LOW-COMPLEXITY CODE EXCITED LINEAR PREDICTION ENCODING

Description

TECHNICAL FIELD

[0001] The present invention relates in general to audio coding, and in particular to code excited linear prediction coding.

BACKGROUND

[0002] Existing stereo, or in general multi-channel, coding techniques require a rather high bit-rate. Parametric stereo is often used at very low bit-rates. However, these techniques are designed for a wide class of generic audio material, i.e. music, speech and mixed content.

[0003] In multi-channel speech coding, very little has been done. Most work has focused on an inter-channel prediction (ICP) approach. ICP techniques utilize the fact that there is correlation between a left and a right channel. Many different methods that reduce this redundancy in the stereo signal are described in the literature, e.g. in [1][2][3].

[0004] The ICP approach models quite well the case where there is only one speaker, however it fails to model multiple speakers and diffuse sound sources (e.g. diffuse background noises). Therefore, encoding a residual of ICP is a must in several cases and puts quite high demands on the required bit-rate.

[0005] Most existing speech codecs are monophonic and are based on the code-excited linear predictive (CELP) coding model. Examples include AMR-NB and AMR-WB (Adaptive Multi-Rate Narrow Band and Adaptive Multi-Rate Wide Band). In this model, i.e. CELP, an excitation signal at an input of a short-term LP synthesis filter is constructed by adding two excitation vectors from adaptive and fixed (innovative) codebooks, respectively. The speech is synthesized by feeding the two properly chosen vectors from these codebooks through the short-term synthesis filter. The optimum excitation sequence in a codebook is chosen using an analysis-by-synthesis search procedure in which the error between the original and synthesized speech is minimized according to a perceptually weighted distortion measure.

[0006] There are two types of fixed codebooks. A first type of codebook is the so-called stochastic codebooks. Such a codebook often involves substantial physical storage. Given the index in a codebook, the excitation vector is obtained by conventional table lookup. The size of the codebook is therefore limited by the bit-rate and the complexity.

[0007] A second type of codebook is an algebraic codebook. By contrast to the stochastic codebooks, algebraic codebooks are not random and require virtually no storage. An algebraic codebook is a set of indexed code vectors whose amplitudes and positions of the pulses constituting the k^th code vector are derived directly from the corresponding index k. This requires virtually no memory requirements. Therefore, the size of algebraic codebooks is not limited by memory requirements. Additionally, the algebraic codebooks are well suited for efficient search procedures.

[0008] It is important to note that a substantial and often also major part of the speech codec available bits are allocated to the fixed codebook excitation encoding. For instance, in the AMR-WB standard, the amount of bits allocated to the fixed codebook procedures ranges from 36% up to 76%. Additionally, it is the fixed codebook excitation search that represents most of the encoder complexity.

[0009] In [7], a multi-part fixed codebook including an individual fixed codebook for each channel and a shared codebook common to all channels is used. With this strategy it is possible to have a good representation of the inter-channel correlations. However, this comes at an extent of increased complexity as well as storage. Additionally, the required bit rate to encode the fixed codebook excitations is quite large because in addition to each channel codebook index one needs also to transmit the shared codebook index. In [8] and [9], similar methods for encoding multi-channel signals are described where the encoding mode is made dependent on the degree of correlation of the different channels. These techniques are already well known from Left/Right and Mid/Side encoding, where switching between the two encoding modes is dependent on a residual, thus dependent on correlation.

[0010] In [10], a method for encoding multichannel signals is described which generalizes different elements of a single channel linear predictive codec. The method has the disadvantage of requiring an enormous amount of computations rendering it unusable in real-time applications such as conversational applications. Another disadvantage of this technology is the amount of bits needed in order to encode the various decorrelation filters used for encoding.

[0011] Another disadvantage with the previously cited solutions described above is their incompatibility towards existing standardized monophonic conversational codecs, in the sense that no monophonic signal is separately encoded thus prohibiting the ability to directly decode a monophonic only signal.

[0012] In [11], multiple stage audio encoding/decoding of a multi-pulse signal is disclosed. An auxiliary multi-pulse setting circuit sets candidates of pulse positions so that the pulse positions to which no pulse is located are selected in an auxiliary multi-phase searching circuit prior to the pulse positions at which pulses have already been encoded in a multi-pulse searching circuit. The auxiliary multi-phase searching circuit generates an auxiliary multi-pulse signal according to the candidates and encodes the auxiliary multi-pulse signal so that the difference between the reproduced audio signal which is obtained by driving a linear predictive filter synthesis filter with the auxiliary multi-pulse signal and an input audio signal is minimized similarly to the multi-pulse searching circuit.

SUMMARY

[0013] A general problem with prior art speech coding is that it requires high bit rates and complex encoders.

[0014] A general object of the present invention is thus to provide improved methods and devices for speech coding. A subsidiary object of the present invention is to provide CELP methods and devices having reduced requirement in terms of bit rates and encoder complexity.

[0015] The above objects are achieved by methods and devices according to the enclosed patent claims wherein the invention is defined by methods for encoding and decoding audio signals according to claims 1 and 12, respectively, and by an encoder according to claim 18 an a decoder according to claim 29. In general words, excitation signals of a first signal encoded by CELP are used to derive a limited set of candidate excitation signals for a second signal, different from the first signal. Preferably, the second signal is correlated with the first signal. In a particular embodiment, the limited set of candidate excitation signals is derived by a rule, which was selected from a predetermined set of rules based on the encoded first signal and/or the second signal. Preferably, pulse locations of the excitation signals of the first encoded signal are used for determining the set of candidate excitation signals. More preferably, the pulse locations of the set of candidate excitation signals are positioned in the vicinity of the pulse locations of the excitation signals of the first encoded signal. The first and second signals may be multi-channel signals of a common speech or audio signal.

[0016] One advantage with the present invention is that the coding complexity is reduced. Furthermore, in the case of multi-channel signals, the required bit rate for transmitting coded signals is reduced.

[0017] Another advantage of the invention is the possibility to be implemented as an extension to existing speech codecs with very few modifications.

BRIEF DESCRIPTION OF THE DRAWINGS

[0018] The invention, together with further objects and advantages thereof, may best be understood by making reference to the following description taken together with the accompanying drawings, in which:

FIG. 1A is a schematic illustration of a code excited linear prediction model;

FIG. 1B is a schematic illustration of a process of deriving an excitation signal;

FIG. 1C is a schematic illustration of an embodiment of an excitation signal for use in a code excited linear prediction model;

FIG. 2 is a block scheme of an encoder and decoder according to the code excited linear prediction model;

FIG. 3A is a diagram illustrating one embodiment of a principle of selecting candidate excitation signals according to the present invention;

FIG. 3B is a diagram illustrating another embodiment of a principle of selecting candidate excitation signals according to the present invention;

FIG. 4 illustrates a possibility to reduce required data entities according to an embodiment of the present invention;

FIG. 5A is a block scheme of an embodiment of encoders and decoders for two signals according to the present invention;

FIG. 5B is a block scheme of another embodiment of encoders and decoders for two signals according to the present invention;

FIG. 6 is a block scheme of an example of encoders and decoders for re-encoding of a signal.

FIG. 7 is a block scheme of an example of encoders and decoders for parallel encoding of a signal for different bit rates;

FIG. 8 is a diagram illustrating the perceptual quality achieved by embodiments of the present invention;

FIG. 9 is a flow diagram of the main steps of an embodiment of an encoding method according to the present invention;

FIG. 10 is a flow diagram of the main steps of a re-encoding method and

FIG. 11 is a flow diagram of the main steps of an embodiment of a decoding method according to the present invention.

DETAILED DESCRIPTION

[0019] A general CELP speech synthesis model is depicted in Fig. 1A. A fixed codebook 10 comprises a number of candidate excitation signals 30, characterized by a respective index k. In the case of an algebraic codebook, the index k alone characterizes the corresponding candidate excitation signal 30 completely. Each candidate excitation signal 30 comprises a number of pulses 32 having a certain position and amplitude. An index k determines a candidate excitation signal 30 that is amplified in an amplifier 11 giving rise to an output excitation signal c_k(n) 12. An adaptive codebook 14, which is not the primary subject of the present invention, provides an adaptive signal v(n), via an amplifier 15. The excitation signal c_k(n) and the adaptive signal v(n) are summed in an adder 17, giving a composite excitation signal u(n). The composite excitation signal u(n) influences the adaptive codebook for subsequent signals, as indicated by the dashed line 13.

