Abstract: A scalable encoding apparatus wherein stereo audio signals can be scalable encoded by use of a CELP encoding to improve the encoding efficiency. In the apparatus, an adder and a multiplier obtain an average of first and second channel signals as a monophonic signal. A CELP encoding part performs a CELP encoding of the monophonic signal. A first channel difference information encoding part performs an encoding of the first channel signal in conformance with the CELP encoding and obtains a difference between a resulting encoded parameter and an encoded parameter outputted from the CELP encoding part. The first channel difference information encoding part then encodes this difference and outputs the resulting encoded parameter.

Abstract: A method of evaluating the efficiency of a noise-reducing function adapted to be applied to audio signals and comprising a preliminary step of obtaining a predefined test audio signal X[m] containing a noise-free wanted signal, a noisy signal Xb[m] obtained by adding a predefined noise signal to the test signal X[m], and a processed signal Y[m], obtained by applying the noise-reducing function to the noisy signal Xb[m], is remarkable in that it includes a loudness measuring step (E3, E4) for some or all the frames m of the aforementioned signals X[m], Xb[m] and Y[m].

Abstract: A speech dialog system includes a signal input unit that receives an acoustic input signal. A voice activity detector compares a portion of the received signal to a noise estimate to determine if the signal includes voice activity. A speech recognizer processes signals containing voice activity to determine if the signal contains speech. An output unit modifies signals when output of the system substantially coincides with the delivered speech.

Abstract: A modulation device including: a modulation unit for modulating a carrier in an audible sound range by an encoded transmission signal to generate a modulated signal; a masker sound generation unit for generating a masker signal outputted as a masker sound for making the modulated signal harder to hear when transmitted with the modulated signal; and an acoustic signal generation unit for inserting the masker signal in the modulated signal to generate an acoustic signal.

Abstract: An action acquiring unit acquires action information corresponding to operation information from a first storage unit for a first user, and stores acquired action information in a third storage unit. A receiving unit receives, via a network, action information of a second user from an external device. A situation acquiring unit acquires, from a second storage, a communication situation corresponding to received action information. A writing unit writes the action information of the first user indicated by the communication situation to the third storage unit, additionally.

Abstract: Automatic level control of speech portions of an audio signal is provided. An audio signal is received in the form of a sequence of samples and may contain speech portion and non-speech portions. The sequence of samples is divided into a sequence of sub-frames. Multiple sub-frames adjacent to a present sub-frame are examined to determine a peak value of samples in the sub-frames. A gain factor is computed for the present sub-frame based on the peak value and a desired maximum value for said speech portion, and each sample in the present sub-frame is amplified by the gain factor. In an embodiment, variations in filtered energy values of multiple sub-frames enable determination of whether a sub-frame corresponds to a speech or non-speech/noise portion.

Abstract: A signal separating device that is inputted with signals formed by mixing plural signals and separates the signals into individual signals includes a signal converting unit that converts input signals into signals in the time-frequency domain and generates observation spectrograms and a signal separating unit that generates separated results from the observation spectrograms generated by the signal converting unit. The signal separating unit interprets the observation spectrograms as observation signals subjected to convolutive mixtures in the time-frequency domain and generates separated results by executing processing for solving convolutive mixtures in the time-frequency domain.

Abstract: A multi-modal human computer interface (HCI) receives a plurality of available information inputs concurrently, or serially, and employs a subset of the inputs to determine or infer user intent with respect to a communication or information goal. Received inputs are respectively parsed, and the parsed inputs are analyzed and optionally synthesized with respect to one or more of each other. In the event sufficient information is not available to determine user intent or goal, feedback can be provided to the user in order to facilitate clarifying, confirming, or augmenting the information inputs.

Abstract: The present invention relates to a method for encoding an audio signal. In a first embodiment a model relating to temporal masking of sound provided to a human ear is provided. A temporal masking index is determined in dependence upon a received audio signal and the model using a forward and a backward masking function. Using a psychoacoustic model a masking threshold is determined in dependence upon the temporal masking index. Finally, the audio signal is encoded in dependence upon the masking threshold. The method has been implemented using the MPEG-1 psychoacoustic model 2. Semiformal listening test showed that using the method for encoding an audio signal according to the present invention the subjective high quality of the decoded compressed sounds has been maintained while the bit rate was reduced by approximately 10%. In a second embodiment, the inharmonic structure of audio signals is modeled and incorporated into the MPEG-1 psychoacoustic model 2.

