Motorola Corporation

Continental Automotive Systems Corporation

Freescale Semiconductor,inc.

General Dynamics Decision Systems, Inc.

David L Barron

William Chunhung Yip

Suat Yeldener

Sean S You

James Brian Piket

James A Stephens

Daniel S Rokusek

Jianming Song

Jonathan Michael Lasch

Edward Srenger

Kyle K Iwai

Andrew W Hardell

Andrew Kirby

Eduardo M Molieri

Susan D Stephens

Paul Lopez Kennedy

A speech signal is received at an input. At least one electrical value associated with the received speech signal is tracked. A dynamic adjustment of the speech signal is determined. The dynamic adjustment is selected at least in part so as to minimize a distortion and minimize an over-amplification of the speech signal based at least in part upon an analysis of the at least one electrical value. The dynamic adjustment is further selected to obtain a desired output signal characteristic for the speech signal presented at an output. The dynamic adjustment value is applied to the speech signal and the adjusted speech signal is presented at the output. The gain of the signal can also be limited to prevent over-amplification.

Apparatus and method for automatic gain control

Methods and systems for optimizing and audio communication system, including detecting a surrounding noise profile, determining one or more states of one or more noise-related conditions corresponding to the surrounding noise profile, and associating the one or more states of the one or more noise-related conditions to the surrounding noise profile.

Systems and methods for optimizing an audio communication system

A continuous stream of noise is created from a plurality of input signals. A smoothing spectrum estimate is continuously calculated from the continuous stream of noise. Noise is responsively removed from a selected one of the plurality of input signals using the smoothing spectrum estimate. The removal of the noise from the selected input signal is performed substantially synchronously and in time alignment with the creating of the continuous stream of noise and the calculating of the smoothing spectrum estimate.

Apparatus and method for noise removal by spectral smoothing

Residual frequency components of a reference signal are suppressed from an error signal. A magnitude of the frequency domain representation of the reference signal is divided by a magnitude of the frequency domain representation of LMS-filtered representation of the error signal to obtain a frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal. The frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal is multiplied by the frequency domain ratio of the frequency domain representation of the reference signal to the frequency domain representation of the LMS-filtered signal, to obtain a frequency domain signal having reduced residual frequency components of the reference signal.

Method and apparatus for a frequency domain echo suppression filter

Regardless of the presence of echo or uplink speech, a microphone signal is received at a small filter and this small filter is functionally and/or physically separate from an adaptive echo canceller filter. The signal is applied to the small filter and an error signal is determined from the signal utilizing the small filter. The small filter continuously adapts the received signals. A first adaptation factor is determined based at least upon the error signal and the microphone signal according to a first signal analysis approach and a second adaptation factor is determined based at least upon the microphone signal according to a second signal analysis approach. The first adaption factor is compared to the second adaptation factor and one of the first adaptation factor or the second adaptation factor is selected based upon at least one predetermined criteria. The selected adaptation factor is applied to the echo canceller filer to control the convergence of the echo canceller filter.

Apparatus and method for convergence control

Methods and systems for selectively switching between microphones in a plurality of microphones are disclosed, including providing a first state that corresponds to one or more microphones selected from the plurality of microphones. A subset of microphones may be selected from the plurality of microphones in response to determining an average power of an input signal for each of the plurality of microphones. A second state may be identified that includes at least one of the subset of microphones in response to evaluating the average powers of the input signals for the subset of microphones against a predetermined condition. A transition from the first state to the second state may be delayed in response to determining a transition delay time corresponding to the first state.

Systems and methods for selectively switching between multiple microphones

A method for reducing a computational complexity of an m-stage adaptive filter is provided by updating recursively forward and backward error prediction square terms for a first portion of a length of the adaptive filter, and keeping the updated forward and backward error prediction square terms constant for a second portion of the length of the adaptive filter.

Reduced complexity recursive least square lattice structure adaptive filter by means of limited recursion of the backward and forward error prediction squares

A method for reducing a computational complexity of an m-stage adaptive filter is provided by expanding a weighted sum of forward prediction error squares into a corresponding binomial expansion series, expanding a weighted sum of backward prediction error squares into a corresponding binomial expansion series, and determining coefficient updates of the adaptive filter with the weighted sums of forward and backward prediction error squares approximated by a select number of terms of their corresponding binomial expansion series.

