Title of Invention	METHOD FOR CODING A DIGITAL AUDIO FRAME AND AN AUDIO CODER TO IMPLEMENT THE SAID METHOD
Abstract	A method of coding a digital audio signal frame (S) as a binary output sequence (Φ), in which a maximum number Nmax of coding bits is defined for a set of parameters that can be calculated accordingto the signal frame, which set it composed of a first and of a second subset,the method comprising the following steps: - Calculating (1) the parameters of the first subset, and coding these parameters on a number NO of coding bits such that NO < Nmax; - determining (12) an allocation of Nmax-NO coding bits for the parameters of the second subset; and - ranking (13) the Nmax - NO coding bits allocated to the parameters of the second subset in a determined order, in which the allocation and/or the order of ranking of the Nmax - NO coding bits is determined as a function of the coded parameters of the first subset, the method furthermore comprising the following steps in response to the indication of a number N of bits of the binary output sequence that are available for the coding of said set of parameters, with NO < N ≤ Nmax: - selecting the second subset's parameters to which are allocated the N - NO 'coding bits ranked first in said order; - calculating (9) the selected parameters of the second subset, and coding these parameters so as to produce said N - NO coding bits ranked first; and - inserting (10) into the output sequence the NO coding bits of the first subset as well as the N - NO coding bits of the selected parameters of the second subset. 2. The method as claimed in claim 1, in which the order of ranking 'of the coding bits allocated -to the parameters of the second subset is variable form one

Title of Invention

METHOD FOR CODING A DIGITAL AUDIO FRAME AND AN AUDIO CODER TO IMPLEMENT THE SAID METHOD

Abstract

A method of coding a digital audio signal frame (S) as a binary output sequence (Φ), in which a maximum number Nmax of coding bits is defined for a set of parameters that can be calculated accordingto the signal frame, which set it composed of a first and of a second subset,the method comprising the following steps: - Calculating (1) the parameters of the first subset, and coding these parameters on a number NO of coding bits such that NO < Nmax; - determining (12) an allocation of Nmax-NO coding bits for the parameters of the second subset; and - ranking (13) the Nmax - NO coding bits allocated to the parameters of the second subset in a determined order, in which the allocation and/or the order of ranking of the Nmax - NO coding bits is determined as a function of the coded parameters of the first subset, the method furthermore comprising the following steps in response to the indication of a number N of bits of the binary output sequence that are available for the coding of said set of parameters, with NO < N ≤ Nmax: - selecting the second subset's parameters to which are allocated the N - NO 'coding bits ranked first in said order; - calculating (9) the selected parameters of the second subset, and coding these parameters so as to produce said N - NO coding bits ranked first; and - inserting (10) into the output sequence the NO coding bits of the first subset as well as the N - NO coding bits of the selected parameters of the second subset. 2. The method as claimed in claim 1, in which the order of ranking 'of the coding bits allocated -to the parameters of the second subset is variable form one

