a device, receiving a speech input designating a segment of the content to be excerpted and transferring the excerpted content to a predetermined location for storage and retrieval.

Abstract: A system is provided with a conversation support means. A conversation support means creates a conversation response, and outputs it in a sound, a character, etc. A conversation response is created in a manner that combines words by inserting a reference keyword as a leading keyword in the response sentence model prepared separately. A conversation support means retrieves the reference keyword beforehand provided in conversation support by dictionary collation from the conversation entry content made by a sound, a manual entry, etc. by a user. Furthermore, the retrieved reference keyword themselves or another reference keyword associated with the retrieved reference keyword are handled as a leading keyword. A series of user conversation contents inputted by the conversation support are accumulated as a base data for determining a user interest. The base data is analyzed to determine a user interest for providing suitable information service.

Abstract: A system and method for providing automatic and coordinated sharing of conversational resources, e.g., functions and arguments, between network-connected servers and devices and their corresponding applications. In one aspect, a system for providing automatic and coordinated sharing of conversational resources includes a network having a first and second network device, the first and second network device each comprising a set of conversational resources, a dialog manager for managing a conversation and executing calls requesting a conversational service, and a communication stack for communicating messages over the network using conversational protocols, wherein the conversational protocols establish coordinated network communication between the dialog managers of the first and second network device to automatically share the set of conversational resources of the first and second network device, when necessary, to perform their respective requested conversational service.

Abstract: To increase channel capacity, mobile phone carriers have deployed speech coders, such as Advanced MultiBand Excitation coding (AMBE), in networks to reduce the bit rate of each call. One undesired consequence of employing such speech coders is that the voice quality can be much worse as compared to higher bit-rate speech coders. A method or corresponding apparatus in an example embodiment of the present invention performs voice quality enhancement transparently within a network by detecting use of a coder applying rate reduction to a speech signal and known to have an adverse effect on a coded speech signal. Upon detection of the use of such coder, the coded speech signal is corrected based on components introduced into the coded speech signal due to the rate reduction. As a result of applying the voice quality enhancement, adverse effects of speech coders can be reduced, while maintaining high quality voice signals.

Abstract: A speech enhancement system controls the gain of an excitation signal to prevent uncontrolled gain adjustments. The system includes a first device that converts sound waves into operational signals. An ambient noise estimator is linked to the first device and an echo canceller. The ambient noise estimator estimates how loud a background noise would be near the first device before or after an echo cancellation. The system then compares the ambient noise estimate to a current ambient noise estimate near the first device to control a gain of an excitation signal.

Abstract: There is provided an audio encoding device for correcting the component having insufficient encoding capability in the core layer by an extended layer. In this device, a core layer encoding unit (101) encodes an audio signal, an extended layer encoding unit (150) encodes an encoding residual of the core layer encoding unit (101), a characteristic correction inverse filter (102 arranged at the pre-stage of an LPC synthesis filter (104) subjects the component having insufficient encoding capability in the core layer to the inverse characteristic correction process, and a characteristic correction filter (105) arranged at the post-stage of the LPC synthesis filter (104) performs a process for characteristic correction of the synthesis signal inputted from the LPC synthesis filter (104).

Abstract: A system and method is described for compensating for the effects of a corrupted Continuously Variable Delta Slope Modulation (CVSD) decoder memory state on a decoded audio signal. In accordance with the system and method, a first estimated step size associated with a first frame of the decoded audio signal is calculated and a second estimated step size associated with a replacement frame generated to conceal bit errors in the first frame of the decoded audio signal is calculated. At least a second frame of the decoded audio signal is then modified based on the first estimated step size and the second estimated step size.

Abstract: Disclosed are new methods and apparatus particularly suited for applications in a vehicle, to provide a wide range of information, and the safe input of data to a computer controlling the vehicle subsystems or “Telematic” communication using for example GM's “ONSTAR” or cellular based data sources. Preferred embodiments utilize new programmable forms of tactile touch screens and displays employing tactile physical selection or adjustment means which utilize direct optical data input. A revolutionary form of dashboard or instrument panel results which is stylistically attractive, lower in cost, customizable by the user, programmable in both the tactile and visual sense, and with the potential of enhancing interior safety and vehicle operation. Non-automotive applications of the invention are also disclosed, for example means for general computer input using touch screens and home automation systems.

