The speech encoding apparatus (220) has a first layer encoding section (2201), a first layer decoding section (2202), a delay section (2203), a subtracting section (104), a frequency domain transforming section (101), a second layer encoding section (105) and a multiplexing section (106).

Encoding device, decoding device, and method thereof

Methods and apparatus for generating an anti-noise signal and equalizing a reproduced audio signal (e.g., a far-end telephone signal) are described, wherein the generating and the equalizing are both based on information from an acoustic error signal.

Systems, methods, devices, apparatus, and computer program products for audio equalization

The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.

Method and apparatus for suppressing wind noise

A rule-based end-pointer isolates spoken utterances contained within an audio stream from background noise and non-speech transients. The rule-based end-pointer includes a plurality of rules to determine the beginning and/or end of a spoken utterance based on various speech characteristics. The rules may analyze an audio stream or a portion of an audio stream based upon an event, a combination of events, the duration of an event, or a duration relative to an event. The rules may be manually or dynamically customized depending upon factors that may include characteristics of the audio stream itself, an expected response contained within the audio stream, or environmental conditions.

Speech end-pointer

Active noise cancellation is combined with spectrum modification of a reproduced audio signal to enhance intelligibility.

Systems, methods, apparatus, and computer-readable media for automatic control of active noise cancellation

A voice enhancement logic improves the perceptual quality of a processed voice. The voice enhancement system includes a noise detector and a noise attenuator. The noise detector detects a wind buffet and a continuous noise by modeling the wind buffet. The noise attenuator dampens the wind buffet to improve the intelligibility of an unvoiced, a fully voiced, or a mixed voice segment.

System for Suppressing Wind Noise

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.

Bandwidth extension of narrowband speech

A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range. The speech enhancement system may be built into, may be a unitary part of, or may be configured to interface other systems that process audio or high frequency signals.

High frequency compression integration

A speech enhancement system that improves the intelligibility and the perceived quality of processed speech includes a frequency transformer and a spectral compressor. The frequency transformer converts speech signals from the time domain to the frequency domain. The spectral compressor compresses a pre-selected portion of the high frequency band and maps the compressed high frequency band to a lower band limited frequency range.

System for improving speech intelligibility through high frequency compression

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.

Xueman Li

Papers

System for Suppressing Wind Noise

Bandwidth extension of narrowband speech

High frequency compression integration

System for improving speech intelligibility through high frequency compression

Speech enhancement through partial speech reconstruction