Abstract: A high-speed search method in a speech encoder using an order character of LSP (Line Spectrum Pair) parameters in an LSP parameter quantizer using SVQ (Split Vector Quantization) used in a low-speed transmission speech encoder, includes the steps of rearranging a codebook according to an element value of a reference row for determining a range of code vectors to be searched; and determining a search range by using an order character between a given target vector and an arranged code vector to obtain an optimal code vector. The method gives effects of reducing computational complexity required to search the codebook without signal distortion in quantizing the LSP parameters of the speech encoder using SVQ, and reducing computational complexity without loss of tone quality in G.729 fixed codebook search by performing candidate selection and search on the basis of the correlation value size of the pulse position index.

Abstract: A system and method for separating a mixture of audio signal into desired audio signals (430) (e.g., speech) and a noise sign (440) is disclosed. Microphones (310, 320) are positioned to receive the mixed audio signals, and an independent component analysis (ICA) processes (212) the sound mixture using stability constraints. The ICA process (508) uses predefined characteristics of the desired speech signal to identify and isolate a target sound signal (430). Filter coefficients are adapted with a learning rule and filter weight update dynamics are stabilized to assist convergence to a stable separated ICA signal result. The separated signals may be peripherally-processed to further reduce noise effects using post-processing (214) and pre-processing (220, 230) techniques and information. The proposed system is designed and easily adaptable for implementation on DSP units or CPUs in audio communication hardware environments.

Abstract: The present invention provides a process for separating a good quality information signal from a noisy acoustic environment. The separation process uses a set of at least two spaced-apart transducers to capture noise and information components. The transducer signals, which have both a noise and information component, are received into a separation process. The separation process generates one channel that is substantially only noise, and another channel that is a combination of noise and information. An identification process is used to identify which channel has the information component. The noise signal is then used to set process characteristics that are applied to the combination signal to efficiently reduce or eliminate the noise component. In this way, the noise is effectively removed from the combination signal to generate a good qualify information signal. The information signal may be, for example, a speech signal, a seismic signal, a sonar signal, or other acoustic signal.

Abstract: An apparatus and method for mapping CELP parameters between a source codec and a destination codec. The apparatus includes an LSP mapping module, an adaptive codebook mapping module coupled to the LSP mapping module, and a fixed codebook mapping module coupled to the LSP mapping module and the adaptive codebook mapping module. The LSP mapping module includes an LP overflow module and an LSP parameter modification module. The adaptive codebook mapping module includes a first pitch gain codebook. The fixed codebook mapping module includes a first target processing module, a pulse search module, a fixed codebook gain estimation module, a pulse position searching module.

Abstract: A privacy apparatus adds a privacy sound into the environment, thereby confusing listeners as to which of the sounds is the real source. The privacy sound may be based on the speaker's own voice or may be based on another voice. At least one characteristic of the speaker (such as a characteristic of the speaker's speech) may be identified. The characteristic may then be used to access a database of the speaker's own voice or another's voice, and to form one or more voice streams to form the privacy sound. The privacy sound may thus permit disruption of the ability to understand the source speech of the user by eliminating segregation cues that the auditory system uses to interpret speech.

Abstract: A household appliance with a simple to use selected operating language includes activating the programs by taking into account the special characteristics of specific languages. The household appliance includes a display unit that displays all information in a simple language. The selected language, the selected programming of the household appliance, and the selected additional functions in addition to error messages are shown on the display unit so that operational mistakes are basically prevented.

Abstract: An audio encoder regulates quality and bitrate with a control strategy. The strategy includes several features. First, an encoder regulates quantization using quality, minimum bit count, and maximum bit count parameters. Second, an encoder regulates quantization using a noise measure that indicates reliability of a complexity measure. Third, an encoder normalizes a control parameter value according to block size for a variable-size block. Fourth, an encoder uses a bit-count control loop de-linked from a quality control loop. Fifth, an encoder addresses non-monotonicity of quality measurement as a function of quantization level when selecting a quantization level. Sixth, an encoder uses particular interpolation rules to find a quantization level in a quality or bit-count control loop. Seventh, an encoder filters a control parameter value to smooth quality. Eighth, an encoder corrects model bias by adjusting a control parameter value in view of current buffer fullness.

Abstract: The present invention provides a means for an end user to customize a speech-based user interface, such as those used in status and control for such applications as messaging, unified communications, automobile control, and many others. The invention consists of two major components: a graphical configuration component, and a run-time component. The graphical configuration component allows a user to quickly and efficiently: specify the information they wish to be included in spoken summaries; restructure command menus or vocabulary structures to meet their personal requirements, moving commonly-used commands to the “top” of a hierarchy or vocabulary structure, and rarely-used commands to the bottom layers; create additional commands to access information relating to functions external to the system, e.g. access data from the web, or control another device or system, via the web; and save the customization for use by the run-time program.

Abstract: Methods and devices for dynamically adjusting a multi-band signal-modification profile based on a psychoacoustic model are disclosed. In one arrangement, the encoding parameter side information is used to estimate encoding noise of an encoded signal. The signal spectrum of the signal is estimated. Adjustments to the multi-band signal-modification profile are determined using the estimated noise and signal spectrum and a psychoacoustic profile.