Full Text	The present invention relates to method for coding a digital audio frame and an audio coder to implement the said method.The invention further relates to devices for coding and decoding audio signals, intended in particular to sit within applications of transmission or storage of digitized and compressed audio signals (speech and/or sounds). More particularly, this invention pertains to audio coding systems having the capacity to provide varied bit rates, also referred to as multirate coding systems. Such systems are distinguished from fixed rate coders by their capacity to modify the bit rate of the coding, possibly during processing, this being especially suited to transmission over heterogeneous access networks: be they networks of IP type mixing fixed and mobile access, high bit rates (ADLS) , low bit rates (RTC, GPRS modems) or involving terminals with variable capacities (mobiles, PCs, etc.). Essentially, two categories of multirate coders are distinguished: that of "switchable" multirate coders and that of "hierarchical" coders. "Switchable" multirate coders rely on a coding architecture belonging to a technologic'al family (temporal coding or frequency coding, for example: CELP, sinusoidal, or by transform), in which an indication of bit rate is simultaneously supplied to the coder and to the decoder. The coder uses this information to select the parts of the algorithm and the tables relevant to the bit rate chosen. The decoder operates in a symmetric manner. Numerous switchable multirate coding structures have been proposed for audio coding. Such is the case for example with mobile coders standardized by the 3GPP organization ("3rd Generation Partnership Project"), NB-AMR ("Narrow Band Adaptive Multirate", Technical Specification 3GPP TS 26.090, version 5.0.0, June 2002) in the telephone band, or WB-AMR ("Wide Band Adaptive Multirate", Technical Specification 3GPP TS 26.190, version 5.1.0, December 2001) in wideband. These coders operate over fairly wide bit rate ranges (4.75 to 12.2 kbit/s for NB-AMR, and 6.60 to 23.85 kbit/s for WB-AMR), with a fairly sizeable granularity (8 bit rates for NB-AMR and 9 for WB-AMR) . However, the price to be paid for this flexibility is a rather considerable complexity of structure: to be able to host all these bit rates, these coders must support numerous different options, varied quantization tables etc. The performance curve increases progressively with bit rate, but the progress is not linear and certain bit rates are in essence better optimized than others. In so-called "hierarchical" coding systems, also referred to as "scalable", the binary data arising from the coding operation are distributed into successive layers. A base layer, also called the "kernel", is formed of the binary elements that are absolutely necessary for the decoding of the binary train, and determine a minimum quality of decoding. The subsequent layers make it possible to progressively improve the quality of the signal arising from the decoding operation, each new layer bringing new information which, utilized by the decoder, supplies a signal of increasing quality at output. One of the particular features of hierarchical coding is the possibility offered of intervening at any level whatsoever of the transmission or storage chain so as to delete a part of the binary train without having to supply any particular indication to the coder or to the decoder. The decoder uses the binary information that it receives and produces a signal of corresponding quality. The field of hierarchical coding structures has given rise likewise to much work. Certain hierarchical coding structures operate on the basis of one type of coder alone, designed to deliver hierarchized coded information. When the additional layers improve the quality of the output signal without modifying the bandwidth, one speaks rather of "embedded coders" (see for example R.D. Lacpvo et al., "Embedded CELP Coding for Variable Bit-Rate Between 6.4 and 9.6 kbit/s, Proc ICASSP 1991, pp. 681-686) . Coders of this type do not however allow large 'gaps between the lowest and the highest bit rate proposed. The hierarchy is often used to progressively increase the bandwidth of the signal: the kernel supplies a baseband signal, for Example telephonic (300-3400 Hz), and the subsequent layers allow the coding of additional frequency bands (for example, wide band up to 7 kHz, HiFi band up to 20 kHz or intermediate, etc.). The subband coders or coders using a time/frequency transformation such as described in the documents "Subband/trahsform coding using filter banks designs based on time domain aliasing cancellation" by J.P. Princen et al. (:Proc. IEEE ICASSP-87, pp. 2161- 2164) and "High Quality Audio Transform Coding at 64 kbit/s", by Y. Mahieux et al. (IEEE Trans. Commun., Vol. 42, No. 11, November 1994, pp. 3010-3019), lend themselves particularly to such operations. Moreover, a different coding technique is frequently used for the kernel and for the module or modules coding the additional layers, one then speaks of various coding stages, each stage consisting of a subcoder. The subcoder of the stage of a given level will be able either to code parts of the signal that are not coded by the previous stages, or to code the coding residual of the previous stage, the residual is obtained by subtracting the decoded signal from the original signal. The advantage of such structures it that they make it possible to go down, to relatively low bit rates with sufficient quality, while producing good quality at high bit rate. Specifically, the techniques -used for low bit rates are not generally effective at high bit rates and vice versa.; Such structures making it possible to use two different technologies (for example CELP and time/frequency transform, etc.) are especially effective for sweeping large bit rate ranges. However, the hierarchical coding structures proposed in the prior art define precisely the_bit. rate allocated to eacn of the intermediate layers. Each layer corresponds to the encoding of certain parameters, and the granularity of the hierarchical binary train depends on the bit rate allocated to these parameters (typically a layer can contain of the order of a few tens of bits per frame, a signal frame consisting of a certain number of samples of the signal over a given duration, the example described later considering a frame of 960 samples corresponding to 60 ms of signal). Moreover, when the bandwidth of the decoded signals can vary according to the level of the layers of binary elements, the modification of the line bit rate may produce artifacts that impede listening. The present invention has the aim in particular of proposing a multirate coding solution which alleviates the drawbacks cited in the case of the use of existing hierarchical and switchable codings. The invention thus proposes a method of coding a digital audio signal frame as a binary output sequence, in wnich a maximum number Nmax of coding bits is defined for a set of parameters that can be calculated according to the signal frame, which set is composed of a first and of a second subset. The proposed method comprises the following steps: calculating the; parameters of the first subset, and coding these parameters on a number NO of coding bits such that NO determining an allocation of Nmax - NO coding bits for the parameters of the second subset; and ranking the Nmax - NO coding bits allocated to the parameters of the second subset in a determined order. The allocation and/or the order of ranking of the Nmax - NO coding bits are determined as a function of the coded parameters] of the first subset. The coding method furthermore comprises the following steps in response to the indication of a number N of bits of the binary output sequence that are available for the coding of said set of parameters, with NO selecting the second subset's parameters to which are allocated the N - NO coding bits ranked first in said order; calculating the selected parameters of the second subset, and coding these parameters so as to produce said N - nO coding bits ranked first; and inserting into the output sequence the NO coding bits of the first subset as well as the N - NO coding bits of the selected parameters of the second subset. The method according to the invention makes it possible to define a multirate coding, which will operate at least in a range corresponding for each frame to a number of bits ranging' from NO to Nmax. It may thus be considered that the notion of pre- established bit rates which is related to the existing hierarchical and switchable codings is replaced by a notion of "cursor", making it possible to freely vary the bit rate between a minimum value (that may possibly correspond to a number of bits N less than NO) and a maximum value (corresponding to Nmax). These extreme values are potentially far apart. The method offers good performance in terms of effectiveness of coding regardless of the bit rate chosen. Advantageously, the [number N of bits of the binary output sequence is strictly less than Nmax. What is noteworthy about the coder is then that the allocation of the bits that is employed makes no reference to the actual output bit rate of the coder, but to another number Nmax agreed with the decoder. It is however possible to fix Nmax = N as a function of the instantaneous bit, rate available on a transmission channel. The output sequence of a switchable multirate coder such as this may be processed by a decoder which does not receive the entire sequence, so long as it is capable of retrieving the structure of the coding bits of the second subset by virtue of the knowledge of Nmax. Another case where it is possible to have N = Nmax is that of the storage of;audio data at the maximum coding rate. When reading N' bits of this content stored at lower bit rate, the decoder would be capable of retrieving the structure of the coding bits of the second subset as long a>s N' > NO. The order of ranking of the coding bits allocated to the parameters of the second subset may be a preestablished order. In a preferred embodiment, the order of ranking of the coding bits allocated to the parameters of the second subset is variable. It may in particular be an order of decreasing importance determined as a function of at least the coded parameters of the first subset. Thus the decoder which receives a binary sequence of N' bits for the frame, with NO deduce this order from the NO bits received for the coding of the first subset. The allocation of the Nmax - NO bits to the coding of the parameters of the second subset may be carried out in a fixed manner (in this case, the order of ranking of these bits will be dependent at least on the coded parameters of the first subset). In a preferred embodiment, the allocation of the Nmax - NO bits to the coding of the parameters of the second subset is a function of the coded parameters of the first subset. Advantageously, this order of ranking of the coding bits allocated to the parameters of the second subset is determined with the aid of at least one psychoacoustic criterion as a function of the coded parameters of the first; subset. The parameters of the second subset pertain to spectral bands of the signal. In this case, the method advantageously comprises a step of estimating a spectral envelope of the coded signal on the basis of the coded parameters of the first subset, and a step of calculating a curve of frequency masking by applying an auditory perception model to the estimated spectral envelope, and the psychoacoustic criterion makes reference to the level of the estimated spectral envelope with respect to the masking curve in each spectral band. In a mode of implementation, the coding bits are ordered in the output sequence in such a way that the NO coding' bits of the first subset precede the N - NO coding bits of the selected parameters of the second subset and that the respective coding bits of the selected parameters of the second subset appear therein in the order determined for said coding bits. This makes it possible, in the case where the binary sequence is truncated, to receive the most important part. The number N may vary from one frame ' to another, in particular as a function for example of the available capacity of the transmission resource. The multirate audio coding according to the present invention may be used according to a very flexible hierarchical or switchable mode, since any number of bits to be transmitted chosen freely between NO and Nmax may be selected at any moment, that is to say frame by frame. The coding of the parameters of the first subset may be at variable bit rate, thereby varying the number NO from one frame to another. This allows best adjustment of the distribution of the bits as a function of the frames to be coded. In a mode of implementation, the first subset comprises parameters calculated by a coder kernel. Advantageously, the coder kernel has a lower frequency band of operation than the bandwidth of the signal to be coded, and the first subset furthermore comprises energy levels of the audio signal that are associated with frequency bands higher than the operating band of the coder kernel. This type of structure is that of a hierarchical coder with two levels, which delivers for example via the coder kernel a coded signal of a quality deemed to be sufficient and which, as a function of the bit rate available, supplements the coding performed by the coder kernel with additional information arising from the method of coding according to the invention. Preferably, the coding bits of the first subset are then ordered in the output sequence in such a way that the coding bits of the parameters calculated by the coder kernel are immediately followed by the coding bits of the energy levels associated with the higher frequency bands. This ensures one and the same bandwidth for the successively coded frames as long as the decoder receives enough bits to be in possession of information of the coder kernel and coded energy levels associated with the higher frequency bands. In a mode of implementation, a signal of difference between the signal to be coded and a synthesis signal derived from the coded parameters produced by the coder kernel is estimated, and the first subset furthermore comprises energy levels of the difference signal that are associated with frequency bands included in the operating band of the coder kernel. A second aspect of the invention pertains to a method of decoding a binary input sequence so as to synthesize ~a"~digital audio signal corresponding to the decoding of a frame coded according to the method of coding of the invention. According to this method, a maximum number Nmax of coding bits is defined for a set of parameters for describing a signal frame, which set is composed of a first and a second subset. The input sequence comprises, for a signal frame, a number N' of coding bits for the set of parameters, with N' decoding method according to the invention comprises the following steps: extracting, from said N' bits of the input sequence, a number NO of coding bits of the parameters of the first subset if NO recovering the parameters of the first subset on the basis of said NO coding bits extracted; determining an allocation of Nmax - NO coding bits for the parameters of the second subset; and ranking the Nmax - NO coding bits allocated to the parameters of the second subset in a determined order. The allocation and/or the order of ranking of the Nmax - NO coding bits are determined as a function of the recovered parameters of the first subset. The decoding method furthermore comprises the following steps: selecting the second subset's parameters to which are allocated the N' - NO coding bits ranked first in said order; extracting, from said N' bits of the input sequence, N' - NO coding bits of the selected parameters of the second subset; recovering the selected parameters