[0020] The composite excitation signal u(n) is used as input signal to a transform 1/A(z) in a linear prediction synthesis section 20, resulting in a "predicted" signal ŝ(n) 21, which, typically after post-processing 22, is provided as the output from the CELP synthesis procedure.

[0021] The CELP speech synthesis model is used for analysis-by-synthesis coding of the speech signal of interest. A target signal s(n), i.e. the signal that is going to be resembled is provided. A long-term prediction is made by use of the adaptive codebook, adjusting a previous coding to the present target signal, giving an adaptive signal v(n)=g_p u(n-δ). The remaining difference is the target for the fixed codebook excitation signal, whereby a codebook index k corresponding to an entry C_k should minimize the difference according to typically an objective function, e.g. a mean square measure. In general, the algebraic codebook is searched by minimizing the mean square error between the weighted input speech and the weighted synthesis speech. The fixed codebook search, aims to find the algebraic codebook entry C_k corresponding to index k, such that

is maximized. The matrix H is a filtering matrix whose elements are derived from the impulse response of a weighting filter. y₂ is a vector of components which are dependent on the signal to be encoded.

[0022] This fixed codebook procedure can be illustrated as in Fig. 1B, where an index k selects an entry C_k from the fixed codebook 10 as excitation signal 12. In a stochastic fixed codebook, the index k typically serves as an input to a table look-up, while in an algebraic fixed codebook, the excitation signal 12 are derived directly from the index k. In general the multi-pulse excitation can be written as:

[0023] Where p_i,k are the pulses positions for index k, while b_i,k are the individual pulses amplitudes and P is the number of pulses and δ is the Dirac pulse function:

[0024] Fig. 1C illustrates an example of a candidate excitation signal 30 of the fixed codebook 10. The candidate excitation signal 30 is characterized by a number of pulses 32, in this example 8 pulses. The pulses 32 are characterized by their position P(1)-P(8) and their amplitude, which in a typical algebraic fixed codebook is either + 1 or -1.

[0025] In an exemplary encoder/decoder system for a single channel, the CELP model is typically implemented as illustrated in Fig. 2. The different parts corresponding to the different functions of the CELP synthesis model of Fig. 1A are given the same reference numbers, since the parts mainly are characterized by their function and typically not in the same degree by their actual implementation. For instance, error weighting filters, usually present in an actual implementation of a linear prediction analysis by synthesis are not represented.

[0026] A signal to be encoded s(n) 33 is provided to an encoder unit 40. The encoder unit comprises a CELP synthesis block 25 according to the above discussed principles. (Post-processing is omitted in order to facilitate the reading of the figure.) The output from the CELP synthesis block 25 is compared with the signal s(n) in a comparator block 31. A difference 37, which may be weighted by a weighting filter, is provided to an codebook optimization block 35, which is arranged according to any prior-art principles to find an optimum or at least reasonably good excitation signal c_k(n) 12. The codebook optimization block 35 provides the fixed codebook 10 with the corresponding index k. When the final excitation signal is found, the index k and the delay δ of the adaptive codebook 12 are encoded in an index encoder 38 to provide an output signal 45 representing the index k and the delay δ.

[0027] The representation of the index k and the delay δ is provided to a decoder unit 50. The decoder unit comprises a CELP synthesis block 25 according to the above discussed principles. (Post-processing is also here omitted in order to facilitate the reading of the figure.) The representation of index k and delay δ are decoded in an index decoder 53, and index k and delay δ are provided as input parameters to the fixed codebook and the adaptive code, respectively, resulting in a synthesized signal ŝ(n) 21, which is supposed to resemble the original signal s(n).

[0028] The representation of the index k and the delay δ can be stored for a shorter or longer time anywhere between the encoder and decoder, enabling e.g. audio recordings storing requiring relatively small storing capability.

[0029] The present invention is related to speech and in general audio coding. In a typical case, it deals with cases where a main signal s_M(n) has been encoded according to the CELP technique and the desire is to encode another signal s_S(n). The other signal could be a signal corresponding to another channel, e.g. stereo, multi-channel 5.1, etc.

[0030] This invention is thus directly applicable to stereo and in general multi-channel coding for speech in teleconferencing applications. The application of this invention can also include audio coding as part of an open-loop or closed-loop content dependent encoding.

[0031] There should preferably exist a correlation between the main signal and the other signal, in order for the present invention to operate in optimal conditions. However, the existence of such correlation is not a mandatory requirement for the proper operation of the invention. In fact, the invention can be operated adaptively and made dependent on the degree of correlation between the main signal and the other signal. Since there exist no causal relationship between a left and right channel in stereo applications, the main signal s_M(n) is often chosen as the sum signal and s_S(n) as the difference signal of the left and right channels.

[0032] The presumption of the present invention is that the main signal s_M(n) is available in a CELP encoded representation. One basic idea of the present invention is to limit the search in the fixed codebook during the encoding of the other signal s_S(n) to a subset of candidate excitation signals. This subset is selected dependent on the CELP encoding of the main signal. In a preferred embodiment, the pulses of the candidate excitation signals of the subset are restricted to a set of pulse positions that are dependent on the pulse positions of the main signal. This is equivalent to defining constrained candidate pulse locations. The set of available pulse positions can typically be set to the pulse positions of the main signal plus neighboring pulse positions.

[0033] This reduction of the number of candidate pulses reduces dramatically the computational complexity of the encoder.

[0034] Below, an illustrative example is given for the general case of two channel signals. However, this is easily extended to multiple channels. However, in the case of multiple channels, the target may be different given different weighting filters on each channel, but also the targets on each channels may be delayed with respect to each other.

[0035] A main channel and a side channel can be constructed by



where s_L(n) and s_R(n) are the input of the left and right channel respectively. One can clearly see that even if the left and right channel were a delayed version of each other, then this would not be the case for the main and the side channel, since in general these would contain information from both channels.

[0036] In the following, it is assumed that the main channel is the first encoded channel and that the pulses locations for the fixed codebook excitation for that encoding are available.

[0037] The target for the side signal fixed codebook excitation encoding is computed as the difference between the side signal and the adaptive codebook excitation:

where g_Pν(n) is the adaptive codebook excitation and s_C(n) is the target signal for adaptive codebook search.

[0038] In the present embodiment, the number of potential pulse positions of the candidate excitation signals are defined relative to the main signal pulse positions. Since they are only a fraction of all possible positions, the amount of bits required for encoding the side signal with an excitation signal within this limited set of candidate excitation signals is therefore largely reduced, compared with the case where all pulse positions may occur.

[0039] The selection of the pulses candidate positions relatively to the main pulse position is fundamental in determining the complexity as well as the required bit-rate.

[0040] For example, if the frame length is L and if the number of pulses in the main signal encoding is N, then one would need roughly N*log2(L) bits to encode the pulse positions. However for encoding the side signal, if one retains only the main signal pulse positions as candidates, and the number of pulses in candidate excitation signals for the side signal is P, then one needs roughly P*log2(N) bits. For reasonable numbers for N, P and L, this corresponds to quite a reduction in bit rate requirements.

[0041] One interesting aspect is when the pulse positions for the side signal are set equal to the pulse positions of the main signal. Then there is no encoding of the pulse positions needed and only encoding of the pulse amplitudes is needed. In the case of algebraic code books with pulses having +1/-1 amplitudes, then only the signs (N bits) need to be encoded.

[0042] If we denote by P_M(i), i =1,···n, the main signal pulse positions. The pulse positions of candidate excitation signals for the side signal are selected based on the main signal pulse positions and possible additional parameters. The additional parameters may consist of time delay between the two channels and/or difference of adaptive codebook index.

[0043] In this embodiment, the set of pulse positions for the side signal candidate excitation signal is constructed as

where J(i,k) denote some delay index. This means that each mono pulse position generates a set of pulse positions used for constructing the candidate excitation signals for the side signal pulse search procedure. This is illustrated in Fig. 3A. Here, P_M denotes the pulse positions of the excitation signal for the main signal, and

denotes possible pulse positions of the candidate excitation signals for the side signal analysis.

[0044] This of course is optimal with highly correlated signals. For low correlated or uncorrelated signals the inverse strategy would be adopted. This consists in taking the pulses candidates as all pulses not belonging to the set

[0045] Since this is a complementary case, it is easily understood by those skilled in the art that both strategies are similar and only the correlated case will be described in more detail.

[0046] It is easily seen that the position and number of pulse candidates is dependent on the delay index J(i,k). The delay index may be made dependent on the effective delay between the two channels and/or the adaptive codebook index. In Fig. 3A, k max=3, and J(i, k) = J(k)∈ {-1,0,+1}.

