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 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 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 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 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: The invention relates to a method for determining voice activity in a signal section of an audio signal. The result, i.e., whether voice activity is present in the section of the signal thus observed, depends upon spectral and temporal stationarity of the signal section and/or prior signal sections. In a first step, the method determines whether there is spectral stationarity in the observed signal section. In a second step, the method determines whether there is temporal stationarity in the signal section in question. The final decision as to the presence of voice activity in the signal section observed depends upon the initial values of both steps.

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 method and system for identifying, saving and utilizing bookmarks in a voice-command platform. The system allows a user to bookmark objects specified within voice-markup filed resulting in the ability to directly recall the object rather than the voice-markup file as a whole. The system of the invention also provides a user of the voice command platform with a list of proposed bookmark names that are appropriate for the object. Once a user selects a bookmark, the platform may determine that a voice command is a bookmark command, such as a request to save a given voice command navigation point in a centralized list of bookmarks for the user, or to recall a navigation point from the user's centralized list, and the platform may respond to the bookmark command accordingly. The system improves accuracy in the use of bookmarks by proposing bookmark names for a given navigation point that avoid confusion with established grammars.

Abstract: A method for tracking pitch signal, including receiving a detected pitch signal that consists of a succession of pitch values, and for each current pitch value in the detected signal perform the following steps: constructing sub-sequences of consistent pitch values from neighboring pitch values. Next, calculating significance of the sub-sequences, and selecting a sub-sequence or a collection of consistent subsequences with highest significance. If the current pitch value is not consistent with the sub-sequence with highest significance, smoothing the current pitch value by diving it or multiplying it by an integer value>1, so as to render it consistent with the sub-sequence with highest significance.

Abstract: A method (300, 400) of and server (200) for providing assistance with control of a subscriber device is described. The method comprises receiving an instruction message (303) via a network that corresponds to spoken instructions from the subscriber device; converting the spoken instructions to control commands (309); providing a control message corresponding to the control commands (313); and sending the control message to the subscriber device (315), thereby assisting with the control of the subscriber device.

Abstract: With respect to audio signal coding and decoding apparatuses, there is provided a coding apparatus that enables a decoding apparatus to reproduce an audio signal even through it does not use all of data from the coding apparatus, and a decoding apparatus corresponding to the coding apparatus. A quantization unit constituting a coding apparatus includes a first sub-quantization unit comprising sub-quantization units for low-band, intermediate-band, and high-band; a second sub-quantization unit for quantizing quantization errors from the first sub-quantization unit; and a third sub-quantization unit for quantizing quantization errors which have been processed by the first sub-quantization unit and the second sub-quantization unit.

Abstract: The present invention provides a speech recognition apparatus having high speech recognition performance and capable of performing speech recognition in a highly efficient manner. A matching unit 14 calculates the scores of words selected by a preliminary word selector 13 and determines a candidate for a speech recognition result on the basis of the calculated scores. A control unit 11 produces word connection relationships among words included in a word series employed as a candidate for the speech recognition result and stores them into a word connection information storage unit 16. A reevaluation unit 15 corrects the word connection relationships one by one. On the basis of the corrected word connection relationships, the control unit 11 determines the speech recognition result. A word connection managing unit 21 limits times allowed for a boundary between words represented by the word connection relationships to be located thereat.

Abstract: A speaker authentication system includes a data fuser operable to fuse voiceprint match attempt results with additional information to assist in authenticating a speaker providing audio input. In other aspects, the system includes a data store of speaker voiceprints and a voiceprint matching module adapted to receive an audio input and operable to attempt to assist in authenticating a speaker by matching the audio input to at least one of the speaker voiceprints. The voiceprint matching module adjusts a confidence of voiceprint match attempt results by at least one of: (a) a number of utterance repetitions upon which a matching speaker voiceprint has been trained; or (b) a passage of time since a training occurrence associated with a matching speaker voiceprint.

Abstract: An efficient optimization algorithm is provided for multipulse speech coding systems. The efficient algorithm performs computations using the contribution of the non-zero pulses of the excitation function and not the zeroes of the excitation function. Accordingly, efficiency improvements of 87% to 99% are possible with the efficient optimization algorithm.

Abstract: A method and data structure are provided for efficiently storing asymmetric clustering models. The models are stored by storing a first level record for a word identifier and two second level records, one for a word identifier and one for a cluster identifier. An index to the second level word record and an index to the second level cluster record are stored in the first level record. Many of the records in the data structure include both cluster sub-model parameters and word sub-model parameters.

Abstract: A method of modeling speech distinctions within computer-animated talking heads that utilize the manipulation of speech production articulators for selected speech segments. Graphical representations of voice characteristics and speech production characteristics are generated in response to said speech segment. By way of example, breath images are generated such as particle-cloud images, and particle-stream images to represent the voiced characteristics such as the presence of stops and fricatives, respectively. The coloring on exterior portions of the talking head is displayed in response to selected voice characteristics such as nasality. The external physiology of the talking head is modulated, such as by changing the width and movement of the nose, the position of the eyebrows, and movement of the throat in response to the voiced speech characteristics such as pitch, nasality, and voicebox vibration, respectively.

Abstract: Systems and methods are provided for gathering audience measurement data relating to receipt of and/or exposure to audio data by an audience member. Audio data is monitored to detect a monitoring code. Based on detection of the monitoring code, a signature characterizing the audio data is extracted.