Abstract: The invention relates to transform coding/decoding of a digital audio signal represented by a succession of frames, using windows of different lengths. For the coding within the meaning of the invention, it is sought to detect (51) a particular event, such as an attack, in a current frame (Ti): and, at least if said particular event is detected at the start of the current frame (53), a short window (54) is directly applied in order to code (56) the current frame (Ti) without applying a transition window. Thus, the coding has a reduced delay in relation to the prior art. In addition, an ad hoc processing is applied during decoding in order to compensate for the direct passage from a long window to a short window during coding.

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: The invention relates to compression coding and/or decoding of digital signals, in particular by vector variable-rate quantisation defining a variable resolution. For this purpose an impulsion dictionary comprises: for a given dimension, increasing resolution dictionaries imbricated into each other and, for a given dimension, a union of: a totality (D?i<N>) of code-vectors produced, by inserting elements taken in a final set (A) into smaller dimension code-vectors according to a final set of predetermined insertion rules (F1) and a second totality of code-vectors (Y?) which are not obtainable by insertion into the smaller dimension code vectors according to said set of the insertion rules.

Abstract: The invention concerns a method for trained discrimination and attenuation of echoes of a digital audio signal generated from a transform coding, which consists, for each current frame of the signal. In comparing (A) in real time, in at least one frequency band a variable derived from one characteristic of the echo generating signal with that of a non-echo generating signal at a threshold value, and deducing therefrom (B) the existence or non-existence (C) of an echo derived from the transform coding, discriminating the existence of the echo and defining (D) a false alarm zone in the high-energy parts of the digital audio signal, determining an initial processing and attenuating the echoes (E) in the parts complementary to the low-energy false alarm zone and inhibiting (F) the attenuation of echoes in the false alarm zone. The invention is applicable to the technology of coders/decoders in particular hierarchical coders/decoders.

Abstract: A method of bitrate switching on decoding an audio signal coded by a audio coding system, said decoding comprising a post-processing step depending on the bitrate. On switching from an initial bitrate to a final bitrate, said method includes a transition step of continuous change from a signal at the initial bitrate to a signal at the final bitrate, one or both of said signals being post-processed. Application to transmission of VoIP speech and/or audio signals in data packet networks.

Abstract: A method of concealing transmission error in a digital audio signal in which 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 operators estimated in this way, the method being characterized in that the energy of the synthesized signal as generated in this way is controlled by means of a gain that is computed and adapted sample by sample.

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: 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 maximum of Nmax bits for encoding is defined for a set of parameters which may be calculated from a signal frame. The parameters for a first sub-set are calculated and encoded with N0 bits, where N0<Nmax. The allocation of Nmax?N0 encoding bits for the parameters of a second sub-set are determined and the encoding bits allocated to the parameters for the second sub-set are classified. The allocation and/or order of classification of the encoding bits are determined as a function of the encoding parameters for the first sub-set. For a total of N available bits for the encoding of the total parameters (N0<N=Nmax), the parameters for the second sub-set allocated the N?N0 encoding bits classified the first in said order are selected. Said selected parameters are calculated and encoded to give the N?N0 bits. The N0 encoding bits for the first sub-set and the N?N0 encoding bits for the selected parameters for the second sub-set are finally introduced into the output sequence of the encoder.

Abstract: The invention relates to compression coding and/or decoding of digital signals, in particular by vector variable-rate quantisation defining a variable resolution. For this purpose an impulsion dictionary comprises: for a given dimension, increasing resolution dictionaries imbricated into each other and, for a given dimension, a union of: a totality (D?i<N>) of code-vectors produced, by inserting elements taken in a final set (A) into smaller dimension code-vectors according to a final set of predetermined insertion rules (F1) and a second totality of code-vectors (Y?) which are not obtainable by insertion into the smaller dimension code vectors according to said set of the insertion rules.

Abstract: A maximum of Nmax bits for encoding is defined for a set of parameters which may be calculated from a signal frame. The parameters for a first sub-set are calculated and encoded with N0 bits, where N0<Nmax. The allocation of Nmax?N0 encoding bits for the parameters of a second sub-set are determined and the encoding bits allocated to the parameters for the second sub-set are classified. The allocation and/or order of classification of the encoding bits are determined as a function of the encoding parameters for the first sub-set. For a total of N available bits for the encoding of the total parameters (N0<N=Nmax), the parameters for the second sub-set allocated the N?N0 encoding bits classified the first in said order are selected. Said selected parameters are calculated and encoded to give the N?N0 bits. The N0 encoding bits for the first sub-set and the N?N0 encoding bits for the selected parameters for the second sub-set are finally introduced into the output sequence of the encoder.

Abstract: An improved EX-CELP or RCELP encoding scheme is proposed, in which, at the encoder side, a speech signal is perceptually weighted signal prior to entering a time scale modification module, then the modified signal is transformed into another domain, such as the speech or LP short-term residual domain, using the corresponding inverse filtering operation directly or possibly combined with another processing, for instance a short-term LP filtering. A shift function is calculated in the time scale modification process to associate the position of each sample in the modified signal with its original position before the modification. The positions of the samples in the modified signal that correspond to sub-frame boundaries of the original signal are evaluated to switch filters for the inverse filtering at the appropriate instants. Therefore, the synchronization between the inverse filters and the modified signal is maintained.

Abstract: A method of concealing transmission error in a digital audio signal in which 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 operators estimated in this way, the method being characterized in that the energy of the synthesized signal as generated in this way is controlled by means of a gain that is computed and adapted sample by sample.