[0047] In Fig. 3B, another slightly different selection of pulse positions is made. Here k max = 3 , but J(i,k) = J(k)∈ {0,+1,+2}.

[0048] Anyone skilled in the art realizes that the rules how to select the pulse positions can be constructed in many various manners. The actual rule to use may be adapted to the actual implementation. The important characteristics are, however, that the pulse positions candidates are selected dependent on the pulse positions resulting from the main signal analysis following a certain rule. This rule may be unique and fixed or may be selected from a set of predetermined rules dependent on e.g. the degree of correlation between the two channels and/or the delay between the two channels.

[0049] Dependent on the rule used, the set of pulse candidates of the side signal is constructed. The set of the side signal pulse candidates is in general very small compared to the entire frame length. This allows reformulating the objective maximization problem based on a decimated frame.

[0050] In the general case, the pulses are searched by using, for example, the depth-first algorithm described in [5] or by using an exhaustive search if the number of candidate pulses is really small. However, even with a small number of candidates it is recommended to use a fast search procedure.

[0051] A backward filtered signal is in general pre-computed using

[0052] The matrix Φ = H^TH is the matrix of correlations of h(n) (the impulse response of a weighting filter), elements of which are computed by

[0053] The objective function can therefore be written as

[0054] Given the set of possible candidate pulse positions on the side signal, only a subset of indices of the backward filtered vector d and the matrix Φ are needed. The set of candidate pulses can be sorted in ascending order



are the candidate pulses positions and p is their number. It should be noted that p is always less than, and typically much less than, the frame length L.

[0055] If we denote the decimated signal

[0056] And the decimated correlations matrix Φ₂

[0057] Φ₂ is symmetric and is positive definite. We can directly write

where c'_k' is the new algebraic code vector. The index becomes k' which is a new entry in a reduced size codebook.

[0058] The summary of these decimation operations is illustrated in Fig. 4. In the top of the figure, a reduction of an algebraic codebook 10 of ordinary size to a reduced size codebook 10' is illustrated. In the middle, a reduction of a weighting filter covariance matrix 60 of ordinary size to a reduced weighting filter covariance matrix 60' is illustrated. Finally, in the bottom part, a reduction of a backward filtered target 62 of ordinary size to a reduced size backward filtered target 62' is illustrated. Anyone skilled in the art realizes the reduction in complexity that is the result of such a reduction.

[0059] Maximizing the objective function on the decimated signals has several advantages. One of them is the reduction of memory requirements, for instance the matrix Φ₂ needs lower memory. Another advantage is the fact that because the main signal pulse locations are in all cases transmitted to the receiver, the indices of the decimated signals are always available to the decoder. This in turn allows the encoding of the other signal (side) pulse positions relatively to the main signal pulse positions, which consumes much less bits. Another advantage is the reduction in computational complexity since the maximization is performed on decimated signals.

[0060] In Fig. 5A, an embodiment of a system of encoders 40A, 40B and decoders 50A, 50B according to the present invention is illustrated. Many details are similar as those illustrated in Fig. 2 and will therefore not be discussed in detail again, if their functions are essentially unaltered. A main signal 33A s_m(n) is provided to a first encoder 40A. The first encoder 40A operates according to any prior art CELP encoding model, producing an index k_m for the fixed codebook and a delay measure δ_m for the adaptive codebook. The details of this encoding are not of any importance for the present invention and is omitted in order to facilitate the understanding of Fig. 5A. The parameters k_m and δ_m are encoded in a first index encoder 38A, giving representations k*_m and δ*_m of the parameters that are sent to a first decoder 50A. In the first decoder, the representations k*_m and δ*_m are decoded into parameters k_m and δ_m in a first index decoder 53A. From these parameters, the original signal is reproduced according to any CELP decoding model according to prior art. The details of this decoding are not of any importance for the present invention and is omitted in order to facilitate the understanding of Fig. 5A. A reproduced first output signal 21A ŝ_m(n) is provided.

[0061] A side signal 33B s_S(n) is provided as an input signal to a second encoder 40B. The second encoder 40B is to most parts similar as the encoder of Fig. 2. The signals are now given an index "s" to distinguish them from any signals used for encoding the main signal. The second encoder 40B comprises a CELP synthesis block 25¹. According to the present invention, the index k_m or a representation thereof is provided from the first encoder 40A to an input 45 of the fixed codebook 10 of the second encoder 40B. The index k_m is used by a candidate deriving means 47 to extract a reduced fixed codebook 10' according to the above presented principles. The synthesis of the CELP synthesis block 25' of the second encoder 40B is thus based on indices k'_s representing excitation signals c'_k's (n) from the reduced fixed codebook 10'. An index k'_s is thus found to represent a best choice of the CELP synthesis. The parameters k'_s and δ_s are encoded in a second index encoder 38B, giving representations k'*_s and δ*_s of the parameters that are sent to a second decoder 50B.

[0062] In the second decoder 50B, the representations k'*_s and δ*_s are decoded into parameters k'_s and δ_s in a second index decoder 53B. Furthermore, the index parameter k_m is available from the first decoder 50A and is provided to the input 55 of the fixed codebook 10 of the second decoder 50B, in order to enable an extraction by a candidate deriving means 57 of a reduced fixed codebook 10' equal to what was used in the second encoder 40B. From the parameters k'_s and δ_s and the reduced fixed codebook 10', the original side signal is reproduced according to ordinary CELP decoding models 25". The details of this decoding are performed essentially in analogy with Fig. 2, but using the reduced fixed codebook 10' instead. A reproduced side output signal 21B ŝ_s (n) is thus provided.

[0063] Selection of the rule to construct the set of candidate pulses, e.g. the indexing function J(i,k), can advantageously be made adaptive and dependent on additional inter-channel characteristics, such as delay parameters, degree of correlation, etc. In this case, i.e. adaptive rule selection, the encoder has preferably to transmit to the decoder which rule has been selected for deriving the set of candidate pulses for encoding the other signal. The rule selection could for instance be performed by a closed-loop procedure, where a number of rules are tested and the one giving the best result finally is selected.

[0064] Fig. 5B illustrates an embodiment, using the rule selection approach. The mono signal s_m(n) and preferably also the side signal s_s(n) are here additionally provided to a rule selecting unit 39. Alternatively to the mono signal, the parameter k_m representing the mono signal can be used. In the rule selection unit 39, the signals are analysed, e.g. with respect to delay parameters or degree of correlation. Depending on the results, a rule, e.g. represented by an index r is selected from a set of predefined rules. The index of the selected rule is provided to the candidate deriving means 47 for determining how the candidate sets should be derived. The rule index r is also provided to the second index encoder 38B giving a representation r* of the index, which subsequently is sent to the second decoder 50B. The second index decoder 53B decodes the rule index r, which then is used to govern the operation of the candidate deriving means 57.

[0065] In this manner, a set of rules can be provided, which will be suitable for different types of signals. A further flexibility is thus achieved, just by adding a single rule index in the transfer of data.

[0066] The specific rule used as well as the resulting number of candidate side signal pulses are the main parameters governing the bit rate and the complexity of the algorithm.

[0067] The same principles as described above could equally well be applied for re-encoding of one and the same channel. Fig. 6 illustrates an exemplary encoder and decoder, where different parts of a transmission path allows for different bit rates. It is thus applicable as part of a rate transcoding solution. A signal s(n) is provided as an input signal 33A to a first encoder 40A, which produces representations k* and δ* of parameters that are transmitted according to a first bit rate. At a certain place, the available bit rate is reduced, and a re-encoding for lower bit-rates has to be performed. A first decoder 50A uses the representations k* and δ* of parameters for producing a reproduced signal 21A ŝ(n). This reproduced signal 21A ŝ(n) is provided to a second encoder 40B as an input signal 33B. Also the index k from the first decoder 50A is provided to the second encoder 40B. The index k is in analogy with Fig. 6 used for extracting a reduced fixed codebook 10'. The second encoder 40B encodes the signal ŝ(n) for a lower bit rate, giving an index k̂' representing the selected excitation signal c'_k̂, (n). However, this index k̂' is of little use in a distant decoder, since the decoder does not have the information necessary to construct a corresponding reduced fixed codebook. The index k' thus has to be associated with an index k̂, referring to the original codebook 10. This is preferably performed in connection with the fixed codebook 10 and is represented in Fig. 6 by the arrows 41 and 43 illustrating the input of k' and the output of k̂. The encoding of the index k̂ is then performed with reference to a full set of candidate excitation signals.