Reduced complexity recursive least square lattice structure adaptive filter by means of estimating the backward and forward error prediction squares using binomial expansion

A comfort noise generator () suitable for use in a communication system includes a finite impulse response (FIR) filter (), a random number generator (), and a coefficient updater (). The coefficient updater () determines an updated set of filter coefficients () based on the signal frame of the input signal (). The updated set of filter coefficients () is output to the FIR filter (). The FIR filter () shapes a white noise signal () supplied by the random number generator () to provide a simulated background noise signal, or comfort noise signal (). The comfort noise signal () is selectively output from an echo suppression system or corresponding method to overwrite or suppress reflected residual echoes.

Low-complexity comfort noise generator

Methods and apparatus are provided for an echo cancellation system. The echo cancellation system comprises an automatic gain control (AGC), a first scalar, a second scalar, a signal summing stage, and an adaptive filter. The AGC is responsive to a first signal in a reference path and a second signal in a near end path. The signal summing stage is between the first and second scalar in the near end path and the first and second scalars are responsive to the AGC. The adaptive filter is responsive to the first signal and provides a third signal to the signal summing stage that corresponds to an echo signal and is subtracted from a signal in the near end path. The second scalar has a scale rate that is the inverse of the scale rate of the first scalar such that unity scaling occurs on the near end path.

Automatic gain control for an adaptive finite impulse response and method therefore

An echo canceling system receives and transmits audio signals between a far end and a near end. During single talk, which is when only one end is originating audio, the path back to the originator is impeded by echo cancellation and attenuation. When there is double talk, which is when both ends are originating audio, the attenuation is removed, or at least significantly reduced. This is achieved by using ERLE, which itself is a known signal used for other purposes in an echo cancellation system, to provide information as to when double talking is occurring. This allows for stopping the attenuation for the double talk situation, which is the desired result.

Controlling attenuation during echo suppression

A speech recognition system () having a sampler block () and a feature extractor block () for extracting time domain and spectral domain parameters from a spoken input speech into a feature vector. A polynomial expansion block () generates polynomial coefficients from the feature vector. A correlator block (), a sequence vector block (), an HMM table () and a Viterbi block () perform the actual speech recognition based on the speech units stored in a speech unit table () and the HMM word models stored in the HMM table (). The HMM word model that produces the highest probability is determined to be the word that was spoken.

Speech recognition using polynomial expansion and hidden markov models

A voice decoder detects channel errors and loss of cryptographic synchronization using the change in spectral energy between sequential frames. The change in energy between frames is determined between corresponding LSP's of said successive frames and summed together. A running average of the change in energy for a predetermined number of frames is maintained. Current voice frames are eliminated based on the difference between the change in energy associated with the current frame and the running average. Accordingly, offensive audio associated with such channel errors or cryptographic synchronization loss is eliminated.

Voice decoder and method for detecting channel errors using spectral energy evolution

An apparatus comprises a first and a second input and a processor coupled to the inputs. The processor estimates peak and minimum levels of each of the first and second input signals. A first signal regulator is coupled to the first input and to a first output for delivering a first output signal from the first signals. A second signal regulator is coupled to the second input and to a second output for delivering a second output signal from the second audio signals. A gain adjustment device is coupled to the processor and the first and second signal regulators. The gain adjustment device provides control signals to the first and second signal regulators to adjust an output signal level of the first output signal in response to minimum and peak levels of the first and second input signals. The output signal level is continuously variable over a range.

Method and apparatus for volume switched gain control

A new method for Code Excited Linear Predictive (CELP) coding of speech reduces the computational complexity by removing a convolution operation from a recursive loop used to poll the adaptive code book vectors. In a preferred embodiment, an impulse function of a short term perceptually weighted filter is first convolved with perceptual weighted target speech and the result cross-correlated with each vector in the codebook to produce an error function. The vector having the minimum error function is chosen to represent the particular speech frame being examined.

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Scottsdale, AZ, United States of America

David L Barron