Abstract: A device for processing parameters representing Head-Related Transfer Functions includes an input stage configured to receive audio signals of sound sources, a determinor configured to receive reference parameters representing Head-Related Transfer Functions and configured to determine, from the audio signals, position information representing positions and/or directions of the sound sources. A processor is configured to process the audio signals; and an influencer is configured to influence the processing of the audio signals based on the position information yielding an influenced output audio signal.

Abstract: A device for processing audio data includes a summation unit configured to receive a number of audio input signals for generating a summation signal, a filter unit configured to filter the summation signal dependent on filter coefficient resulting in at least two audio output signals. A parameter conversion unit is configured to receive position information, which is representative of spatial positions of sound sources of the audio input signals, and spectral power information which is representative of a spectral power of the audio input signals. The parameter conversion unit is configured to generate the filter coefficients based the position information and the spectral power information. The parameter conversion unit is further configured to receive transfer function parameters and generate the filter coefficients in dependence on the transfer function parameters.

Abstract: A method of generating parameters representing Head-Related Transfer Functions, the method comprising the steps of a) sampling with a sample length (n) a first time-domain HRTF impulse response signal using a sampling rate (fs) yielding a first time-discrete signal, b) transforming the first time-discrete signal to the frequency domain yielding a first frequency-domain signal, c) splitting the first frequency-domain signal into sub-bands, and d) generating a first parameter of the sub-bands based on a statistical measure of values of the sub-bands.

Abstract: A method of generating parameters representing Head-Related Transfer Functions, the method comprising the steps of a) sampling with a sample length (n) a first time-domain HRTF impulse response signal using a sampling rate (fs) yielding a first time-discrete signal, b) transforming the first time-discrete signal to the frequency domain yielding a first frequency-domain signal, c) splitting the first frequency-domain signal into sub-bands, and d) generating a first parameter of the sub-bands based on a statistical measure of values of the sub-bands.

Abstract: An encoding device (1) for converting a first number (M) of input audio channels into a second, smaller number (N) of output audio channels comprises at least one conversion unit (12) for converting a first signal (Lf; Rf; Co) and a second signal (Lr; Rr; Le) into a third signal (L; R; C) and a fourth signal (Ls; Rs; Cs). The third, dominant signal contains most of the signal energy of the first and second signals, while the fourth, residual signal contains the remainder of said signal energy. The encoding device is arranged for using the third signal (L; R; C) to produce an output signal and for outputting the fourth signal (Ls; Rs; Cs). A decoding device (2) for converting a first number (N) of input audio channels into a second, larger number (M) or output audio channels comprises at least one conversion unit (24) for converting a first signal (L; R; C) and a second signal (Ld; Rd; Ld) into a third signal (Lf, Rf; Co) and a fourth signal (Lr; Rr; Le).

Abstract: A device (100) for processing audio data (101), wherein the device (100) comprises a summation unit (102) adapted to receive a number of audio input signals for generating a summation signal, a filter unit (103) adapted to filter said summation signal dependent on filter coefficient, (SF1, SF2) resulting in at least two audio output signals (OS1, OS2), a parameter conversion unit (104) adapted to receive, on the one hand, position information, which is representative of spatial positions of sound sources of said audio input signals, and, on the other hand, spectral power information which is representative of a spectral power of said audio input signals, wherein the parameter conversion unit is adapted to generate said filter coefficients (SF1, SF2) on the basis of the position information and the spectral power information, and wherein the parameter conversion unit (104) is additionally adapted to receive transfer function parameters and generate said filter coefficients in dependence on said transfer functi

Abstract: A method of generating parameters representing Head-Related Transfer Functions, the method comprising the steps of a) sampling with a sample length (n) a first time-domain HRTF impulse response signal using a sampling rate (fs) yielding a first time-discrete signal, b) transforming the first time-discrete signal to the frequency domain yielding a first frequency-domain signal, c) splitting the first frequency-domain signal into sub-bands, and d) generating a first parameter of the sub-bands based on a statistical measure of values of the sub-bands.

Abstract: This document gives a technical description of a multi-channel parametric audio coding system. The goal of this system is to describe an m-channel signal by an n-channel signal, with n<m, and parameters describing a spatial image in order to reconstruct the m-channel signal.

Abstract: A method and a device are described for processing a stereo signal obtained from an encoder, which encodes an N-channel audio signal into spatial parameters (P) and a stereo down-mix comprising first and second stereo signals (Lo, R0). A first signal and a third signal are added in order to obtain a first output signal (Low), wherein the first signal (L0wL) comprises the first stereo signal (Lo) modified by a first complex function (g1), and the third signal (L0wR) comprises the second stereo signal (R0) modified by a third complex function (g3). A second signal and a fourth signal are added to obtain a second output signal (R0w). The fourth signal (R0wR) comprises the second stereo signal (R0) modified by a fourth complex function (g4), and the second signal (R0wL)comprises the first stereo signal (L0) modified by a second complex function (g2).

Abstract: There is described a multi-channel encoder (10; 600) for processing input signals conveyed in N input channels to generate corresponding output signals conveyed in M output channels together with complementary parametric data; M and N are integers wherein N>M. The encoder (10; 600) includes a down-mixer for down-mixing the input signals to generate the corresponding output signals, the encoder also comprising an analyser for processing the input signals to generate the parameter data, said parametric data describing mutual differences between the N channels of input signal to allow for regenerating during decoding one or more of the N channels of input signals from the M channels of output signal. Such an encoder (10; 600) is capable of providing highly efficient data encoding and also of being backwards compatibility with relatively simpler decoders having fewer than N decoding output channels. The invention also concerns decoders (800) compatible with such a multi-channel encoder (10; 600).

