摘要:
An audio apparatus comprises a processor (101) for providing a set of audio channels. A prediction circuit (103) generates a predicted signal for a first channel by adaptive filtering of a second channel by an adaptive filter. An adaptation processor (105) adapts the adaptive filter to minimize a cost function indicative of a difference between the predicted signal and the first channel. A compensation processor (107) then generates a non-predicted signal by compensating the first signal for the predicted signal and a distribution processor (109) generates an output set of audio channels by distributing at least the predicted signal and the non-predicted signal over the output set of audio signals where the distribution is different for the predicted signal and the non-predicted signal. The cross-channel predictive filtering provides signal components that represent different spatial characteristics of the originating sound and which are therefore advantageously distributed differently for the output channels.
摘要:
A method of modifying an audio signal comprises the steps of analyzing the input audio signal (x) so as to produce a set of filter parameters (p) and a residual signal (r), modifying the set of filter parameters (p) so as to produce a modified set of filter parameters (p′), and synthesizing an output audio signal (y) using the modified set of filter parameters (p′) and the residual signal (r). The set of filter parameters (p) comprises poles (λA) and coefficients (a; c). The step of modifying the filter parameters (p) involves interpolating lattice filter reflection coefficients (c) so as to scale the spectral envelope of the audio signal.
摘要:
In a method of encoding input signals (CH1 to CH3; 400 to 450) in a multi-channel encoder (5; 15) to generate corresponding output data having down-mix output signals (610, 620) together with complementary parametric data (600), the method includes a first step of down-mixing input signals (CH1 to CH3; 400 to 450) to generate the corresponding down-mix output signals (610, 620), and a second step of processing the input signals (CH1 to CH3; 400 to 450) during down-mixing to generate the parametric data (600) complementary to the down-mix output signals (610, 620). Processing of the input signals (CH1 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 (CH1 to CH3; 400 to 450) to be subsequently regenerated.
摘要:
Encoding (2) a signal (A) is provided, wherein frequency and amplitude information of at least one sinusoidal component in the signal (A) is determined (20), and sinusoidal parameters (f,a) representing the frequency and amplitude information are transmitted (22), and wherein further a phase jitter parameter (p) is transmitted, which represents an amount of phase jitter that should be added during restoring the sinusoidal component from the transmitted sinusoidal parameters (f,a).
摘要:
Coding (1) of an audio signal is provided including estimating (110) a position of a transient signal component in the audio signal, matching (111,112) a shape function on the transient signal component in case the transient signal component is gradually declining after an initial increase, which shape function has a substantially exponential initial behavior and a substantially logarithmic declining behavior; and including (15) the position and shape parameters describing the shape function in an audio stream (AS).
摘要:
An inverse filtering method, comprising: generating a first filtered signal based on an input signal; and combining the first filtered signal with the input signal for obtaining a residual signal. The generating comprises: generating at least two second filtered signals, each of said second filtered signals not significantly delayed in time relative to each other, the generating being stable and causal; and amplifying at least one of the second filtered signals with a prediction coefficient.
摘要:
Coding of an audio signal is provided where an indicator of the frequency variation of sinusoidal components of the signal is used in the tracking algorithm of a sinusoidal coder where sinusoidal parameters from appropriate sinusoids from consecutive segments are linked. By applying an indicator such as a warp factor or polynomial fitting, more accurate tracks are obtained. As a result, the sinusoids can be encoded more efficiently. Furthermore, a better audio quality can be obtained by improved phase continuation.
摘要:
An encoder (100) for encoding a multi-channel audio signal comprises a prediction processor (101) for generating two residual signals for two signal components of the multi-channel signal by linear prediction which is associated with psycho-acoustic characteristics and which specifically uses psycho-acoustic prediction filters; a rotation processor (105) for rotating the combined signal of the two residual signals to generate a main signal and a side signal, in which the energy of the main signal is maximized and the energy of the side signal is minimized; an encoding processor (109) for encoding the main and preferably the side signal; and an output processor (111) for generating an output signal data, prediction parameters and rotation parameters.
摘要:
A device (2) for changing the pitch of an audio signal (r), such as a speech signal, comprises a sinusoidal analysis unit (21) for determining sinusoidal parameters of the audio signal (r), a parameter production unit (22) for predicting the phase of a sinusoidal component, and a sinusoidal synthesis unit (23) for synthesizing the parameters to produce a reconstructed signal (r′). The parameter production unit (22) receives, for each time segment of the audio signal, the phase of the previous time segment to predict the phase of the current time segment.
摘要:
A hybrid sinusoidal/pulse excitation encoder has been recently proposed for constructing a scalable audio encoder The base layer consisting of data supplied by the sinusoidal encoder retains the main features of the input signal achieving medium to high quality audio at a very low bit rate. Quality can be further enhanced by adding excitation signal layers associated with a decreasing decimation that increasingly model more subtle aspects of the original signal. The invention provides a method of mixing the different excitation signal layers so that the full concept of scalability is realised without compromising the quality of the encoded signals. The mixing is controlled via a quality parameter that weights the significance of previous layers when constructing a new higher layer.