摘要:
A method and a device for processing a stereo signal obtained from an encoder, which codes an N-channel audio signal into spatial parameters (P) and a stereo down-mix comprising first and second stereo signals (L 0 , R 0 ). A first signal and a third signal are added in order to obtain a first output signal (L 0w ), wherein the first signal QL 0wL ) comprises the first stereo signal (L 0 ) modified by a first complex function (g 1 ), and the third signal (L 0wR ) comprises the second stereo signal (R 0 ) modified by a third complex function (g 3 ). A second signal and a fourth signal are added to obtain a second output signal (R 0w ). The fourth signal (R 0wR ) comprises the second stereo signal (R 0 ) modified by a fourth complex function (g 4 ), and the second signal (R 0wL ) comprises the first stereo signal (L 0 ) modified by a second complex function (g 2 ). The complex functions (g 1 ,g 2 ,g 3 ,g 4 ) are functions of the spatial parameters (P) and are chosen such that an energy value of the difference (L 0wL -P 0wL ) between the first signal and the second signal is larger than or equal to the energy value of the sum (L 0wL +R 0wL ) of the first and the second signal and the energy value of the difference (R 0wR -L 0wR ) between the fourth signal and the third signal is larger than or equal to the energy value of the sum (R 0wR +L 0wR ) of the fourth signal and the third signal.
摘要:
A spatial decoder unit (23) is arranged for transforming one or more audio channels (s; 1, r) into a pair of binaural output channels (Ib, rb). The device comprises a parameter conversion unit (234) for converting the spatial parameters (sp) into binaural parameters (bp) containing binaural information. The device additionally comprises a spatial synthesis unit (232) for transforming the audio channels (L, R) into a pair of binaural signals (Lb, Rb) while using the binaural parameters (bp). The spatial synthesis unit (232) preferably operates in a transform domain, such as the QMF domain.
摘要:
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.
摘要:
A device (10) for enhancing a multi-channel (e.g. stereo) audio signal has a parameter adjustment unit (13) for adjusting an original parameter (α, ILD, ICC) which represents an original inter-channel property of the audio signal. The device further comprises a processing unit (11) for processing the audio signal so as to produce an enhanced audio signal having the adjusted parameter (α′, ILD′, ICC′). The device allows stereo widening or other multi-channel signal enhancements without introducing artifacts.
摘要:
A method of synthesizing a first (L) and a second (R) output signal from an input signal (x). The method comprises: filtering (201) the input signal to generate a filtered signal; obtaining a correlation parameter indicative of a desired correlation between the first and second output signals; obtaining a level parameter (c) indicative of a desired level difference between the first and second input signals; and transforming the input signal and the filtered signal by a matrixing operation (203) into the first and second output signals, where the matrixing operation depends on the correlation parameter and the level parameter.
摘要:
A method of encoding input signals (1, r) to generate encoded data (100) is provided. The method involves processing the input signals (1, r) to determine first parameters (Õ 1 , Õ 2 ) describing relative phase difference and temporal difference between the signals (1, r), and applying these first parameters (Õ 1 , Õ 2 ) to process the input signals to generate intermediate signals. The method involves processing the intermediate signals to determine second parameters (±; IID, Á) 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. The method also involves quantizing the first parameters, the second parameters, and dominant and residual signals (m, s) to generate corresponding quantized data for subsequent multiplexing to generate the encoded data (100).
摘要:
There is described a method of encoding input signals (CHI to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data comprising down-mix output signals (610, 620) together with complementary parametric data (600). The method includes a first step of down-mixing input signals (CHI to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CHI to CH3; 400 to 450) during down-mixing to generate said parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CHI to CH3; 400 to 450) involves including information in the down-mix signals (610, 620) which is useable during subsequent decoding of the down-mix output signals (610, 620) and the parametric data (600) to determine at least some parameter data and thereby enabling representations of the input signals (CHI to CH3; 400 to 450) to be subsequently regenerated. Coders for use in the encoder (5; 15) for performing essential signal processing operations therein are also elucidated.
摘要:
Multi-channel audio signals are coded into a monaural audio signal and information allowing to recover the multi-channel audio signal from the monaural audio signal and the information. The information is generated by determining a first portion of the information for a first frequency region of the multi-channel audio signal, and by determining a second portion of the information for a second frequency region of the multi-channel audio signal. The second frequency region is a portion of the first frequency region and thus is a sub-range of the first frequency region. The information is multi-layered enabling a scaling of the decoding quality versus bit rate.
摘要:
The invention relates to a method of decreasing the dynamic range of a signal comprising the steps of:- determining a property of the signal,- determining a limitation parameter based on the property of the signal, - limiting the signal by means of the limitation parameter, - clipping the limited signal.