Abstract: An administration method and system. The method includes receiving by a computing system, a telephone call from an administrator. The computing system presents an audible menu associated with a plurality of computers to the administrator. The computing system receives from the administrator, an audible selection for a computer from the audible menu. The computing system receives from the administrator, an audible verbal command for performing a maintenance operation on the computer. The computing system executes the maintenance operation on the computer. The computing system receives from the computer, confirmation data indicating that the maintenance operation has been completed. The computing system converts the confirmation data into an audible verbal message. The computing system transmits the second audible verbal message to the administrator.

Abstract: A system that incorporates teachings of the present disclosure may operate according to, for example, a method involving recording audio feedback from a plurality of subscribers commenting on media content supplied by a media communication system on at least one of a plurality of media channels, detecting one or more trigger words in the recorded audio feedback having an association with a disruption of one or more media services supplied by the media communication system, selecting one or more network elements of the media communication system in at least one transmission path that supplies media services to one or more of the plurality of subscribers that supplied audio feedback matching the one or more trigger words, and directing the selected one or more network elements to record media content on one or more media channels selected from the plurality of media channels. Other embodiments are disclosed.

Abstract: A method and system for monitoring, evaluating, and improving effectiveness and efficiency for treating chronic medical conditions of a large patient population. The inventive method and system utilizes computerized patient interview and data analysis modules for assessing a patient's condition and indicating a need for medical attention. Patient interviews are regularly conducted by telephone using computer generated questions and voice recognition methods to enter responses into a database. A series of medical questions are developed and presented to the patient. Their answers are recorded and analyzed with respect to the database. Based upon the answers and the analysis thereof, a medical action plan is developed, care instructions provided, and an appointment with a doctor scheduled.

Abstract: A method of remotely controlling operation of a controlled device involves receiving a telephone call from an owner via a telephone network; authenticating the telephone call to establish that the owner is authorized to control the controlled device; interpreting a voice command from the owner that issues instructions to the controlled device; identifying the controlled device based upon the authentication and identification by the owner of the controlled device; converting the voice command to one or more data packets capable of interpretation by the controlled device to execute the command; and delivering the one or more data packets to the controlled device via the Internet. This abstract is not to be considered limiting, since other embodiments may deviate from the features described in this abstract.

Abstract: Methods, systems, and computer program products are provided multilingual administration of enterprise data. Embodiments include retrieving enterprise data; extracting text from the enterprise data for rendering from a digital media file, the extracted text being in a source language; prompting a user to select a target language; receiving from the user a selection of a target language; translating the extracted text in the source language to translated text in the target language; converting the translated text to synthesized speech in the target language; and recording the synthesized speech in the target language in a digital media file.

Abstract: For audio encoding and decoding, in order to enhance coded audio signals, the audio signal is divided into at least a low frequency band and a high frequency band, the high frequency band is divided into at least two high frequency sub-band signals, and parameters are generated that refer at least to the low frequency band signal sections which match best with high-frequency sub-band signals.

Abstract: A method and apparatus for limiting the absolute magnitude of an audio signal. The method may include firstly variable-gain reducing the gain of an audio signal, and then secondly variable-gain reducing the gain of the audio signal faster than the first variable-gain reduction, thereby limiting the absolute magnitude of the audio signal to a threshold. The first variable-gain reduction may include variable-gain reducing the gain of the audio signal in a first stage, and the second variable-gain reduction may include variable-gain reducing the gain of the audio signal in a second stage that reduces the gain faster than the first stage. The second variable-gain reduction may include delaying the audio signal, finding a peak among the delayed audio signal, calculating a fast gain from a found peak, and modifying the delayed audio signal with the calculated fast gain.

Abstract: A two microphone noise reduction system is described. In an embodiment, input signals from each of the microphones are divided into subbands and each subband is then filtered independently to separate noise and desired signals and to suppress non-stationary and stationary noise. Filtering methods used include adaptive decorrelation filtering. A post-processing module using adaptive noise cancellation like filtering algorithms may be used to further suppress stationary and non-stationary noise in the output signals from the adaptive decorrelation filtering and a single microphone noise reduction algorithm may be used to further provide optimal stationary noise reduction performance of the system.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The preset invention includes receiving a downmix signal including at least one object, object information based on attribute of the object, preset information to render the downmix signal and preset attribute information indicating attribute of the preset information; rendering the downmix signal by applying the preset information to all data regions of the downmix signal, if the preset information is included in an extension region of a configuration information region based on the preset attribute information; and rendering the downmix signal by applying the preset information to one corresponding data region of the downmix signal, if the preset information is included in an extension region of a data region based on the preset attribute information.

