Abstract: A method is provided for extending the frequency band of an audio signal during a decoding or improvement process. The method includes obtaining the decoded signal in a first frequency band, referred to as a low band. Tonal components and a surround signal are extracted from the signal from the low-band signal, and the tonal components and the surround signal are combined by adaptive mixing using energy-level control factors to obtain an audio signal, referred to as a combined signal. The low-band decoded signal before the extraction step or the combined signal after the combination step are extended over at least one second frequency band which is higher than the first frequency band. Also proved are a frequency-band extension device which implements the described method and a decoder including a device of this type.

Abstract: A method is provided for resampling an audio-frequency signal in an audio-frequency signal encoding or decoding operation. The resampling is carried out by a method of interpolation of an order greater than one. The method is such that the interpolated samples are obtained by calculating a weighted average of possible interpolation values calculated over a plurality of intervals covering the time location of the sample to be interpolated. A resampling device is provided, which implements the method, and also an encoder and decoder including at least one resampling device.

Abstract: A signal processing device, media, and method are provided, where a signal comprises a succession of samples distributed in successive frames. The processing is implemented during decoding of such a signal in order to replace at least one signal frame lost in decoding, and comprising in particular: a) searching, in a valid signal available to the decoder, for a signal segment of length corresponding to a period set as a function of the valid signal; b) analyzing a spectrum of the segment in order to determine spectral components of the segment; and c) synthesizing at least one replacement frame for the lost frame by construction of a synthesized signal from at least a portion of the spectral components.

Abstract: A method for processing a digital audio signal, including a series of samples distributed in consecutive frames, is implemented when decoding the signal in order to replace at least one signal frame lost during decoding. The method includes the following steps: a) searching, in a valid signal segment available when decoding, for at least one period in the signal, determined in accordance with the valid signal; b) analyzing the signal in the period, in order to determine spectral components of the signal in the period; c) synthesizing at least one frame for replacing the lost frame, by construction of a synthesis signal from: an addition of components selected among the predetermined spectral components, and a noise added to the addition of components. In particular, the amount of noise added to the addition of components is weighted in accordance with voice information of the valid signal, obtained when decoding.

Abstract: A method for resampling an audio-frequency signal with an output sampling frequency, for a current signal frame. The method is used when the preceding frame is sampled at a first sampling frequency which is different from a second sampling frequency of the current frame. The method includes: determining a first and second segments of the signal by adding samples at zero at the end of stored samples of the preceding frame and at the start of samples of the current frame, respectively; obtaining the first resampled segment and the second resampled segment by applying at least one resampling filter respectively to the first segment resampling the first frequency at the output frequency, and to the second segment resampling the second frequency at the output frequency; and combining the overlapping portion of the first and second resampled segments to obtain at least one portion of the resampled current frame.

Abstract: A method and apparatus are provided for controlling the shaping of encoding noise during the ADPCM encoding of a digital audio input signal. The noise-shaping is carried out through the use of feedback that comprises filtering noise. The method includes the following steps: obtaining a parameter for indicating a high spectral dynamic range of the signal, the parameter indicating a risk of instability of the feedback; detecting a risk of instability by comparing the indication parameter to at least one predetermined threshold; limiting the feedback in the event that a risk of instability is detected; and gradually reactivating the feedback over a predetermined number of frames subsequent to the current frame for which the feedback is limited. Also provided is an encoder with feedback, including a control module implementing the control method as described.

Abstract: A method is provided for processing pre-echo attenuation in a digital audio signal generated from a transform coding, wherein, at the decoding point, the method includes: detection of a position of attack in the decoded signal; determination of a pre-echo region preceding the position of attack detected in the decoded signal; calculation of attenuation factors per sub-block of the pre-echo region, according to at least the frame wherein the attack has been detected and the preceding frame; and pre-echo attenuation in the sub-blocks of the pre-echo region by the corresponding damping factors. The method also includes application of a filter for the spectral shaping of the pre-echo region on the current frame up to the detected position of the attack. A device and a decoder including the device are also proved for implementing the method.

Abstract: Methods and apparatus are provided for coding and decoding a digital audio signal. Decoding includes: decoding according to an inverse transform decoding of a previous frame of samples of the digital signal, which is received and coded according to a transform coding; and decoding according to a predictive decoding of a current frame of samples of the digital signal, which is received and coded according to a predictive coding. The predictive decoding of the current frame is a transition predictive decoding which does not use any adaptive dictionary arising from the previous frame. At least one state of the predictive decoding is reinitialized to a predetermined default value, and an add-overlap step combines a signal segment synthesized by predictive decoding of the current frame and a signal segment synthesized by inverse transform decoding, corresponding to a stored segment of the decoding of the previous frame.

Abstract: A method and device for resampling an audio frequency signal in an audio frequency signal coding or decoding. The method includes the following acts for each signal block to be resampled: determining, by adaptive linear prediction, a number of future signal samples, this number being defined as a function of a chosen resampling delay; constructing a resampling support vector from at least samples of the current block and determined future signal samples; applying a resampling filter to the samples of the resampling support vector.

Abstract: A method and device are provided for determining an optimized scale factor to be applied to an excitation signal or a filter during a process for frequency band extension of an audio frequency signal. The band extension process includes decoding or extracting, in a first frequency band, an excitation signal and parameters of the first frequency band including coefficients of a linear prediction filter, generating an excitation signal extending over at least one second frequency band, filtering using a linear prediction filter for the second frequency band. The determination method includes determining an additional linear prediction filter, of a lower order than that of the linear prediction filter of the first frequency band, the coefficients of the additional filter being obtained from the parameters decoded or extracted from the first frequency band and calculating the optimized scale factor as a function of at least the coefficients of the additional filter.

