In one aspect of the present invention, a network gateway is configured to facilitate on line and off line bi-directional communication between a number of near end data and telephony devices with far end data termination devices via a hybrid fiber coaxial network and a cable modem termination system. The described network gateway combines a QAM receiver, a transmitter, a DOCSIS MAC, a CPU, a voice and audio processor, an Ethernet MAC, and a USB controller to provide high performance and robust operation.

Gateway with voice

This invention is a method for removing motion artifacts from devices for sensing bodily parameters, an apparatus, and system for effecting same. The method includes analyzing segments of measured data (100) representing bodily parameters, and possibly noise from motion artifacts. Each segment of measured data may correspond to a single light signal transmitted, and detected after transmission or reflection through bodily tissue. Each data segment is frequency analyzed (110) to determine dominant frequency components. The frequency component which represents at least on bodily parameter of interest (120) is selected for further processing. The segment of data is subdivided into subsegments, each subsegment representing one heartbeat. The subsegments are used to calculate a modified average pulse as a candidate output pulse (130). The candidate output pulse is analyzed to determine whether it is a valid bodily parameter and, if yes, it is output for use in calculating the at least one bodily parameter of interest without any substantial noise degradation. The above method may be applied to red, and infrared pulse oximetry signals prior to calculating pulsatile blood oxygen concentration.

Method, apparatus and system for removing motion artifacts from measurements of bodily parameters

A method and system use an alternative sensor signal received from a sensor other than an air conduction microphone to estimate a clean speech value. The estimation uses either the alternative sensor signal alone, or in conjunction with the air conduction microphone signal. The clean speech value is estimated without using a model trained from noisy training data collected from an air conduction microphone. Under one embodiment, correction vectors are added to a vector formed from the alternative sensor signal in order to form a filter, which is applied to the air conductive microphone signal to produce the clean speech estimate. In other embodiments, the pitch of a speech signal is determined from the alternative sensor signal and is used to decompose an air conduction microphone signal. The decomposed signal is then used to determine a clean signal estimate.

Method and apparatus for multi-sensory speech enhancement

A mobile device includes an air conduction microphone and an alternative sensor that provides an alternative sensor signal indicative of speech. A communication interface permits the mobile device to communicate directly with other mobile devices.

Method and apparatus for multi-sensory speech enhancement on a mobile device

A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.

Voice and Data Exchange Over a Packet Based Network

A method for encoding a speech signal into digital bits including the steps of dividing the speech signal into speech frames representing time intervals of the speech signal, determining voicing information for frequency bands of the speech frames, and determining spectral magnitudes representative of the magnitudes of the spectrum at determined frequencies across the frequency bands. The method further includes quantizing and encoding the spectral magnitudes and the voicing information. The steps of determining, quantizing and encoding the spectral magnitudes is done is such a manner that the spectral magnitudes independent of voicing information are available for later synthesizing.

Spectral magnitude representation for multi-band excitation speech coders

A method for decoding and synthesizing a synthetic digital speech signal from digital bits of the type produced by dividing a speech signal into frames and encoding the speech signal by an MBE based encoder. The method includes the steps of decoding the bits to provide spectral envelope and voicing information for each of the frames, processing the spectral envelope information to determine regenerated spectral phase information for each of the frames based on local envelope smoothness determining from the voicing information whether frequency bands for a particular frame are voiced or unvoiced. The method further includes synthesizing speech components for voiced frequency bands using the regenerated spectral phase information, synthesizing a speech component representing the speech signal in at least one unvoiced frequency band, and synthesizing the speech signal by combining the synthesized speech components for voiced and unvoiced frequency bands.

Synthesis of MBE-based coded speech using regenerated phase information

A method of encoding speech by analyzing a digitized speech signal to determine excitation parameters for the digitized speech signal is disclosed. The method includes dividing the digitized speech signal into at least two frequency bands, determining a first preliminary excitation parameter by performing a nonlinear operation on at least one of the frequency band signals to produce a modified frequency band signal and determining the first preliminary excitation parameter using the modified frequency band signal, determining a second preliminary excitation parameter using a method different from the first method, and using the first and second preliminary excitation parameters to determine an excitation parameter for the digitized speech signal. The method is useful in encoding speech. Speech synthesized using the parameters estimated based on the invention generates high quality speech at various bit rates useful for applications such as satellite voice communication.

Estimation of excitation parameters

An improved speech model and methods for estimating the model parameters, synthesizing speech from the parameters, and quantizing the parameters are disclosed. The improved speech model allows a time and frequency dependent mixture of quasi-periodic, noise-like, and pulse-like signals. For pulsed parameter estimation, an error criterion with reduced sensitivity to time shifts is used to reduce computation and improve performance. Pulsed parameter estimation performance is further improved using the estimated voiced strength parameter to reduce the weighting of frequency bands which are strongly voiced when estimating the pulsed parameters. The voiced, unvoiced, and pulsed strength parameters are quantized using a weighted vector quantization method using a novel error criterion for obtaining high quality quantization. The fundamental frequency and pulse position parameters are efficiently quantized based on the quantized strength parameters. These methods are useful for high quality speech coding and reproduction at various bit rates for applications such as satellite voice communication.

Speech model and analysis, synthesis, and quantization methods

An audio watermarking system conveys information using an audio channel by modulating an audio signal to produce a modulated signal by embedding additional information into the audio signal. Modulating the audio signal includes segmenting the audio signal into overlapping time segments using a non-rectangular analysis window function produce a windowed audio signal, processing the windowed audio signal for a time segment to produce frequency coefficients representing the windowed time segment and having phase values and magnitude values, selecting one or more of the frequency coefficients, modifying phase values of the selected frequency coefficients using the additional information to map the phase values onto a known phase constellation, and processing the frequency coefficients including the modified phase values to produce the modulated signal.

Daniel W. Griffin

Papers

Spectral magnitude representation for multi-band excitation speech coders

Synthesis of MBE-based coded speech using regenerated phase information

Estimation of excitation parameters

Speech model and analysis, synthesis, and quantization methods

Audio Watermarking via Phase Modification