Abstract: The present invention carries out pre-selection on many LPC codevectors stored in an LSF codebook 101 using a weighted Euclidean distortion as a measure and carries out a full-code selection on the LPC codevectors left after the pre-selection using an amount of distortion in a spectral space as a measure. This makes it possible to improve the quantization performance of the LPC parameter vector quantizer and improve the quality of synthesized speech of the speech coder/decoder.

Abstract: An audio data input unit of an encoding device splits an audio data string into contiguous samples of audio data, and a transforming unit transforms the split audio data into spectral data in a frequency domain. A data dividing unit divides the spectral data into a lower frequency band and a higher frequency band at 11.025 kHz (f1) as a boundary. The spectral data in the lower frequency band is quantized and encoded by a first quantizing unit and an encoding unit. A second quantizing unit generates sub information indicating a characteristic of the spectral data in the higher frequency band, and a second encoding unit encodes the sub information. A stream output unit integrates the codes obtained by the first and second encoding units and outputs the integrated one. Here, f1 is a half or less of a sampling frequency f2 at which the audio data string is created.

Abstract: A processor-based system obtaining information about an event from schedule data, and using the information to assist speech recognition of speech occurring during at least a portion of the event.

Abstract: An overflow problem of LSF quantization in G.729 Annex B speech encoding which may lead to non-assignment of a codebook index. Preferred embodiments fix the problem with default or limited random variable assignments or flagging the overflow and adjusting the frame encoding such as by limiting spectral components or changing quantization targets.

Abstract: A method for determining a characteristic data set (“fingerprint”) for a sound signal, the sound signal itself is searched through for characteristic locations, and these characteristics locations are used for producing a characteristic data set. For this the frequency spectrum is evaluated over a time interval, subdivided into frequency bands, and averaged over each frequency band into a value. The fingerprint then consists of data that has been obtained from these values after possible further averagings, wherein only data is included which belongs to certain time segments.

Abstract: Acoustic noise for wireless or landline telephony is reduced through optimal filtering in which each frequency band of every time frame is filtered as a function of the estimated signal-to-noise ratio and the estimated total noise energy for the frame. Non-speech bands, non-speech frames and other special frames are further attenuated by one or more predetermined multiplier values. Noise in a transmitted signal formed of frames each formed of frequency bands is reduced. A respective total signal energy and a respective current estimate of the noise energy for at least one of the frequency bands is determined. A respective local signal-to-noise ratio for at least one of the frequency bands is determined as a function of the respective signal energy and the respective current estimate of the noise energy. A respective smoothed signal-to-noise ratio is determined from the respective local signal-to-noise ratio and another respective signal-to-noise ratio estimated for a previous frame.

Abstract: A method for calculating the amplication factor, which co-determines the volume, for a speech signal transmitted in encoded form includes dividing the speech signal into short temporal signal segments. The individual signal segments are encoded and transmitted separately from each other, and the amplication factor for each signal segment is calculated, transmitted and used by the decoder to reconstruct the signal. The amplication factor is determined by minimizing the value E(g_opt2)=(1?a)*f1(g_opt2)+a*f2(g_opt2), the weighting factor a being determined taking into account both the periodicity and the stationarity of the encoded speech signal.

Abstract: A process of identifying a speaker in coded speech data and a process of searching for the speaker are efficiently performed with fewer computations and with a smaller storage capacity. In an information search apparatus, an LSP decoding section extracts and decodes only LSP information from coded speech data which is read for each block. An LPC conversion section converts the LSP information into LPC information. A Cepstrum conversion section converts the obtained LPC information into an LPC Cepstrum which represents features of speech. A vector quantization section performs vector quantization on the LPC Cepstrum. A speaker identification section identifies a speaker on the basis of the result of the vector quantization. Furthermore, the identified speaker is compared with a search condition in a condition comparison section, and based on the result, the search result is output.

Abstract: When a voice encoding apparatus embeds any data in encoded voice code, the apparatus determines whether data embedding condition is satisfied using a first element code from among element codes constituting the encoded voice code, and a threshold value. If the data embedding condition is satisfied, the apparatus embeds optional data in the encoded voice code by replacing a second element code with the optional data. When a voice decoding apparatus extracts data that has been embedded in encoded voice code, the apparatus determines whether data embedding condition is satisfied using a first element code from among element codes constituting the encoded voice code, and a threshold value. If the data embedding condition is satisfied, the apparatus determines that optional data has been embedded in the second element code portion of the encoded voice code and extracts this embedded data.

Abstract: In a three-stage speech recognition process, a phoneme sequence is first assigned to a speech unit, then those vocabulary entries which are most similar to the phoneme sequence are sought in a selection vocabulary, and finally the speech unit is recognized using a speech unit recognizer which uses, as its vocabulary, the selected vocabulary entries which are most like the phoneme sequence.

