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. The method offers improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filter banks used in high frequency reconstruction (HFR) or parametric stereo systems.
Abstract:
An apparatus and method are disclosed for filtering and performing high frequency reconstruction of an audio signal. The apparatus includes an analysis filter bank, a phase shifter, a high frequency reconstructor, and a synthesis filter bank. The analysis filterbank receives real-valued time domain input audio samples and generates complex valued subband samples. The phase shifter shifts a phase of the complex-valued subband samples by an arbitrary amount. The high frequency reconstructor modifies at least some of the complex valued subband samples. A phase shifter unshifts a phase of the modified complex-valued subband samples by the arbitrary amount. The synthesis filter bank receives the modified complex valued subband samples and generates time domain output audio samples. The analysis filter bank comprises analysis filters that are complex exponential modulated versions of a prototype filter with an arbitrary phase shift.
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:
An apparatus for generating real-valued output audio samples is disclosed. The apparatus includes a memory that stores complex-valued input subband samples, real-valued demodulated samples, and the real-valued output audio samples. The apparatus also incudes a phase shifter that shifts a phase of the complex-valued input subband samples by an amount equal to a previously added phase shift and a complex-valued synthesis filter bank that generates the real-valued output audio samples in response to the complex-valued input subband samples, the real-valued demodulated samples, and prototype filter coefficients.
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. The method offers improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filter banks used in high frequency reconstruction (HFR) or parametric stereo systems.
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:
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 Δf. The system further comprises a nonlinear processing unit (502) configured to determine a set of synthesis subband signals from the set of analysis subband signals using a transposition order P; wherein the set of synthesis subband signals comprises a portion of the set of analysis subband signals phase shifted by an amount derived from the transposition order P; and a synthesis filter bank (504) configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein the synthesis filter bank (504) has a frequency resolution of FΔf ; with F being a resolution factor, with F≧1; wherein the transposition order P is different from the resolution factor F.
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 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:
Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.