Abstract: The present invention relates to a method of encoding and decoding an audio signal. The invention further relates to an arrangement for encoding and decoding an audio signal. The invention further relates to a computer-readable medium comprising a data record indicative of an audio signal and a device for communicating an audio signal having been encoded according to the present invention. By the method of encoding, a double description of the signal is obtained, where the encoding comprises two encoding steps, a first standard encoding and an additional second encoding. The second encoding is able to give a coarse description of the signal, such that a stochastic realization can be made and appropriate parts can be added to the decoded signal from the first decoding. The required description of the second encoder in order to make the realization of a stochastic signal possible requires a relatively low bit rate, while other double/multiple descriptions require a much higher bit rate.

Abstract: There is described a method of encoding a signal (s(n)) in a coder (400) to generate a corresponding encoded bit-stream (x(n); STP). The method comprises steps of: (a) processing the signal (s(n)) to determine main sinusoidal components and transient components thereof to generate corresponding component parameters; (b) processing the signal (s(n)) by removing the sinusoidal and transient components therefrom to generate a residual signal (r(n)); (c) processing the residual signal (r(n)) to determine a spectral representation (PSD) and determining therefrom a spectral broadening measure (SBM); (d) determining from the residual signal (r(n)) spectral envelope parameters by linear prediction; and (e) combining the components parameters together with the spectral envelope parameters and the spectral broadening measure to generate the encoded bit-stream. The method is capable of reducing noise that would otherwise arise were the bitstream to be decoded not subjected to such spectral broadening.

Abstract: An audio encoding device (100) comprises first encoding means (101, 111) for encoding transient signal components and/or sinusoidal signal components of an audio signal (x(n)) and producing a residual signal (z(n)), and second encoding means for encoding the residual signal. The second encoding means comprise filter means (122) for selecting at least two frequency bands of the residual signal. The selected frequency bands (LF, HF) of the residual signal (z(n)) are encoded by a first encoding unit (123) and a second encoding unit (124) respectively. The first encoding unit (123) may comprise a waveform encoder, such as a time-domain encoder, while the second encoding unit (124) may comprise a noise encoder.

Abstract: 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).

Abstract: 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.

Abstract: A method of encoding a digital audio signal, wherein for each time segment the signal is spectrally flattened to obtain a spectrally flattened signal (r) and possibly spectral flattening parameters (LPP). The spectrally flattened signal is modelled by an excitation signal comprising a first partial excitation signal (px) conforming to an excitation signal generated by an RPE or CELP technique, and a second partial excitation signal (PEp) being a set of extra pulses with arbitrary positions and amplitudes. An audio bit stream as comprising the first and second partial excitation signals is generated. The extra pulses can be added to the excitation signal at positions in time that correspond to the time of occurrence of the spike, or preferably at positions in time of an RPE time grid.

Abstract: 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 (?A) 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.

Abstract: A device (1) is arranged for synthesizing sound represented by sets of parameters, each set comprising noise parameters (NP) representing noise components of the sound and optionally also other parameters representing other components, such as transients and sinusoids. Each set of parameters may correspond with a sound channel, such as a MIDI voice. In order to reduce the computational load, the device comprises a selection unit (2) for selecting a limited number of sets from the total number of sets on the basis of a perceptual relevance value, such as the amplitude or energy. The device further comprises a synthesizing unit (3) for synthesizing the noise components using the noise parameters of the selected sets only.

Abstract: 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.

Abstract: 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.

Abstract: The invention relates to a linking unit 100, a parametric encoder 400 and a method for generating linking information L indicating components of consecutive extended segments sp and sc which may be linked together in order to form a sinusoidal track. The segments sp and sc approximate consecutive segments of a sinusoidal audio or speech signal s. The linking unit comprises a calculating unit 120 for generating a similarity matrix S(m,n) in response to received sinusoidal code data and an evaluating unit 140 for receiving and evaluating said similarity matrix S in order to generate said linking information by selecting those pairs of components m,n the similarity of which is maximal. According to the invention the calculating unit 120 is adapted to calculate the similarity matrix S by additionally considering information about the phase consistency between the components of the extended previous segment sp and the extended current segment sc.