Abstract: A noise cancellation apparatus includes a noise estimation module for receiving a noise-containing input speech, and estimating a noise therefrom to output the estimated noise; a first Wiener filter module for receiving the input speech, and applying a first Wiener filter thereto to output a first estimation of clean speech; a database for storing data of a Gaussian mixture model for modeling clean speech; and an MMSE estimation module for receiving the first estimation of clean speech and the data of the Gaussian mixture model to output a second estimation of clean speech. The apparatus further includes a final clean speech estimation module for receiving the second estimation of clean speech from the MMSE estimation module and the estimated noise from the noise estimation module, and obtaining a final Wiener filter gain therefrom to output a final estimation of clean speech by applying the final Wiener filter gain.

Abstract: Disclosed is an apparatus including an unvoiced speech input device, a decision unit and an alarm unit. The unvoiced speech input device receives the unvoiced speech, and the decision unit determines whether or not a signal received from the unvoiced speech input device is an ordinary speech. The alarm unit receives a result of the decision from the decision unit to give an alarm when the result of decision indicates the ordinary speech. The alarm is given to a wearer of the apparatus if he/she has made ordinary speech.

Abstract: Provided is a computer system and a computerized method to automatically generate the synthetic images that simulate the human activities in a particular environment. The program instructions are input in the form of the natural language. Particular columns are provided in the user interface to allow the user to select desired instruction elements from sets of limited candidates. The instruction elements form the program instructions. The system analyzes the program instructions to obtain the standard predetermined time evaluation codes of the instructions. Parameters not include in the input program instructions are generated automatically. Synthetic images are generated by using the input program instructions and the parameters obtained.

Abstract: In order to put the voice transmission function of a data transmission device into practice, the present invention provides an analog processing device for a data transmission device, which includes an analog signal processing unit for performing analog signal processes on an input signal received by the data transmission device and an output signal transmitted by the data transmission device, an audio interface unit coupled to the analog signal processing unit, for transmitting the input signal and the output signal, an output unit coupled to the audio interface unit, for transmitting the output signal from the audio interface unit to an external device, and an input unit coupled to the audio interface unit, for transmitting the input signal from the external device to the audio interface unit.

Abstract: An apparatus and method for encoding and decoding using mutual information between a high band signal and a low band signal to increase a coding efficiency in a portable terminal are provided. The apparatus includes a bandwidth extender for extracting auxiliary information relating to a characteristic of a high band signal using the high band signal and a low band signal and an encoder for encoding residual high band signal obtained by subtracting auxiliary information acquired from the low band signal from auxiliary information acquired from the high band signal.

Abstract: A band-limited voice signal is processed to reduce its spectral envelope or harmonic structure, or both. The resulting reduced signal is moved into a frequency band above the upper limit frequency of the band-limited voice signal, and then combined with the band-limited voice signal to form a band expanded signal with improved quality and comprehensibility, free of unnatural high-frequency resonances and unnaturally strong high-frequency harmonics.

Abstract: A data reproduction device is provided that can achieve seamless reproduction of a stream even at the switching positions of the validity of the bandwidth extension function even in the case where the validity of the bandwidth extension function is switched in the stream.

Abstract: A gaming machine and a gaming system include an engine for interactively advancing a game by a conversation with a player using sounds and texts as media.

Abstract: An audio performance of a media selection is recorded in segments over a communication network. A sender obtains a copy of a media selection that may be divided into media segments for audio recording. The sender can annotate and record a reading of each media segment and any additional commentary. The audio data constituting the “audio performance” is transmitted from a sender telephony device over the communication network to a voice server. The segments of audio data may be collected and arranged in order and assembled with prerecorded segment cues. The audio segments may also be synchronized with digital copies of the media segments. In one implementation, a user, for example, a grandparent, can read a children's book into a telephony device, including personal anecdotes, for page-by-page recording over the communication network for storage at a voice server for later fulfillment to a grandchild in conjunction with a copy of the media selection in physical or electronic form.

Abstract: The invention aims at constructing improved dictionaries of CELP excitation vectors for coding/decoding digital audio signals. Usually, each vector of dimension N comprises pulses capable of occupying N valid positions. The invention concerns the construction of dictionaries with particular structure by: providing a common sequence of pulses forming a base pattern; and assigning the base pattern to each excitation vector of the dictionary, based on one or more occurrences at one or more respective positions among said N valid positions. The invention also concerns a combination of dictionaries thus constructed with optionally standard multipulse dictionaries, by union or summation or cascading.