[0068] In a typical case, a first encoding is made with a bit rate n and the second encoding is made with a bit rate m, where n>m.

[0069] In certain applications, for instance real-time transmission of live content through different types of networks with different capacities (for example teleconferencing), it may also be of interest to provide parallel encodings with differing bit rates, e.g. in situation where real time encoding of the same signal at several different bit-rates is needed in order to accommodate the different types of networks, so-called parallel multirate encoding. Fig. 7 illustrates an exemplary system, where a signal s(n) is provided to both a first encoder 40A and a second encoder 40B. In analogy with previous exemplary encoder and decoder, the second encoder provides a reduced fixed codebook 10' based on an index k_a representing the first encoding. The second encoding is here denoted by the index "b". The second encoder 40B thus becomes independent of the first decoder 50B. Most other parts are in analogy with Fig. 6, however, with adapted indexing.

[0070] For these two applications, re-encoding of the same signal at a lower rate, a substantial reduction in complexity is offered thus allowing the implementation of these applications with low cost hardware.

[0071] An embodiment of the above-described algorithm has been implemented in association with an AMR-WB speech codec. For encoding a side signal, the same adaptive codebook index is used as is used for encoding the mono excitation. The LTP gain as well as the innovation vector gain was not quantized.

[0072] The algorithm for the algebraic codebook was based on the mono pulse positions. As described in e.g. [6], the codebook may be structured in tracks. Except for the lowest mode, the number of tracks is equal to 4. For each mode a certain number of pulses positions is used. For example, for mode 5, i.e. 15.85kbps, the candidate pulse positions are as follows

Table 1. Candidate pulse positions.

Track	Pulse	Positions
1	i0, i4, i8	0, 4, 8, 12, 16, 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 60
2	i1, i5, i9	1, 5, 9, 13, 17, 21, 25, 29, 33, 37, 41, 45, 49, 53, 57, 61
3	i2, i6, i10	2, 6, 10, 14, 18, 22, 26, 30, 34, 38, 42, 46, 50, 54, 58, 62
4	i3, i7, i11	3, 7, 11, 15, 19, 23, 27, 31, 35, 39, 43, 47, 51, 55, 59, 63

[0073] The implemented algorithm retains all the mono pulses as the pulse positions of the side signal, i.e. the pulse positions are not encoded. Only the signs of the pulses are encoded.

Table 2. Side and mono signal pulses.

Track	Side signal pulse	Mono signal pulse
1	p0, p4, p8	i0, i4, i8
2	p1, p5, p9	i1, i5, i9
3	p2, p6, p10	i2, i6, i10
4	p3, p7, p11	i3, i7, i11

[0074] Thus, each pulse will consume only 1 bit for encoding the sign, which leads to a total bit rate equal to the number of mono pulses. In the above example, there are 12 pulses per sub-frame and this leads to a total bit rate equal to 12 bits x 4 x 50 = 2.4 kbps for encoding the innovation vector. This is the same number of bits required for the very lowest AMR-WB mode (2 pulses for the 6.6kbps mode), but in this case we have higher pulses density.

[0075] It should be noted that no additional algorithmic delay is needed for encoding the stereo signal.

[0076] Fig. 8 shows the results obtained with PEAQ [4] for evaluating the perceptual quality. PEAQ has been chosen since to the best knowledge, it is the only tool that provides objective quality measures for stereo signals. From the results, it is clearly seen that the stereo 100 does in fact provide a quality lift with respect to the mono signal 102. The used sound items were quite various, sound 1, S1, is an extract from a movie with background noise, sound 2, S2, is a 1 min radio recording, sound 3, S3, a cart racing sport event, and sound 4, S4, is a real two microphone recording.

[0077] Fig. 9 illustrates an embodiment of an encoding method according to the present invention. The procedure starts in step 200. In step 210, a representation of a CELP excitation signal for a first audio signal is provided. Note that it is not absolutely necessary to provide the entire first audio signal, just the representation of the CELP excitation signal. In step 212, a second audio signal is provided, which is correlated with the first audio signal. A set of candidate excitation signals is derived in step 214 depending on the first CELP excitation signal. Preferably, the pulse positions of the candidate excitation signals are related to the pulse positions of the CELP excitation signal of the first audio signal. In step 216, a CELP encoding is performed on the second audio signal, using the reduced set of candidate excitation signals derived in step 214. Finally, the representation, i.e. typically an index, of the CELP excitation signal for the second audio signal is encoded, using references to the reduced candidate set. The procedure ends in step 299.

[0078] Fig. 10 illustrates an exemplary re-encoding method. The procedure starts in step 200. In step 211, an audio signal is provided. In step 213, a representation of a first CELP excitation signal for the same audio signal is provided. A set of candidate excitation signals is derived in step 215 depending on the first CELP excitation signal. Preferably, the pulse positions of the candidate excitation signals are related to the pulse positions of the CELP excitation signal of the first audio signal. In step 217, a CELP re-encoding is performed on the audio signal, using the reduced set of candidate excitation signals derived in step 215. Finally, the representation, i.e. typically an index, of the second CELP excitation signal for the audio signal is encoded, using references to the nonreduced candidate set, i.e. the set used for the first CELP encoding. The procedure ends in step 299.

[0079] Fig. 11 illustrates an embodiment of a decoding method according to the present invention. The procedure starts in step 200. In step 210, a representation of a first CELP excitation signal for a first audio signal is provided. In step 252, a representation of a second CELP excitation signal for a second audio signal is provided. In step 254, a second excitation signal is derived from the second excitation signal and with knowledge of the first excitation signal. Preferably, a reduced set of candidate excitation signals is derived depending on the first CELP excitation signal, from which a second excitation signal is selected by use of an index for the second CELP excitation signal. In step 256, the second audio signal is reconstructed using the second excitation signal. The procedure ends in step 299.

[0080] The embodiments described above are to be understood as a few illustrative examples of the present invention. It will be understood by those skilled in the art that various modifications, combinations and changes may be made to the embodiments without departing from the scope of the present invention. In particular, different part solutions in the different embodiments can be combined in other configurations, where technically possible. The scope of the present invention is, however, defined by the appended claims.

[0081] The invention allows a dramatic reduction of complexity (both memory and arithmetic operations) as well as bit-rate when encoding multiple audio channels by using algebraic codebooks and CELP.

REFERENCES

[0082] 

[1] H. Fuchs, "Improving joint stereo audio coding by adaptive inter-channel prediction", in Proc. IEEE WASPAA, Mohonk, NY, Oct. 1993.

[2] S.A. Ramprashad, "Stereophonic CELP coding using cross channel prediction", in Proc. IEEE Workshop Speech Coding, pp. 136-138, Sept. 2000.

[3] T. Liebschen, "Lossless audio coding using adaptive multichannel prediction", in Proc. AES 113th Conv., Los Angeles, CA, Oct. 2002.

[4] ITU-R BS.1387

[5] WO 96/28810.

[6] 3GPP TS 26.190, p. 28, table 7

[7] US 2004/0044524 A1

[8] US 2004/0109471 A1

[9] US 2003/0191635 A1

[10] US 6,393,392 B1

[11]US 6,192,334 B1

Claims

1. Method for encoding audio signals, comprising the steps of:

providing a representation (k, k_m, k_a) of a first excitation signal of a code excited linear prediction of a first audio signal (33A)characterised by,

deriving a set (10') of candidate excitation signals (c'(n)) based on said first excitation signal,

providing a second audio signal (33B), said second audio signal (33B) being different from said first audio signal (33A) characterised by

performing a code excited linear prediction encoding of said second audio signal (33B) using said set (10') of candidate excitation signals (c'(n)).

2. Method according to claim 1, wherein said second audio signal (33B) is correlated to said first audio signal (33A).

3. Method according to claim 1 or 2, wherein said step of deriving said set (10') of candidate excitation signals (c'(n)) comprises selecting a rule out of a predetermined set of rules based on said first excitation signal and/or said second audio signal, whereby said set (10') of candidate excitation signals (c'(n)) is derived according to said selected rule.

4. Method according to any of the claims 1 to 3, wherein
said first excitation signal has n pulse locations (P_M) out of a set of N possible pulse locations;
said candidate excitation signals (c'(n)) have pulse locations (P*s) only at a subset of said N possible pulse locations; and
said subset of pulse locations (P*s) being selected based on the n pulse locations (P_M) of said first excitation signal.

