Abstract: A method for estimating a pitch frequency of an audio signal includes computing a first transform of the signal to a frequency domain over a first time interval, and computing a second transform of the signal to the frequency domain over a second time interval, which contains the first time interval. A line spectrum of the signal is found, based on the first and second transforms, the spectrum including spectral lines having respective line amplitudes and line frequencies. A utility function that is periodic in the frequencies of the lines in the spectrum is then computed. This function is indicative, for each candidate pitch frequency in a given pitch frequency range, of a compatibility of the spectrum with the candidate pitch frequency. The pitch frequency of the speech signal is estimated responsive to the utility function.

Abstract: A frame erasure compensation method in a variable-rate speech coder includes quantizing, with a first encoder, a pitch lag value for a current frame and a first delta pitch lag value equal to the difference between the pitch lag value for the current frame and the pitch lag value for the previous frame. A second, predictive encoder quantizes only a second delta pitch lag value for the previous frame (equal to the difference between the pitch lag value for the previous frame and the pitch lag value for the frame prior to that frame). If the frame prior to the previous frame is processed as a frame erasure, the pitch lag value for the previous frame is obtained by subtracting the first delta pitch lag value from the pitch lag value for the current frame. The pitch lag value for the erasure frame is then obtained by subtracting the second delta pitch lag value from the pitch lag value for the previous frame.

Abstract: A portable audio recorder stores voice information files in the form of digital data. Header data is associated with each voice information file. The header data includes the name of the person who dictated the voice information file and the serial number of the recorder unit itself. This information may be used in connection with management of the voice information files in the recorder or after the files are transferred to another device such as a personal computer. The header data may also include the serial number of the recorder and a work facility location to which the file pertains. The user of the recorder may be permitted to select a format for the header data from among a plurality of pre-stored formats. Specific transaction data may be downloaded to the recorder, and used to build a header for a voice file, which is then uploaded with the header to another device.

Abstract: The invention relates to a method and apparatus for recognition processing of continuous words of a group which is structured by a plurality of words such that a recognition result of all of the words which structures the continuous words is effectively and accurately confirmed. All of the continuous words which have been input are recognition processed, the recognition result of all of the continuous words is output, a response from a speaker showing an affirmative/negative recognition result is input and recognition processed. If affirmative is determined, the recognition result at that time is confirmed for all of the continuous words. If negative is determined, for each word from a first to an nth (third in this case) which structures continuous words, the content showing affirmative/negative from the speaker is recognized, affirmative or negative is determined, and the recognition result at that time is confirmed as a recognition processing target word.

Abstract: An apparatus and method for data processing that improves estimation of spectral parameters of speech data and reduces algorithmic delay in a data coding operation. Estimation of spectral parameters is improved by adaptively adjusting a gain function used to enhance data based on whether the data contains information speech and noise or noise only. Delay is reduced by extracting coding parameters using incompletely processed data. This data is formed by multiplying a less current portion of an input data frame with a synthesis window and a more current portion of the data frame with an inverse analysis window, and performing an overlap-add process on the data frame and a similarly processed previous data frame.

Abstract: A bandwidth expanding method and apparatus in which frequency characteristics of high-frequency components of broad band signals can be adjusted to the liking of the user, overflow due to addition is prevented from occurring without power variations being perceived by a user, the number of broad band formants is reduced, and emphasis is attached to the rough structure of the spectrum, so that the produced broad band speech signals can be improved in quality. To this end, in a speech bandwidth expansion device, frequency characteristics of the frequency components not less than 3400 Hz are adjusted by preset alterable parameter values and summed to the original narrow band speech components. If overflow has occurred in a sample, the high-range gain of the sample is lowered to a level below the overflow level before proceeding to addition.

Abstract: An improved thermal design for passively cooled telecommunication repeater housings for use with wire transmission in the local loop outside plant is achieved by replacing the known convection based heat transfer designs with a design based on solid thermal conduction. A thermal chassis includes thermal collection, transfer and distribution members that collect the repeater modules' waste heat through respective thermal interfaces, transfer the waste heat along respective thermal conduction paths to the environmental enclosure, and then distribute the waste heat over a substantial portion of the enclosure's available surface area to form an enlarged thermal interface for convectively transferring the waste heat to the ambient air. Heat transfer is further improved by expanding the enclosure's external surface area and fabricating the distribution members so that they are in permanent and intimate thermal contact with the enclosure's expanded surface area.

Abstract: Home networks low-cost digital interfaces are introduced that integrate entertainment, communication and computing electronics into consumer multimedia. Normally, these are low-cost, easy to use systems, since they allow the user to remove or add any kind of network devices with the bus being active. To improve the user interface a speech unit (2) is proposed that enables all devices (11) connected to the bus system (31) to be controlled by a single speech recognition device. The properties of this device, e.g. the vocabulary can be dynamically and actively extended by the consumer devices (11) connected to the bus system (31). The proposed technology is independent from a specific bus standard, e.g. the IEEE 1394 standard, and is well-suited for all kinds of wired wireless home networks. The speech unit (2) receives data and messages from the device. The speech unit (2) recognizes speaker-dependent commands. A Speech synthesizer synthesizes messages.

Abstract: A signal processing unit is disclosed for selectively routing an unfiltered input signal and a noise reduced version of the unfiltered input signal to an output port in response to a noise power estimate. Routing the unfiltered input signal to the output port when the noise power estimate is less than a noise floor threshold avoids degrading the information content of an input signal having a power level close to the noise floor. A first attenuation factor and a second attenuation factor can be applied to the unfiltered input signal. A method is disclosed for parsing a signal into a plurality of frames, selecting a maximum value for each frame, and averaging the maximum values to form a noise floor threshold.