of the second subset on the basis of said N' - NO coding bits extracted; and synthesizing the signal frame by using the recovered parameters of the first and second subsets. This method of decoding is advantageously associated with procedures for regenerating the parameters which are missing on account of the truncation of the sequence of Nmax bits that is produced, virtually or otherwise, by the coder. A third aspect of the invention pertains to an audio coder, comprisinq means of digital signal processing that are devised to implement a method of coding according to the invention. Another aspect of the invention pertains to an audio decoder, comprising means of digital-signal processing tha are devised to implement a method of decoding according to the invention. Other features and advantages of the present invention will become apparent in the description hereinbelow of nonlimiting exemplary embodiments, with reference to the appended drawings,] in which: figure 1 is a schematic diagram of an exemplary audio coder according to the invention; figure 2 represents a binary output sequence of N bits in a embodiment of the invention; and figure 3 is a schematic diagram of an audio decoder according to the invention. The coder represented I in figure 1 has a hierarchical structure with two coding stages. A first coding stage 1 consists for example of a coder kernel in a telephone band (300-3400 Hz) of CELP type. This coder is in the example considered a G.723.1 coder standardized by the ITU-T ("International Telecommunication Union") in fixed mode at 6.4 kbit/s. It calculated G. 723.1 parameters in accordance with the standard and quantizes them by means, of 192 coding bits P1 per frame of 30 ms. The second coding stage 2, making it possible to increase the bandwidth towards the wide band (50-7000 Hz) , operates on the coding residual E of the first stage, supplied by a subtractor 3 in the diagram of figure 1. A signals synchronization module 4 delays the audio signal frame S by the time taken by the processing of the coder kernel 1. Its output is addressed 'to the subtractor 3 which subtracts from it the synthetic signal S' equal to the output of the decoder kernel operating on the basis of the quantized parameters such as represented by the output bits P1 of the coder kernel. As is usual, the coder 1 incorporates a local decoder supplying S', The audio signal to be coded S has for example a bandwidth of 7 kHz, while being sampled at 16 kHz. A frame consists for example of 960 samples, i.e. 60 ms of signal or two elementary frames of the coder kernel G. 723.1. Since the latter operates on signals sampled at 8 kHz, the signal S is subsampled in a factor 2 at the input of the coder kernel 1. Likewise, the synthetic signal S' is oversampled at 16 kHz at the output of the coder kernel 1. The bit rate of the first stage 1 is 6.4 kbit/s (2 x Nl = 2 x 192 = 384 bits per frame). If the coder has a maximum bit rate of 32 kbit/s (Nmax = 1920 bits per frame), the maximum bit rate of the second stage is 25.6 kbit/s (1920 - 384 = 1536 bits per frame). The second stage 2 operates for example on elementary frames, or subframes, of 20 ms (320 samples at 16 kHz). The second stage 2 comprises a time/frequency transformation module 5, for example of MDCT ("Modified Discrete Cosine Transform") type to which the residual E obtained by the subtractor 3 is addressed. In practice, the manner of operation of the modules 3 and 5 represented in figure 1 may be achieved by performing the following operations for each 20 ms subframe: MDCT transformation of the input signal S delayed by the module 4, which supplies 320 MDCT coefficients. The spectrum being limited to 7225 Hz, only the first 289 MDCT coefficients are different from 0; MDCT transformation of the synthetic signal S'. Since one is dealing with the spectrum of a telephone band signal, only the first 139 MDCT coefficients are different from 0 (up to 3450 Hz)/ and calculation of the spectrum of difference between the previous spectra. The resulting spectrum is distributed into several bands of different widths by a module 6. By way of example, the bandwidth of the G.723.1 codec may be subdivided into 21 bands while the higher frequencies are distributed into 11 additional bands. In these 11 additional bands, the residual E is identical to the input signal S. A module 7 performs the coding of the spectral envelope of the residual E. It begins by calculating the energy of the MDCT coefficients of each band of the difference spectrum. These energies are hereinbelow referred to as "scale factors". The 32 scale factors constitute the spectral envelope of the difference signal. The module 7 then proceeds to their quantization in two parts. The first part corresponds to the telephone band (first 21 bands, from 0 to 3450 Hz), the second to the high bands (last 11 bands, from 3450 to 7225 Hz) . In each part, the first scale factor is quantized on an absolute basis, and the subsequent ones on a differential basis, by using a conventional Huffman coding with variable bit rate. These 32 scale factors are quantized on a variable number N2(i) of bits P2 for each subframe of rank i (i = 1, 2, 3) . The quantized scale factors are denoted FQ in figure 1. The quantization bits P1, P2 of the first subset consisting of the quantized parameters of the coder kernel 1 and the quantized scale factors FQ are variable in number NO = (2 x N1) + N2(l) + N2(2) + N2(3). The difference Nmax - NO = 1536 - N2(l) - N2{2) - N2(3) is available to quantize the spectra of the bands more finely. A module 8 normalizes the MDCT coefficients distributed into bands by the module 6, by dividing them by the quantized scale factors FQ respectively determined for these bands. The spectra thus normalized are supplied to the quantization module 9 which uses a vector quantization scheme of known type. The quantization bits, arising from the module 9 are denoted P3 in figure 1. An output multiplexer 10 gathers together the bits P1, P2 and P3 arising from the modules 1, 7 and 9 to form the binary output sequence F of the coder. In accordance with the invention, the total number of bits N of the output sequence representing a current frame is not necessarily equal to Nmax. It may be less than the latter. However, the allocation of the quantization bits to the bands is performed on the basis of the number Nmax. In the diagram of figure 1, this allocation is performed for each subframe by the module 12 on the basis of the number Nmax - NO, of the quantized scale factors FQ and of a spectral masking curve calculated by a module 11. The manner of operation of the latter module 11 is as follows. It firstly determines an approximate value of the original spectral envelope of the signal S on the basis of that of the difference signal, such as quantized by the module 7, and of that which it determines with the same resolution for the synthetic signal S' resulting from the coder kernel. These last two envelopes are also determinable by a decoder which is provided only with the parameters of the aforesaid first subset. Thus the estimated spectral envelope of the signal S will also be available to the decoder. Thereafter, the module 11 calculates a spectral masking curve by applying, in a manner known per se, a model of band by band auditlory perception to the original estimated spectral envelope. This curve 11 gives a masking level for each band considered. The module 12 carries out a dynamic allocation of the Nmax - NO remaining bits of the sequence F among the 3 x 32 bands of the three MDCT transformations of the difference signal. In the implementation of the invention set forth here, as a function of a criterion of psychoacoustic perceptual importance making reference to the level of the spectral envelope estimated with respect to the masking curve in each band, a bit rate proportional to this level is allocated to each band. Other ranking criteria would be useable. Subsequent to this allocation of bits, the module 9 knows how many bits are to be considered for the quantization of each band in each subframe. Nevertheless, if N not necessarily all be used. An ordering of the bits representing the bands is performed by a module 13 as a function of a criterion of perceptual importance. The module 13 ranks the 3 x 32 bands in an order of decreasing importance which may be the decreasing order of the signal-to-mask ratios (ratio between the estimated spectral envelope and the masking curve in each band) . This order is used for the construction of the binary sequence F in accordance with the invention. As a function of the desired number N of bits in the sequence F for the coding of the current frame, the bands which are to be quantized by the module 9 are determined by selecting the bands ranked first by the module 13 and by keeping for each band selected a number of bits such as is determined by the module 12. Then the MDCT coefficients of each band selected are quantized by the module 9, for example with the aid of a vector quantizer, in accordance with the allocated number of bits, so as to produce a total number of bits equal to N - NO. The output multiplexer 10 builds the binary sequence F consisting of the first N bits qf the following ordered sequence represented in figure 2 (case N = Nmax): a/ firstly the binary trains corresponding to the two G. 723.1 frames (384 bits'); b/ next the bits for quantizing the scale factors, for the three subframes (i = 1, 2, 3), from the 22nd spectral band (first band beyond the telephone band) to the 32nd band (variable rate Huffman coding); c/ next the bits for quantizing the scale factors, for the three subframes (i = 1, 2, 3), from the 1st spectral band to the 21st band (variable rate Huffman coding); d/ and finally the indices Mc1, Mc2, Mc96 of vector quantization of the 96 bands in order of perceptual importance, from the most important band to the least important band, while complying with the order determined by the module 13. By placing first (a and b) the G.723.1 parameters and the scale factors of the high bands it is possible to retain the same bandwidth for the signal restorable by the decoder regardless of the actual bit rate beyond a minimum value corresponding to the reception of these groups a and b. This minimum value, sufficient for the Huffman coding of the 3 x 11 = 33 scale factors of the high bands in addition to the G.723.1 coding, is for example 8 kbit/s. The method of coding hereinabove allows a decoding of the frame if the decoder receives N' bits with NO from one frame to another. A decoder according to the invention, corresponding to this example, is illustrated by figure 3. A demultiplexer 20 separates the sequence of bits received F so as to extract therefrom the coding bits P1 and P2. The 384 bits P1 are supplied to the decoder kernel 21 of G.723.1 type so that the latter synthesizes two frames of the base signal S' in the telephone band. The bits P2 are decoded according to the Huffman algorithm by a module 22 which thus recovers the quantized scale factors FQ for each of the 3 subframes. A module 23 calculating the masking curve, identical to the module 11 of the coder of figure 1, receives the base signal S' and the quantized scale factors FQ and produces the spectral masking levels for each of the 96 bands. On the basis of these masking levels, of the quantized scale factors FQ and of the knowledge of the number Nmax (as well as of that of the number NO which is deduced from the Huffman decoding of the bits P2 by the module 22), a module 24 determines an allocation of bits in the same manner as the module 12 of figure 1. Furthermore, a module 25 proceeds to the ordering of the bands according to the same ranking criterion as the module 13 described with reference to figure 1. According to the information supplied by the modules 24 and 25, the module 26 extracts the bits P3 of the input sequence F' and synthesizes the normalized MDCT coefficients relating to the bands represented in the sequence F' If appropriate (N' standardized MDCT coefficients relating to the missing bands may furthermore be synthesized by interpolation or extrapolation as described hereinbelow (module 27) . These missing bands may have been eliminated by the coder on account of a truncation to N may have been eliminated in the course of transmission (N' The standardized MDCT coefficients, synthesized by the module 26 and/or the module 27, are multiplied by their respective quantized scale factors (multiplier 28) before being presented to the module 29 which performs the frequency/time transformation which is the inverse of the MDCT transformation operated by the module 5 of the coder. The temporal correction signal which results therefrom is added to the synthetic signal S' delivered by the decoder kernel 21 (adder 30) to produce the output audio signal S of the decoder. It should be noted that the decoder will be able to synthesize a signal S even in cases where it does not receive the first NO bits of the sequence. It is sufficient for it to receive the 2 x Nl bits corresponding to the part a of the listing hereinabove, the decoding then being in a "degraded" mode. Only this degraded mode does not use the MDCT synthesis to obtain the decoded signal. To ensure the switching with no break between this mode and the other modes, the decoder performs three MDCT analyses followed by three MDCT syntheses, allowing the updating of the memories of the MDCT transformation. The output signal contains a signal of telephone band quality. If the first 2 x Nl bits are not even received, the decoder considers the corresponding frame as having been erased and can use a known algorithm for conceiving erased frames. If the decoder receives the 2 x Nl bits corresponding to part a plus bits of part b (high bands of the three spectral envelopes), it can begin to synthesize a wide band signal. It can in particular proceed as follows. 1/ The module 22 recovers the parts of the three spectral envelopes received. 2/ The bands not received have their scale factors temporarily set to zero. 3/ The low parts of the spectral envelopes are calculated on the basis of the MDCT analyses performed on the signal obtained after the G.723.1 decoding, and the module 23 calculates the three masking curves on the envelopes thus obtained. 4/ The spectral envelope is corrected so as to regularize it by avoiding the nulls due to the bands not received; the zero values in the high part of the spectral envelopes FQ are for example replaced by a hundredth of the value of the masking curve calculated previously, so that they remain inaudible. The complete spectrum of the low bands and the spectral envelope of the high bands are known at this juncture. 5/ The module 27 then generates the high spectrum. The fine structure of these bands is generated by reflection of the fine structure of its known neighborhood before weighting by the scale factors (multipliers 28) . In the case where none of the bits P3 is received, the "known neighborhood" corresponds to the spectrum of the signal S' produced by the G.723.1 decoder kernel. Its "reflection" can consist in copying the value of the standardized MDCT spectrum, possibly with its variations being attenuated in proportion to the distance away from the "known neighborhood". 