5. Method according to claim 4, wherein pulse locations (P*s) of said subset of pulse locations are positioned at positions p_j, where index j is within intervals {i+L, i+K}, where i is an index of said n pulse locations, K and L are integers and K>L.

6. Method according to claim 5, wherein K=1 and L=-1.

7. Method according to any of the claims 1 to 6, wherein said code excited linear prediction of said second audio signal (33B) is performed with a global search within said set (10') of candidate excitation signals.

8. Method according to any of the claims 1 to 7, comprising the further steps of:

encoding a second excitation signal of said code excited linear prediction of said second audio signal (33B) with reference to said set (10') of candidate excitation signals; and

providing said encoded second excitation signal together with said representation (k, k_m, k_a) of said first excitation signal.

9. Method according to claim 3 and claim 8, comprising the further step of providing data representing an identification of said selected rule together with said representation (k, k_m, k_a) of said first excitation signal.

10. Method according to any of the claims 1 to 7, comprising the further step of:

encoding a second excitation signal of said code excited linear prediction of said second audio signal (33B) with reference to a set (10) of candidate excitation signals having N possible pulse locations.

11. Method according to any of the claims 4 to 10, wherein the second excitation signal has m pulse locations, where m<n.

12. Method for decoding of audio signals (33A, 33B), comprising the step of:

providing a representation (k, k_m, k_a) of a first excitation signal of a code excited linear prediction of a first audio signal (33A),

providing a representation (k'_s) of a second excitation signal of a code excited linear prediction of a second audio signal (33B), said second audio signal (33B) being different from said first audio signal (33A);

said second excitation signal being one of a set (10') of candidate excitation signals;

said set (10') of candidate excitation signals being based on said first excitation signal;

deriving said second excitation signal (c'_k's (n)) from said representation (k'_s) of said second excitation signal and based on information related to said set (10') of candidate excitation signals; and

reconstructing said second audio signal (ŝ_s(n)) by prediction filtering said second excitation signal (c'_k's (n)).

13. Method according to claim 12, wherein said second audio signal (33B) is correlated to said first audio signal (33A) .

14. Method according to claim 12 or 13, wherein said information related to said set (10') of candidate excitation signals comprises identification of a rule out of a pre-determined set of rules, said rule determining derivation of said set (10') of candidate excitation signals.

15. Method according to any of the claims 12 to 14, wherein
said first excitation signal has n pulse locations (P_M) out of a set of N possible pulse locations;
said candidate excitation signals have pulse locations (P*s) only at a subset of said N possible pulse locations; and
said subset of pulse locations (P*s) being selected based on the n pulse locations (P_M) of said first excitation signal.

16. Method according to claim 15, wherein pulse locations (P*s) of said subset of pulse locations are positioned at positions p_j, where index j is within intervals {i+L, i+K}, where i is an index of said n pulse locations, K and L are integers and K>L.

17. Method according to claim 16, wherein K=1 and L=-1.

18. Encoder (40B) for audio signals, comprising:

means (45) for providing a representation (k, k_m, k_a) of a first excitation signal of a code excited linear prediction of a first audio signal (33A)characterised by,

means (47) for deriving a set (10') of candidate excitation signals, connected to receive said representation (k, k_m, k_a) of said first excitation signal, said set (10') of candidate excitation signals being based on said first excitation signal,

means for providing a second audio signal (33B), said second audio signal (33B) being different from said first audio signal (33A), characterised by

means (25') for performing a code excited linear prediction connected to receive said second audio signal (33B) and a representation of said set (10') of candidate excitation signals, said means (25') for performing a code excited linear prediction being arranged for performing a code excited linear prediction of said second audio signal (33B) using said set (10') of candidate excitation signals.

19. Encoder according to claim 18, wherein said second audio signal (33B) is correlated to said first audio signal (33A).

20. Encoder according to claim 18 or 19, wherein said means (47) for deriving a set (10') of candidate excitation signals is arranged to select a rule out of a predetermined set of rules based on said first excitation signal and/or said second audio signal and to derive said set (10') of candidate excitation signals (c'(n)) according to said selected rule.

21. Encoder according to any of the claims 18 to 20, wherein
said first excitation signal has n pulse locations (P_M) out of a set of N possible pulse locations;
said candidate excitation signals have pulse locations (P*s) only at a subset of said N possible pulse locations; and
said subset of pulse locations (P*s) being selected based on the n pulse locations (P_M) of said first excitation signal.

22. Encoder according to claim 21, wherein pulse locations (P*s) of said subset of pulse locations are positioned at positions p_j, where index j is within intervals {i+L, i+K}, where i is an index of said n pulse locations, K and L are integers and K>L.

23. Encoder according to claim 22, wherein K=1 and L=-1.

24. Encoder according to any of the claims 18 to 23, wherein said means (25') for performing code excited linear prediction of said second audio signal (33B) is arranged to perform a global search within said set (10') of candidate excitation signals.

25. Encoder according to any of the claims 18 to 24, further comprising:

means (38B) for encoding a second excitation signal of said code excited linear prediction of said second audio signal (33B) with reference to said set (10') of candidate excitation signals; and

means for providing said encoded second excitation signal together with said representation (k, k_m, k_a) of said first excitation signal.

26. Encoder according to claim 25 and 20, further comprising:

means for providing data representing an identification of said selected rule together with said representation (k, k_m, k_a) of said first excitation signal.

27. Encoder according to any of the claims 18 to 24, further comprising:

means (38B) for encoding a second excitation signal of said code excited linear prediction of said second audio signal (33B) with reference to a set (10) of candidate excitation signals having N possible pulse locations.

28. Encoder according to any of the claims 21 to 27, wherein the second excitation signal has m pulse locations, where m<n.

29. Decoder (50B) for audio signals, comprising:

means (55) for providing a representation (k_m) of a first excitation signal of a code excited linear prediction of a first audio signal (33A),

means (53B) for providing a representation (k'_s) of a second excitation signal of a code excited linear prediction of a second audio signal (33B), said second audio signal (33B) being different from said first audio signal (33A);

said second excitation signal being one of a set (10') of candidate excitation signals;

said set (10') of candidate excitation signals being based on said first excitation signal;

means (57) for deriving said second excitation signal, connected to receive information associated with said representation (k_m) of a first excitation signal and said representation (k'_s) of said second excitation signal, said means (57) for deriving being arranged to derive said second excitation signal (c'_k's (n)) from said representation (k'_s) of said second excitation signal and based on information related to said set (10') of candidate excitation signals; and

means (25") for reconstructing said second audio signal (ŝ_s(n)) by prediction filtering said second excitation signal (c'_k's (n)).

30. Decoder according to claim 29, wherein said second audio signal (33B) is correlated to said first audio signal (33A).

31. Decoder according to claim 29 or 30, wherein said information related to said set (10') of candidate excitation signals comprises identification of a rule out of a pre-determined set of rules, said rule determining derivation of said set (10') of candidate excitation signals.

32. Decoder according to any of the claims 29 to 31, wherein
said first excitation signal has n pulse locations (P_M) out of a set of N possible pulse locations;
said candidate excitation signals have pulse locations (P*s) only at a subset of said N possible pulse locations; and
said subset of pulse locations (P*s) being selected based on the n pulse locations (P_M) of said first excitation signal.

33. Decoder according to claim 32, wherein pulse locations (P_M) of said subset of pulse locations are positioned at positions p_j, where index j is within intervals {i+L, i+K}, where i is an index of said n pulse locations, K and L are integers and K>L.

34. Decoder according to claim 33, wherein K=1 and L=-1.

Ansprüche

1. Verfahren zum Codieren von Audiosignalen, folgende Schritte umfassend:

Bereitstellen einer Repräsentation (k, k_m, k_a) eines ersten Anregungssignals einer codeangeregten linearen Prädiktion eines ersten Audiosignals (33A),

Ableiten einer Menge (10') von Anregungssignal-Kandidaten (c'(n)) auf der Basis des ersten Anregungssignals,

Bereitstellen eines zweiten Audiosignals (33B), wobei das zweite Audiosignal (33B) vom ersten Audiosignal (33A) verschieden ist,

gekennzeichnet durch

Ausführen einer codeangeregten linearen Prädiktionscodierung des zweiten Audiosignals (33B) unter Verwendung der Menge (10') von Anregungssignal-Kandidaten (c' (n)).