Abstract: A source encoder encodes audio signals into increasing quality layers defined in bit planes. Each bit plane has a data unit that includes a beginning partition having one or more contiguous refinement bits, a second partition having one or more contiguous coded significance bits, a third partition having one or more contiguous sign boundary mark bits, and a fourth partition having one or more contiguous coded sign bits. A channel encoder encodes the bit planes into respective columns containing multiple rows. Unequal error protection coding is provided according to the quality of each layer such that each row has row and column channel protection codes for the respective row and column that correspond to the respective quality layer. For the corresponding row and column, each row contains the row channel protection codes and either the compressed audio data from the respective layer or the column channel protection codes.

Abstract: The present invention is a system and method that improves upon voice activity detection by packetizing actual noise signals, typically background noise. In accordance with the present invention an access network receives an input voice signal (including noise) and converts the input voice signal into a packetized voice signal. The packetized voice signal is transmitted via a network to an egress network. The egress network receives the packetized voice signal, converts the packetized voice signal into an output voice signal, and outputs the output voice signal. The egress network also extracts and stores noise packets from the received packetized voice signal and converts the packetized noise signal into an output noise signal. When the access network ceases to receive the input voice signal while the call is still ongoing, the access network instructs the egress network to continually output the output noise signal.

Abstract: A method and system are provided for acquiring information about communication among nodes [110, 210] in a network [100, 200] by intercepting chunks of data in the network by a tap [120, 220] located among the nodes [110, 210]. A file [740] of data, including characteristics [400] of the intercepted chunks may be produced. The data may be converted into at least one time series and processed to produce prosody information. The prosody information may be used by prosody analysis.

Abstract: A decoding device (30a) comprises a narrow-band decoding unit (31) operable to reproduce a PCM signal (P1) from a narrow-band bit stream included in a wide-band bit stream (S0), a wide-band decoding unit (32) operable to reproduce a PCM signal (P2) having a frequency band which is wider than that of the PCM signal (P1) reproduced by the narrow-band decoding unit (31) from the narrow-band bit stream and a band expanding bit stream included in the wide band bit stream (S0) and a selecting unit (34) operable to select either the PCM signal (P1) reproduced by the narrow-band decoding unit (31) or the PCM signal (P2) reproduced by the wide-band decoding unit (32), and to output the selected sound digital signal.

Abstract: A recognizing target vocabulary comparing unit calculates a compared likelihood of a recognizing target vocabulary, i.e., a compared likelihood of a registered vocabulary, by using the time series of the amount of characteristics of an input speech. An environment adaptive noise model comparing unit calculates a compared likelihood of a noise model adaptive to a noise environment, i.e., a compared likelihood of environmental noise. A rejection determining unit compares the likelihood of the registered vocabulary with the likelihood of the environmental noise, and determines whether or not the input speech is the noise. When it is determined that the input speech is the noise, a noise model adapting unit adaptively updates an environment adaptive noise model by using the input speech. Thus, the environment adaptive noise model matches to a real environment and the rejection determination can be performed for a noise input with high accuracy.

Abstract: A person can use a portable electronic device to electronically purchase or otherwise request a product, service or other deliverable related to audio programming to which the person is listening at the time they initiate the request. The request is fulfilled by a service that analyzes the audio content to identify the deliverable the person desires.

Abstract: A synthesis of automated speech recognition (voice to text) technology and a knowledge-based analysis of the concepts and contexts of the free text therefrom enable a directed-vocabulary look up index to be used in conjunction with the speech recognition technology thus enabling medical dictation to be transcribed in real time without elaborate training of the dictator or the speech recognition technology. Thus, caregivers can create and review Computer-Based Patient Records in the necessary timeframe consistent with good patient care. The Computer-Based Patient Records can be linked to other applications such as prescription cross checking, lab test results, payer regulations, etc.

Abstract: A speech processing unit assigns priority either to voice guidance processing or to speech recognition processing to be carried out previously, when a speech input requesting for the speech recognition processing is accepted while the voice guidance processing is being carried out. It can solve a problem of a conventional speech processing unit in that when a user operates a speech input button requesting for the speech recognition processing, the currently output voice guidance is interrupted, or the voice guidance scheduled to be output is not produced, thereby hindering the user from obtaining truly necessary information.

Abstract: A method of maintaining state information within a voice browser can include establishing a voice browser session with a client. The voice browser can be in communication with an application, wherein the voice browser and the application are remotely located from the client. State information, which corresponds to the voice browser session and application, can be received from the application. The state information can be stored as non-persistent data within the voice browser. The non-persistent data can be provided to the application to continue a transaction managed by the application.

Abstract: A communication system for sending a sequence of symbols on a communication link. The system includes a transmitter for placing information indicative of the sequence of symbols on the communication link and a receiver for receiving the information placed on the communication link by the transmitter. The transmitter includes a clock for defining successive frames, each of the frames including M time intervals, where M is an integer greater than 1. A modulator modulates each of M carrier signals with a signal related to the value of one of the symbols thereby generating a modulated carrier signal corresponding to each of the carrier signals. The modulated carriers are combined into a sum signal which is transmitted on the communication link. The carrier signals include first and second carriers, the first carrier having a different bandwidth than the second carrier.