Abstract: An improved service access and heat transfer design for passively cooled telecommunication repeater housings for use with wire transmission in the local loop outside plant is achieved by a cover, sealable to the housing's sidewall, removable to provide field replaceable, plug-in access to the repeater modules and voltage protector assemblies wherein the voltage protector assemblies can be installed and removed without first removing the repeater modules protected by those voltage protector assemblies and by replacing the known convection based heat transfer designs with a design based on solid thermal conduction. Thermal sleeves, which mount the repeater modules, collect the repeater modules' waste heat through thermal interfaces, transfer the waste heat along thermal conduction paths to the housing's sidewalls, and then distribute the waste heat over a substantial portion of the housing's sidewalls.

Abstract: Formants, corresponding to input speech units based either on a known text or the results of a speech recognition procedure, are generated from a formant synthesizer. A frequency response is generated based on the synthesized formants. A second frequency response is generated based on a speech signal which is received and which corresponds to utterances of speech units. The synthesized formants are modified based on a comparison of the frequency response corresponding to the synthesized formants and specific proportional characteristics of a frequency response of the input speech signal. In one illustrative embodiment, the comparison is then recalculated and further modifications are made accordingly to improve accuracy. In one illustrative embodiment, time aligning and frequency warping are utilized as modification functions.

Abstract: The present invention is embodied in a system and method for performing spectral analysis of a digital signal having a discrete duration by spectrally decomposing the digital signal at predefined frequencies uniformly distributed over a sampling frequency interval into complex frequency coefficients so that magnitude and phase information at each frequency is immediately available to produce a modulated complex lapped transform (MCLT). The present invention includes a MCLT processor, an acoustic echo cancellation device and a noise reducer integrated with an encoder/decoder device.

Abstract: An apparatus and method for speech compression includes dividing the speech spectrum into a plurality of frames, assigning frame classifications to the plurality of frames, and determining the speech modeling parameters based on the assigned frame classification. The voiced part of the speech spectrum and the unvoiced part of the speech spectrum are synthesized separately using an Analysis by Synthesis allowing a correct correspondence between voiced and unvoiced parts of the reconstructed signal. Particularly, a frequency response of a special simulated signal based on the previous and current frames is used as an approximating function. The simulated signal is synthesized at the encoder side in the way it will be generated at the decoder side. Also, a better of two encoding methods is selected to encode the spectral magnitudes. A wavelet encoder and an inter-frame predictive encoder illustrate the invention's efficient, yet accurate reconstruction of synthesized digital speech.

Abstract: A system controller (106) includes a speech encoder (107) that encodes a low bit rate digital voice message. The speech encoder sets values of words of a header of the encoded message. The values of the words define a quantity of frames in the voice message, N, and define a vocoder rate used for the encoded message. The speech encoder sets a state of each indicator in each frame status field of N frame status fields that are transmitted after the header of the encoded message. The speech encoder assembles N frame data fields, wherein each of the frame data fields comprises a set of data words. The N frame data fields follow the N frame status fields. Each set of data words conforms to at least one of the vocoder rate and the states of the indicators. A decoder (3310) decodes the encoded low bit rate digital message.

Abstract: In order to achieve low error rates in a speech recognition system, for example, in a system employing rank-based decoding, we discriminate the most confusable incorrect leaves from the correct leaf by lowering their ranks. That is, we increase the likelihood of the correct leaf of a frame, while decreasing the likelihoods of the confusable leaves. In order to do this, we use the auxiliary information from the prediction of the neighboring frames to augment the likelihood computation of the current frame. We then use the residual errors in the predictions of neighboring frames to discriminate between the correct (best) and incorrect leaves of a given frame. We present a new methodology that incorporates prediction error likelihoods into the overall likelihood computation to improve the rank position of the correct leaf.

Abstract: A half duplex switching device includes an input connection for receiving an input audio signal, and classification module coupled to the input connection. The classification module provides an output which indicates a classification of the input signal based upon a density of the input audio signal, an energy level of the input audio signal, and classification data provided with the input audio signal. A switching device is coupled to the classification module and determines if the received input audio signal contains speech signals based upon the output of the classification module. The communication receiving device can be used in both communication systems which provide continuous speech signals, and communication systems which remove silence and only provide speech signals.

Abstract: A system and method for IP-based telephone communication utilizing speech-generated text is disclosed. An embodiment of the present invention includes an interface to the Internet for sending and receiving voice and video signals. In addition, the embodiment generates text signals corresponding to voice signals generated by the user. A transmission signal is generated from the text signals and the voice signals for transmission over the Internet. Further, the embodiment of the present invention includes an application which is capable of receiving video and/or speech-generated data transmitted by another device, and concurrently displaying the speech-generated data and video to a user. The speech-generated data may be converted to an audio signal and applied to a speaker concurrently with the video and speech-generated data. In this way, a user is capable of easily communicating speech-generated information to another user during periods of voice signal loss.

Abstract: A system and method for speech recognition includes a speaker-independent set of stored word representations derived from speech of many users deemed to be typical speakers and for use by all users, and may further include speaker-dependent sets of stored word representations specific to each user. Utterances from a user which match stored words in either set according to the ordering rules are reported as words.

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

Abstract: A balanced spectrum limiter for limiting the frequency spectrum of signals transmitted through a telephone or communication system includes one or more filters and an energy surge protection circuit. Each filter includes a pair of inductors wound around a single core and a capacitor connected between the two inductors. The energy surge protection circuit can be any type of energy surge protection circuit known in the art. The range of frequencies which can be transmitted is determined by the filter. The filter is removably connected to the energy surge protection circuit so that filters of varying parameters may be easily interchanged.