2. Verfahren nach Anspruch 1, worin das zweite Audiosignal (33B) mit dem ersten Audiosignal (33A) korreliert ist.

3. Verfahren nach Anspruch 1 oder 2, worin der Schritt des Ableitens der Menge (10') von Anregungssignal-Kandidaten (c'(n)) das Auswählen einer Regel aus einer vorbestimmten Menge von Regeln auf der Basis des ersten Anregungssignals und/oder des zweiten Audiosignals umfasst, wodurch die Menge (10') von Anregungssignal-Kandidaten (c'(n)) gemäß der ausgewählten Regel abgeleitet ist.

4. Verfahren nach einem der Ansprüche 1 bis 3, worin
das erste Anregungssignal n Impulspositionen (P_M) aus einer Menge von N möglichen Impulspositionen hat;
die Anregungssignal-Kandidaten (c'(n)) Impulspositionen (P*s) nur in einer Teilmenge der N möglichen Impulspositionen haben; und
die Teilmenge von Impulspositionen (P*s) auf der Basis der n Impulspositionen (P_M) des ersten Anregungssignals ausgewählt wird.

5. Verfahren nach Anspruch 4, worin Impulspositionen (P*s) der Teilmenge von Impulspositionen an Positionen p_j positioniert sind, wo der Index j innerhalb der Intervalle {i+L, i+K} liegt, wo i ein Index der n Impulspositionen ist, wo K und L Ganzzahlen sind und K>L.

6. Verfahren nach Anspruch 5, worin K=1 und L=-1.

7. Verfahren nach einem der Ansprüche 1 bis 6, worin die codeangeregte lineare Prädiktion des zweiten Audiosignals (33B) mit einer Globalsuche innerhalb der Menge (10') von Anregungssignal-Kandidaten ausgeführt wird.

8. Verfahren nach einem der Ansprüche 1 bis 7, folgende weitere Schritte umfassend:

Codieren eines zweiten Anregungssignals der codeangeregten linearen Prädiktion des zweiten Audiosignals (33B) mit Bezug auf die Menge (10') von Anregungssignal-Kandidaten; und

Bereitstellen des codierten zweiten Anregungssignals zusammen mit der Repräsentation (k, k_m, k_a) des ersten Anregungssignals.

9. Verfahren nach Anspruch 3 und Anspruch 8, den weiteren Schritt umfassend, Daten bereitzustellen, die eine Identifikation der ausgewählten Regel zusammen mit der Repräsentation (k, k_m, k_a) des ersten Anregungssignals repräsentieren.

10. Verfahren nach einem der Ansprüche 1 bis 7, den folgenden weiteren Schritt umfassend:

Codieren eines zweiten Anregungssignals der codeangeregten linearen Prädiktion des zweiten Audiosignals (33B) mit Bezug auf eine Menge (10) von Anregungssignal-Kandidaten mit N möglichen Impulspositionen.

11. Verfahren nach einem der Ansprüche 4 bis 10, worin das zweite Anregungssignal m Impulspositionen hat mit m<n.

12. Verfahren zum Decodieren von Audiosignalen (33A, 33B), den folgenden Schritt umfassend:

Bereitstellen einer Repräsentation (k, k_m, k_a) eines ersten Anregungssignals einer codeangeregten linearen Prädiktion eines ersten Audiosignals (33A),

Bereitstellen einer Repräsentation (k'_s) eines zweiten Anregungssignals einer codeangeregten linearen Prädiktion eines zweiten Audiosignals (33B), wobei das zweite Audiosignal (33B) vom ersten Audiosignal (33A) verschieden ist;

wobei das zweite Anregungssignal eines einer Menge (10') von Anregungssignal-Kandidaten ist;

wobei die Menge (10') von Anregungssignal-Kandidaten auf dem ersten Anregungssignal basiert;

Ableiten des zweiten Anregungssignals (c'_k's (n)) aus der Repräsentation (k'_s) des zweiten Anregungssignals und auf der Basis von Information, die die Menge (10') von Anregungssignal-Kandidaten betrifft; und

Rekonstruieren des zweiten Audiosignals (Ŝ_s (n)) durch Prädiktionsfilterung des zweiten Anregungssignals (c'_k's (n)).

13. Verfahren nach Anspruch 12, worin das zweite Audiosignal (33B) mit dem ersten Audiosignal (33A) korreliert ist.

14. Verfahren nach Anspruch 12 oder 13, worin die die Menge (10') von Anregungssignal-Kandidaten betreffende Information die Identifikation einer Regel aus einer vorbestimmten Menge von Regeln umfasst, wobei die Regel die Ableitung der Menge (10') von Anregungssignal-Kandidaten bestimmt.

15. Verfahren nach einem der Ansprüche 12 bis 14, worin
das erste Anregungssignal n Impulspositionen (P_M) aus einer Menge von N möglichen Impulspositionen hat;
die Anregungssignal-Kandidaten Impulspositionen (P*s) nur in einer Teilmenge der N möglichen Impulspositionen haben; und
die Teilmenge von Impulspositionen (P*s) auf der Basis der n Impulspositionen (P_M) des ersten Anregungssignals ausgewählt wird.

16. Verfahren nach Anspruch 15, worin Impulspositionen (P*s) der Teilmenge von Impulspositionen an Positionen p_j positioniert sind, wo der Index j innerhalb der Intervalle {i+L, i+K} liegt, wo i ein Index der n Impulspositionen ist, wo K und L Ganzzahlen sind und K>L.

17. Verfahren nach Anspruch 16, worin K=1 und L=-1.

18. Codierer (40B) für Audiosignale, umfassend:

Mittel (45) zum Bereitstellen einer Repräsentation (k, k_m, k_a) eines ersten Anregungssignals einer codeangeregten linearen Prädiktion eines ersten Audiosignals (33A),

Mittel (47) zum Ableiten einer Menge (10') von Anregungssignal-Kandidaten, angeschlossen zum Empfangen der Repräsentation (k, k_m, k_a) des ersten Anregungssignals, wobei die Menge (10') von Anregungssignal-Kandidaten auf dem ersten Anregungssignal basiert,

Mittel zum Bereitstellen eines zweiten Audiosignals (33B), wobei das zweite Audiosignal (33B) vom ersten Audiosignal (33A) verschieden ist,

gekennzeichnet durch

Mittel (25') zum Ausführen einer codeangeregten linearen Prädiktion, angeschlossen zum Empfangen des zweiten Audiosignals (33B) und einer Repräsentation der Menge (10') von Anregungssignal-Kandidaten, wobei das Mittel (25') zum Ausführen einer codeangeregten linearen Prädiktion dazu angeordnet ist, eine codeangeregte lineare Prädiktion des zweiten Audiosignals (33B) unter Verwendung der Menge (10') von Anregungssignal-Kandidaten auszuführen.

19. Codierer nach Anspruch 18, worin das zweite Audiosignal (33B) mit dem ersten Audiosignal (33A) korreliert ist.

20. Codierer nach Anspruch 18 oder 19, worin das Mittel (47) zum Ableiten einer Menge (10') von Anregungssignal-Kandidaten dazu angeordnet ist, eine Regel aus einer vorbestimmten Menge von Regeln auf der Basis des ersten Anregungssignals und/oder des zweiten Audiosignals auszuwählen und die Menge (10') von Anregungssignal-Kandidaten (c'(n)) gemäß der ausgewählten Regel abzuleiten.

21. Codierer nach einem der Ansprüche 18 bis 20, worin
das erste Anregungssignal n Impulspositionen (P_M) aus einer Menge von N möglichen Impulspositionen hat;
die Anregungssignal-Kandidaten Impulspositionen (P*s) nur in einer Teilmenge der N möglichen Impulspositionen haben; und
die Teilmenge von Impulspositionen (P*s) auf der Basis der n Impulspositionen (P_M) des ersten Anregungssignals ausgewählt wird.

22. Codierer nach Anspruch 21, worin Impulspositionen (P*s) der Teilmenge von Impulspositionen an Positionen p_j positioniert sind, wo der Index j innerhalb der Intervalle {i+L, i+K} liegt, wo i ein Index der n Impulspositionen ist, wo K und L Ganzzahlen sind und K>L.

23. Codierer nach Anspruch 22, worin K=1 und L=-1.

24. Codierer nach einem der Ansprüche 18 bis 23, worin das Mittel (25') zum Ausführen von codeangeregter linearer Prädiktion des zweiten Audiosignals (33B) dazu angeordnet ist, eine Globalsuche innerhalb der Menge (10') von Anregungssignal-Kandidaten auszuführen.

