Abstract: Modifying a panned source in an audio signal comprising a plurality of channel signals is disclosed. Portions associated with the panned source are identified in at least selected ones of the channel signals. The identified portions are modified based at least in part on a user input.

Abstract: An audio equivalent of a video zoom feature for video recording and communication applications, as well as video post production processes. The audio zoom may operate in conjunction with a video zoom feature or independently. The audio zoom may be achieved by controlling reverberation effects of a signal, controlling a gain of the signal, as well as controlling the width of a directional beam from which is used to select the particular audio component to focus on. The audio zoom may operate in response to user input, such as a user selection of a particular direction, or automatically based a current environment or other factors.

Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.

Abstract: Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

Abstract: An audio signal is processed to derive primary and ambient components of the signal. The signal is first transformed to generate frequency-domain subband signals. Primary and ambient components are separated by comparing frequency subband content using a complex-valued similarity metric, wherein one of the primary and ambient components is determined to be the residual after the other is identified using the similarity metric.

Abstract: A system and method are disclosed for transient detection and modification in audio signals. Digital signal processing techniques are used to detect transients and modify an audio signal to enhance or suppress such transients, as desired. A transient audio event is detected in a first portion of the audio signal. A graded response to the detected transient audio event is determined. The first portion of the audio signal is modified in accordance with the graded response. The extent of enhancement or suppression (as applicable) may be determined at least in part by a measure of the significance or magnitude of the transient.

Abstract: Modifying an audio signal comprising a plurality of channel signals is disclosed. At least selected ones of the channel signals are transformed into a time-frequency domain. The at least selected ones of the channel signals are compared in the time-frequency domain to identify corresponding portions of the channel signals that are not correlated or are only weakly correlated across channels. The identified corresponding portions of said channel signals are modified.

Abstract: A system and method are disclosed for transient detection and modification in audio signals. Digital signal processing techniques are used to detect transients and modify an audio signal to enhance or suppress such transients, as desired. A transient audio event is detected in a first portion of the audio signal. A graded response to the detected transient audio event is determined. The first portion of the audio signal is modified in accordance with the graded response. The extent of enhancement or suppression (as applicable) may be determined at least in part by a measure of the significance or magnitude of the transient.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone.

Abstract: A frequency band of an audio input signal is analyzed to determine if a transient is present. When transients are detected, modifications are made to the intensity levels corresponding to the frequency band for a brief time period.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

Abstract: Separating a source in a stereo signal having a left channel and a right channel includes transforming the signal into a short-time transform domain; classifying portions of the signals having similar panning coefficients; segregating a selected one of the classified portions of the signals corresponding to the source; and reconstructing the source from the selected portions of the signals.

Abstract: A method for obtaining a clean speech signal in a communication system having a transducer for receiving a clean speech signal from a user and having a pair of loudspeakers for providing an output signal to the user. The output signal contains loudspeaker signals which interfere with the clean speech signal, the loudspeaker signals traveling through acoustic paths to reach the transducer. The transducer receives an input signal containing the loudspeaker signals and the clean speech signal. The method includes a number of steps, namely, performing a short time Fourier transform (STFT) on the input signal to obtain at least one frequency component, performing a short time Fourier transform (STFT) on the loudspeaker signals to obtain frequency components, summing the frequency components to obtain an interference sum, and subtracting the interference sum from the at least one frequency component to obtain the clean speech signal for translation into a time domain.