Abstract: Audio sources are coded by recognizing different classes of audio such as speech and music. The classes are used to select between coding algorithms and to provide object definitions. Objects have abstract and concrete classes which may further rely on parameters produced by linear prediction and subband filters to provide a frame-based bit stream of information. Each object in the bit stream has layers of information such as basic bit rate, coding parameters and enhancement parameters. The layers of information in each object allow altering selected parameters to manipulate audio signals.

Abstract: A method of and a device for speech signal digital coding are described, where spectral parameters are quantized at each frame in order to exploit the actual correlation inside a frame or between contiguous frames. The quantization devices (DQ) recognize strongly correlated signal periods by using a first set of indexes (j.sub.1), representing the parameters and provided by the spectral analysis circuits (ABT, ALT), and in these periods they convert the same indexes into a second set of indexes (j4) which can be coded with a lower number of bits and which is inserted into the coded signal in place of the first set.

Abstract: An optimum excitation signal for each subframe is determined in a speech coder based on analysis-by-synthesis techniques and operating on frames of samples divided into a number of subframes. The excitation signal includes a shape contribution (innovation) and an amplitude contribution (gain) which are quantized separately. A circuit (IT) for gain quantization includes means (QU) for determining a gain index for each subframe; a comparison logic network (CFR) for detecting the maximum value taken by the gain index in the frame; and means for computing a normalized index for each subframe as a difference between the maximum index and the gain index relevant to that subframe. The coded signal includes the coded values of the maximum index and of the normalized indexes as information on the gain relevant to a frame.

Abstract: In a radio communication system, in particular a mobile communication system, variable-rate information flows originating from different sources (CV1, CS1; CV2, CS2) and relative to a same communication are transmitted on a same radio channel. Each mobile station (M1 . . . Mh) and the fixed part (B1 . . . Bn; RNC) of the system comprise a unit for the control of the variable-rate transmission, which dynamically allocates the available bits to the different streams by taking into account the needs of the sources (CV1, CS1; CV2, CS2), the conditions of the channel (6) and the system occupancy. (FIG. 1).

Abstract: The set of possible excitation signals is subdivided into a plurality of subsets, the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network, while the others supply a contribution which, when added to that of the first subset, causes a rate increase by successive steps. At the receiving side, a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate, while for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used.

Abstract: The set of possible excitation signals is subdivided into a plurality of subsets, the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network, while the others supply a contribution which, when added to that of the first subset, causes a rate increase by successive steps. At the receiving side, a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate, while for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used.

Abstract: A low-delay audio signal coding system, using analysis-by-synthesis techniques, has circuitry for adapting the spectral parameters and the prediction order of synthesis filters, and of perceptual weighting filters in the order at each frame, starting from the reconstructed signal relevant to the previous frame. In the case of a CELP coder, gain controls are also provided to adapt, starting from the reconstructed sinal, a factor, bound to the average power of the input signal, of the gain by which the innovation vectors are weighted.

Abstract: According to the coding method, the digital speech signal is split into subbands; each subband is coded independently of the others. The first two subbands undergo an adaptive linear prediction operation yielding a residual, normalized with respect to a quantized r.m.s. value; the residual samples are quantized and coded by a variable bit number. As to the third subband the samples normalized with respect to a quantized gain term are directly quantized and coded by a variable bit number. While splitting the subband signals into sample blocks, for the first two subbands the coefficient vectors of subband linear prediction filters relevant to each block are also determined by vector quantization and linear prediction inverse filtering operations; quantized r.m.s. values are also determined to calculate the numbers of bits permitting subband signals coding. The coded signal consists of subband signal quantization level indices, of r.m.s.

Abstract: This method provides a filtering of digital samples of speech signal by a linear-prediction inverse filter, whose coefficients are chosen out of a codebook of quantized filter coefficient vectors, obtaining a residual signal subdivided into vectors. The weighted mean-square error made in quantizing said vectors with quantized residual vectors contained in a codebook and forming excitation waveforms is computed.The coding signal for each block of samples consists of the coefficient vector index chosen for the inverse filter as well as of the indices of the vectors of the excitation waveforms which have generated minimum weighted mean-square error. During the decoding phase, a synthesis filter, having the same coefficients as chosen for the inverse filter, is excited by quantized-residual vectors chosen during the coding phase (FIGS. 1, 2).