Abstract:
Systems and methods are described for dynamically suppressing non-linear distortion for a device, such as a speakerphone. A device may receive a signal, where the device has non-linear distortion at a predetermined frequency. The received signal may be analyzed to compute a tone strength parameter and a band level. The received signal may be filtered such that a spectrum of the input signal is dynamically limited by reducing suppression of the non-linear distortion when the tone strength parameter is in a lower portion of a predetermined range and increasing suppression of the non-linear distortion when the tone strength parameter is in an upper portion of the predetermined range, the predetermined range of the tone strength parameter corresponding to a loudness range of the device.
Abstract:
A method for altering an audio signal of interest in a multi-channel soundfield representation of an audio environment, the method including the steps of: (a) extracting the signal of interest from the soundfield representation; (b) determining a residual soundfield signal; (c) inputting a further associated audio signal, which is associated with the signal of interest; (d) transforming the associated audio signal into a corresponding associated soundfield signal compatable with the residual soundfield; and (e) combining the residual soundfield signal with the associated soundfield signal to produce an output soundfield signal.
Abstract:
A method of processing data comprises processing first frequency-domain audio or video data using signal processing of a first type, and transforming the processed first frequency-domain audio or video data to processed time-domain audio or video data using a transform which is the inverse of the first transform, and transforming the processed time-domain audio or video data using a second transform which is matched to a second type of signal processing. The method further comprises identifying time-domain audio or video data for which signal processing of the first type, after transformation using the second transform, would yield satisfactory results. The method further comprises transforming the identified time-domain audio or video data to frequency-domain identified audio or video data using the second transform, instead of using the first transform, and processing the identified frequency-domain audio or video data using signal processing of the first type.
Abstract:
Improved audio data processing method and systems are provided. Some implementations involve dividing frequency domain audio data into a plurality of subbands and determining amplitude modulation signal values for each of the plurality of subbands. A band-pass filter may be applied to the amplitude modulation signal values in each subband, to produce band-pass filtered amplitude modulation signal values for each subband. The band-pass filter may have a central frequency that exceeds an average cadence of human speech. A gain may be determined for each subband based, at least in part, on a function of the amplitude modulation signal values and the band-pass filtered amplitude modulation signal values. The determined gain may be applied to each subband.
Abstract:
A method in a soundfield-capturing endpoint and the capturing endpoint that comprises a microphone array capturing soundfield, and an input processor pre-processing and performing auditory scene analysis to detect local sound objects and positions, de-clutter the sound objects, and integrate with auxiliary audio signals to form a de-cluttered local auditory scene that has a measure of plausibility and perceptual continuity. The input processor also codes the resulting de-cluttered auditory scene to form coded scene data comprising mono audio and additional scene data to send to others. The endpoint includes an output processor generating signals for a display unit that displays a summary of the de-cluttered local auditory scene and/or a summary of activity in the communication system from received data, the display including a shaped ribbon display element that has an extent with locations on the extent representing locations and other properties of different sound objects.
Abstract:
In one embodiment, a sound field is mapped by extracting spatial angle information, diffusivity information, and optionally, sound level information. The extracted information is mapped for representation in the form of a Riemann sphere, wherein spatial angle varies longitudinally, diffusivity varies latitudinally, and level varies radially along the sphere. A more generalized mapping employs mapping the spatial angle and diffusivity information onto a representative region exhibiting variations in direction of arrival that correspond to the extracted spatial information and variations in distance that correspond to the extracted diffusivity information.