Abstract: An encoder includes a segmentation unit for segmenting an audio or speech signal into at least one segment and a calculation unit for calculating sinusoidal code data in the form of frequency and amplitude data of a given extension from the segment such that the extension approximates the segment for a given criterion. The calculation of the sinusoidal code data ?ki, dji and eji for the segment x(n) is carried out according to the following extension {circumflex over (x)}: x ? = ? i = 1 L ? ? ? j = 0 J - 1 ? ? [ d j i ? f j ? ( n ) ? cos ? ( ? i ? ( n ) ) + e j i ? f j ? ( n ) ? sin ( ? i ? ( n ) ] . Fig . ? 1.

Abstract: 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).

Abstract: 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 a second filtered signal, the generating being stable and causal; amplifying a of the second filtered signals with a prediction coefficient; obtaining the first filtered signal based on the a second filtered signal storing a first signal related to the input signal in a buffer; and retrieving from the buffer a delayed signal. Further a synthesis filtering method, an inverse filter device and a synthesis filter device are provided.

Abstract: In a sinusoidal audio encoder it is known to use different time scales for analyzing different parts of the frequency spectrum. In prior art encoders sub-band filtering is used to split the input signal into a number of sub bands. By splitting the input signal into sub-bands, it can happen that a signal component at the boundary of two sub-bands results in a representation in both sub-band signals. This double representation of signal components can lead to several problems when coding these components. According to the present invention it is proposed to use preventing means (46, 48, 58, 68; 88, 92, 96) to avoid signal components to have multiple representations.

Abstract: Coding of an audio signal (x) is provided where an indicator (ab P1k) of the frequency variation of sinusoidal components of the signal is used in the tracking algorithm of a sinusoidal coder (13) 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.

Abstract: A known encoder 100 comprises a segmentation unit 110 for segmenting an audio or speech signal s into at least one segment x(n) and a calculation unit 120 for calculating sinusoidal code data in the form of frequency and amplitude data of a given extension {circumflex over (x)}(n) from the segment x(n) such that the extension {circumflex over (x)}(n) approximates the segment x(n) as good as possible for a given criterion. It is the object of the invention to improve the known encoder such that the calculation of said sinusoidal code data can be carried out in a simpler and cheaper way.

Abstract: The invention relates to a parametric encoder for encoding an audio or speech signal into sinusoidal code data. Such parametric encoders typically comprise a segmentation unit 120 for segmenting said signal s into at least one single scale segment xm(n) with m=1 . . . M and for outputting the samples xm(0), . . . ,xm(L−1) of said segment xm(n) and comprise a sinusoidal estimation unit 140 for estimating the sinusoidal code data representing said segment xm(n) from said samples. It is the object of the invention to improve a parametric encoder and method such that the achievement of a required time-frequency resolution trade-off is facilitated. This is achieved by embodying the segmentation unit 120 such that it carries out a frequency-warping operation in order to transform the output samples xm(0), . . .

Abstract: The invention relates to a linking unit 100, a parametric encoder 400 and a method for generating linking information L indicating components of consecutive extended segments sp and sc which may be linked together in order to form a sinusoidal track. The segments sp and sc approximate consecutive segments of a sinusoidal audio or speech signal s. The linking unit comprises a calculating unit 120 for generating a similarity matrix S(m,n) in response to received sinusoidal code data and an evaluating unit 140 for receiving and evaluating said similarity matrix S in order to generate said linking information by selecting those pairs of components m,n the similarity of which is maximal. According to the invention the calculating unit 120 is adapted to calculate the similarity matrix S by additionally considering information about the phase consistency between the components of the extended previous segment sp and the extended current segment sc.

Abstract: 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).