Abstract: A system and method that enhances spoken utterances by capturing one or more microphone signals. The system and method estimates a plurality of echo paths from each of the one or more microphone signals and synthesizes a speech reinforcement signal in response to and corresponding to the one or more microphone signals. The system and method concatenates portions of the synthesized reinforcement signal with the captured microphone signals and processes the captured microphone signals in response to the estimated plurality of echo paths.

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 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: 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: 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: 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 sub-band processing system that reduces computational complexity and memory requirements includes a processor and a local or distributed memory. Logic stored in the memory partitions a frequency spectrum of bins into a smaller number of sub-bands. The logic enables a lossy compression by designating a magnitude and a designated or derived phase of each bin in the frequency spectrum as representative. The logic renders a lossless compression by decompressing the lossy compressed data and providing lost data based on original spectral relationships contained within the frequency spectrum.

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 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 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: 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 present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.

Abstract: A speech enhancement system improves the perceptual quality of a processed voice signal. The system improves the perceptual quality of a voice signal by removing unwanted noise components from a voice signal. The system removes undesirable signals that may result in the loss of information. The system receives and analyzes signals to determine whether an undesired random or persistent signal corresponds to one or more modeled noises. When one or more noise components are detected, the noise components are substantially removed or dampened from the signal to provide a less noisy voice signal.

Abstract: A sub-band processing system that reduces computational complexity and memory requirements includes a processor and a local or distributed memory. Logic stored in the memory partitions a frequency spectrum of bins into a smaller number of sub-bands. The logic enables a lossy compression by designating a magnitude and a designated or derived phase of each bin in the frequency spectrum as representative. The logic renders a lossless compression by decompressing the lossy compressed data and providing lost data based on original spectral relationships contained within the frequency spectrum.

Abstract: A voice enhancement logic improves the perceptual quality of a processed voice. The voice enhancement system includes a passing tire hiss noise detector and a passing tire hiss noise attenuator. The passing tire hiss noise detector detects a passing tire hiss noise by modeling the passing tire hiss. The passing tire hiss noise attenuator dampens the passing tire hiss noise to improve the intelligibility of a speech signal.

Abstract: A sub-band processing system that reduces computational complexity and memory requirements includes a processor and a local or distributed memory. Logic stored in the memory partitions a frequency spectrum of bins into a smaller number of sub-bands. The logic enables a lossy compression by designating a magnitude and a designated or derived phase of each bin in the frequency spectrum as representative. The logic renders a lossless compression by decompressing the lossy compressed data and providing lost data based on original spectral relationships contained within the frequency spectrum.

Abstract: A speech enhancement system improves the perceptual quality of a processed voice signal. The system improves the perceptual quality of a voice signal by removing unwanted noise components from a voice signal. The system removes undesirable signals that may result in the loss of information. The system receives and analyzes signals to determine whether an undesired random or persistent signal corresponds to one or more modeled noises. When one or more noise components are detected, the noise components are substantially removed or dampened from the signal to provide a less noisy voice signal.

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.