Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: An adaptive equalization system that adjusts the spectral shape of a speech signal based on an intelligibility measurement of the speech signal may improve the intelligibility of the output speech signal. Such an adaptive equalization system may include a speech intelligibility measurement module, a spectral shape adjustment module, and an adaptive equalization module. The speech intelligibility measurement module is configured to calculate a speech intelligibility measurement of a speech signal. The spectral shape adjustment module is configured to generate a weighted long-term speech curve based on a first predetermined long-term average speech curve, a second predetermined long-term average speech curve, and the speech intelligibility measurement. The adaptive equalization module is configured to adapt equalization coefficients for the speech signal based on the weighted long-term speech curve.

Abstract: In a system and method for noise estimation with music detection described herein provides for generating a music classification for music content in an audio signal. The music detector may classify the audio signal as music or non-music. The non-music signal may be considered to be signal and noise. An adaption rate may be adjusted responsive to the generated music classification. A noise estimate is calculated applying the adjusted adaption rate. The system and method may mitigate the noise modeling algorithms being misled by the music components.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

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 system and method for dynamic residual noise shaping configured to reduce hiss noise in an audio signal. The system and method may detect an amount and type of hiss noise. The system and method may limit calculated noise suppression gains responsive to the detected amount and type of hiss noise. The limited noise suppression gains may be applied to the audio signal and may reduce the hiss noise.

Abstract: The suppression of off-axis audio in an audio environment is provided. Off-axis audio may be considered audio that does not originate from a region of interest. The off-axis audio is suppressed by comparing a phase difference between signals from two microphones to a target slope of the phase difference between signals originating from the region of interest. The target slope can be adapted to allow the region of interest to move with the location of a human speaker such as a driver.

Abstract: In a system and method for encoding a sound field the orientation of a computing device may be detected. Several orientation indications may be used to detect the computing device orientation. The detected orientation may be relative to a sound field that is a spatial representation of an audible environment associated with the computing device. Microphones associated with the computing device may be selected in order to receive the sound field based on the detected orientation. The received sound field may be processed and encoded with associated descriptive information.

Abstract: A system and method for dynamically mixing audio signals may calculate a signal amplitude for each of two or more audio signals. The signal amplitude may be the absolute value of the audio signal. A signal sum may be calculated using each of the two or more signal amplitudes. Each of the two or more signal amplitudes may be smoothed. The signal sum may be smoothed. The smoothing may be a filter or a leaky integrator. A respective mixing gain may be calculated for each of the two or more audio signals using a respective ratio of each of the two or more smoothed signal amplitudes and the smoothed signal sum. Each of the two or more audio signals may be gain adjusted responsive to the respective mixing gain. Each of the two or more gain adjusted audio signals may be mixed to create an output signal.

Abstract: A system and method for speech reinforcement may determine the spatial location of an audio source and the spatial location of a listener. An audio signal generated by the audio source may be captured. The spatial location, relative to the listener, of two or more audio transducers that emit a reinforcing audio signal to reinforce the audio signal may be determined. The captured audio signal, responsive to the spatial location of the audio source, the spatial location of the listener and the spatial location of the two or more audio transducers to generate the reinforcing audio signal, such that, when emitted by the two of more audio transducers, the listener perceives a source of the reinforcing audio signal to be spatially located in substantially the spatial location of the audio source thereby reinforcing the audio signal.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: A system and method for audio bandwidth dependent noise suppression may detect the audio bandwidth of an audio signal responsive to one or more audio indicators. The audio indicators may include the audio sampling rate and characteristics of an associated compression format. Noise suppression gains may be calculated responsive to the audio signal. Noise suppression gains may mitigate undesirable noise in the reproduced output signal. The noise suppression gains may be modified responsive to the detected audio bandwidth. Less noise reduction may be desirable when more audio bandwidth is available. The modified noise suppression gains may be applied to the audio signal.

