A novel method, free flow acoustophoresis (FFA), capable of continuous separation of mixed particle suspensions into multiple outlet fractions is presented. Acoustic forces are utilized to separate particles based on their size and density. The method is shown to be suitable for both biological and nonbiological suspended particles. The microfluidic separation chips were fabricated using conventional microfabrication methods. Particle separation was accomplished by combining laminar flow with the axial acoustic primary radiation force in an ultrasonic standing wave field. Dissimilar suspended particles flowing through the 350-μm-wide channel were thereby laterally translated to different regions of the laminar flow profile, which was split into multiple outlets for continuous fraction collection. Using four outlets, a mixture of 2-, 5-, 8-, and 10-μm polystyrene particles was separated with between 62 and 94% of each particle size ending up in separate fractions. Using three outlets and three particle siz...

Free flow acoustophoresis: Microfluidic-based mode of particle and cell separation

A method and apparatus for determining the location of a communication unit in a CDMA system includes in a first embodiment, sending a location request via a spread spectrum signal to the subscriber (140), and receiving in return a subscriber signal including a response message showing a receive time of a particular symbol of the base's spreading sequence and a transmit time of a particular symbol of the subscriber's spreading sequence. The base (130), along with other receiving base(s) (140), also receives a predetermined symbol of the subscriber spreading sequence, and each determines a respective receive time of the predetermined symbol. The received information is then processed, along with known base location and delay information, to determine the subscriber location. If insufficient number of bases are capable of communicating with the subscriber, for example due to high loading/interference, auxiliairy bases (121) are also provided for receiving from or transmitting to the subscriber.

Method and apparatus for location finding in a cdma system

An apparatus and method for the robust recognition of speech during a call in a noisy environment is presented. Specific background noise models are created to model various background noises which may interfere in the error free recognition of speech. These background noise models are then used to determine which noise characteristics a particular call has. Once a determination has been made of the background noise in any given call, speech recognition is carried out using the appropriate background noise model.

Selective noise/channel/coding models and recognizers for automatic speech recognition

A method of identifying a relationship path from a user or from a start entity to a target entity by establishing a computer platform including an algorithm for determining the shortest path; establishing a central database of combined public and private information comprising a plurality of items of contact information; assigning a unique, relational identifier to each item of contact information in said central database; obtaining at least one client database of contact information comprising a plurality of items of contact information; assigning a unique, relational identifier to each item of contact information in said client database; loading said client database into said central database; integrating said central database with said client database; providing a user interface to said platform for a user to enter start or user entities and a target entity; determining whether relationship paths exist to said target entity; and identifying said relationship paths.

System and method for identifying relationship paths to a target entity

A method and system is disclosed for non-parametric speech conversion. A text-to-speech (TTS) synthesis system may include hidden Markov model (HMM) HMM based speech modeling for both synthesizing output speech. A converted HMM may be initially set to a source HMM trained with a voice of a source speaker. A parametric representation of speech may be extract from speech of a target speaker to generate a set of target-speaker vectors. A matching procedure, carried out under a transform that compensates for speaker differences, may be used to match each HMM state of the source HMM to a target-speaker vector. The HMM states of the converted HMM may be replaced with the matched target-speaker vectors. Transforms may be applied to further adapt the converted HMM to the voice of target speaker. The converted HMM may be used to synthesize speech with voice characteristics of the target speaker.

Method and system for non-parametric voice conversion

A Vector-Sum Excited Linear Predictive Coding (VSELP) speech coder provides improved quality and reduced complexity over a typical speech coder. VSELP uses a codebook which has a predefined structure such that the computations required for the codebook search process can be significantly reduced. This VSELP speech coder uses single or multi-segment vector quantizer of the reflection coefficients based on a Fixed-Point-Lattice-Technique (FLAT). Additionally, this speech coder uses a pre-quantizer to reduce the vector codebook search complexity and a high-resolution scalar quantizer to reduce the amount of memory needed to store the reflection coefficient vector codebooks. Resulting in a high quality speech coder with reduced computations and storage requirements.

Multi-segment vector quantizer for a speech coder suitable for use in a radiotelephone

In a statistical based speech recognition system, one of the key issues is the selection of the Hidden Markov Model that best matches a given sequence of feature observations. The problem is usually addressed by the calculation of the maximum likelihood, ML, state sequence by means of a Viterbi or other decoder. Noise or inadequate training can produce a ML sequence associated with a Hidden Markov Model other than the correct model. The method of the present invention provides improved robustness by combining the standard ML state sequence score (416) with an additional path score (418) derived from the dynamics of the ML score as a function of time. These two scores, when combined, form a hybrid metric (420) that, when used with the decoder, optimizes selection of the correct Hidden Markov Model (422).

Method, apparatus, and radio optimizing Hidden Markov Model speech recognition

Analysis by synthesis calculates a difference by subtracting (130) synthesized speech from input speech. The synthesized speech is formed by exciting long and short term filters (124,126) with excitation vectors from a codebook store (114) which is searched by codebook generation (120). A weighting filter (132) is applied to the difference signal and the weighted difference is used to calculate an energy measure (134) which is used to control the codebook search (140). The weighting filter is an Rth-order filter controlled with calculated coefficients. The method for calculating coefficients models the frequency response of L Pth-order filters by a single Rth-order filter, where the order R ~ LxP. This method increases the control of a speech coder filter without an increase in hardware complexity.

Method for generating a spectral noise weighting filter for use in a speech coder

An improved speech coder provides a more natural sounding replication of speech by modifying the mean-squared error criterion for the selected speech coder parameters. Specifically, the modification emphasizes the signal components that the speech coder has difficulty matching, i.e. the high frequencies. This emphasis is constrained to certain limitations to avoid over-emphasizing the speech.

Speech coding method and apparatus using mean squared error modifier for selected speech coder parameters using VSELP techniques

An input speech signal is encoded as one or more reflection coefficients. To reduce storage requirements, the reflection coefficients are scalar quantized by storing an N-bit code rather than the entire reflection coefficient. An exemplary value for N is 8. A table is provided having 2N reflection coefficient values. The N-bit code is used to look up reflection coefficient values from the table. To reduce spectral distortion due to scalar quantization, the reflection coefficient values in the table are non-linearly scaled.

Matthew A Hartman

Papers

Multi-segment vector quantizer for a speech coder suitable for use in a radiotelephone

Method, apparatus, and radio optimizing Hidden Markov Model speech recognition

Method for generating a spectral noise weighting filter for use in a speech coder

Speech coding method and apparatus using mean squared error modifier for selected speech coder parameters using VSELP techniques

Method of storing reflection coeffients in a vector quantizer for a speech coder to provide reduced storage requirements