Abstract: A method, apparatus and system that receives speech commands at a remote control device microphone, digitizes those input speech commands, compresses the digitized speech commands, multiplexes control commands with the compressed digitized speech commands, and transmits the compressed digitized speech commands to an electronic device, such as a digital home communication terminal (DCHT). The electronic device decompresses and interprets the speech commands to allow the remote control operator to control the electronic device. Because speech recognition is performed at the electronic device, rather than at the remote control device, the remote control does not have to interpret and transmit infrared signals that represent user commands. This simplifies the processing and voice recognition capabilities required by the remote control.

Abstract: A speech reference enrollment method involves requesting a user speak a word; detecting a first utterance; requesting the user speak the word; detecting a second utterance; determining a first similarity between the first utterance and the second utterance; when the first similarity is less than a predetermined similarity, requesting the user speak the word; detecting a third utterance; determining a second similarity between the first utterance and the third utterance; and when the second similarity is greater than or equal to the predetermined similarity, creating a reference.

Abstract: A system and method are disclosed for processing an audio signal including separating the audio signal into a plurality of streams which group sounds from a same source prior to classification and analyzing each separate stream to determine phoneme-level classification. One or more words of the audio signal may then be outputted.

Abstract: A method and apparatus is provided for generating speech that sounds more natural. In one embodiment, word prominence and latent semantic analysis are used to generate more natural sounding speech. A method for generating speech that sounds more natural may comprise generating synthesized speech having certain word prominence characteristics and applying a semantically-driven word prominence assignment model to specify word prominence consistent with the way humans assign word prominence. A speech representative of a current sentence is generated. The determination is made whether information in the current sentence is new or previously given in accordance with a semantic relationship between the current sentence and a number of preceding sentences. A word prominence is assigned to a word in the current sentence in accordance with the information determination.

Abstract: An adaptive variable bit rate audio encoder and method that examines audio level information and detects various information in the audio data in a psychoacoustic model to create a quantization value and assign a mode tag to a single frame of audio. A bit rate is assigned according to one of three modes, a self-adaptive mode which is free-running and takes direction only from the characteristics of the incoming audio signal, a managed mode which is controlled by rules set from a statistical multiplexer, and a combination of self-adaptive and managed in which control rules from the statistical multiplexer act to maintain limits on the self-adaptive mode.

Abstract: A method for objective speech quality assessment that accounts for phonetic contents, speaking styles or individual speaker differences by distorting speech signals under speech quality assessment. By using a distorted version of a speech signal, it is possible to compensate for different phonetic contents, different individual speakers and different speaking styles when assessing speech quality. The amount of degradation in the objective speech quality assessment by distorting the speech signal is maintained similarly for different speech signals, especially when the amount of distortion of the distorted version of speech signal is severe. Objective speech quality assessment for the distorted speech signal and the original undistorted speech signal are compared to obtain a speech quality assessment compensated for utterance dependent articulation.

Abstract: A scalable audio codec processes, quantizes and encodes audio signals into an embedded audio bitstream of bit-planes each having a data unit. The data unit has a beginning refinement bits partition, a second significance bits partition, a third sign boundary mark bits partition, and a fourth sign bits partition. The second and fourth partitions form a boundary for the third partition. The quantizing uses a variable length coding algorithm. The third partition is an invalid codeword for a predetermined encoding method being used to encode. The codec uses a decoder to decode the embedded audio bitstream of bit-planes using Reversible exponential Golomb (Exp-Golomb) codes in a Reversible Variable Length Code (RVLC) algorithm to produce quantized data of weighted subbands. An inverse quantizer dequantizes the quantized data into audio signals.

Abstract: Disclosed is an objective speech quality assessment technique that reflects the impact of distortions which can dominate overall speech quality assessment by modeling the impact of such distortions on subjective speech quality assessment, thereby, accounting for language effects in objective speech quality assessment.

Abstract: Roughly described, a natural language interpretation system that provides commands to a back-end application in response to user input is modified to separate out user interaction functions into a user interaction subsystem. The user interaction subsystem can include an interaction block that is specific to each particular I/O agency, and which converts user input received from that agency into an agency-independent form for providing to the natural language interpretation system. The user interaction subsystem also can take results from the back-end application and clarification requests and other dialoguing from the natural language interpretation system, both in device-independent form, and convert them for forwarding to the particular I/O agency.

Abstract: Method and apparatus detect voice activity for spectrum or power efficiency purposes. The method determines and tracks the instant, minimum and maximum power levels of the input signal. The method selects a first range of signals to be considered as noise, and a second range of signals to be considered as voice. The method uses the selected voice, noise and power levels to calculate a log likelihood ratio (LLR). The method uses the LLR to determine a threshold, then uses the threshold for differentiating between noise and voice.

Abstract: An audio encoder regulates quality and bitrate with a control strategy. The strategy includes several features. First, an encoder regulates quantization using quality, minimum bit count, and maximum bit count parameters. Second, an encoder regulates quantization using a noise measure that indicates reliability of a complexity measure. Third, an encoder normalizes a control parameter value according to block size for a variable-size block. Fourth, an encoder uses a bit-count control loop de-linked from a quality control loop. Fifth, an encoder addresses non-monotonicity of quality measurement as a function of quantization level when selecting a quantization level. Sixth, an encoder uses particular interpolation rules to find a quantization level in a quality or bit-count control loop. Seventh, an encoder filters a control parameter value to smooth quality. Eighth, an encoder corrects model bias by adjusting a control parameter value in view of current buffer fullness.