Abstract: An audio encoder or encoding method receives a plurality of input channels and generates one or more audio output channels and one or more parameters describing desired spatial relationships among a plurality of audio channels that may be derived from the one or more audio output channels, by detecting changes in signal characteristics with respect to lime in one or more of the plurality of audio input channels, identifying as auditory event boundaries changes in signal characteristics with respect to lime in the one or more of the plurality of audio input channels, an audio segment between consecutive boundaries constituting an auditory event in the channel or channels, and generating all or some of the one or more parameters al least partly in response to auditory events and/or the degree of change in signal characteristics associated with the auditory event boundaries. An auditory-event-responsive audio upmixer or upmixing method is also disclosed.

Abstract: The present invention relates to a transmitting apparatus, a transmitting method, a receiving apparatus, a receiving method, a transceiver, a communication apparatus and method, a recording medium, and a program in which high quality voice can be decoded. A cellular telephone 421-1 outputs coded voice data and also supplies uncoded voice sample data to a switching center 423 while a telephone call is not made. Based on voice data used for the previous calculation processing and newly input voice data, the switching center 423 performs calculation processing for quality-improving data for improving the quality of voice to be output from a cellular telephone 421-2 that receives the coded voice data. The switching center 423 stores the optimal quality-improving data as a user information database in association with the cellular telephone 421-1. The cellular telephone 421-2 decodes the coded voice data based on the optimal quality-improving data supplied from the switching center 423.

Abstract: A frame compensation method is provided. The method includes: obtaining a length of a lost frame and a length of a correct frame; determining that the length of the correct frame is integral power of 2 times of the length of the lost frame, and then obtaining a data sequence with the same length as the length of the lost frame according to the correct frame; and compensating the lost frame according to the data sequence to obtain a compensated data frame. A frame compensation system is also provided. Lost frames in various formats are compensated according to correct frames in various formats, so that the limitation of the related art that a lost frame in a single format can be merely compensated according to a correct frame in a single format is eliminated, and the effect of the compensated data frames is better than that of filling comfort noises.

Abstract: A relay device 20 duplicates speech data received from a communication terminal that is engaged in voice communication with another communication terminal. The duplicated speech data is transmitted to and is stored at a media processing device 40. Media processing device 40 builds a database for speech synthesis based on the stored speech data.

Abstract: Provided is a residual signal coding/decoding apparatus and method. The residual signal coding apparatus includes a transformer, a band splitter, a pulse searcher, and a pulse quantizer. The transformer transforms time-domain residual signals into a frequency domain to output transform coefficients. The band splitter splits the transform coefficients into bands to output the transform coefficients. The pulse searcher searches the transform coefficients for the respective bands to select optimal pulses and output parameters of the optimal pulses. The pulse quantizer quantizes the parameters of the optimal pulses.

Abstract: A decoding method and device are provided. The spectrum parameter of a current bad data frame is determined. Specifically, a number of continuous bad frames that occur currently is determined. A spectrum parameter of a good data frame before the current bad data frame is determined. And a constant mean value of a spectrum parameter is determined. Then, the spectrum parameter of the good data frame is adaptively shifted towards the constant mean value of the spectrum parameter according to the number of the continuous bad data frames to calculate and obtain spectrum parameter information of the current bad frame. When the continuous bad data frames occur, the relevance between the spectrum parameter of the nearest good frame and the spectrum parameter of the current bad frame is gradually reduced, so that more accurate spectrum parameter of the current bad data frame can be obtained, thereby obtaining a better speech quality under a same code rate and a same frame error rate.

Abstract: Disclosed are systems, methods and computer-readable media for enabling speech processing in a user interface of a device. The method includes receiving an indication of a field and a user interface of a device, the indication also signaling that speech will follow, receiving the speech from the user at the device, the speech being associated with the field, transmitting the speech as a request to public, common network node that receives and processes speech, processing the transmitted speech and returning text associated with the speech to the device and inserting the text into the field. Upon a second indication from the user, the system processes the text in the field as programmed by the user interface. The present disclosure provides a speech mash up application for a user interface of a mobile or desktop device that does not require expensive speech processing technologies.

Abstract: A system for providing multi-language conference is provided. The system includes conference terminals and a multipoint control unit. The conference terminals are adapted to process a speech of a conference site, transmitting the processed speech to the multipoint control unit, process an audio data received from the multipoint control unit and output it. At least one of the conference terminals is an interpreting terminal adapted to interpret the speech of the conference according to the audio data transmitted from the multipoint control unit, process the interpreted audio data and output the processed audio data. The multipoint control unit is adapted to perform a sound mixing process of the audio data from the conference terminals in different sound channels according to language types, and then sends mixed audio data after the sound mixing process to the conference terminals.