25. Codierer nach einem der Ansprüche 18 bis 24, außerdem umfassend:

Mittel (38B) zum Codieren eines zweiten Anregungssignals der codeangeregten linearen Prädiktion des zweiten Audiosignals (33B) mit Bezug auf die Menge (10') von Anregungssignal-Kandidaten; und

Mittel zum Bereitstellen des codierten zweiten Anregungssignals zusammen mit der Repräsentation (k, k_m, k_a) des ersten Anregungssignals.

26. Codierer nach Anspruch 25 und 20, außerdem umfassend:

Mittel zum Bereitstellen von Daten, die eine Identifikation der ausgewählten Regel zusammen mit der Repräsentation (k, k_m, k_a) des ersten Anregungssignals repräsentieren.

27. Codierer nach einem der Ansprüche 18 bis 24, außerdem umfassend:

Mittel (38B) zum Codieren eines zweiten Anregungssignals der codeangeregten linearen Prädiktion des zweiten Audiosignals (33B) mit Bezug auf eine Menge (10) von Anregungssignal-Kandidaten mit N möglichen Impulspositionen.

28. Codierer nach einem der Ansprüche 21 bis 27, worin das zweite Anregungssignal m Impulspositionen hat und m<n.

29. Decodierer (50B) für Audiosignale, umfassend:

Mittel (55) zum Bereitstellen einer Repräsentation (k_m) eines ersten Anregungssignals einer codeangeregten linearen Prädiktion eines ersten Audiosignals (33A),

Mittel (53B) zum Bereitstellen einer Repräsentation (k'_s) eines zweiten Anregungssignals einer codeangeregten linearen Prädiktion eines zweiten Audiosignals (33B), wobei das zweite Audiosignal (33B) vom ersten Audiosignal (33A) verschieden ist;

wobei das zweite Anregungssignal eines einer Menge (10') von Anregungssignal-Kandidaten ist;

wobei die Menge (10') von Anregungssignal-Kandidaten auf dem ersten Anregungssignal basiert;

Mittel (57) zum Ableiten des zweiten Anregungssignals, angeschlossen zum Empfangen von Information, die mit der Repräsentation (k_m) eines ersten Anregungssignals und der Repräsentation (k'_s) des zweiten Anregungssignals assoziiert ist, wobei das Mittel (57) zum Ableiten dazu angeordnet ist, das zweite Anregungssignal (c'_k's (n)) aus der Repräsentation (k'_s) des zweiten Anregungssignals abzuleiten und auf der Basis von Information, die die Menge (10') von Anregungssignal-Kandidaten betrifft; und

Mittel (25") zum Rekonstruieren des zweiten Audiosignals (ŝ_s(n))durch Prädiktionsfilterung des zweiten Anregungssignals (c'_k's (n)).

30. Decodierer nach Anspruch 29, worin das zweite Audiosignal (33B) mit dem ersten Audiosignal (33A) korreliert ist.

31. Decodierer nach Anspruch 29 oder 30, worin die Information, die die Menge (10') von Anregungssignal-Kandidaten betrifft, Identifikation einer Regel aus einer vorbestimmten Menge von Regeln umfasst, wobei die Regel die Ableitung der Menge (10') von Anregungssignal-Kandidaten bestimmt.

32. Decodierer nach einem der Ansprüche 29 bis 31, worin
das erste Anregungssignal n Impulspositionen (P_M) aus einer Menge von N möglichen Impulspositionen hat;
die Anregungssignal-Kandidaten Impulspositionen (P*s) nur in einer Teilmenge der N möglichen Impulspositionen haben; und
die Teilmenge von Impulspositionen (P*s) auf der Basis der n Impulspositionen (P_M) des ersten Anregungssignals ausgewählt wird.

33. Decodierer nach Anspruch 32, worin Impulspositionen (P_M) der Teilmenge von Impulspositionen an Positionen p_j positioniert sind, wo der Index j innerhalb der Intervalle {i+L, i+K} liegt, wo i ein Index der n Impulspositionen ist, wo K und L Ganzzahlen sind und K>L.

34. Decodierer nach Anspruch 33, worin K=1 und L=-1 ist.

Revendications

1. Procédé destiné à coder des signaux audio, comportant les étapes ci-dessous consistant à :

fournir une représentation (k, k_m, k_a) d'un premier signal d'excitation d'une prédiction linéaire à excitation par code d'un premier signal audio (33A) ;

calculer un ensemble (10') de signaux d'excitation candidats (c'(n)) sur la base dudit premier signal d' excitation ;

fournir un second signal audio (33B), ledit second signal audio (33B) étant distinct dudit premier signal audio (33A),

caractérisé par l'étape ci-dessous consistant à :

mettre en oeuvre un codage par prédiction linéaire à excitation par code dudit second signal audio (33B) en faisant appel audit ensemble (10') de signaux d'excitation candidats (c'(n)).

2. Procédé selon la revendication 1, dans lequel ledit second signal audio (33B) est corrélé avec ledit premier signal audio (33A).

3. Procédé selon la revendication 1 ou 2, dans lequel ladite étape consistant à calculer ledit ensemble (10') de signaux d'excitation candidats (c'(n)) comporte l'étape consistant à sélectionner une règle parmi un ensemble de règles prédéterminé sur la base dudit premier signal d'excitation et/ou dudit second signal audio, moyennant quoi ledit ensemble (10') de signaux d'excitation candidats (c'(n)) est obtenu selon ladite règle sélectionnée.

4. Procédé selon l'une quelconque des revendications 1 à 3, dans lequel
ledit premier signal d'excitation présente n emplacements d'impulsion (P_M) sur un ensemble de N emplacements d'impulsion possibles ;
lesdits signaux d'excitation candidats (c' (n)) présentent des emplacements d'impulsion (P*s) uniquement au niveau d'un sous-ensemble desdits N emplacements d'impulsion possibles ; et
ledit sous-ensemble d'emplacements d'impulsion (P*_s) étant sélectionné sur la base des n emplacements d'impulsion (P_M) dudit premier signal d'excitation.

5. Procédé selon la revendication 4, dans lequel les emplacements d'impulsion (P*_s) dudit sous-ensemble d'emplacements d'impulsion sont positionnés au niveau de positions p_j, où l'indice j se situe au sein des intervalles {i + L, i + K}, où i est un indice desdits n emplacements d'impulsion, K et L sont des nombres entiers et K > L.

6. Procédé selon la revendication 5, dans lequel K = 1 et L = -1.

7. Procédé selon l'une quelconque des revendications 1 à 6, dans lequel ladite prédiction linéaire à excitation par code dudit second signal audio (33B) est mise en oeuvre avec une recherche globale dans ledit ensemble (10') de signaux d'excitation candidats.

8. Procédé selon l'une quelconque des revendications 1 à 7, comportant en outre les étapes ci-dessous consistant à :

coder un second signal d'excitation de ladite prédiction linéaire à excitation par code dudit second signal audio (33B) en référence audit ensemble (10') de signaux d'excitation candidats ; et

fournir ledit second signal d'excitation codé avec ladite représentation (k, k_m, k_a) dudit premier signal d'excitation.

9. Procédé selon les revendications 3 et 8, comportant en outre l'étape consistant à fournir des données représentant une identification de ladite règle sélectionnée avec ladite représentation (k, k_m, k_a) dudit premier signal d'excitation.

10. Procédé selon l'une quelconque des revendications 1 à 7, comportant en outre l'étape consistant à :

coder un second signal d'excitation de ladite prédiction linéaire à excitation par code dudit second signal audio (33B) en référence à un ensemble (10) de signaux d'excitation candidats présentant N emplacements d'impulsion possibles.

11. Procédé selon l'une quelconque des revendications 4 à 10, dans lequel le second signal d'excitation présente m emplacements d'impulsion, où m < n.

12. Procédé destiné à décoder des signaux audio (33A, 33B), comportant les étapes ci-dessous consistant à :

fournir une représentation (k, k_m, k_a) d'un premier signal d'excitation d'une prédiction linéaire à excitation par code d'un premier signal audio (33A) ;

fournir une représentation (k'_s) d'un second signal d'excitation d'une prédiction linéaire à excitation par code d'un second signal audio (33B), ledit second signal audio (33B) étant distinct dudit premier signal audio (33A) ;

ledit second signal d'excitation étant l'un d'un ensemble (10') de signaux d'excitation candidats ;

ledit ensemble (10') de signaux d'excitation candidats étant basé sur ledit dudit premier signal d' excitation ;

calculer ledit second signal d'excitation (C'_K's, (n)) à partir de ladite représentation (k'_s) dudit second signal d'excitation et sur la base d'informations connexes audit ensemble (10') de signaux d'excitation candidats ; et

reconstruire ledit second signal audio (Ŝ_s (n)) par le biais d'un filtrage de prédiction dudit second signal d'excitation (C'_K's, (n)).

