Abstract: 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.

Abstract: Wind and other noise is suppressed in a signal by adaptively changing characteristics of a filter. The filter characteristics are changed in response to the noise content of the signal over time using a history of noise content. Filter characteristics are changed according to a plurality of reference filters, the characteristics of which are chosen to optimally attenuate or amplify signals in a range of frequencies.

Abstract: 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.

Abstract: 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.

Abstract: An input signal is received. A plurality of electrical characteristics from the input signal is obtained. A plurality of acoustic features is determined from the obtained electrical characteristics and each of the acoustic features being different from the others. At least some of the acoustic features are compared to a plurality of predetermined criteria. Based upon the comparing of the acoustic features to the plurality of predetermined criteria, it is determined when the signal is a voice signal or a noise signal.

Abstract: 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.

Abstract: An input signal from the input source is received. An attenuation of an echo canceller filter is calculated using at least the input signal. A spectral component of a frequency band of an echo suppressor is adjusted to perform enhanced suppression using the calculated attenuation. A comfort noise factor is calculated using at least the input signal and the calculated attenuation. The comfort noise to the output of the echo suppressor is adjusted to obtain a modified input signal.

Abstract: Wind and other noise is suppressed in a signal by adaptively changing characteristics of a filter. The filter characteristics are changed in response to the noise content of the signal over time using a history of noise content. Filter characteristics are changed according to a plurality of reference filters, the characteristics of which are chosen to optimally attenuate or amplify signals in a range of frequencies.

Abstract: 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.

Abstract: 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.

Abstract: A method and apparatus to reduce background noise in speech signals in order to improve the quality and intelligibility of processed speech. In mobile communications environment, speech signals are degraded by additive random noise. A randomness of the noise, which is often described in terms of its first and second order statistics, make it difficult to remove much of the noise without introducing background artifacts. This is particularly true for lower signal to background noise ratios. The method and apparatus provides noise reduction without any knowledge of the signal to background noise ratio.

Abstract: A method and apparatus to reduce background noise in speech signals in order to improve the quality and intelligibility of processed speech. In mobile communications environment, speech signals are degraded by additive random noise. A randomness of the noise, which is often described in terms of its first and second order statistics, make it difficult to remove much of the noise without introducing background artifacts. This is particularly true for lower signal to background noise ratios. The method and apparatus provides noise reduction without any knowledge of the signal to background noise ratio.

Abstract: A lost frame recovery technique for LPC-based systems employs interpolation of parameters from previous and subsequent good frames, selective attenuation of frame energy when the energy of a subframe exceeds a threshold, and energy tapering in the presence of multiple successive lost frames.

Abstract: A “multi-stage” method of estimating pitch in a speech encoder (FIG. 2). In a first stage of the method, a set of candidate pitch values is selected, such as by using a cost function that operates on said speech signal (steps 21-23). In a second stage of the method, a best candidate is selected. Specifically, in the second stage, pitch values calculated from previous speech segments are used to calculate an average pitch value (step 25). Then, depending on whether the average pitch value is short or long, one of two different analysis-by-synthesis (ABS) processes is then repeated for each candidate, such that for each iteration, a synthesized signal is derived from that pitch candidate and compared to a reference signal to provide an error value. A time domain ABS process is used if the average pitch is short (step 27), whereas a frequency domain ABS process is used if the average pitch is long (step 28).

Abstract: A “multi-stage” method of estimating pitch in a speech encoder (FIG. 2). In a first stage of the method, a set of candidate pitch values is selected, such as by using a cost function that operates on said speech signal (steps 21-23). In a second stage of the method, a best candidate is selected. Specifically, in the second stage, pitch values calculated from previous speech segments are used to calculate an average pitch value (step 25). Then, depending on whether the average pitch value is short or long, one of two different analysis-by-synthesis (ABS) processes is then repeated for each candidate, such that for each iteration, a synthesized signal is derived from that pitch candidate and compared to a reference signal to provide an error value. A time domain ABS process is used if the average pitch is short (step 27), whereas a frequency domain ABS process is used if the average pitch is long (step 28).

Abstract: A speech coding algorithm interpolates groups speech frames into speech frame pairs, and quantizes each frame of the pair according to a different algorithm. The spectral amplitudes of the second frame are quantized by dividing them into two portions and quantizing one portion and then quantizing a difference between the two portions. The spectral amplitudes of the first frame of the pair are quantized by first converting to a fixed dimension, then interpolating between previous and subsequent frames, then selecting interpolated values in accordance with a mean squared error approach.

Abstract: A voicing probability determination method is provided for estimating a percentage of unvoiced and voiced energy for each harmonic within each of a plurality of bands of a speech signal spectrum. Initially, a synthetic speech spectrum is generated based on the assumption that speech is purely voiced. The original and synthetic speech spectra are then divided into plurality of bands. The synthetic and original speech spectra are compared harmonic by harmonic, and a voicing determination is made based on this comparison. In one embodiment, each harmonic of the original speech spectrum is assigned a voicing decision as either completely voiced or unvoiced by comparing the difference with an adaptive threshold. If the difference for each harmonic is less than the adaptive threshold, the corresponding harmonic is declared as voiced; otherwise the harmonic is declared as unvoiced. The voicing probability for each band is then computed based on the amount of energy in the voiced harmonics in that decision band.

Abstract: A voicing probability determination method is provided for estimating a percentage of unvoiced and voiced energy for each harmonic within each of a plurality of bands of a speech signal spectrum. Initially, a synthetic speech spectrum is generated based on the assumption that speech is purely voiced. The original and synthetic speech spectra are then divided into plurality of bands. The synthetic and original speech spectra are compared harmonic by harmonic, and a voicing determination is made based on this comparison. In one embodiment, each harmonic of the original speech spectrum is assigned a voicing decision as either completely voiced or unvoiced by comparing the difference with an adaptive threshold. If the difference for each harmonic is less than the adaptive threshold, the corresponding harmonic is declared as voiced; otherwise the harmonic is declared as unvoiced. The voicing probability for each band is then computed based on the amount of energy in the voiced harmonics in that decision band.

Abstract: A voicing probability determination method is provided for estimating a percentage of unvoiced and voiced energy for each harmonic within each of a plurality of bands of a speech signal spectrum. Initially, a synthetic speech spectrum is generated based on the assumption that speech is purely voiced. The original and synthetic speech spectra are then divided into plurality of bands. The synthetic and original speech spectra are compared harmonic by harmonic, and a voicing determination is made based on this comparison. In one embodiment, each harmonic of the original speech spectrum is assigned a voicing decision as either completely voiced or unvoiced by comparing the difference with an adaptive threshold. If the difference for each harmonic is less than the adaptive threshold, the corresponding harmonic is declared as voiced; otherwise the harmonic is declared as unvoiced. The voicing probability for each band is then computed based on the amount of energy in the voiced harmonics in that decision band.

Abstract: An adaptive time-domain post-filtering technique is based on the modified Yule-Walker filter. This technique eliminates the problem of spectral tilt in speech spectrum that can be applied to various speech coders. The new post-filter has a flat frequency response at the formant peaks of speech spectrum. Information is gathered about the relation between poles and formants and then the formants and their bandwidths are estimated. The information about the formants and their bandwidths is then used to design the modified Yule-Walker filter based on a least squares fit in time domain.