Abstract: A system that performs automatic gain control (AGC) using different decay rates. The system may select a slow decay rate to track a loudness level within speech (e.g., within an utterance), improving audio quality and maintaining dynamic range for an individual voice, while selecting a fast decay rate to track the loudness level after a gap of silence (e.g., no voice activity detected for a duration of time) or during large level changes (e.g., actual speech loudness is lower than estimated speech loudness for a duration of time). This improves an accuracy of the loudness estimate and therefore a responsiveness of the automatic gain control, resulting in an improved user experience.

Abstract: 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.

Abstract: 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.

Abstract: The system provides a technique for suppressing or eliminating tonal noise in and input signal. The system operates on the input signal at a plurality of frequency bins and uses information generated at a prior bin to assist in calculating values at subsequent bins. The system first identifies peaks in a signal and then determines if the peaks are from tonal effects. This can be done by comparing the estimated background noise of a current bin to the smoothed background noise of the same bin. The smoothed background noise can be calculated using an asymmetric IIR filter. When the ratio of the current background noise estimate to the currently calculated smoothed background noise is far greater than 1, tonal noise is assumed. When tonal noise is found, a number of suppression techniques can be applied to reduce the tonal noise, including gain suppression with fixed floor factor, an adaptive floor factor gain suppression technique, and a random phase technique.

Abstract: The present system proposes a technique called the spectro-temporal varying technique, to compute the suppression gain. This method is motivated by the perceptual properties of human auditory system; specifically, that the human ear has higher frequency resolution in the lower frequencies band and less frequency resolution in the higher frequencies, and also that the important speech information in the high frequencies are consonants which usually have random noise spectral shape. A second property of the human auditory system is that the human ear has lower temporal resolution in the lower frequencies and higher temporal resolution in the higher frequencies. Based on that, the system uses a spectro-temporal varying method which introduces the concept of frequency-smoothing by modifying the estimation of the a posteriori SNR. In addition, the system also makes the a priori SNR time-smoothing factor depend on frequency.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: The system describes a karaoke system that enhances the experience of singing along with music, but without the need to display the lyrics. The system includes a combination of a vocal track reducer and an echo canceller, decision logic for determining when a person is talking or singing (double-talk detector) and a method for “ducking” (i.e., attenuating) the vocal track when the singing is detected. No special CD or DVD with lyric tracks is required, making the system capable of working with CD, mp3, AM, FM, HD radio, satellite radio signals, or any other suitable content source. The result is that any content source may potentially be used as a karaoke soundtrack without any pre-modification.

Abstract: 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.

Abstract: The system describes a karaoke system that enhances the experience of singing along with music, but without the need to display the lyrics. The system includes a combination of a vocal track reducer and an echo canceller, decision logic for determining when a person is talking or singing (double-talk detector) and a method for “ducking” (i.e., attenuating) the vocal track when the singing is detected. No special CD or DVD with lyric tracks is required, making the system capable of working with CD, mp3, AM, FM, HD radio, satellite radio signals, or any other suitable content source. The result is that any content source may potentially be used as a karaoke soundtrack without any pre-modification.

Abstract: The present system proposes a technique called the spectro-temporal varying technique, to compute the suppression gain. This method is motivated by the perceptual properties of human auditory system; specifically, that the human ear has higher frequency resolution in the lower frequencies band and less frequency resolution in the higher frequencies, and also that the important speech information in the high frequencies are consonants which usually have random noise spectral shape. A second property of the human auditory system is that the human ear has lower temporal resolution in the lower frequencies and higher temporal resolution in the higher frequencies. Based on that, the system uses a spectro-temporal varying method which introduces the concept of frequency-smoothing by modifying the estimation of the a posteriori SNR. In addition, the system also makes the a priori SNR time-smoothing factor depend on frequency.

Abstract: The system provides a technique for suppressing or eliminating tonal noise in and input signal. The system operates on the input signal at a plurality of frequency bins and uses information generated at a prior bin to assist in calculating values at subsequent bins. The system first identifies peaks in a signal and then determines if the peaks are from tonal effects. This can be done by comparing the estimated background noise of a current bin to the smoothed background noise of the same bin. The smoothed background noise can be calculated using an asymmetric IIR filter. When the ratio of the current background noise estimate to the currently calculated smoothed background noise is far greater than 1, tonal noise is assumed. When tonal noise is found, a number of suppression techniques can be applied to reduce the tonal noise, including gain suppression with fixed floor factor, an adaptive floor factor gain suppression technique, and a random phase technique.

Abstract: 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.

Abstract: 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.