6/ After inverse MDCT transformation (29) and addition (30) of :he resulting correction signal to the output signal of the decoder kernel, the wide band synthesized signal is obtained. In the case where the decoder also receives part at least of the low spectral envelope of the difference signal (part c) , it may or may not take this information into account to refine the spectral envelope in step 3. If the decoder 10 receives enough bits P3 to decode at least the MDCT coefficients of the most important band, ranked first in the part d of the sequence, then the module 26 recovers certain of the normalized MDCT coefficients according to the allocation and ordering that are indicated by the modules 24 and 25. These MDCT coefficients therefore, need not be interpolated as in step 5 hereinabove. For the other bands, the process of steps 1 to 6 is applicable by the module 27 in the same manner as previously; the knowledge of the MDCT coefficients received for certain bands allowing more reliable interpolation in step 5. The bands not received may vary from one MDCT subframe to the next. The "known neighborhood" of a missing band may correspond to the same band in another subframe where . it is not missing, and/or to one or more bands closest in the frequency domain in the course of the same subframe. It is also possible to regenerate an MDCT spectrum missing from a band for a subframe by calculating a weighted sum of contributions evaluated on the basis of several bands/subframes of the "known neighborhood". Insofar as the actual bit rate of N' bits per frame places the last bit of a given frame arbitrarily, the last coded parameter transmitted may, according to case, be transmitted completely or partially. Two cases may then arise: either the coding structure adopted makes it possible to utilize the partial information received (case of scalar quantizers, or of vector quantization with partitioned dictionaries), or it does not allow it and the parameter not fully received is processed like the other parameters not received. It is noted that, for this latter case. if the order of the bits varies with each frame, the number of bits thus lost is variable and the selection of N' bits will produce on average, over the whole set of frames decoded, a better quality than that which would be obtained with a smaller number of bits. WE CLAIM: 1. A method of coding a digital audio signal frame (S) as a binary output sequence (F), in which a maximum number Nmax of coding bits is defined for a set of parameters that can be calculated accordingto the signal frame, which set it composed of a first and of a second subset,the method comprising the following steps: Calculating (1) the parameters of the first subset, and coding these parameters on a number NO of coding bits such that NO determining (12) an allocation of Nmax-NO coding bits for the parameters of the second subset; and ranking (13) the Nmax - NO coding bits allocated to the parameters of the second subset in a determined order, in which the allocation and/or the order of ranking of the Nmax - NO coding bits is determined as a function of the coded parameters of the first subset, the method furthermore comprising the following steps in response to the indication of a number N of bits of the binary output sequence that are available for the coding of said set of parameters, with NO selecting the second subset's parameters to which are allocated the N - NO 'coding bits ranked first in said order; calculating (9) the selected parameters of the second subset, and coding these parameters so as to produce said N - NO coding bits ranked first; and inserting (10) into the output sequence the NO coding bits of the first subset as well as the N - NO coding bits of the selected parameters of the second subset. 2. The method as claimed in claim 1, in which the order of ranking 'of the coding bits allocated -to the parameters of the second subset is variable form one frame to another. 3. The method as claimed in claim 1 or 2, in which N 4. The method as claimed in any one of the preceding claims, in which the order of ranking of the coding bits allocated to the parameters of the second subset is an order of decreasing importance determined as a function of at least the coded parameters of the first subset. 5. The method as, claimed in claim 4, in which the order of ranking of the coding bits allocated to the parameters of the second subset is determined with the aid of at least one psychoacoustic criterion as a function of the coded parameters of the first subset. 6. The method as claimed in claim 5, in which the parameters of the second subset pertain to spectral bands of the signal, in which a spectral envelope of the coded signal is estimated on the basis of the coded parameters of the first subset, in which a curve of frequency masking is calculated by applying an auditory perception model to the estimated spectral envelope, and in which the psychoacoustic criterion makes reference to the level of the estimated spectral envelope with respect to the masking curve in each spectral band. 7. The method as claimed in any one of claims 4 to 6, in which Nmax = N. 8. The method as claimed in any one of the preceding claims, in which the coding bits are ordered in the output sequence in such a way that the NO coding bits of the first subset precede the N - NO coding bits of the selected parameters of the second subset and that the respective coding bits of the selected parameters of the second subset appear therein in the order determined for said coding bits. 9. The method as claimed in any one of the preceding claims, in which the number N varies from one frame to another. 10. The method as claimed in any one of the preceding claims, in which the coding of the parameters of the first subset is at variable bit rate, thereby varying the number NO from one frame to another. 11. The method as claimed in any one of the preceding claims, in which the first subset comprises parameters calculated by a coder kernel (1). 12. The method as claimed in claim 11, in which the coder kernel (1) has a lower frequency band of operation than the bandwidth of the signal to be coded, and in which the first subset furthermore comprises energy levels of the audio signal that are associated with frequency bands higher than the operating band of the coder kernel. 13. The method as claimed in each of claims 8 and 12, in which the coding bits of the first subset are ordered in the output sequence in such a way that the coding bits of the parameters calculated by the coder kernel are immediately followed by the coding bits of the energy levels associated with the higher frequency bands. 14. The method as claimed in any one of claims 11 to 13, in which a signal of difference between the signal to be coded and a synthesis signal derived from the coded parameters produced by the coder kernel is estimated, and in which the first subset furthermore comprises energy levels of the difference signal that are associated with frequency bands included in the operating band of the coder kernel 15. The method as claimed in claim 8 and any one of claims 12 to 14, in which the coding bits of the first subset are ordered in the output sequence in such a way that the coding bits of the parameters calculated by the coder kernel (1) are followed by the coding bits of the energy levels associated with the frequency band. 16. A method of decoding a binary input sequence (F) so as to synthesize a digital audio signal (S), in which a maximum number Nmax of coding bits is defined for a set of parameters for describing a signal frame, which set is composed of a first and a second subset, the input sequence comprising, for a signal frame, a number N' of coding bits for said set of parameters, with N' = Nmax, the method comprising the following steps: extracting (21), from said N' bits of the input sequence, a number No of coding bits of the parameters of the first subset if NO recovering (23) the parameters of the first subset on the basis of said NO coding bits extracted; determining (24) an allocation of Nmax - NO coding bits for the parameters of the second subset; and ranking (25) the NmaX - No coding bits allocated to the parameters of the second subset in a determined order, in which the allocation and/or the order of ranking of the Nmax - NO coding bits is determined as a function of the recovered parameters of the first subset, the method furthermore comprising time following steps: selecting the second subset's parameters to which are allocated the N'- NO coding bits -ranked first in said order; extracting (26), from said N' bits of the input sequence, N' - NO coding bits of the selected parameters of the second subset; recovering (29) the selected parameters of the second subset on the basis of said N' - NO coding bits extracted; and synthesizing (30) the signal frame by using the recovered parameters of the first and second subsets. 17. The method as claimed in claim 16, in which the order of ranking of the coding bits allocated to the parameters of the second subset is variable from one frame to another. 18. The method as claimed in claim 16 or 17, in which N' Nmax. 19. The method as claimed in any one of claims 16 to 18, in which the order of ranking of the coding bits allocated to the parameters of the second subset is an order of decreasing importance determined as a function of at least the recovered parameters of the first subset. 20. The method as claimed in claim 19, in which the order of ranking of the coding bits allocated to the parameters of the second subset is determined with the aid of at least one psychoacoustic criterion as a function of the recovered parameters of the first subset. 21. The method as claimed in claim 20, in which the parameters of the second subset pertain to spectral bands of the signal, in which a spectral envelope of the signal is estimated on the basis of the recovered parameters of the first subset, in which a curve of frequency masking is calculated by applying an auditory perception model to the estimated spectral envelope, and in which the psychoacoustic criterion makes reference to the level of the estimated spectral envelope with respect to the masking curve in each spectral band. 22. The method as claimed in any one of claims 16 to 21, in which the NO coding bits of the parameters of the first subset are extracted from the N' bits received at positions of the sequence which precede the positions from which are extracted the N' - NO coding bits of the selected parameters of the second subset. 23. The method as claimed in any one of claims 16 to 22, in which, to synthesize the signal frame, nonselected parameters of the second subset are estimated by interpolation on the basis of at least selected parameters recovered on the basis of said N' - NO coding bits extracted. 24. The method as claimed in any one of claims 16 to 23, in which the first subset comprises input parameters of a decoder kernel (21). 25. The method as claimed in claim 24, in which the decoder kernel (21) has a lower frequency band of operation than the bandwidth of the signal to be synthesized, and in which the first subset furthermore comprises energy levels of the audio signal that are associated with frequency bands higher than the operating band of the decoder kernel. 26. The method as claimed in each of claims 22 and 25, in which the coding bits of the first subset in the input sequence are ordered in such a way that the coding bits of the input parameters of the decoder kernel (21) are immediately followed by the coding bits of the energy levels associated with the higher frequency bands. 27. The method as claimed in claim 26, comprising the following steps if the N' bits of the input sequence (F') are limited to the coding bits of the input parameters of the decoder kernel (21) and to part at least of the coding bits of the energy levels associated with the higher frequency bands: extracting from the input sequence the coding bits of the input parameters of the decoder kernel and said part of the coding bits of the energy levels; synthesizing a base signal (S') in the decoder kernel and recovering energy levels associated with the higher frequency bands on the basis of said extracted coding bits; calculating a spectrum of the base signal; assigning an energy level to each higher band with which is associated an uncoded energy level in the input sequence; synthesizing spectral components for each higher frequency band on the basis of the corresponding energy level and of the spectrum of the base signal in at least one band of said spectrum; applying a transformation into the time domain to the synthesized spectral components so as to obtain a base signal correction signal; and adding together the base signal and the correction signal so as to synthesize the signal frame. 28. The method as claimed in claim 27, in which the energy level assigned to a higher band with which is associated an uncoded energy level in the input sequence is a fraction of a perceptual masking level calculated in accordance with the spectrum of the base signal and the energy levels recovered on the basis of the extracted coding bits. 29. The method as claimed in any one of claims 24 to 28, in which a base signal (S') is synthesized in the decoder kernel, and in which the first subset furthermore comprises energy levels of a signal of difference between the signal to be synthesized and the base signal that are associated with frequency bands included in the operating band of the coder kernel. 30. The method as claimed in any one of claims 25, 26 and 29, in which, for NO parameters of the second subset that pertain to spectral components in frequency bands are estimated with the aid of a calculated spectrum of the base signal and/or selected parameters recovered on the basis of said N' 31. The method as claimed in claim 30, in which the unselected parameters of the second subset in a frequency band are estimated with the aid of a spectral neighborhood of said band, which neighborhood is determined on the basis of the N' coding bits of the input sequence. 32. The method as claimed in claim 22 and any one of claims 25 to 31, in which the coding bits of the input parameters of the decoder kernel (21) are extracted from the N' bits received at positions of the sequence which precede the positions from which are extracted the coding bits of the energy levels associated with the frequency bands. 33. The method as claimed in any one of claims 16 to 32, in which the number N' varies from one frame to another. 34. The method as claimed in any one of claims 16 to 33, in which the number NO varies from one frame to another. 35. An audio coder, comprising means of digital signal processing that are; devised to implement a method of coding as claimed in any one of claims 1 to 15. 36. An audio decoder, comprising means of digital signal processing tihat are devised to implement a method of decoding as claimed in any one of claims 16 to 34. A method of coding a digital audio signal frame (S) as a binary output sequence (Φ), in which a maximum number Nmax of coding bits is defined for a set of parameters that can be calculated accordingto the signal frame, which set it composed of a first and of a second subset,the method comprising the following steps: - Calculating (1) the parameters of the first subset, and coding these parameters on a number NO of coding bits such that NO - determining (12) an allocation of Nmax-NO coding bits for the parameters of the second subset; and - ranking (13) the Nmax - NO coding bits allocated to the parameters of the second subset in a determined order, in which the allocation and/or the order of ranking of the Nmax - NO coding bits is determined as a function of the coded parameters of the first subset, the method furthermore comprising the following steps in response to the indication of a number N of bits of the binary output sequence that are available for the coding of said set of parameters, with NO - selecting the second subset's parameters to which are allocated the N - NO 'coding bits ranked first in said order; - calculating (9) the selected parameters of the second subset, and coding these parameters so as to produce said N - NO coding bits ranked first; and - inserting (10) into the output sequence the NO coding bits of the first subset as well as the N - NO coding bits of the selected parameters of the second subset. 2. The method as claimed in claim 1, in which the order of ranking 'of the coding bits allocated -to the parameters of the second subset is variable form one

