Abstract:
An apparatus for decoding an encoded audio signal, comprises a spectral domain audio decoder (112) for generating a first decoded representation of a first set of first spectral portions, the decoded representation having a first spectral resolution; a parametric decoder (114) for generating a second decoded representation of a second set of second spectral portions having a second spectral resolution being lower than the first spectral resolution; a frequency regenerator (116) for regenerating every constructed second spectral portion having the first spectral resolution using a first spectral portion and spectral envelope information for the second spectral portion; and a spectrum time converter (118) for converting the first decoded representation and the reconstructed second spectral portion into a time representation.
Abstract:
An apparatus for generating a frequency enhancement signal (130) comprises: a signal generator (200) for generating an enhancement signal from a core signal (120, 110), the enhancement signal comprising an enhancement frequency range not included in the core signal, wherein a current time portion (320, 340) of the enhancement signal or the core signal comprises subband signals for a plurality of subbands; a controller (800) for calculating the same smoothing information (802) for the plurality of subband signals of the enhancement frequency range or the core signal, and wherein the signal generator (200) is configured for smoothing the plurality of subband signals of the enhancement frequency range or the core signal using the same smoothing information (802).
Abstract:
The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value (311, 312) for a first frequency subband (205) of an audio signal is described. The first banded tonality value (311, 312) is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal. The method comprises determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of the audio signal; determining a set of bin tonality values (341 ) for the set of frequency bins using the set of transform coefficients, respectively; and combining a first subset of two or more of the set of bin tonality values (341) for two or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby yielding the first banded tonality value (311, 312) for the first frequency subband.
Abstract:
The present document relates to the technical field of audio coding, decoding and processing. It specifically relates to methods of recovering high frequency content of an audio signal from low frequency content of the same audio signal in an efficient manner. A method for determining a first banded tonality value (311, 312) for a first frequency subband (205) of an audio signal is described. The first banded tonality value (311, 312) is used for approximating a high frequency component of the audio signal based on a low frequency component of the audio signal. The method comprises determining a set of transform coefficients in a corresponding set of frequency bins based on a block of samples of the audio signal; determining a set of bin tonality values (341 ) for the set of frequency bins using the set of transform coefficients, respectively; and combining a first subset of two or more of the set of bin tonality values (341) for two or more corresponding adjacent frequency bins of the set of frequency bins lying within the first frequency subband, thereby yielding the first banded tonality value (311, 312) for the first frequency subband.
Abstract:
The invention relates to a background noise estimator and a method therein, for estimation of background noise in an audio signal. The method comprises obtaining at least one parameter associated with an audio signal segment, such as a frame or part of a frame, based on a first linear prediction gain, calculated as a quotient between a residual signal from a 0th-order linear prediction and a residual signal from a 2nd-order linear prediction for the audio signal segment; and, a second linear prediction gain calculated as a quotient between a residual signal from a 2nd-order linear prediction and a residual signal from a 16th-order linear prediction for the audio signal segment. The method further comprises determining whether the audio signal segment comprises a pause based at least on the obtained at least one parameter; and, updating a background noise estimate based on the audio signal segment when the audio signal segment comprises a pause.
Abstract:
Embodiments of the present invention relate to signal processing. Methods for enhancing intelligibility of speech content in an audio signal are disclosed. One of the methods comprises obtaining reference loudness of the audio signal. The method further comprises enhancing the intelligibility of the speech content by adjusting partial loudness of the audio signal based on the reference loudness and a degree of the intelligibility. Corresponding systems and computer program products are also disclosed.