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. The system comprises an analysis filter bank (501) comprising an analysis transformation unit (601) having a frequency resolution of Δf; and an analysis window (611) having a duration of D.sub.A; the analysis filter bank (501) being configured to provide a set of analysis subband signals from the low frequency component of the signal; a nonlinear processing unit (502, 650) configured to determine a set of synthesis subband signals based on a portion of the set of analysis subband signals, wherein the portion of the set of analysis subband signals is phase shifted by a transposition order T; and a synthesis filter bank (504) comprising a synthesis transformation unit (602) having a frequency resolution of QΔf; and a synthesis window (612) having a duration of D.sub.S; the synthesis filter bank (504) being configured to generate the high frequency component of the signal from the set of synthesis subband signals; wherein Q is a frequency resolution factor with Q≧1 and smaller than the transposition order T; and wherein the value of the product of the frequency resolution Δf and the duration D.sub.A of the analysis filter bank is selected based on the frequency resolution factor Q.

The present document relates to multichannel audio coding and more precisely to techniques for discrete multichannel audio encoding and decoding. In particular, the present document relates to systems and method for coding soundfields. An audio encoder (200) configured to encode a frame of a soundfield signal (110) comprising a plurality of audio signals is described. The audio encoder (200) comprises a transform determination unit (203, 204) configured to determine an energy-compacting orthogonal transform (V) based on the frame of the soundfield signal (110). Furthermore, the encoder (200) comprises a transform unit (202) configured to apply the energy-compacting orthogonal transform (V) to the frame of the soundfield signal (110), and configured to provide a frame of a rotated soundfield signal (112) comprising a plurality of rotated audio signals (E1, E2, E3). The audio encoder (200) comprises a waveform encoding unit (103) configured to encode a first rotated audio signal (E1) of the plurality of rotated audio signals (E1, E2, E3), and a parametric encoding unit (104) configured to determine a set of spatial parameters (ae2, be2) for determining a second rotated audio signal (E2) of the plurality of rotated audio signals (E1, E2, E3) based on the first rotated audio signal (E1).

An encoder for encoding a parametric spectral representation (f) of auto-regressive coefficients that partially represent an audio signal. The encoder includes a low-frequency encoder configured to quantize elements of a part of the parametric spectral representation that correspond to a low-frequency part of the audio signal. It also includes a high-frequency encoder configured to encode a high-frequency part (f.sup.H) of the parametric spectral representation (f) by weighted averaging based on the quantized elements ({circumflex over (f)}.sup.L) flipped around a quantized mirroring frequency ({circumflex over (f)}.sub.m), which separates the low-frequency part from the high-frequency part, and a frequency grid determined from a frequency grid codebook in a closed-loop search procedure. Described are also a corresponding decoder, corresponding encoding/decoding methods and UEs including such an encoder/decoder.

An acoustic signal coding apparatus includes a subband classifier that classifies subbands obtained by dividing a frequency-domain spectrum into a plurality of perceptually important first-category subbands and the other subbands referred to as second-category subbands according to at least one of measures in terms of energy and peak property, a subband peak-algebraic vector quantization (SBP-AVQ) vector generator that generates an SBP-AVQ vector by collecting a maximum peak from each first-category subband, outputs the generated SBP-AVQ vector, and outputs peak position information indicating the positions of the maximum peaks, a bit distributor that distributes bits for AVQ coding to the SBP-AVQ vector and the second-category subband vector, and an AVQ coder that performs AVQ coding on the SBP-AVQ vector and the second-category subband vector.

A microphone assembly includes an acoustic sensor and a voice activity detector on an integrated circuit coupled to an external-device interface. The acoustic sensor produces an electrical signal representative of acoustic energy detected by the sensor. A filter bank separates data representative of the acoustic energy into a plurality of frequency bands. A power tracker obtains a power estimate for at least one band, including a first estimate based on relatively fast changes in a power metric of the data and a second estimate based on relatively slow changes in a power metric of the data. The presence of voice activity in the electrical signal is based upon the power estimate.

The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor. To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.

An application-specific integrated circuit for accelerated encoding and decoding and a method, which are related to the technical field of Bluetooth mobile communication. The application-specific integrated circuit for accelerated encoding and decoding includes: a hardware accelerator, wherein the hardware accelerator includes a pre-processing and pronation processing module, which performs a pre-processing and pronation processing of data, a discrete Fourier transform module is used for performing a multi-level discrete Fourier transform, in an accelerated low-delay modified discrete cosine transform operation LD-MDCT and/or an accelerated the low-delay inverse modified discrete cosine transform operation LD-IMDCT. The application-specific integrated circuit for accelerated encoding and decoding and a method of the present invention adopts an ASIC application-specific integrated circuit, and adopts multi-level discrete Fourier transforms, so that the complex operations are completed by the ASIC application-specific integrated circuit.

There is inter alia disclosed an apparatus for spatial audio encoding comprising: means for analysing a plurality of spatial audio parameter sets associated with a frame of one or more audio signals, wherein the plurality of spatial audio parameter sets are associated with a plurality of subframes, a plurality of frequency sub bands and a plurality of sound source directions for the frame of the one or more audio signals; and means for determining from the analysis of the plurality of spatial audio parameter sets at least one spatial audio parameter set for subframes of the frame of the one or more audio signals.

The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S. The subband processing unit performs a block based nonlinear processing wherein the magnitude of samples of the synthesis subband signal are determined from the magnitude of corresponding samples of the analysis subband signal and a predetermined sample of the analysis subband signal. In addition, the system comprises a synthesis filterbank configured to generate the time stretched and/or frequency transposed signal from the synthesis subband signal.

An encoding method includes dividing a to-be-encoded time-domain signal into a low band signal and a high band signal, performing encoding on the low band signal to obtain a low frequency encoding parameter, performing encoding on the high band signal to obtain a high frequency encoding parameter, obtaining a synthesized high band signal, performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal, and calculating a high frequency gain based on the high band signal and the short-time filtering signal.