Signal-dependent companding system and method to reduce quantization noise

    公开(公告)号:US10861475B2

    公开(公告)日:2020-12-08

    申请号:US15775000

    申请日:2016-10-27

    Inventor: Arijit Biswas

    Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A method of processing an audio signal comprises receiving an audio signal, classifying the audio signal as one of pure sinusoidal, hybrid, or pure transient signal using two defined threshold values, and selectively applying a companding operation by switching between a companding off mode, a companding on mode, and an average companding mode, comprising selecting between the companding on mode and the average companding mode for a classified hybrid signal using a companding rule that uses a temporal sharpness measure in a frequency domain.

    MDCT-domain error concealment
    22.
    发明授权

    公开(公告)号:US10424305B2

    公开(公告)日:2019-09-24

    申请号:US15533625

    申请日:2015-12-08

    Abstract: An error-concealing audio decoding method comprises: receiving a packet comprising a set of MDCT coefficients encoding a frame of time-domain samples of an audio signal; identifying the received packet as erroneous; generating estimated MDCT coefficients to replace the set of MDCT coefficients of the erroneous packet, based on corresponding MDCT coefficients associated with a received packet directly preceding the erroneous packet; assigning signs of a first subset of MDCT coefficients of the estimated MDCT coefficients, wherein the first subset comprises such MDCT coefficients that are associated with tonal-like spectral bins, to coincide with signs of corresponding MDCT coefficients of said preceding packet; randomly assigning signs of a second subset of MDCT coefficients of the estimated MDCT coefficients, wherein the second subset comprises MDCT coefficients associated with noise-like spectral bins; replacing the erroneous packet by a concealment packet containing the estimated MDCT coefficients and the signs assigned.

    Speech/Dialog Enhancement Controlled by Pupillometry

    公开(公告)号:US20190057694A1

    公开(公告)日:2019-02-21

    申请号:US15998796

    申请日:2018-08-16

    Inventor: Arijit Biswas

    Abstract: The present disclosure relates to methods for processing a decoded audio signal and for selectively applying speech/dialog enhancement to the decoded audio signal. The present disclosure also relates to a method of operating a headset for computer-mediated reality. A method of processing a decoded audio signal comprises obtaining a measure of a cognitive load of a listener that listens to a rendering of the audio signal, determining whether speech/dialog enhancement shall be applied based on the obtained measure of the cognitive load, and performing speech/dialog enhancement based on the determination. A method of operating a headset for computer-mediated reality comprises obtaining eye-tracking data of a wearer of the headset, determining a measure of a cognitive load of the wearer of the headset based on the eye-tracking data, and outputting an indication of the cognitive load of the wearer of the headset. The present disclosure further relates to corresponding apparatus and systems, and to methods of operating such apparatus and systems.

    Method and apparatus for controlling enhancement of low-bitrate coded audio

    公开(公告)号:US11929085B2

    公开(公告)日:2024-03-12

    申请号:US17270053

    申请日:2019-08-29

    CPC classification number: G10L19/24

    Abstract: Described herein is a method of low-bitrate coding of audio data and generating enhancement metadata for controlling audio enhancement of the low-bitrate coded audio data at a decoder side, including the steps of: (a) core encoding original audio data at a low bitrate to obtain encoded audio data; (b) generating enhancement metadata to be used for controlling a type and/or amount of audio enhancement at the decoder side after core decoding the encoded audio data; and (c) outputting the encoded audio data and the enhancement metadata. Described is further an encoder configured to perform said method. Described is moreover a method for generating enhanced audio data from low-bitrate coded audio data based on enhancement metadata and a decoder configured to perform said method.

    METHODS AND SYSTEM FOR WAVEFORM CODING OF AUDIO SIGNALS WITH A GENERATIVE MODEL

    公开(公告)号:US20220392458A1

    公开(公告)日:2022-12-08

    申请号:US17770035

    申请日:2020-10-16

    Abstract: Described herein is a method of waveform decoding, the method including the steps of: (a) receiving, by a waveform decoder, a bitstream including a finite bitrate representation of a source signal; (b) waveform decoding the finite bitrate representation of the source signal to obtain a waveform approximation of the source signal; (c) providing the waveform approximation of the source signal to a generative model that implements a probability density function, to obtain a probability distribution for a reconstructed signal of the source signal; and (d) generating the reconstructed signal of the source signal based on the probability distribution. Described are further a method and system for waveform coding and a method of training a generative model.

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