Abstract: A system processes signals exchanged between a near end terminal and a far end terminal over a communication path. The system selects the processing characteristics of a near end terminal based on characteristics of the communication path. The communication path may include the near and the far end terminal and their respective codecs, and the networks. The system may select processing characteristics of the near end terminal based on characteristics of the communication path. Selecting the processing characteristics may include selecting a predetermined operation. The system may adapt a selected operation based on a real time analysis of the communication path.

Abstract: A voice enhancement system is provided for improving the perceptual quality of a processed voice signal. The system improves the perceptual quality of a received voice signal by removing unwanted noise from a voice signal recorded by a microphone or from some other source. Specifically, the system removes sounds that occur within the environment of the signal source but which are unrelated to speech. The system is especially well adapted for removing transient road noises from speech signals recorded in moving vehicles. Transient road noises include common temporal and spectral characteristics that can be modeled. A transient road noise detector employs such models to detect the presence of transient road noises in a voice signal. If transient road noises are found to be present, a transient road noise attenuator is provided to remove them from the signal.

Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, an adaptive filter, and signal reinforcement logic. The adaptive filter may track one or more fundamental frequencies in the input signal and outputs a filtered signal. The filtered signal may approximately reproduce the input signal approximately delayed by an integer multiple of the signal's fundamental frequencies. The reinforcement logic combines the input signal and the filtered signal output to produce an enhanced signal output.

Abstract: A speech enhancement system enhances transitions between speech and non-speech segments. The system includes a background noise estimator that approximates the magnitude of a background noise of an input signal that includes a speech and a non-speech segment. A slave processor is programmed to perform the specialized task of modifying a spectral tilt of the input signal to match a plurality of expected spectral shapes selected by a Codec.

Abstract: A speech enhancement system improves speech conversion within an encoder and decoder. The system includes a first device that converts sound waves into operational signals. A second device selects a template that represents an expected signal model. The selected template models speech characteristics of the operational signals through a speech codebook that is further accessed in a communication channel.

Abstract: A speech enhancement system controls the gain of an excitation signal to prevent uncontrolled gain adjustments. The system includes a first device that converts sound waves into operational signals. An ambient noise estimator is linked to the first device and an echo canceller. The ambient noise estimator estimates how loud a background noise would be near the first device before or after an echo cancellation. The system then compares the ambient noise estimate to a current ambient noise estimate near the first device to control a gain of an excitation signal.

Abstract: A speech signal isolation system configured to isolate and reconstruct a speech signal transmitted in an environment in which frequency components of the speech signal are masked by background noise. The speech signal isolation system obtains a noisy speech signal from an audio source. The noisy speech signal may then be fed through a neural network that has been trained to isolate and reconstruct a clean speech signal from against background noise. Once the noisy speech signal has been fed through the neural network, the speech signal isolation system generates an estimated speech signal with substantially reduced noise.

Abstract: A voice activity detection process is robust to a low and high signal-to-noise ratio speech and signal loss. A process divides an aural signal into one or more bands. Signal magnitudes of frequency components and the respective noise components are estimated. A noise adaptation rate modifies estimates of noise components based on differences between the signal to the estimated noise and signal variability.

Abstract: A signal enhancement system improves the understandability of speech or other audio signals. The system reinforces selected parts of the signal, may attenuate selected parts of the signal, and may increase SNR. The system includes delay logic, a partitioned adaptive filter, and signal reinforcement logic. The partitioned adaptive filter may track and enhance the fundamental frequency and harmonics in the input signal. The partitioned filter output signals may approximately reproduce the input signal, delayed by an integer multiple of the period of the fundamental frequency of the input signal. The reinforcement logic combines the input signal and the filtered signals to produce an enhanced output signal.

Abstract: A speech synthesis system synthesizes a speech signal corresponding to an input speech signal based on a spectral envelope of the input speech signal. A glottal pulse generator generates a time series of glottal pulses, that are processed into a glottal pulse magnitude spectrum. A shaping circuit shapes the glottal pulse magnitude spectrum based on the spectral envelope and generates a shaped glottal pulse magnitude spectrum. A harmonic null adjustment circuit reduces harmonic nulls in the shaped glottal pulse magnitude spectrum and generates a null-adjusted synthesized speech spectrum. An inverse transform circuit generates a null-adjusted time-series speech signal. An overlap and add circuit synthesizes the speech signal based on the null-adjusted time-series speech signal.