Full Text

The present invention relates to method for coding a digital
audio frame and an audio coder to implement the said method.The
invention further relates to devices for coding and decoding
audio signals, intended in particular to sit within applications
of transmission or storage of digitized and compressed audio
signals (speech and/or sounds).
More particularly, this invention pertains to audio coding
systems having the capacity to provide varied bit rates, also
referred to as multirate coding systems. Such systems are
distinguished from fixed rate coders by their capacity to modify
the bit rate of the coding, possibly during processing, this
being especially suited to transmission over heterogeneous access
networks: be they networks of IP type mixing fixed and mobile
access, high bit rates (ADLS) , low bit rates (RTC, GPRS modems)
or involving terminals with variable capacities (mobiles, PCs,
etc.).
Essentially, two categories of multirate coders are
distinguished: that of "switchable" multirate coders and that of
"hierarchical" coders.
"Switchable" multirate coders rely on a coding architecture
belonging to a technologic'al family (temporal coding or frequency
coding, for example: CELP, sinusoidal, or by transform), in which
an indication of bit rate is simultaneously supplied to the coder
and to the decoder. The coder uses this information to select the
parts of the algorithm and the tables relevant to the bit rate
chosen. The decoder operates in a symmetric manner. Numerous
switchable multirate coding structures have been proposed for
audio coding. Such is the case for example with mobile coders
standardized by the 3GPP organization ("3rd Generation
Partnership Project"), NB-AMR ("Narrow Band
Adaptive Multirate", Technical Specification 3GPP
TS 26.090, version 5.0.0, June 2002) in the telephone
band, or WB-AMR ("Wide Band Adaptive Multirate",
Technical Specification 3GPP TS 26.190, version 5.1.0,
December 2001) in wideband. These coders operate over
fairly wide bit rate ranges (4.75 to 12.2 kbit/s for
NB-AMR, and 6.60 to 23.85 kbit/s for WB-AMR), with a
fairly sizeable granularity (8 bit rates for NB-AMR and
9 for WB-AMR) . However, the price to be paid for this
flexibility is a rather considerable complexity of
structure: to be able to host all these bit rates,
these coders must support numerous different options,
varied quantization tables etc. The performance curve
increases progressively with bit rate, but the progress
is not linear and certain bit rates are in essence
better optimized than others.
In so-called "hierarchical" coding systems, also
referred to as "scalable", the binary data arising from
the coding operation are distributed into successive
layers. A base layer, also called the "kernel", is
formed of the binary elements that are absolutely
necessary for the decoding of the binary train, and
determine a minimum quality of decoding.
The subsequent layers make it possible to progressively
improve the quality of the signal arising from the
decoding operation, each new layer bringing new
information which, utilized by the decoder, supplies a
signal of increasing quality at output.
One of the particular features of hierarchical coding
is the possibility offered of intervening at any level
whatsoever of the transmission or storage chain so as
to delete a part of the binary train without having to
supply any particular indication to the coder or to the
decoder. The decoder uses the binary information that
it receives and produces a signal of corresponding
quality.
The field of hierarchical coding structures has given
rise likewise to much work. Certain hierarchical coding
structures operate on the basis of one type of coder
alone, designed to deliver hierarchized coded
information. When the additional layers improve the
quality of the output signal without modifying the
bandwidth, one speaks rather of "embedded coders" (see
for example R.D. Lacpvo et al., "Embedded CELP Coding
for Variable Bit-Rate Between 6.4 and 9.6 kbit/s, Proc
ICASSP 1991, pp. 681-686) . Coders of this type do not
however allow large 'gaps between the lowest and the
highest bit rate proposed.
The hierarchy is often used to progressively increase
the bandwidth of the signal: the kernel supplies a
baseband signal, for Example telephonic (300-3400 Hz),
and the subsequent layers allow the coding of
additional frequency bands (for example, wide band up
to 7 kHz, HiFi band up to 20 kHz or intermediate,
etc.). The subband coders or coders using a
time/frequency transformation such as described in the
documents "Subband/trahsform coding using filter banks
designs based on time domain aliasing cancellation" by
J.P. Princen et al. (:Proc. IEEE ICASSP-87, pp. 2161-
2164) and "High Quality Audio Transform Coding at
64 kbit/s", by Y. Mahieux et al. (IEEE Trans. Commun.,
Vol. 42, No. 11, November 1994, pp. 3010-3019), lend
themselves particularly to such operations.
Moreover, a different coding technique is frequently
used for the kernel and for the module or modules
coding the additional layers, one then speaks of
various coding stages, each stage consisting of a
subcoder. The subcoder of the stage of a given level
will be able either to code parts of the signal that
are not coded by the previous stages, or to code the
coding residual of the previous stage, the residual is
obtained by subtracting the decoded signal from the
original signal.
The advantage of such structures it that they make it
possible to go down, to relatively low bit rates with
sufficient quality, while producing good quality at
high bit rate. Specifically, the techniques -used for
low bit rates are not generally effective at high bit
rates and vice versa.;
Such structures making it possible to use two different
technologies (for example CELP and time/frequency
transform, etc.) are especially effective for sweeping
large bit rate ranges.
However, the hierarchical coding structures proposed in
the prior art define precisely the_bit. rate allocated
to eacn of the intermediate layers. Each layer
corresponds to the encoding of certain parameters, and
the granularity of the hierarchical binary train
depends on the bit rate allocated to these parameters
(typically a layer can contain of the order of a few
tens of bits per frame, a signal frame consisting of a
certain number of samples of the signal over a given
duration, the example described later considering a
frame of 960 samples corresponding to 60 ms of signal).
Moreover, when the bandwidth of the decoded signals can
vary according to the level of the layers of binary
elements, the modification of the line bit rate may
produce artifacts that impede listening.
The present invention has the aim in particular of
proposing a multirate coding solution which alleviates
the drawbacks cited in the case of the use of existing
hierarchical and switchable codings.
The invention thus proposes a method of coding a
digital audio signal frame as a binary output sequence,
in wnich a maximum number Nmax of coding bits is
defined for a set of parameters that can be calculated
according to the signal frame, which set is composed of
a first and of a second subset. The proposed method
comprises the following steps:
calculating the; parameters of the first subset,
and coding these parameters on a number NO of
coding bits such that NO determining an allocation of Nmax - NO coding bits
for the parameters of the second subset; and
ranking the Nmax - NO coding bits allocated to the
parameters of the second subset in a determined
order.
The allocation and/or the order of ranking of the
Nmax - NO coding bits are determined as a function of
the coded parameters] of the first subset. The coding
method furthermore comprises the following steps in
response to the indication of a number N of bits of the
binary output sequence that are available for the
coding of said set of parameters, with NO selecting the second subset's parameters to which
are allocated the N - NO coding bits ranked first
in said order;
calculating the selected parameters of the second
subset, and coding these parameters so as to
produce said N - nO coding bits ranked first; and
inserting into the output sequence the NO coding
bits of the first subset as well as the N - NO
coding bits of the selected parameters of the
second subset.
The method according to the invention makes it possible
to define a multirate coding, which will operate at
least in a range corresponding for each frame to a
number of bits ranging' from NO to Nmax.
It may thus be considered that the notion of pre-
established bit rates which is related to the existing
hierarchical and switchable codings is replaced by a
notion of "cursor", making it possible to freely vary
the bit rate between a minimum value (that may possibly
correspond to a number of bits N less than NO) and a
maximum value (corresponding to Nmax). These extreme
values are potentially far apart. The method offers
good performance in terms of effectiveness of coding
regardless of the bit rate chosen.
Advantageously, the [number N of bits of the binary
output sequence is strictly less than Nmax. What is
noteworthy about the coder is then that the allocation
of the bits that is employed makes no reference to the
actual output bit rate of the coder, but to another
number Nmax agreed with the decoder.
It is however possible to fix Nmax = N as a function of
the instantaneous bit, rate available on a transmission
channel. The output sequence of a switchable multirate
coder such as this may be processed by a decoder which
does not receive the entire sequence, so long as it is
capable of retrieving the structure of the coding bits
of the second subset by virtue of the knowledge of
Nmax.
Another case where it is possible to have N = Nmax is
that of the storage of;audio data at the maximum coding
rate. When reading N' bits of this content stored at
lower bit rate, the decoder would be capable of
retrieving the structure of the coding bits of the
second subset as long a>s N' > NO.
The order of ranking of the coding bits allocated to
the parameters of the second subset may be a
preestablished order.