Abstract: A system and method for enhancing comprehensibility through spatialization may receive two or more audio signals each associated with one of two or more audio sources. A respective panning gain may be calculated for each of the two or more audio signals using a count of the total number of audio sources and a count of a total number of output channels. The respective panning gain may be calculated further responsive to audio source metadata associated with each of two or more audio sources. Each of the two or more audio signals may be gain adjusted responsive to the respective panning gain. Each of the two or more gain adjusted audio signals may be mixed to create two or more output channels wherein a reproduction of the output channels produces enhanced comprehensibility.

Abstract: A noise estimation control system may limit increases of a stored background noise estimate in response to a detected noise feedback situation. The system receives an input audio signal detected within a space, and a reference audio signal that is transmitted by a speaker as an aural signal into the space. A signal processor processes the input audio signal and the reference audio signal to determine a coherence value based on an amount of the aural signal that is included in the input audio signal. The signal processor also calculates an amount to adjust the stored background noise estimate based on the coherence value and a determined background noise level of the input audio signal.

Abstract: A system and method for processing an audio signal captured from a microphone may reproduce a known audio signal with an audio transducer into an acoustic space. The known audio signal may include content from one or more audio sources. A microphone audio signal may be captured from the acoustic space where the microphone audio signal comprises the known audio signal and one or more unknown audio signals. Processing control information may be accessed. The known audio signal may be reduced in the microphone audio signal responsive to the processing control information where the processing control information indicates one or more characteristics of a downstream audio processor that processes the microphone audio signal.

Abstract: In a system and method for maintaining the spatial stability of a sound field a background noise estimate may be estimated for each of a first signal and a second signal. A first gain coefficient may be calculated responsive to the first audio signal and the background noise estimate of the first audio signal. A second gain coefficient may be calculated responsive to the second signal and the background noise estimate of the second signal. The gain coefficients may be calculated using one or more gain coefficient calculators. A common gain coefficient may be selected from one of the first gain coefficient and the second gain coefficient. The selected common gain coefficient may be one that results in a least amount of audio signal modification and may be applied to each of the first signal and the second signal.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: In an adaptive phase discovery system a first audio signal is received via a first microphone and a second signal is received via a second microphone. Corresponding audio frames of the first and second signals are each transformed into the frequency domain and a plurality of frequency sub-bands are generated. A phase is determined for each frequency sub-band in each signal. Instantaneous phase differences are determined between the signals at each of the frequency sub-bands. Lower frequency instantaneous phase differences are filtered over time to determine current phase differences at lower frequencies. When SNR is high in lower frequency sub-bands, lower frequency sub-band phase differences are tracked to the higher frequency sub-bands. The tracked higher frequency phase differences are filtered over time to determine phase differences for the current frame. The phase differences may be used to rotate phases in each sub-band and sum signals and/or to reject off-axis signals.

Abstract: A system and method for acoustic management that includes improving the sound quality of two or more audio processing modules in an acoustic environment may receive first control parameters from a first audio processing module. Receiving second control parameters from a second audio processing module. An audio processing interaction may be derived between with the first audio processing module and the second audio processing module determined from the first control parameters and the second control parameters. The first control parameters and the second control parameters may be modified responsive to the derived audio processing interaction.

Abstract: A system and method for dynamically mixing audio signals may calculate a signal amplitude for each of two or more audio signals. The signal amplitude may be the absolute value of the audio signal. A signal sum may be calculated using each of the two or more signal amplitudes. Each of the two or more signal amplitudes may be smoothed. The signal sum may be smoothed. The smoothing may be a filter or a leaky integrator. A respective mixing gain may be calculated for each of the two or more audio signals using a respective ratio of each of the two or more smoothed signal amplitudes and the smoothed signal sum. Each of the two or more audio signals may be gain adjusted responsive to the respective mixing gain. Each of the two or more gain adjusted audio signals may be mixed to create an output signal.