Abstract: A latency reduction system in a virtual bass processing system performs harmonic transposition on low frequency components of an audio signal to generate transposed data indicative of harmonics of the audio signal. The system uses a base transposition factor greater than two, and generates the harmonics in response to frequency-domain values determined by forward and inverse transform stages that use asymmetric analysis and synthesis windows. The system combines a virtual bass signal with the delayed wide band audio signal through analysis filter banks having filter coefficient truncated Nyquist filters. The virtual bass signal may lag the delayed wide band audio signal when combining with the audio signal to further reduce the latency caused by the harmonic transposition. The virtual bass input signal may be directly routed from a CQMF analysis filter bank of a preceding Hybrid filter bank stage, in order to avoid the delay associated with a Nyquist filter bank.

Abstract: An apparatus for generating a high frequency audio signal that includes an analyzer for analyzing an input signal to determine a transient information adaptively. Additionally a spectral converter is provided for converting the input signal into an input spectral representation. A spectral processor processes the input spectral representation to generate a processed spectral representation including values for higher frequencies than the input spectral representation. A time converter is configured for converting the processed spectral representation to a time representation, wherein the spectral converter or the time converter are controllable to perform a frequency domain oversampling for the first portion of the input signal having the transient information associated and to not perform the frequency domain oversampling for the second portion of the input signal not having the associated transient information.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal. In particular, a system configured to generate a high frequency component of a signal from a low frequency component of the signal is described. The system may comprise an analysis filter bank (501) configured to provide a set of analysis subband signals from the low frequency component of the signal; wherein the set of analysis subband signals comprises at least two analysis subband signals; wherein the analysis filter bank (501) has a frequency resolution of ?f.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilising high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: The present document relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), and to digital effect processors, e.g. so-called exciters, where generation of harmonic distortion adds brightness to the processed signal.

Abstract: In some embodiments, a virtual bass generation method including steps of: performing harmonic transposition on low frequency components of an input audio signal (typically, bass frequency components expected to be inaudible during playback of the input audio signal using an expected speaker or speaker set) to generate transposed data indicative of harmonics (which are expected to be audible during playback, using the expected speaker(s), of an enhanced version of the input audio which includes the harmonics); generating an enhancement signal in response to the transposed data; and generating an enhanced audio signal by combining (e.g., mixing) the enhancement signal with the input audio signal. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

Abstract: The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a (64) channel filter bank using a prototype filter length of (640) coefficients and a system delay of (319) samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip.

Abstract: The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a 64 channel filter bank using a prototype filter length of 640 coefficients and a system delay of 319 samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilising high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: A method for determining an inverse filter for altering the frequency response of a loudspeaker so that with the inverse filter applied in the loudspeaker's signal path the inverse-filtered loudspeaker output has a target frequency response, and optionally also applying the inverse filter in the signal path, and a system configured (e.g., a general or special purpose processor programmed and configured) to determine an inverse filter. In some embodiments, the inverse filter corrects the magnitude of the loudspeaker's output. In other embodiments, the inverse filter corrects both the magnitude and phase of the loudspeaker's output. In some embodiments, the inverse filter is determined in the frequency domain by applying eigenfilter theory or minimizing a mean square error expression by solving a linear equation system.

Abstract: An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilising high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems, and offers significantly reduced computational complexity. This is accomplished by means of frequency translation or folding in the subband domain, preferably integrated with spectral envelope adjustment in the same domain. The concept of dissonance guard-band filtering is further presented. The proposed invention offers a low-complexity, intermediate quality HFR method useful in speech and natural audio coding applications.

Abstract: The present invention relates to a new method and apparatus for improvement of High Frequency Reconstruction (HFR) techniques using frequency translation or folding or a combination thereof. The proposed invention is applicable to audio source coding systems. This is accomplished by means of frequency translation in the frequency domain with spectral envelope adjustment in the same domain. The proposed invention offers a low-complexity HFR method useful in speech and natural audio coding applications.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: An apparatus for processing a plurality of real-valued subband signals using a first real-valued subband signal and a second real-valued subband signal to provide at least a complex-valued subband signal comprises a multiband filter for providing an intermediate real-valued subband signal and a calculator for providing the complex-valued subband signal by combining a real-valued subband signal from the plurality of real-valued subband signals and the intermediate subband signal.