In a preferred embodiment, the order of ranking of the
coding bits allocated to the parameters of the second
subset is variable. It may in particular be an order of
decreasing importance determined as a function of at
least the coded parameters of the first subset. Thus
the decoder which receives a binary sequence of N' bits
for the frame, with NO deduce this order from the NO bits received for the
coding of the first subset.
The allocation of the Nmax - NO bits to the coding of
the parameters of the second subset may be carried out
in a fixed manner (in this case, the order of ranking
of these bits will be dependent at least on the coded
parameters of the first subset).
In a preferred embodiment, the allocation of the
Nmax - NO bits to the coding of the parameters of the
second subset is a function of the coded parameters of
the first subset.
Advantageously, this order of ranking of the coding
bits allocated to the parameters of the second subset
is determined with the aid of at least one
psychoacoustic criterion as a function of the coded
parameters of the first; subset.
The parameters of the second subset pertain to spectral
bands of the signal. In this case, the method
advantageously comprises a step of estimating a
spectral envelope of the coded signal on the basis of
the coded parameters of the first subset, and a step of
calculating a curve of frequency masking by applying an
auditory perception model to the estimated spectral
envelope, and the psychoacoustic criterion makes
reference to the level of the estimated spectral
envelope with respect to the masking curve in each
spectral band.
In a mode of implementation, the coding bits are
ordered in the output sequence in such a way that the
NO coding' bits of the first subset precede the N - NO
coding bits of the selected parameters of the second
subset and that the respective coding bits of the
selected parameters of the second subset appear therein
in the order determined for said coding bits. This
makes it possible, in the case where the binary
sequence is truncated, to receive the most important
part.
The number N may vary from one frame ' to another, in
particular as a function for example of the available
capacity of the transmission resource.
The multirate audio coding according to the present
invention may be used according to a very flexible
hierarchical or switchable mode, since any number of
bits to be transmitted chosen freely between NO and
Nmax may be selected at any moment, that is to say
frame by frame.
The coding of the parameters of the first subset may be
at variable bit rate, thereby varying the number NO
from one frame to another. This allows best adjustment
of the distribution of the bits as a function of the
frames to be coded.
In a mode of implementation, the first subset comprises
parameters calculated by a coder kernel.
Advantageously, the coder kernel has a lower frequency
band of operation than the bandwidth of the signal to
be coded, and the first subset furthermore comprises
energy levels of the audio signal that are associated
with frequency bands higher than the operating band of
the coder kernel. This type of structure is that of a
hierarchical coder with two levels, which delivers for
example via the coder kernel a coded signal of a
quality deemed to be sufficient and which, as a
function of the bit rate available, supplements the
coding performed by the coder kernel with additional
information arising from the method of coding according
to the invention.
Preferably, the coding bits of the first subset are
then ordered in the output sequence in such a way that
the coding bits of the parameters calculated by the
coder kernel are immediately followed by the coding
bits of the energy levels associated with the higher
frequency bands. This ensures one and the same
bandwidth for the successively coded frames as long as
the decoder receives enough bits to be in possession of
information of the coder kernel and coded energy levels
associated with the higher frequency bands.
In a mode of implementation, a signal of difference
between the signal to be coded and a synthesis signal
derived from the coded parameters produced by the coder
kernel is estimated, and the first subset furthermore
comprises energy levels of the difference signal that
are associated with frequency bands included in the
operating band of the coder kernel.
A second aspect of the invention pertains to a method
of decoding a binary input sequence so as to synthesize
~a"~digital audio signal corresponding to the decoding of
a frame coded according to the method of coding of the
invention. According to this method, a maximum number
Nmax of coding bits is defined for a set of parameters
for describing a signal frame, which set is composed of
a first and a second subset. The input sequence
comprises, for a signal frame, a number N' of coding
bits for the set of parameters, with N' decoding method according to the invention comprises
the following steps:
extracting, from said N' bits of the input
sequence, a number NO of coding bits of the
parameters of the first subset if NO recovering the parameters of the first subset on
the basis of said NO coding bits extracted;
determining an allocation of Nmax - NO coding bits
for the parameters of the second subset; and
ranking the Nmax - NO coding bits allocated to the
parameters of the second subset in a determined
order.
The allocation and/or the order of ranking of the
Nmax - NO coding bits are determined as a function of
the recovered parameters of the first subset. The
decoding method furthermore comprises the following
steps:
selecting the second subset's parameters to which
are allocated the N' - NO coding bits ranked first
in said order;
extracting, from said N' bits of the input
sequence, N' - NO coding bits of the selected
parameters of the second subset;
recovering the selected parameters of the second
subset on the basis of said N' - NO coding bits
extracted; and
synthesizing the signal frame by using the
recovered parameters of the first and second
subsets.
This method of decoding is advantageously associated
with procedures for regenerating the parameters which
are missing on account of the truncation of the
sequence of Nmax bits that is produced, virtually or
otherwise, by the coder.
A third aspect of the invention pertains to an audio
coder, comprisinq means of digital signal processing
that are devised to implement a method of coding
according to the invention.
Another aspect of the invention pertains to an audio
decoder, comprising means of digital-signal processing
tha are devised to implement a method of decoding
according to the invention.
Other features and advantages of the present invention
will become apparent in the description hereinbelow of
nonlimiting exemplary embodiments, with reference to
the appended drawings,] in which:
figure 1 is a schematic diagram of an exemplary
audio coder according to the invention;
figure 2 represents a binary output sequence of N
bits in a embodiment of the invention; and
figure 3 is a schematic diagram of an audio
decoder according to the invention.
The coder represented I in figure 1 has a hierarchical
structure with two coding stages. A first coding stage
1 consists for example of a coder kernel in a telephone
band (300-3400 Hz) of CELP type. This coder is in the
example considered a G.723.1 coder standardized by the
ITU-T ("International Telecommunication Union") in
fixed mode at 6.4 kbit/s. It calculated G. 723.1
parameters in accordance with the standard and
quantizes them by means, of 192 coding bits P1 per frame
of 30 ms.
The second coding stage 2, making it possible to
increase the bandwidth towards the wide band
(50-7000 Hz) , operates on the coding residual E of the
first stage, supplied by a subtractor 3 in the diagram
of figure 1. A signals synchronization module 4 delays
the audio signal frame S by the time taken by the
processing of the coder kernel 1. Its output is
addressed 'to the subtractor 3 which subtracts from it
the synthetic signal S' equal to the output of the
decoder kernel operating on the basis of the quantized
parameters such as represented by the output bits P1 of
the coder kernel. As is usual, the coder 1 incorporates
a local decoder supplying S',
The audio signal to be coded S has for example a
bandwidth of 7 kHz, while being sampled at 16 kHz. A
frame consists for example of 960 samples, i.e. 60 ms
of signal or two elementary frames of the coder kernel
G. 723.1. Since the latter operates on signals sampled
at 8 kHz, the signal S is subsampled in a factor 2 at
the input of the coder kernel 1. Likewise, the
synthetic signal S' is oversampled at 16 kHz at the
output of the coder kernel 1.
The bit rate of the first stage 1 is 6.4 kbit/s
(2 x Nl = 2 x 192 = 384 bits per frame). If the coder
has a maximum bit rate of 32 kbit/s (Nmax = 1920 bits
per frame), the maximum bit rate of the second stage is
25.6 kbit/s (1920 - 384 = 1536 bits per frame). The
second stage 2 operates for example on elementary
frames, or subframes, of 20 ms (320 samples at 16 kHz).
The second stage 2 comprises a time/frequency
transformation module 5, for example of MDCT ("Modified
Discrete Cosine Transform") type to which the residual
E obtained by the subtractor 3 is addressed. In
practice, the manner of operation of the modules 3 and
5 represented in figure 1 may be achieved by performing
the following operations for each 20 ms subframe:
MDCT transformation of the input signal S delayed
by the module 4, which supplies 320 MDCT
coefficients. The spectrum being limited to
7225 Hz, only the first 289 MDCT coefficients are
different from 0;
MDCT transformation of the synthetic signal S'.
Since one is dealing with the spectrum of a
telephone band signal, only the first 139 MDCT
coefficients are different from 0 (up to 3450 Hz)/
and
calculation of the spectrum of difference between
the previous spectra.
The resulting spectrum is distributed into several
bands of different widths by a module 6. By way of
example, the bandwidth of the G.723.1 codec may be
subdivided into 21 bands while the higher frequencies
are distributed into 11 additional bands. In these 11
additional bands, the residual E is identical to the
input signal S.
A module 7 performs the coding of the spectral envelope
of the residual E. It begins by calculating the energy
of the MDCT coefficients of each band of the difference
spectrum. These energies are hereinbelow referred to as
"scale factors". The 32 scale factors constitute the
spectral envelope of the difference signal. The module
7 then proceeds to their quantization in two parts. The
first part corresponds to the telephone band (first 21
bands, from 0 to 3450 Hz), the second to the high bands
(last 11 bands, from 3450 to 7225 Hz) . In each part,
the first scale factor is quantized on an absolute
basis, and the subsequent ones on a differential basis,
by using a conventional Huffman coding with variable
bit rate. These 32 scale factors are quantized on a
variable number N2(i) of bits P2 for each subframe of
rank i (i = 1, 2, 3) .
The quantized scale factors are denoted FQ in figure 1.
The quantization bits P1, P2 of the first subset
consisting of the quantized parameters of the coder kernel
1 and the quantized scale factors FQ are variable in
number NO = (2 x N1) + N2(l) + N2(2) + N2(3). The
difference Nmax - NO = 1536 - N2(l) - N2{2) - N2(3) is
available to quantize the spectra of the bands more
finely.
A module 8 normalizes the MDCT coefficients distributed
into bands by the module 6, by dividing them by the
quantized scale factors FQ respectively determined for
these bands. The spectra thus normalized are supplied
to the quantization module 9 which uses a vector
quantization scheme of known type. The quantization
bits, arising from the module 9 are denoted P3 in
figure 1.
An output multiplexer 10 gathers together the bits P1,
P2 and P3 arising from the modules 1, 7 and 9 to form
the binary output sequence F of the coder.
In accordance with the invention, the total number of
bits N of the output sequence representing a current
frame is not necessarily equal to Nmax. It may be less
than the latter. However, the allocation of the
quantization bits to the bands is performed on the
basis of the number Nmax.
In the diagram of figure 1, this allocation is
performed for each subframe by the module 12 on the
basis of the number Nmax - NO, of the quantized scale
factors FQ and of a spectral masking curve calculated
by a module 11.
The manner of operation of the latter module 11 is as
follows. It firstly determines an approximate value of
the original spectral envelope of the signal S on the
basis of that of the difference signal, such as
quantized by the module 7, and of that which it
determines with the same resolution for the synthetic
signal S' resulting from the coder kernel. These last
two envelopes are also determinable by a decoder which
is provided only with the parameters of the aforesaid
first subset. Thus the estimated spectral envelope of
the signal S will also be available to the decoder.
Thereafter, the module 11 calculates a spectral masking
curve by applying, in a manner known per se, a model of
band by band auditlory perception to the original
estimated spectral envelope. This curve 11 gives a
masking level for each band considered.
The module 12 carries out a dynamic allocation of the
Nmax - NO remaining bits of the sequence F among the
3 x 32 bands of the three MDCT transformations of the
difference signal. In the implementation of the
invention set forth here, as a function of a criterion
of psychoacoustic perceptual importance making
reference to the level of the spectral envelope
estimated with respect to the masking curve in each
band, a bit rate proportional to this level is
allocated to each band. Other ranking criteria would be
useable.
Subsequent to this allocation of bits, the module 9
knows how many bits are to be considered for the
quantization of each band in each subframe.
Nevertheless, if N not necessarily all be used. An ordering of the bits
representing the bands is performed by a module 13 as a
function of a criterion of perceptual importance. The
module 13 ranks the 3 x 32 bands in an order of
decreasing importance which may be the decreasing order
of the signal-to-mask ratios (ratio between the
estimated spectral envelope and the masking curve in
each band) . This order is used for the construction of
the binary sequence F in accordance with the invention.
As a function of the desired number N of bits in the
sequence F for the coding of the current frame, the
bands which are to be quantized by the module 9 are
determined by selecting the bands ranked first by the
module 13 and by keeping for each band selected a
number of bits such as is determined by the module 12.
Then the MDCT coefficients of each band selected are
quantized by the module 9, for example with the aid of
a vector quantizer, in accordance with the allocated
number of bits, so as to produce a total number of bits
equal to N - NO.
The output multiplexer 10 builds the binary sequence F
consisting of the first N bits qf the following ordered
sequence represented in figure 2 (case N = Nmax):
a/ firstly the binary trains corresponding to the two
G. 723.1 frames (384 bits');
b/ next the bits for quantizing the scale
factors, for the three subframes (i = 1, 2, 3),
from the 22nd spectral band (first band beyond the
telephone band) to the 32nd band (variable rate
Huffman coding);
c/ next the bits for quantizing the scale
factors, for the three subframes (i = 1, 2, 3),
from the 1st spectral band to the 21st band
(variable rate Huffman coding);
d/ and finally the indices Mc1, Mc2, Mc96 of
vector quantization of the 96 bands in order of
perceptual importance, from the most important
band to the least important band, while complying
with the order determined by the module 13.
By placing first (a and b) the G.723.1 parameters and
the scale factors of the high bands it is possible to
retain the same bandwidth for the signal restorable by
the decoder regardless of the actual bit rate beyond a
minimum value corresponding to the reception of these
groups a and b. This minimum value, sufficient for the
Huffman coding of the 3 x 11 = 33 scale factors of the
high bands in addition to the G.723.1 coding, is for
example 8 kbit/s.
The method of coding hereinabove allows a decoding of
the frame if the decoder receives N' bits with
NO from one frame to another.
A decoder according to the invention, corresponding to
this example, is illustrated by figure 3. A
demultiplexer 20 separates the sequence of bits
received F so as to extract therefrom the coding bits
P1 and P2. The 384 bits P1 are supplied to the decoder
kernel 21 of G.723.1 type so that the latter
synthesizes two frames of the base signal S' in the
telephone band. The bits P2 are decoded according to
the Huffman algorithm by a module 22 which thus
recovers the quantized scale factors FQ for each of the
3 subframes.
A module 23 calculating the masking curve, identical to
the module 11 of the coder of figure 1, receives the
base signal S' and the quantized scale factors FQ and
produces the spectral masking levels for each of the 96
bands. On the basis of these masking levels, of the
quantized scale factors FQ and of the knowledge of the
number Nmax (as well as of that of the number NO which
is deduced from the Huffman decoding of the bits P2 by
the module 22), a module 24 determines an allocation of
bits in the same manner as the module 12 of figure 1.
Furthermore, a module 25 proceeds to the ordering of
the bands according to the same ranking criterion as
the module 13 described with reference to figure 1.
According to the information supplied by the modules 24
and 25, the module 26 extracts the bits P3 of the input
sequence F' and synthesizes the normalized MDCT
coefficients relating to the bands represented in the
sequence F' If appropriate (N' standardized MDCT coefficients relating to the missing
bands may furthermore be synthesized by interpolation
or extrapolation as described hereinbelow (module 27) .
These missing bands may have been eliminated by the