Abstract: The invention relates to a method for extending the frequency band of an audio signal during a decoding or improvement process comprising a step of decoding or extracting, in a first so-called low frequency band, an excitation signal and coefficients of a linear prediction filter. The method comprises the following steps: —obtaining a signal (UHB2(k), E403)) extended in at least a second frequency band higher than the first frequency band from an oversampled excitation signal extended in at least a second frequency band (UHB1(k), E401); —scaling (E406) the extended signal by means of a gain defined by subframe on the basis of an energy ratio of a frame and of a subframe; —filtering (E404) said scaled extended signal with a linear prediction filter of which the coefficients are derived from the coefficients of the low frequency band filter. The invention also relates to a frequency band extension device implementing the described method and a decoder comprising such a device.

Abstract: A method and apparatus for the parametric encoding of a stereo digital-audio signal. The method includes encoding a mono signal produced by downmixing applied to the stereo signal and encoding spatialization information of the stereo signal. Downmixing includes determining, for a predetermined set of frequency sub-bands, a phase difference between two stereo channels; obtaining an intermediate channel by rotating a first predetermined channel of the stereo signal through an angle obtained by reducing the phase difference; determining the phase of the mono signal from the phase of the signal that is the sum of the intermediate channel and the second stereo signal, and from a phase difference between, on the one hand, the signal that is the sum of the intermediate channel and the second channel and, on the other hand, the second channel of the stereo signal. Also provided are a decoding method, an encoder and a decoder.

Abstract: A signal processing device, media, and method are provided, where a signal comprises a succession of samples distributed in successive frames. The processing is implemented during decoding of such a signal in order to replace at least one signal frame lost in decoding, and comprising in particular: a) searching, in a valid signal available to the decoder, for a signal segment of length corresponding to a period set as a function of the valid signal; b) analyzing a spectrum of the segment in order to determine spectral components of the segment; and c) synthesizing at least one replacement frame for the lost frame by construction of a synthesized signal from at least a portion of the spectral components.

Abstract: An encoder and a method for encoding a digital signal are provided. The method includes encoding a preceding frame of samples of the digital signal according to a predictive encoding process, and encoding a current frame of samples of the digital signal according to a transform encoding process. The method is implemented such that a first portion of the current frame is also encoded by predictive encoding that is limited relative to the predictive encoding of the preceding frame by reusing at least one parameter of the predictive encoding of the preceding frame and only encoding the parameters of said first portion of the current frame that are not reused. A decoder and a decoding method are also provided, which correspond to the described encoding method.

Abstract: A method is provided for processing attenuation of pre-echo in a digital audio signal decoded by transform decoding. The method includes the following acts: decomposition of the decoded signal into at least two sub-signals according to a pre-determined decomposition criterion; calculation of attenuation factors per sub-signal and per sample of a previously determined pre-echo zone; attenuation of pre-echo in the pre-echo zone of each of the sub-signals by applying attenuation factors to the sub-signals; and production of the attenuated signal by addition of the attenuated sub-signals. Also provided are a processing device implementing the acts of the described method, and a decoder including such a device.

Abstract: A parametric coding method and apparatus are provided for coding a multichannel digital audio signal. The method includes a coding step for coding a signal from a channel reduction matrixing of the multichannel signal. The coding method also includes: obtaining, for each frame of predetermined length, spatial information parameters of the multichannel signal; dividing the spatial information parameters into a plurality of blocks of parameters; selecting a block of parameters as a function of the index of the current frame; and coding the block of parameters selected for the current frame.

Abstract: A method is provided for processing pre-echo attenuation in a digital audio signal generated from a transform coding, wherein, at the decoding point, the method includes: detection of a position of attack in the decoded signal; determination of a pre-echo region preceding the position of attack detected in the decoded signal; calculation of attenuation factors per sub-block of the pre-echo region, according to at least the frame wherein the attack has been detected and the preceding frame; and pre-echo attenuation in the sub-blocks of the pre-echo region by the corresponding damping factors. The method also includes application of a filter for the spectral shaping of the pre-echo region on the current frame up to the detected position of the attack. A device and a decoder including the device are also proved for implementing the method.

Abstract: A method is provided for hierarchical coding of a digital audio signal comprising, for a current frame of the input signal: a core coding, delivering a scalar quantization index for each sample of the current frame and at least one enhancement coding delivering indices of scalar quantization for each coded sample of an enhancement signal. The enhancement coding comprises a step of obtaining a filter for shaping the coding noise used to determine a target signal and in that the indices of scalar quantization of said enhancement signal are determined by minimizing the error between a set of possible values of scalar quantization and said target signal. The coding method can also comprise a shaping of the coding noise for the core bitrate coding. A coder implementing the coding method is also provided.

Abstract: A method of hierarchical coding of a digital audio frequency input signal into several frequency sub-bands, including a core coding of the input signal according to a first throughput and at least one enhancement coding of higher throughput, of a residual signal. The core coding uses a binary allocation according to an energy criterion. The method includes for the enhancement coding: calculating a frequency-based masking threshold for at least part of the frequency bands processed by the enhancement coding; determining a perceptual importance per frequency sub-band as a function of the masking threshold and as a function of the number of bits allocated for the core coding; binary allocation of bits in the frequency sub-bands processed by the enhancement coding, as a function of the perceptual importance determined; and coding the residual signal according to the bit allocation. Also provided are a decoding method, a coder and a decoder.

Abstract: The invention concerns an encoder for an input audio signal (S(z)) comprising a combination module combining the input audio signal with an intermediate counter-reaction signal forming a modified input signal and a quantification module scalable for the rate (91) of said modified input signal, delivering a binary raster of quantification indexes of a predetermined rate.