Abstract: The invention proposes the synthesis of a signal consisting of consecutive blocks. It proposes more particularly, on receipt of such a signal, to replace, by synthesis, lost or erroneous blocks of this signal. To this end, it proposes an attenuation of the overvoicing during the generation of a signal synthesis. More particularly, a voiced excitation is generated on the basis of the pitch period (T) estimated or transmitted at the previous block, by optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting groups (A?,B?,C?,D?) of at least two samples and inverting positions of samples in the groups, randomly (B?,C?) or in a forced manner. An over-harmonicity in the excitation generated is thus broken and the effect of overvoicing in the synthesis of the generated signal is thereby attenuated.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: A module (402) for binary coding of a signal envelope, comprising coding module (502) for coding a variable length first mode. The coding module for coding the first mode incorporates an envelope saturation detector and said coding module (402) further includes a second coding module (503) for coding a second mode in parallel with the coding module (502) for coding the first mode and a mode selector (504) adapted to select one of the two coding modes as a function of a code length criterion and of the result from the envelope saturation detector.

Abstract: A method and decoder are provided for parametrically decoding a stereo digital audio signal.

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 of concealing transmission error in a digital audio signal, wherein a signal that has been decoded after transmission is received, the samples decoded while the transmitted data is valid are stored, at least one short-term prediction operator and one long-term prediction operator are estimated as a function of stored valid samples, and any missing or erroneous samples in the decoder signal are generated using the estimated operators. The energy of the synthesized signal that is thus generated is controlled by means of a gain that is computed and adapted sample by sample.

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: Decoder for an audio signal coded by a coder including a long-term prediction filter wherein the decoder comprises: a block (211) for detecting transmission frame losses; a module (222) for calculating values of an error indication function representative of the cumulative adaptive excitation error during decoding following said transmission frame loss, an arbitrary value being assigned to said adaptive excitation gain for the lost frame; a module (213) for calculating an error indication parameter from said values of the error indication function; a comparator (214) for comparing said error indication parameter to at least one given threshold; and a discriminator (215) adapted to determine as a function of the results supplied by the comparator (214) a value of at least one adaptive excitation gain to be used by the decoder.

Abstract: The invention relates to the processing of a signal that is compression encoded (COD) according to a predetermined encoding type applying a quantification operation (Q) and then decoded (DEC) so that the quantification noise is present in the decoded signal (S*). The signal processing of the invention comprises applying a quantification noise reduction (TBQ) to the decoded signal (S), preferably in the following manner: first obtaining information (INF) on the type of compression encoding, selecting a model for the reduction of the quantification noise adapted to said information by estimating the quantification noise (BQ) that the encoding may have generated; and applying to the decoded signal (S*) a processing for reducing the quantification noise (FIL) according to the selected model.

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 is provided for attenuating pre-echoes in a digital audio signal generated from a transform encoding, comprising, upon decoding and for a current frame of said digital audio signal: defining a concatenated signal from at least the reconstructed signal of the current frame, dividing said concatenated signal into subunits of samples having a predetermined length, calculating the time envelope of the concatenated signal, detecting the transition of the time envelope towards a high-energy area, determining the low-energy sub-units preceding a subunit in which a transition has been detected, and an attenuation step in said determined subunits. The attenuation is carried out according to an attenuation factor calculated for each of the determined subunits, based on the time envelope of the concatenated signal. The invention also relates to a device for implementing said method, and to a decoder including such a device.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: The invention proposes the synthesis of a signal consisting of consecutive blocks. It proposes more particularly, on receipt of such a signal, to replace, by synthesis, lost or erroneous blocks of this signal. To this end, it proposes an attenuation of the overvoicing during the generation of a signal synthesis. More particularly, a voiced excitation is generated on the basis of the pitch period (T) estimated or transmitted at the previous block, by optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting groups (A?,B?,C?,D?) of at least two samples and inverting positions of samples in the groups, randomly (B?,C?) or in a forced manner. An over-harmonicity in the excitation generated is thus broken and the effect of overvoicing in the synthesis of the generated signal is thereby attenuated.

Abstract: The present invention relates to signal modification before pitch period repetition for the synthesis of blocks lost on decoding digital audio signals. The effects of repetition of transitories, such as the plosives of a speech signal, are avoided by comparing the samples of a pitch period with those of the previous pitch period. The signal is modified preferentially by taking the minimum between a current sample (e(3)) of the last pitch period (Tj) and at least one sample (e(2?T0) of approximately the same position in the previous pitch period (Tj?1).

Abstract: The invention relates to a method of processing bit errors in a bit frame emanating from a digital audio coder, comprising a step of receiving a current bit frame liable to comprise bit errors. According to the invention, the bit frame comprises sensitive bits to be protected which are catalogued in at least one category according to the type of parameter that they code and the method furthermore comprises the steps of receiving protection bits, of reading the sensitive bits received in the current bit frame, the number of sensitive bits being lower than the number of bits of the bit frame, of detecting bit errors as a function of said protection bits received and of said sensitive bits received and in the event of detecting at least one erroneous bit in said bit frame, of modifying the current bit frame before decoding, as a function of the category in which the erroneous bit is catalogued.

Abstract: The present invention relates to a method of transmission error concealment in a digital signal split up into a plurality of successive frames associated with different time intervals in which, on reception, the signal might comprise erased frames and valid frames and in order to replace at least the first erased frame (N) after a valid frame, at least two steps are performed, a first step (E1) of preparation not producing any missing sample and comprising at least one analysis of a valid decoded signal and a second step (E2) of concealment producing the missing samples of the signal corresponding to said erased frame. The first step and the second step are executed in different time intervals. The invention relates also to a device of concealment implementing the method according to the invention as well as a decoder comprising such a device. The invention allows the distribution of the error concealment complexity on different time intervals.

Abstract: The invention relates to the processing of a digital signal originating from a decoder and a noise reduction post-processing step, including, in particular, limitation of distortion introduced by the post-processing step in order to deliver a corrected output signal (SOUT), assigning said corrected output signal (SOUT) with: a current amplitude having an intermediary value between a current amplitude value of the post-processed signal (SPOST) and a corresponding current amplitude value of the decoded signal (S?MIC), or the current amplitude of the post-processed signal (SPOST), according to the respective values of the current amplitude of the post-processed signal (SPOST) and by the corresponding current amplitude of the decoded signal (S?MIC).

Abstract: The invention relates to a method for scalar quantization-based coding of the samples of a digital audio signal (S), the samples being coded over a pre-determined number of bits in order to obtain a binary frame of quantization indices (IPCM), the coding being carried out according to an amplitude compression law, where a pre-determined number of least significant bits are not taken into account in the binary frame of quantization indices. The coding method comprises the steps for storing (27) at least a part of the least significant bits that are not taken into account in the quantization index binary frame and for determination (28) of an enhancement bit stream (IEXT) comprising at least one bit thus stored.

Abstract: The invention relates to the processing of a signal that is compression encoded (COD) according to a predetermined encoding type applying a quantification operation (Q) and then decoded (DEC) so that the quantification noise is present in the decoded signal (S*). The signal processing of the invention comprises applying a quantification noise reduction (TBQ) to the decoded signal (S), preferably in the following manner: first obtaining information (INF) on the type of compression encoding, selecting a model for the reduction of the quantification noise adapted to said information by estimating the quantification noise (BQ) that the encoding may have generated; and applying to the decoded signal (S*) a processing for reducing the quantification noise (FIL) according to the selected model.

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.