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: A method, system and a computer program product for an automated interpretation and translation are disclosed. An automated interpretation occurs by receiving language-based content from a user. The received language-based content is processed to interpret the received language-based content into a target language. Also, a presence of a cultural sensitivity in the received language-based content is detected. Further, an appropriate guidance for dealing with the detected cultural sensitivity is provided.

Abstract: A method of estimating noise in data containing voice information and noise includes receiving the data as a sequence of input values; transforming the data by applying a first non linear mapping to the input values wherein the derivative function of the mapping decreases in magnitude as the input values increase in magnitude smoothing the transformed data; and transforming the smoothed transformed data by applying a second non linear mapping that is opposite to the first non linear mapping, to determine an estimate of the noise in the inputted data.

Abstract: Provided is an audio encoding and decoding apparatus and method for improving a compression ratio while maintaining sound quality when sinusoidal waves of an audio signal are connected and encoded. The audio encoding method includes connecting sinusoidal waves of an input audio signal, converting a frequency of each of the connected sinusoidal waves to a psychoacoustic frequency, performing a first encoding operation for encoding the psychoacoustic frequency, performing a second encoding operation for encoding an amplitude of each of the connected sinusoidal waves, and outputting an encoded audio signal by mixing the encoding result of the first encoding operation and the encoding result of the second encoding operation.

Abstract: An audio encoding and decoding apparatus and a method thereof, capable of improving compression efficiency, by using coefficients that are stable over a period of time and in a range of frequency bands, are provided. The audio encoding method divides an input audio signal into frames having lengths different from each other; obtaining at least one magnitude in relation to each of the frames having different lengths; and encoding the magnitude. The audio decoding method separates at least one encoded magnitude in relation to each of frames having different lengths, based on the frame length; decoding each of the separated encoded magnitudes; and restoring an audio signal by using the decoded magnitude.

Abstract: A method, system, and computer program product for assisting individuals with vision impairment in their selection of items that are typically displayed in a list.

Abstract: A method of providing navigational information includes processing destination information spoken by a mobile processing system user. The processed voice information is transmitted to a remote center wirelessly. The processed voice information is voice recognition analyzed to recognize components of die destination information spoken. The remote center generates a list of hypothetical recognized components of the destination information listed by confidence levels. The list of hypothetical recognized components is displayed with confidence levels at the remote center for selective checking by a human data center operator, A component set is selected based on the confidence levels and accuracy of the selected set is confirmed, by interactive voice exchanges. A destination is determined from confirmed components of the destination information.

Abstract: A method of extracting a voice command produced in an enclosed or partially enclosed environment, includes providing an impulse response signal of the enclosed or partially enclosed environment; recording the voice command and ambient sounds; and using the impulse response signal to extract the recorded voice command.

Abstract: The present invention disclose modular speech-to-speech translation systems and methods that provide adaptable platforms to enable verbal communication between speakers of different languages within the context of specific domains. The components of the preferred embodiments of the present invention includes: (1) speech recognition; (2) machine translation; (3) N-best merging module; (4) verification; and (5) text-to-speech. Characteristics of the speech recognition module here are that the modules are structured to provide N-best selections and multi-stream processing, where multiple speech recognition engines may be active at any one time. The N-best lists from the one or more speech recognition engines may be handled either separately or collectively to improve both recognition and translation results. A merge module is responsible for integrating the N-best outputs of the translation engines along with confidence/translation scores to create a ranked list or recognition-translation pairs.

Abstract: When an unreceivable audio sampling frequency is transmitted from an audio data transmitting device or received at an audio data receiving device, frequency changing processing is executed inside an HDMI LSI of the transmitter side or the receiver side to change the unreceivable audio sampling frequency to a frequency that can be received at the audio data receiving device based on EDID information retained in the audio data receiving device, and mute processing of the audio information is executed to prevent generation of strange sounds.

Abstract: Sound signals from sound sources present in multiple directions are accepted as inputs of multiple channels, and signal of each channel is transformed into a signal on a frequency axis. A phase component of the transformed signal is calculated for each identical frequency, and phase difference between the multiple channels is calculated. An amplitude component of the transformed signal is calculated, and a noise component is estimated from the calculated amplitude component. An SN ratio for each frequency is calculated on the basis of the amplitude component and the estimated noise component, and frequencies at which the SN ratios are larger than a predetermined value are extracted. Difference between arrival distances is calculated on the basis of the phase difference at selected frequency, and the arrival direction in which it is estimated that the target sound source is present is calculated.

Abstract: A digital signal is processed by splitting it into at least two frequency subbands and the two subband signals are downsampled. A filter is applied in at least one of the subband signals. At least one of the phase and magnitude of the subband filtered signals is matched in the transition frequency band between the two subbands.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.