13. Procédé selon la revendication 12, dans lequel ledit second signal audio (33B) est corrélé avec ledit premier signal audio (33A).

14. Procédé selon la revendication 12 ou 13, dans lequel lesdites informations connexes audit ensemble (10') de signaux d'excitation candidats comportent une identification d'une règle parmi un ensemble de règles prédéterminé, ladite règle déterminant le calcul dudit ensemble (10') de signaux d'excitation candidats.

15. Procédé selon l'une quelconque des revendications 12 à 14, dans lequel
ledit premier signal d'excitation présente n emplacements d'impulsion (P_M) sur un ensemble de N emplacements d'impulsion possibles ;
lesdits signaux d'excitation candidats présentent des emplacements d'impulsion (P*_s) uniquement au niveau d'un sous-ensemble desdits N emplacements d'impulsion possibles ; et
ledit sous-ensemble d'emplacements d'impulsion (P*_s) étant sélectionné sur la base des n emplacements d'impulsion (P_M) dudit premier signal d'excitation.

16. Procédé selon la revendication 15, dans lequel les emplacements d'impulsion (P*_s) dudit sous-ensemble d'emplacements d'impulsion sont positionnés au niveau de positions p_j, où l'indice j se situe au sein des intervalles {i + L, i + K}, où i est un indice desdits n emplacements d'impulsion, K et L sont des nombres entiers et K > L.

17. Procédé selon la revendication 16, dans lequel K = 1 et L=-1.

18. Codeur (40B) de signaux audio, comportant :

un moyen (45) pour fournir une représentation (k, k_m, k_a) d'un premier signal d'excitation d'une prédiction linéaire à excitation par code d'un premier signal audio (33A) ;

un moyen (47) pour calculer un ensemble (10') de signaux d'excitation candidats, connecté de manière à recevoir ladite représentation (k, k_m, k_a) dudit premier signal d'excitation, ledit ensemble (10') de signaux d'excitation candidats étant basé sur ledit dudit premier signal d'excitation,

un moyen pour fournir un second signal audio (33B), ledit second signal audio (33B) étant distinct dudit premier signal audio (33A) ;

caractérisé par :

un moyen (25') pour mettre en oeuvre une prédiction linéaire à excitation par code, connecté de manière à recevoir ledit second signal audio (33B) et une représentation dudit ensemble (10') de signaux d'excitation candidats, ledit moyen (25') pour mettre en oeuvre une prédiction linéaire à excitation par code étant agencé de manière à mettre en oeuvre une prédiction linéaire à excitation par code dudit second signal audio (33B) en faisant appel audit ensemble (10') de signaux d'excitation candidats.

19. Codeur selon la revendication 18, dans lequel ledit second signal audio (33B) est corrélé avec ledit premier signal audio (33A).

20. Codeur selon la revendication 18 ou 19, dans lequel ledit moyen (47) pour calculer un ensemble (10') de signaux d'excitation candidats est agencé de manière à sélectionner une règle parmi un ensemble de règles prédéterminé sur la base dudit premier signal d'excitation et/ou dudit second signal audio et à calculer ledit ensemble (10') de signaux d'excitation candidats (c'(n)) selon ladite règle sélectionnée.

21. Codeur selon l'une quelconque des revendications 18 à 20, dans lequel
ledit premier signal d'excitation présente n emplacements d'impulsion (P_M) sur un ensemble de N emplacements d'impulsion possibles ;
lesdits signaux d'excitation candidats présentent des emplacements d'impulsion (P*_s) uniquement au niveau d'un sous-ensemble desdits N emplacements d'impulsion possibles ; et
ledit sous-ensemble d'emplacements d'impulsion (P*_s) étant sélectionné sur la base des n emplacements d'impulsion (P_M) dudit premier signal d'excitation.

22. Codeur selon la revendication 21, dans lequel les emplacements d'impulsion (P*_s) dudit sous-ensemble d'emplacements d'impulsion sont positionnés au niveau de positions p_j, où l'indice j se situe au sein des intervalles {i + L, i + K}, où i est un indice desdits n emplacements d'impulsion, K et L sont des nombres entiers et K > L.

23. Codeur selon la revendication 22, dans lequel K = 1 et L= -1.

24. Codeur selon l'une quelconque des revendications 18 à 23, dans lequel ledit moyen (25') pour mettre en oeuvre une prédiction linéaire à excitation par code dudit second signal audio (33B) est agencé de manière à mettre en oeuvre une recherche globale dans ledit ensemble (10') de signaux d'excitation candidats.

25. Codeur selon l'une quelconque des revendications 18 à 24, comportant en outre :

un moyen (38B) pour coder un second signal d'excitation de ladite prédiction linéaire à excitation par code dudit second signal audio (33B) en référence audit ensemble (10') de signaux d'excitation candidats ; et

un moyen pour fournir ledit second signal d'excitation codé avec ladite représentation (k, k_m, k_a) dudit premier signal d'excitation.

26. Codeur selon les revendications 25 et 20, comportant en outre :

un moyen pour fournir des données représentant une identification de ladite règle sélectionnée avec ladite représentation (k, k_m, k_a) dudit premier signal d'excitation.

27. Codeur selon l'une quelconque des revendications 18 à 24, comportant en outre :

un moyen (38B) pour coder un second signal d'excitation de ladite prédiction linéaire à excitation par code dudit second signal audio (33B) en référence à un ensemble (10) de signaux d'excitation candidats présentant N emplacements d'impulsion possibles.

28. Codeur selon l'une quelconque des revendications 21 à 27, dans lequel le second signal d'excitation présente m emplacements d'impulsion, où m < n.

29. Décodeur (50B) de signaux audio, comportant :

un moyen (55) pour fournir une représentation (k_m) d'un premier signal d'excitation d'une prédiction linéaire à excitation par code d'un premier signal audio (33A) ;

un moyen (53B) pour fournir une représentation (k'_s) d'un second signal d'excitation d'une prédiction linéaire à excitation par code d'un second signal audio (33B), ledit second signal audio (33B) étant distinct dudit premier signal audio (33A) ;

ledit second signal d'excitation étant l'un d'un ensemble (10') de signaux d'excitation candidats ;

ledit ensemble (10') de signaux d'excitation candidats étant basé sur ledit dudit premier signal d' excitation ;

un moyen (57) pour calculer ledit second signal d'excitation, connecté de manière à recevoir des informations associées à ladite représentation (k_m) d'un premier signal d'excitation et à ladite représentation (k'_s) dudit second signal d'excitation, ledit moyen (57) pour calculer étant agencé à calculer ledit second signal d'excitation (C'_K's, (n)) à partir de ladite représentation (k'_s) dudit second signal d'excitation et sur la base d'informations connexes audit ensemble (10') de signaux d'excitation candidats ; et

un moyen (25") pour reconstruire ledit second signal audio (Ŝ_s (n)) par le biais d'un filtrage de prédiction dudit second signal d'excitation (C'_K's, (n)).

30. Décodeur selon la revendication 29, dans lequel ledit second signal audio (33B) est corrélé avec ledit premier signal audio (33A).

31. Décodeur selon la revendication 29 ou 30, dans lequel lesdites informations connexes audit ensemble (10') de signaux d'excitation candidats comportent une identification d'une règle parmi un ensemble de règles prédéterminé, ladite règle déterminant le calcul dudit ensemble (10') de signaux d'excitation candidats.

32. Décodeur selon l'une quelconque des revendications 29 à 31, dans lequel
ledit premier signal d'excitation présente n emplacements d'impulsion (P_M) sur un ensemble de N emplacements d'impulsion possibles ;
lesdits signaux d'excitation candidats présentent des emplacements d'impulsion (P*_s) uniquement au niveau d'un sous-ensemble desdits N emplacements d'impulsion possibles ; et
ledit sous-ensemble d'emplacements d'impulsion (P*_s) étant sélectionné sur la base des n emplacements d'impulsion (P_M) dudit premier signal d'excitation.

33. Décodeur selon la revendication 32, dans lequel les emplacements d'impulsion (P_M) dudit sous-ensemble d'emplacements d'impulsion sont positionnés au niveau de positions p_j, où l'indice j se situe au sein des intervalles {i + L, i + K}, où i est un indice desdits n emplacements d'impulsion, K et L sont des nombres entiers et K > L.

34. Décodeur selon la revendication 33, dans lequel K = 1 et L = -1.

Drawing