coder on account of a truncation to N may have been eliminated in the course of transmission
(N' The standardized MDCT coefficients, synthesized by the
module 26 and/or the module 27, are multiplied by their
respective quantized scale factors (multiplier 28)
before being presented to the module 29 which performs
the frequency/time transformation which is the inverse
of the MDCT transformation operated by the module 5 of
the coder. The temporal correction signal which results
therefrom is added to the synthetic signal S' delivered
by the decoder kernel 21 (adder 30) to produce the
output audio signal S of the decoder.
It should be noted that the decoder will be able to
synthesize a signal S even in cases where it does not
receive the first NO bits of the sequence.
It is sufficient for it to receive the 2 x Nl bits
corresponding to the part a of the listing hereinabove,
the decoding then being in a "degraded" mode. Only this
degraded mode does not use the MDCT synthesis to obtain
the decoded signal. To ensure the switching with no
break between this mode and the other modes, the
decoder performs three MDCT analyses followed by three
MDCT syntheses, allowing the updating of the memories
of the MDCT transformation. The output signal contains
a signal of telephone band quality. If the first 2 x Nl
bits are not even received, the decoder considers the
corresponding frame as having been erased and can use a
known algorithm for conceiving erased frames.
If the decoder receives the 2 x Nl bits corresponding
to part a plus bits of part b (high bands of the three
spectral envelopes), it can begin to synthesize a wide
band signal. It can in particular proceed as follows.
1/ The module 22 recovers the parts of the three
spectral envelopes received.
2/ The bands not received have their scale factors
temporarily set to zero.
3/ The low parts of the spectral envelopes are
calculated on the basis of the MDCT analyses
performed on the signal obtained after the G.723.1
decoding, and the module 23 calculates the three
masking curves on the envelopes thus obtained.
4/ The spectral envelope is corrected so as to
regularize it by avoiding the nulls due to the
bands not received; the zero values in the high
part of the spectral envelopes FQ are for example
replaced by a hundredth of the value of the
masking curve calculated previously, so that they
remain inaudible. The complete spectrum of the low
bands and the spectral envelope of the high bands
are known at this juncture.
5/ The module 27 then generates the high spectrum.
The fine structure of these bands is generated by
reflection of the fine structure of its known
neighborhood before weighting by the scale factors
(multipliers 28) . In the case where none of the
bits P3 is received, the "known neighborhood"
corresponds to the spectrum of the signal S'
produced by the G.723.1 decoder kernel. Its
"reflection" can consist in copying the value of
the standardized MDCT spectrum, possibly with its
variations being attenuated in proportion to the
distance away from the "known neighborhood".
6/ After inverse MDCT transformation (29) and
addition (30) of :he resulting correction signal
to the output signal of the decoder kernel, the
wide band synthesized signal is obtained.
In the case where the decoder also receives part at
least of the low spectral envelope of the difference
signal (part c) , it may or may not take this
information into account to refine the spectral
envelope in step 3.
If the decoder 10 receives enough bits P3 to decode at
least the MDCT coefficients of the most important band,
ranked first in the part d of the sequence, then the
module 26 recovers certain of the normalized MDCT
coefficients according to the allocation and ordering
that are indicated by the modules 24 and 25. These MDCT
coefficients therefore, need not be interpolated as in
step 5 hereinabove. For the other bands, the process of
steps 1 to 6 is applicable by the module 27 in the same
manner as previously; the knowledge of the MDCT
coefficients received for certain bands allowing more
reliable interpolation in step 5.
The bands not received may vary from one MDCT subframe
to the next. The "known neighborhood" of a missing band
may correspond to the same band in another subframe
where . it is not missing, and/or to one or more bands
closest in the frequency domain in the course of the
same subframe. It is also possible to regenerate an
MDCT spectrum missing from a band for a subframe by
calculating a weighted sum of contributions evaluated
on the basis of several bands/subframes of the "known
neighborhood".
Insofar as the actual bit rate of N' bits per frame
places the last bit of a given frame arbitrarily, the
last coded parameter transmitted may, according to
case, be transmitted completely or partially. Two cases
may then arise:
either the coding structure adopted makes it
possible to utilize the partial information
received (case of scalar quantizers, or of vector
quantization with partitioned dictionaries),
or it does not allow it and the parameter not
fully received is processed like the other
parameters not received. It is noted that, for
this latter case. if the order of the bits varies
with each frame, the number of bits thus lost is
variable and the selection of N' bits will produce
on average, over the whole set of frames decoded,
a better quality than that which would be obtained
with a smaller number of bits.
WE CLAIM:
1. A method of coding a digital audio signal frame (S) as
a binary output sequence (F), in which a maximum number
Nmax of coding bits is defined for a set of parameters that
can be calculated accordingto the signal frame, which set
it composed of a first and of a second subset,the method
comprising the following steps:
Calculating (1) the parameters of the first subset, and
coding these parameters on a number NO of coding bits
such that NO determining (12) an allocation of Nmax-NO coding bits
for the parameters of the second subset; and
ranking (13) the Nmax - NO coding bits allocated to the
parameters of the second subset in a determined
order,
in which the allocation and/or the order of ranking of the
Nmax - NO coding bits is determined as a function of the
coded parameters of the first subset, the method
furthermore comprising the following steps in response to
the indication of a number N of bits of the binary output
sequence that are available for the coding of said set of
parameters, with NO selecting the second subset's parameters to which are
allocated the N - NO 'coding bits ranked first in said
order;
calculating (9) the selected parameters of the
second subset, and coding these parameters so as to
produce said N - NO coding bits ranked first;
and
inserting (10) into the output sequence the NO coding
bits of the first subset as well as the N - NO coding
bits of the selected parameters of the second subset.
2. The method as claimed in claim 1, in which the order of
ranking 'of the coding bits allocated -to the parameters of
the second subset is variable form one
frame to another.
3. The method as claimed in claim 1 or 2, in which
N 4. The method as claimed in any one of the preceding
claims, in which the order of ranking of the coding
bits allocated to the parameters of the second subset
is an order of decreasing importance determined as a
function of at least the coded parameters of the first
subset.
5. The method as, claimed in claim 4, in which the
order of ranking of the coding bits allocated to the
parameters of the second subset is determined with the
aid of at least one psychoacoustic criterion as a
function of the coded parameters of the first subset.
6. The method as claimed in claim 5, in which the
parameters of the second subset pertain to spectral
bands of the signal, in which a spectral envelope of
the coded signal is estimated on the basis of the coded
parameters of the first subset, in which a curve of
frequency masking is calculated by applying an auditory
perception model to the estimated spectral envelope,
and in which the psychoacoustic criterion makes
reference to the level of the estimated spectral
envelope with respect to the masking curve in each
spectral band.
7. The method as claimed in any one of claims 4 to 6,
in which Nmax = N.
8. The method as claimed in any one of the preceding
claims, in which the coding bits are ordered in the
output sequence in such a way that the NO coding bits
of the first subset precede the N - NO coding bits of
the selected parameters of the second subset and that
the respective coding bits of the selected parameters
of the second subset appear therein in the order
determined for said coding bits.
9. The method as claimed in any one of the preceding
claims, in which the number N varies from one frame to
another.
10. The method as claimed in any one of the preceding
claims, in which the coding of the parameters of the
first subset is at variable bit rate, thereby varying
the number NO from one frame to another.
11. The method as claimed in any one of the preceding
claims, in which the first subset comprises parameters
calculated by a coder kernel (1).
12. The method as claimed in claim 11, in which the
coder kernel (1) has a lower frequency band of
operation than the bandwidth of the signal to be coded,
and in which the first subset furthermore comprises
energy levels of the audio signal that are associated
with frequency bands higher than the operating band of
the coder kernel.
13. The method as claimed in each of claims 8 and 12,
in which the coding bits of the first subset are
ordered in the output sequence in such a way that the
coding bits of the parameters calculated by the coder
kernel are immediately followed by the coding bits of
the energy levels associated with the higher frequency
bands.
14. The method as claimed in any one of claims 11 to
13, in which a signal of difference between the signal
to be coded and a synthesis signal derived from the
coded parameters produced by the coder kernel is
estimated, and in which the first subset furthermore
comprises energy levels of the difference signal that are
associated with frequency bands included in the operating
band of the coder kernel
15. The method as claimed in claim 8 and any one of claims
12 to 14, in which the coding bits of the first subset are
ordered in the output sequence in such a way that the
coding bits of the parameters calculated by the coder
kernel (1) are followed by the coding bits of the energy
levels associated with the frequency band.
16. A method of decoding a binary input sequence (F) so as
to synthesize a digital audio signal (S), in which a
maximum number Nmax of coding bits is defined for a set of
parameters for describing a signal frame, which set is
composed of a first and a second subset, the input sequence
comprising, for a signal frame, a number N' of coding bits
for said set of parameters, with N' = Nmax, the method
comprising the following steps:
extracting (21), from said N' bits of the input
sequence, a number No of coding bits of the parameters
of the first subset if NO recovering (23) the parameters of the first subset on
the basis of said NO coding bits extracted;
determining (24) an allocation of Nmax - NO coding bits
for the parameters of the second subset; and
ranking (25) the NmaX - No coding bits allocated to the
parameters of the second subset in a determined order,
in which the allocation and/or the order of ranking of the
Nmax - NO coding bits is determined as a function of the
recovered parameters of the first subset, the method
furthermore comprising time following steps:
selecting the second subset's parameters to which are
allocated the N'- NO coding bits -ranked first in said
order;
extracting (26), from said N' bits of the input
sequence, N' - NO coding bits of the selected
parameters of the second subset;
recovering (29) the selected parameters of the second
subset on the basis of said N' - NO coding bits
extracted; and
synthesizing (30) the signal frame by using the
recovered parameters of the first and second subsets.
17. The method as claimed in claim 16, in which the order
of ranking of the coding bits allocated to the parameters of
the second subset is variable from one frame to another.
18. The method as claimed in claim 16 or 17, in which N' Nmax.
19. The method as claimed in any one of claims 16 to 18, in
which the order of ranking of the coding bits allocated to
the parameters of the second subset is an order of
decreasing importance determined as a function of at least
the recovered parameters of the first subset.
20. The method as claimed in claim 19, in which the order
of ranking of the coding bits allocated to the parameters of
the second subset is determined with the aid of at least one
psychoacoustic criterion as a function of the recovered
parameters of the first subset.
21. The method as claimed in claim 20, in which the
parameters of the second subset pertain to spectral bands of
the signal, in which a spectral envelope of the signal is
estimated on the basis of the recovered parameters of the
first subset, in which a curve of frequency masking is
calculated by applying an auditory perception model to the
estimated spectral envelope,
and in which the psychoacoustic criterion makes
reference to the level of the estimated spectral
envelope with respect to the masking curve in each
spectral band.
22. The method as claimed in any one of claims 16 to
21, in which the NO coding bits of the parameters of
the first subset are extracted from the N' bits
received at positions of the sequence which precede the
positions from which are extracted the N' - NO coding
bits of the selected parameters of the second subset.
23. The method as claimed in any one of claims 16 to
22, in which, to synthesize the signal frame,
nonselected parameters of the second subset are
estimated by interpolation on the basis of at least
selected parameters recovered on the basis of said
N' - NO coding bits extracted.
24. The method as claimed in any one of claims 16 to
23, in which the first subset comprises input
parameters of a decoder kernel (21).
25. The method as claimed in claim 24, in which the
decoder kernel (21) has a lower frequency band of
operation than the bandwidth of the signal to be
synthesized, and in which the first subset furthermore
comprises energy levels of the audio signal that are
associated with frequency bands higher than the
operating band of the decoder kernel.
26. The method as claimed in each of claims 22 and 25,
in which the coding bits of the first subset in the
input sequence are ordered in such a way that the
coding bits of the input parameters of the decoder
kernel (21) are immediately followed by the coding bits
of the energy levels associated with the higher
frequency bands.
27. The method as claimed in claim 26, comprising the
following steps if the N' bits of the input sequence
(F') are limited to the coding bits of the input
parameters of the decoder kernel (21) and to part at
least of the coding bits of the energy levels
associated with the higher frequency bands:
extracting from the input sequence the coding bits
of the input parameters of the decoder kernel and
said part of the coding bits of the energy levels;
synthesizing a base signal (S') in the decoder
kernel and recovering energy levels associated
with the higher frequency bands on the basis of
said extracted coding bits;
calculating a spectrum of the base signal;
assigning an energy level to each higher band with
which is associated an uncoded energy level in the
input sequence;
synthesizing spectral components for each higher
frequency band on the basis of the corresponding
energy level and of the spectrum of the base
signal in at least one band of said spectrum;
applying a transformation into the time domain to
the synthesized spectral components so as to
obtain a base signal correction signal; and
adding together the base signal and the correction
signal so as to synthesize the signal frame.
28. The method as claimed in claim 27, in which the
energy level assigned to a higher band with which is
associated an uncoded energy level in the input
sequence is a fraction of a perceptual masking level
calculated in accordance with the spectrum of the base
signal and the energy levels recovered on the basis of
the extracted coding bits.
29. The method as claimed in any one of claims 24 to
28, in which a base signal (S') is synthesized in the
decoder kernel, and in which the first subset
furthermore comprises energy levels of a signal of
difference between the signal to be synthesized and the
base signal that are associated with frequency bands
included in the operating band of the coder kernel.
30. The method as claimed in any one of claims 25, 26
and 29, in which, for NO parameters of the second subset that pertain to
spectral components in frequency bands are estimated
with the aid of a calculated spectrum of the base
signal and/or selected parameters recovered on the
basis of said N' 31. The method as claimed in claim 30, in which the
unselected parameters of the second subset in a
frequency band are estimated with the aid of a spectral
neighborhood of said band, which neighborhood is
determined on the basis of the N' coding bits of the
input sequence.
32. The method as claimed in claim 22 and any one of
claims 25 to 31, in which the coding bits of the input
parameters of the decoder kernel (21) are extracted
from the N' bits received at positions of the sequence
which precede the positions from which are extracted
the coding bits of the energy levels associated with
the frequency bands.
33. The method as claimed in any one of claims 16 to
32, in which the number N' varies from one frame to
another.
34. The method as claimed in any one of claims 16 to
33, in which the number NO varies from one frame to
another.
35. An audio coder, comprising means of digital signal
processing that are; devised to implement a method of
coding as claimed in any one of claims 1 to 15.
36. An audio decoder, comprising means of digital
signal processing tihat are devised to implement a
method of decoding as claimed in any one of claims 16
to 34.