Abstract: A system extends the bandwidth of a narrowband speech signal into a wideband spectrum. The system includes a high-band generator that generates a high frequency spectrum based on a narrowband spectrum. A background noise generator generates a high frequency background noise spectrum based on a background noise within the narrowband spectrum. A summing circuit linked to the high-band generator and the background noise generator combines the high frequency spectrum and narrowband spectrum and the high frequency background noise spectrum.

Abstract: A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherence levels of the signals.

Abstract: A system improves speech intelligibility by reconstructing speech segments. The system includes a low-frequency reconstruction controller programmed to select a predetermined portion of a time domain signal. The low-frequency reconstruction controller substantially blocks signals above and below the selected predetermined portion. A harmonic generator generates low-frequency harmonics in the time domain that lie within a frequency range controlled by a background noise modeler. A gain controller adjusts the low-frequency harmonics to substantially match the signal strength to the time domain original input signal.

Abstract: A speech enhancement system improves the speech quality and intelligibility of a speech signal. The system includes a time-to-frequency converter that converts segments of a speech signal into frequency bands. A signal detector measures the signal power of the frequency bands of each speech segment. A background noise estimator measures a background noise detected in the speech signal. A dynamic noise reduction controller dynamically models the background noise in the speech signal. The speech enhancement renders a speech signal perceptually pleasing to a listener by dynamically attenuating a portion of the noise that occurs in a portion of the spectrum of the speech signal.

Abstract: A system allocates channel bandwidth based on the data received from a plurality of remote sources. A de-multiplexer/priority circuit separates two or more different data streams into their components parts. A stream modification driver modifies one or more characteristics of the data received from the de-multiplexer/priority circuit based on a priority assigned to the data by the de-multiplexer/priority circuit. The de-multiplexer/priority circuit determines the data transfer rates for each of the different data streams based on the assigned priority.

Abstract: A noise suppression system reduces low-frequency noise in a speech signal using linear predictive coefficients in an adaptive filter. A digital filter may update or adapt a limited set of linear predictive coefficients on a sample-by-sample basis. The linear predictive coefficients may be used to provide an error signal based on a difference between the speech signal and a delayed speech signal. The error signal represents an enhanced speech signal having attenuated and normalized low-frequency noise components.

Abstract: A speech enhancement system improves the perceptual quality of an aural signal. A receiver detects and receives an unvoiced signal, a fully voiced signal, or a mixed voice remote signal. A coherence processor identifies the similarities or differences between a local signal and the remote signal. A cancellation processor or controller dampens reflected signals that may be part of the local signal.

Abstract: To reliably and consistently detect desirable sounds, a system detects the presence of wind noise based on the power levels of audio signals. A first transducer detects sound originating from a first direction and a second transducer detects sound originating from a second direction. The power levels of the sound are compared. When the power level of the sound received from the second transducer is less than the power level of the sound received from the first transducer by a predetermined value, wind noise may be present. A signal processor may generate an output from one or a combination of the audio signals, based on a wind noise detection.

Abstract: A system detects a speech segment that may include unvoiced, fully voiced, or mixed voice content. The system includes a digital converter that converts a time-varying input signal into a digital-domain signal. A window function passes signals within a programmed aural frequency range while substantially blocking signals above and below the programmed aural frequency range when multiplied by an output of the digital converter. A frequency converter converts the signals passing within the programmed aural frequency range into a plurality of frequency bins. A background voice detector estimates the strength of a background speech segment relative to the noise of selected portions of the aural spectrum. A noise estimator estimates a maximum distribution of noise to an average of an acoustic noise power of some of the plurality of frequency bins.

Abstract: A system extends the high-frequency spectrum of a narrow band audio signal in the time domain. The system extends the harmonics of vowels by introducing a non linearity in a narrow band signal. Extended consonants are generated by a random-noise generator. The system differentiates the vowels from the consonants by exploiting predetermined features of a speech signal.