Abstract: Method for processing a stereo signal comprising: Encoding á N-channel audio signal in a stereo signal (Lo, Ro) and spatial parameters (wl, wr), processing the stereo signal using the spatial parameters for generating a processed stereo signal (low, Row). The matrix of the processed stereo signal can be discribed as the matrix of the stereo signal, multiplied by a filter matrix (H) which element are filter functions (H1, H2, H3, H4) operated with spatial parameters (wl, wr) and a constant (a). The filter functions are time invariant and selected so that the matrix is invertible.

Abstract: A method of encoding input signals (l, r) to generate encoded data (100) is provided. The method involves processing the input signals (l, r) to determine first parameters (?1,?2) describing relative phase difference and temporal difference between the signals (l, r), and applying these first parameters (?1, ?2) to process the input signals to generate intermediate signals. The method involves processing the intermediate signals to determine second parameters (?; IID,?) describing angular rotation of the first intermediate signals to generate a dominant signal (m) and a residual signal (s), the dominant signal (m) having a magnitude or energy greater than that of the residual signal (s). These second parameters are applicable to process the intermediate signals to generate the dominant (m) and residual (s) signals.

Abstract: Parametric stereo coders use perceptually relevant parameters of the input signal to describe spatial properties. One of these parameters is the phase difference between the input signals (ITD or IPD). This time difference only determines the relative time difference between the input signals, without any information about how these time differences should be divided over the output signals in the decoder. An additional parameter is included in the encoded signal that describes how the ITD or IPD should be distributed between the output channels.

Abstract: The audio reproduction apparatus (100) for sports training purposes comprises a tempo derivation unit (103) for deriving a selected tempo (T) on the basis of a data signal (d1, d2, d3) e.g. from a sports measurement device such as a heart rate meter; and an audio conditioning unit (104) arranged to deliver based on the input audio signal the output audio signal, with a tempo within a predefined accepted deviation from the selected tempo (T), whereby the audio conditioning unit (104) comprises a tempo calculation unit (106) arranged to calculate an input tempo (TI) of the input audio signal, and the audio conditioning unit (104) is arranged to deliver the output signal in dependence of the input tempo (T).

Abstract: The encoder transforms the audio signals (x(n),y(n)) from the time domain to audio signal (X(k),Y(k)) in the frequency domain, and determines the cross-correlation function (Ri, Pi) in the frequency domain. A complex coherence value (Qi) is calculated by summing the (complex) cross-correlation function values (Ri, Pi) in the frequency domain. The inter-channel phase difference (IPDi) is estimated by the argument of the complex coherence value (Qi), and the inter-channel coherence (ICi) is estimated by the absolute value of the complex coherence value (Qi). In the prior art a computational intensive Inverse Fast Fourier Transformation and search for the maximum value of the cross-correlation function (Ri; Pi) in the time domain are required.

Abstract: The invention describes a system (1) for performing automatic dubbing on an incoming audio-visual stream (2). The system (1) comprises means (3, 7) for identifying the speech content in the incoming audio-visual stream (2), a speech-to-text converter (13) for converting the speech content into a digital text format (14), a translating system (15) for translating the digital text (14) into another language or dialect; a speech synthesizer (19) for synthesizing the translated text (18) into a speech output (21), and a synchronizing system (9, 12, 22, 23, 26, 31, 33, 34, 35) for synchronizing the speech output (21) to an outgoing audio-visual stream (28). Moreover the invention describes an appropriate method for performing automatic dubbing on an audio-visual stream (2).

Abstract: A method of encoding a multi-channel audio signal including at least a first signal component (LF), a second signal component (LR) and a third signal component (RF). The method comprises the steps of encoding the first and second signal components by a first parametric encoder (202) resulting in a first encoded signal (L) and a first set of encoding parameters (P2); encoding the first encoded signal and a further signal (R) by a second parametric encoder (201), resulting in a second encoded signal (T) and a second set of encoding parameters (P1), where the further signal is derived from at least the third signal component; and representing the multi-channel audio signal at least by a resulting encoded signal (T) derived from at least the second encoded signal, by the first set of encoding parameters and by the second set of encoding parameters.

Abstract: A method of generating a monaural signal (S) comprising a combination of at least two input audio channels (L, R) is disclosed. Corresponding frequency components from respective frequency spectrum representations for each audio channel (L(k), R(k)) are summed (46) to provide a set of summed frequency components (S(k)) for each sequential segment. For each frequency band (i) of each of sequential segment, a correction factor (m(i)) is calculated (45) as function of a sum of energy of the frequency components of the summed signal in the band formula (I) and a sum of components of the input audio channels in the band formula (II). Each summed frequency component is corrected (47) as a function of the correction factor (m(i)) for the frequency band of said component.

Abstract: Parametric stereo coders use perceptually relevant parameters of the input signal to describe spatial properties. One of these parameters is the phase difference between the input signals (ITD or IPD). This time difference only determines the relative time difference between the input signals, without any information about how these time differences should be divided over the output signals in the decoder. An additional parameter is included in the encoded signal that describes how the ITD or IPD should be distributed between the output channels. To this goal the delay between a computed monaural signal and one of the input signals is used.

Abstract: A method of filtering an information signal (x(t)), the method comprising modifying frequency domain components (X(k,n)) of the information signal according to a desired filter response (F(k,n)); wherein the step of modifying frequency domain components further comprises modifying (105) frequency domain components of a first frame of said information signal according to a first actual filter response (F?(k,n)), the first actual filter response being a function (?) of the desired filter response and information (108) related to a previous frame of the information signal.