A method of coding a digital audio signal frame (S) as
a binary output sequence (Φ), in which a maximum number
Nmax of coding bits is defined for a set of parameters that
can be calculated accordingto the signal frame, which set
it composed of a first and of a second subset,the method
comprising the following steps:
- Calculating (1) the parameters of the first subset, and
coding these parameters on a number NO of coding bits
such that NO - determining (12) an allocation of Nmax-NO coding bits
for the parameters of the second subset; and
- ranking (13) the Nmax - NO coding bits allocated to the
parameters of the second subset in a determined
order,
in which the allocation and/or the order of ranking of the
Nmax - NO coding bits is determined as a function of the
coded parameters of the first subset, the method
furthermore comprising the following steps in response to
the indication of a number N of bits of the binary output
sequence that are available for the coding of said set of
parameters, with NO - selecting the second subset's parameters to which are
allocated the N - NO 'coding bits ranked first in said
order;
- calculating (9) the selected parameters of the
second subset, and coding these parameters so as to
produce said N - NO coding bits ranked first;
and
- inserting (10) into the output sequence the NO coding
bits of the first subset as well as the N - NO coding
bits of the selected parameters of the second subset.
2. The method as claimed in claim 1, in which the order of
ranking 'of the coding bits allocated -to the parameters of
the second subset is variable form one

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Patent Number

233763

Indian Patent Application Number

1174/KOLNP/2005

PG Journal Number

15/2099

Publication Date

10-Apr-2009

Grant Date

08-Apr-2009

Date of Filing

20-Jun-2005

Name of Patentee

FRANCE TELECOM

Applicant Address

6, PLACE D'ALLERAY, 75015 PARIS

Inventors:

#	Inventor's Name	Inventor's Address
1	MASSALOUK DOMINIQUE	53, RUE DU PRE DE SAINT-MAUR, 22700 PERROS-GUIREC
2	KOVESI BALAZS	16, CHEMIN DU MOULIN A VENT, 22300 LANNION

PCT International Classification Number

G01L 19/14

PCT International Application Number

PCT/FR2003/003870

PCT International Filing date

2003-12-22

PCT Conventions:

#	PCT Application Number	Date of Convention	Priority Country
1	03 00164	2003-01-08	France